Adding upstream version 5.10.209.upstream/5.10.209upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 1383 insertions, 0 deletions

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
new file mode 100644
index 000000000..84cf190aa
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -0,0 +1,1383 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <linux/spinlock.h>
+#include <sound/compress_driver.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/of_device.h>
+#include <sound/pcm_params.h>
+#include "q6asm.h"
+#include "q6routing.h"
+#include "q6dsp-errno.h"
+
+#define DRV_NAME	"q6asm-fe-dai"
+
+#define PLAYBACK_MIN_NUM_PERIODS    2
+#define PLAYBACK_MAX_NUM_PERIODS   8
+#define PLAYBACK_MAX_PERIOD_SIZE    65536
+#define PLAYBACK_MIN_PERIOD_SIZE    128
+#define CAPTURE_MIN_NUM_PERIODS     2
+#define CAPTURE_MAX_NUM_PERIODS     8
+#define CAPTURE_MAX_PERIOD_SIZE     4096
+#define CAPTURE_MIN_PERIOD_SIZE     320
+#define SID_MASK_DEFAULT	0xF
+
+/* Default values used if user space does not set */
+#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
+#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
+#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
+#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
+
+#define ALAC_CH_LAYOUT_MONO   ((101 << 16) | 1)
+#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
+
+enum stream_state {
+	Q6ASM_STREAM_IDLE = 0,
+	Q6ASM_STREAM_STOPPED,
+	Q6ASM_STREAM_RUNNING,
+};
+
+struct q6asm_dai_rtd {
+	struct snd_pcm_substream *substream;
+	struct snd_compr_stream *cstream;
+	struct snd_codec codec;
+	struct snd_dma_buffer dma_buffer;
+	spinlock_t lock;
+	phys_addr_t phys;
+	unsigned int pcm_size;
+	unsigned int pcm_count;
+	unsigned int pcm_irq_pos;       /* IRQ position */
+	unsigned int periods;
+	unsigned int bytes_sent;
+	unsigned int bytes_received;
+	unsigned int copied_total;
+	uint16_t bits_per_sample;
+	uint16_t source; /* Encoding source bit mask */
+	struct audio_client *audio_client;
+	uint32_t next_track_stream_id;
+	bool next_track;
+	uint32_t stream_id;
+	uint16_t session_id;
+	enum stream_state state;
+	uint32_t initial_samples_drop;
+	uint32_t trailing_samples_drop;
+	bool notify_on_drain;
+};
+
+struct q6asm_dai_data {
+	struct snd_soc_dai_driver *dais;
+	int num_dais;
+	long long int sid;
+};
+
+static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
+	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
+				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,
+	.rate_min =             8000,
+	.rate_max =             48000,
+	.channels_min =         1,
+	.channels_max =         4,
+	.buffer_bytes_max =     CAPTURE_MAX_NUM_PERIODS *
+				CAPTURE_MAX_PERIOD_SIZE,
+	.period_bytes_min =	CAPTURE_MIN_PERIOD_SIZE,
+	.period_bytes_max =     CAPTURE_MAX_PERIOD_SIZE,
+	.periods_min =          CAPTURE_MIN_NUM_PERIODS,
+	.periods_max =          CAPTURE_MAX_NUM_PERIODS,
+	.fifo_size =            0,
+};
+
+static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
+				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_192000,
+	.rate_min =             8000,
+	.rate_max =             192000,
+	.channels_min =         1,
+	.channels_max =         8,
+	.buffer_bytes_max =     (PLAYBACK_MAX_NUM_PERIODS *
+				PLAYBACK_MAX_PERIOD_SIZE),
+	.period_bytes_min =	PLAYBACK_MIN_PERIOD_SIZE,
+	.period_bytes_max =     PLAYBACK_MAX_PERIOD_SIZE,
+	.periods_min =          PLAYBACK_MIN_NUM_PERIODS,
+	.periods_max =          PLAYBACK_MAX_NUM_PERIODS,
+	.fifo_size =            0,
+};
+
+#define Q6ASM_FEDAI_DRIVER(num) { \
+		.playback = {						\
+			.stream_name = "MultiMedia"#num" Playback",	\
+			.rates = (SNDRV_PCM_RATE_8000_192000|		\
+					SNDRV_PCM_RATE_KNOT),		\
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |		\
+					SNDRV_PCM_FMTBIT_S24_LE),	\
+			.channels_min = 1,				\
+			.channels_max = 8,				\
+			.rate_min =     8000,				\
+			.rate_max =	192000,				\
+		},							\
+		.capture = {						\
+			.stream_name = "MultiMedia"#num" Capture",	\
+			.rates = (SNDRV_PCM_RATE_8000_48000|		\
+					SNDRV_PCM_RATE_KNOT),		\
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |		\
+				    SNDRV_PCM_FMTBIT_S24_LE),		\
+			.channels_min = 1,				\
+			.channels_max = 4,				\
+			.rate_min =     8000,				\
+			.rate_max =	48000,				\
+		},							\
+		.name = "MultiMedia"#num,				\
+		.id = MSM_FRONTEND_DAI_MULTIMEDIA##num,			\
+	}
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+	88200, 96000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+	.count = ARRAY_SIZE(supported_sample_rates),
+	.list = supported_sample_rates,
+	.mask = 0,
+};
+
+static const struct snd_compr_codec_caps q6asm_compr_caps = {
+	.num_descriptors = 1,
+	.descriptor[0].max_ch = 2,
+	.descriptor[0].sample_rates = {	8000, 11025, 12000, 16000, 22050,
+					24000, 32000, 44100, 48000, 88200,
+					96000, 176400, 192000 },
+	.descriptor[0].num_sample_rates = 13,
+	.descriptor[0].bit_rate[0] = 320,
+	.descriptor[0].bit_rate[1] = 128,
+	.descriptor[0].num_bitrates = 2,
+	.descriptor[0].profiles = 0,
+	.descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
+	.descriptor[0].formats = 0,
+};
+
+static void event_handler(uint32_t opcode, uint32_t token,
+			  void *payload, void *priv)
+{
+	struct q6asm_dai_rtd *prtd = priv;
+	struct snd_pcm_substream *substream = prtd->substream;
+
+	switch (opcode) {
+	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			q6asm_write_async(prtd->audio_client, prtd->stream_id,
+				   prtd->pcm_count, 0, 0, 0);
+		break;
+	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+		prtd->state = Q6ASM_STREAM_STOPPED;
+		break;
+	case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
+		prtd->pcm_irq_pos += prtd->pcm_count;
+		snd_pcm_period_elapsed(substream);
+		if (prtd->state == Q6ASM_STREAM_RUNNING)
+			q6asm_write_async(prtd->audio_client, prtd->stream_id,
+					   prtd->pcm_count, 0, 0, 0);
+
+		break;
+		}
+	case ASM_CLIENT_EVENT_DATA_READ_DONE:
+		prtd->pcm_irq_pos += prtd->pcm_count;
+		snd_pcm_period_elapsed(substream);
+		if (prtd->state == Q6ASM_STREAM_RUNNING)
+			q6asm_read(prtd->audio_client, prtd->stream_id);
+
+		break;
+	default:
+		break;
+	}
+}
+
+static int q6asm_dai_prepare(struct snd_soc_component *component,
+			     struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	struct q6asm_dai_data *pdata;
+	struct device *dev = component->dev;
+	int ret, i;
+
+	pdata = snd_soc_component_get_drvdata(component);
+	if (!pdata)
+		return -EINVAL;
+
+	if (!prtd || !prtd->audio_client) {
+		dev_err(dev, "%s: private data null or audio client freed\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+	prtd->pcm_irq_pos = 0;
+	/* rate and channels are sent to audio driver */
+	if (prtd->state) {
+		/* clear the previous setup if any  */
+		q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
+		q6asm_unmap_memory_regions(substream->stream,
+					   prtd->audio_client);
+		q6routing_stream_close(soc_prtd->dai_link->id,
+					 substream->stream);
+	}
+
+	ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
+				       prtd->phys,
+				       (prtd->pcm_size / prtd->periods),
+				       prtd->periods);
+
+	if (ret < 0) {
+		dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n",
+							ret);
+		return -ENOMEM;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+				       FORMAT_LINEAR_PCM,
+				       0, prtd->bits_per_sample, false);
+	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
+				      FORMAT_LINEAR_PCM,
+				      prtd->bits_per_sample);
+	}
+
+	if (ret < 0) {
+		dev_err(dev, "%s: q6asm_open_write failed\n", __func__);
+		goto open_err;
+	}
+
+	prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+	ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
+			      prtd->session_id, substream->stream);
+	if (ret) {
+		dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);
+		goto routing_err;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = q6asm_media_format_block_multi_ch_pcm(
+				prtd->audio_client, prtd->stream_id,
+				runtime->rate, runtime->channels, NULL,
+				prtd->bits_per_sample);
+	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
+							   prtd->stream_id,
+							   runtime->rate,
+							   runtime->channels,
+							   prtd->bits_per_sample);
+
+		/* Queue the buffers */
+		for (i = 0; i < runtime->periods; i++)
+			q6asm_read(prtd->audio_client, prtd->stream_id);
+
+	}
+	if (ret < 0)
+		dev_info(dev, "%s: CMD Format block failed\n", __func__);
+	else
+		prtd->state = Q6ASM_STREAM_RUNNING;
+
+	return ret;
+
+routing_err:
+	q6asm_cmd(prtd->audio_client, prtd->stream_id,  CMD_CLOSE);
+open_err:
+	q6asm_unmap_memory_regions(substream->stream, prtd->audio_client);
+	q6asm_audio_client_free(prtd->audio_client);
+	prtd->audio_client = NULL;
+
+	return ret;
+}
+
+static int q6asm_dai_trigger(struct snd_soc_component *component,
+			     struct snd_pcm_substream *substream, int cmd)
+{
+	int ret = 0;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+				       0, 0, 0);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		prtd->state = Q6ASM_STREAM_STOPPED;
+		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+				       CMD_EOS);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+				       CMD_PAUSE);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static int q6asm_dai_open(struct snd_soc_component *component,
+			  struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
+	struct q6asm_dai_rtd *prtd;
+	struct q6asm_dai_data *pdata;
+	struct device *dev = component->dev;
+	int ret = 0;
+	int stream_id;
+
+	stream_id = cpu_dai->driver->id;
+
+	pdata = snd_soc_component_get_drvdata(component);
+	if (!pdata) {
+		dev_err(dev, "Drv data not found ..\n");
+		return -EINVAL;
+	}
+
+	prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
+	if (prtd == NULL)
+		return -ENOMEM;
+
+	prtd->substream = substream;
+	prtd->audio_client = q6asm_audio_client_alloc(dev,
+				(q6asm_cb)event_handler, prtd, stream_id,
+				LEGACY_PCM_MODE);
+	if (IS_ERR(prtd->audio_client)) {
+		dev_info(dev, "%s: Could not allocate memory\n", __func__);
+		ret = PTR_ERR(prtd->audio_client);
+		kfree(prtd);
+		return ret;
+	}
+
+	/* DSP expects stream id from 1 */
+	prtd->stream_id = 1;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		runtime->hw = q6asm_dai_hardware_playback;
+	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		runtime->hw = q6asm_dai_hardware_capture;
+
+	ret = snd_pcm_hw_constraint_list(runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				&constraints_sample_rates);
+	if (ret < 0)
+		dev_info(dev, "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)
+		dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = snd_pcm_hw_constraint_minmax(runtime,
+			SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+			PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
+			PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
+		if (ret < 0) {
+			dev_err(dev, "constraint for buffer bytes min max ret = %d\n",
+				ret);
+		}
+	}
+
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+		SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+	if (ret < 0) {
+		dev_err(dev, "constraint for period bytes step ret = %d\n",
+								ret);
+	}
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+		SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+	if (ret < 0) {
+		dev_err(dev, "constraint for buffer bytes step ret = %d\n",
+								ret);
+	}
+
+	runtime->private_data = prtd;
+
+	snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
+
+	runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
+
+
+	if (pdata->sid < 0)
+		prtd->phys = substream->dma_buffer.addr;
+	else
+		prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static int q6asm_dai_close(struct snd_soc_component *component,
+			   struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+	if (prtd->audio_client) {
+		if (prtd->state)
+			q6asm_cmd(prtd->audio_client, prtd->stream_id,
+				  CMD_CLOSE);
+
+		q6asm_unmap_memory_regions(substream->stream,
+					   prtd->audio_client);
+		q6asm_audio_client_free(prtd->audio_client);
+		prtd->audio_client = NULL;
+	}
+	q6routing_stream_close(soc_prtd->dai_link->id,
+						substream->stream);
+	kfree(prtd);
+	return 0;
+}
+
+static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_soc_component *component,
+					   struct snd_pcm_substream *substream)
+{
+
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+	if (prtd->pcm_irq_pos >= prtd->pcm_size)
+		prtd->pcm_irq_pos = 0;
+
+	return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int q6asm_dai_mmap(struct snd_soc_component *component,
+			  struct snd_pcm_substream *substream,
+			  struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct device *dev = component->dev;
+
+	return dma_mmap_coherent(dev, vma,
+			runtime->dma_area, runtime->dma_addr,
+			runtime->dma_bytes);
+}
+
+static int q6asm_dai_hw_params(struct snd_soc_component *component,
+			       struct snd_pcm_substream *substream,
+			       struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+	prtd->pcm_size = params_buffer_bytes(params);
+	prtd->periods = params_periods(params);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		prtd->bits_per_sample = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		prtd->bits_per_sample = 24;
+		break;
+	}
+
+	return 0;
+}
+
+static void compress_event_handler(uint32_t opcode, uint32_t token,
+				   void *payload, void *priv)
+{
+	struct q6asm_dai_rtd *prtd = priv;
+	struct snd_compr_stream *substream = prtd->cstream;
+	unsigned long flags;
+	u32 wflags = 0;
+	uint64_t avail;
+	uint32_t bytes_written, bytes_to_write;
+	bool is_last_buffer = false;
+
+	switch (opcode) {
+	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+		spin_lock_irqsave(&prtd->lock, flags);
+		if (!prtd->bytes_sent) {
+			q6asm_stream_remove_initial_silence(prtd->audio_client,
+						    prtd->stream_id,
+						    prtd->initial_samples_drop);
+
+			q6asm_write_async(prtd->audio_client, prtd->stream_id,
+					  prtd->pcm_count, 0, 0, 0);
+			prtd->bytes_sent += prtd->pcm_count;
+		}
+
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		break;
+
+	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+		spin_lock_irqsave(&prtd->lock, flags);
+		if (prtd->notify_on_drain) {
+			if (substream->partial_drain) {
+				/*
+				 * Close old stream and make it stale, switch
+				 * the active stream now!
+				 */
+				q6asm_cmd_nowait(prtd->audio_client,
+						 prtd->stream_id,
+						 CMD_CLOSE);
+				/*
+				 * vaild stream ids start from 1, So we are
+				 * toggling this between 1 and 2.
+				 */
+				prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
+			}
+
+			snd_compr_drain_notify(prtd->cstream);
+			prtd->notify_on_drain = false;
+
+		} else {
+			prtd->state = Q6ASM_STREAM_STOPPED;
+		}
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		break;
+
+	case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
+		spin_lock_irqsave(&prtd->lock, flags);
+
+		bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
+		prtd->copied_total += bytes_written;
+		snd_compr_fragment_elapsed(substream);
+
+		if (prtd->state != Q6ASM_STREAM_RUNNING) {
+			spin_unlock_irqrestore(&prtd->lock, flags);
+			break;
+		}
+
+		avail = prtd->bytes_received - prtd->bytes_sent;
+		if (avail > prtd->pcm_count) {
+			bytes_to_write = prtd->pcm_count;
+		} else {
+			if (substream->partial_drain || prtd->notify_on_drain)
+				is_last_buffer = true;
+			bytes_to_write = avail;
+		}
+
+		if (bytes_to_write) {
+			if (substream->partial_drain && is_last_buffer) {
+				wflags |= ASM_LAST_BUFFER_FLAG;
+				q6asm_stream_remove_trailing_silence(prtd->audio_client,
+						     prtd->stream_id,
+						     prtd->trailing_samples_drop);
+			}
+
+			q6asm_write_async(prtd->audio_client, prtd->stream_id,
+					  bytes_to_write, 0, 0, wflags);
+
+			prtd->bytes_sent += bytes_to_write;
+		}
+
+		if (prtd->notify_on_drain && is_last_buffer)
+			q6asm_cmd_nowait(prtd->audio_client,
+					 prtd->stream_id, CMD_EOS);
+
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		break;
+
+	default:
+		break;
+	}
+}
+
+static int q6asm_dai_compr_open(struct snd_soc_component *component,
+				struct snd_compr_stream *stream)
+{
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct q6asm_dai_data *pdata;
+	struct device *dev = component->dev;
+	struct q6asm_dai_rtd *prtd;
+	int stream_id, size, ret;
+
+	stream_id = cpu_dai->driver->id;
+	pdata = snd_soc_component_get_drvdata(component);
+	if (!pdata) {
+		dev_err(dev, "Drv data not found ..\n");
+		return -EINVAL;
+	}
+
+	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+	if (!prtd)
+		return -ENOMEM;
+
+	/* DSP expects stream id from 1 */
+	prtd->stream_id = 1;
+
+	prtd->cstream = stream;
+	prtd->audio_client = q6asm_audio_client_alloc(dev,
+					(q6asm_cb)compress_event_handler,
+					prtd, stream_id, LEGACY_PCM_MODE);
+	if (IS_ERR(prtd->audio_client)) {
+		dev_err(dev, "Could not allocate memory\n");
+		ret = PTR_ERR(prtd->audio_client);
+		goto free_prtd;
+	}
+
+	size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
+			COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+				  &prtd->dma_buffer);
+	if (ret) {
+		dev_err(dev, "Cannot allocate buffer(s)\n");
+		goto free_client;
+	}
+
+	if (pdata->sid < 0)
+		prtd->phys = prtd->dma_buffer.addr;
+	else
+		prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
+
+	snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
+	spin_lock_init(&prtd->lock);
+	runtime->private_data = prtd;
+
+	return 0;
+
+free_client:
+	q6asm_audio_client_free(prtd->audio_client);
+free_prtd:
+	kfree(prtd);
+
+	return ret;
+}
+
+static int q6asm_dai_compr_free(struct snd_soc_component *component,
+				struct snd_compr_stream *stream)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+
+	if (prtd->audio_client) {
+		if (prtd->state) {
+			q6asm_cmd(prtd->audio_client, prtd->stream_id,
+				  CMD_CLOSE);
+			if (prtd->next_track_stream_id) {
+				q6asm_cmd(prtd->audio_client,
+					  prtd->next_track_stream_id,
+					  CMD_CLOSE);
+			}
+		}
+
+		snd_dma_free_pages(&prtd->dma_buffer);
+		q6asm_unmap_memory_regions(stream->direction,
+					   prtd->audio_client);
+		q6asm_audio_client_free(prtd->audio_client);
+		prtd->audio_client = NULL;
+	}
+	q6routing_stream_close(rtd->dai_link->id, stream->direction);
+	kfree(prtd);
+
+	return 0;
+}
+
+static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
+					      struct snd_compr_stream *stream,
+					      struct snd_codec *codec,
+					      int stream_id)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	struct q6asm_flac_cfg flac_cfg;
+	struct q6asm_wma_cfg wma_cfg;
+	struct q6asm_alac_cfg alac_cfg;
+	struct q6asm_ape_cfg ape_cfg;
+	unsigned int wma_v9 = 0;
+	struct device *dev = component->dev;
+	int ret;
+	union snd_codec_options *codec_options;
+	struct snd_dec_flac *flac;
+	struct snd_dec_wma *wma;
+	struct snd_dec_alac *alac;
+	struct snd_dec_ape *ape;
+
+	codec_options = &(prtd->codec.options);
+
+	memcpy(&prtd->codec, codec, sizeof(*codec));
+
+	switch (codec->id) {
+	case SND_AUDIOCODEC_FLAC:
+
+		memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
+		flac = &codec_options->flac_d;
+
+		flac_cfg.ch_cfg = codec->ch_in;
+		flac_cfg.sample_rate = codec->sample_rate;
+		flac_cfg.stream_info_present = 1;
+		flac_cfg.sample_size = flac->sample_size;
+		flac_cfg.min_blk_size = flac->min_blk_size;
+		flac_cfg.max_blk_size = flac->max_blk_size;
+		flac_cfg.max_frame_size = flac->max_frame_size;
+		flac_cfg.min_frame_size = flac->min_frame_size;
+
+		ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+							   stream_id,
+							   &flac_cfg);
+		if (ret < 0) {
+			dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	case SND_AUDIOCODEC_WMA:
+		wma = &codec_options->wma_d;
+
+		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+		wma_cfg.sample_rate =  codec->sample_rate;
+		wma_cfg.num_channels = codec->ch_in;
+		wma_cfg.bytes_per_sec = codec->bit_rate / 8;
+		wma_cfg.block_align = codec->align;
+		wma_cfg.bits_per_sample = prtd->bits_per_sample;
+		wma_cfg.enc_options = wma->encoder_option;
+		wma_cfg.adv_enc_options = wma->adv_encoder_option;
+		wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+		if (wma_cfg.num_channels == 1)
+			wma_cfg.channel_mask = 4; /* Mono Center */
+		else if (wma_cfg.num_channels == 2)
+			wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+		else
+			return -EINVAL;
+
+		/* check the codec profile */
+		switch (codec->profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			wma_cfg.fmtag = 0x161;
+			wma_v9 = 1;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10:
+			wma_cfg.fmtag = 0x166;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_PRO:
+			wma_cfg.fmtag = 0x162;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+			wma_cfg.fmtag = 0x163;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			wma_cfg.fmtag = 0x167;
+			break;
+
+		default:
+			dev_err(dev, "Unknown WMA profile:%x\n",
+				codec->profile);
+			return -EIO;
+		}
+
+		if (wma_v9)
+			ret = q6asm_stream_media_format_block_wma_v9(
+					prtd->audio_client, stream_id,
+					&wma_cfg);
+		else
+			ret = q6asm_stream_media_format_block_wma_v10(
+					prtd->audio_client, stream_id,
+					&wma_cfg);
+		if (ret < 0) {
+			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	case SND_AUDIOCODEC_ALAC:
+		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
+		alac = &codec_options->alac_d;
+
+		alac_cfg.sample_rate = codec->sample_rate;
+		alac_cfg.avg_bit_rate = codec->bit_rate;
+		alac_cfg.bit_depth = prtd->bits_per_sample;
+		alac_cfg.num_channels = codec->ch_in;
+
+		alac_cfg.frame_length = alac->frame_length;
+		alac_cfg.pb = alac->pb;
+		alac_cfg.mb = alac->mb;
+		alac_cfg.kb = alac->kb;
+		alac_cfg.max_run = alac->max_run;
+		alac_cfg.compatible_version = alac->compatible_version;
+		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
+
+		switch (codec->ch_in) {
+		case 1:
+			alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
+			break;
+		case 2:
+			alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
+			break;
+		}
+		ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+							   stream_id,
+							   &alac_cfg);
+		if (ret < 0) {
+			dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	case SND_AUDIOCODEC_APE:
+		memset(&ape_cfg, 0x0, sizeof(ape_cfg));
+		ape = &codec_options->ape_d;
+
+		ape_cfg.sample_rate = codec->sample_rate;
+		ape_cfg.num_channels = codec->ch_in;
+		ape_cfg.bits_per_sample = prtd->bits_per_sample;
+
+		ape_cfg.compatible_version = ape->compatible_version;
+		ape_cfg.compression_level = ape->compression_level;
+		ape_cfg.format_flags = ape->format_flags;
+		ape_cfg.blocks_per_frame = ape->blocks_per_frame;
+		ape_cfg.final_frame_blocks = ape->final_frame_blocks;
+		ape_cfg.total_frames = ape->total_frames;
+		ape_cfg.seek_table_present = ape->seek_table_present;
+
+		ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+							  stream_id,
+							  &ape_cfg);
+		if (ret < 0) {
+			dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
+				      struct snd_compr_stream *stream,
+				      struct snd_compr_params *params)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	int dir = stream->direction;
+	struct q6asm_dai_data *pdata;
+	struct device *dev = component->dev;
+	int ret;
+
+	pdata = snd_soc_component_get_drvdata(component);
+	if (!pdata)
+		return -EINVAL;
+
+	if (!prtd || !prtd->audio_client) {
+		dev_err(dev, "private data null or audio client freed\n");
+		return -EINVAL;
+	}
+
+	prtd->periods = runtime->fragments;
+	prtd->pcm_count = runtime->fragment_size;
+	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
+	prtd->bits_per_sample = 16;
+
+	if (dir == SND_COMPRESS_PLAYBACK) {
+		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
+				params->codec.profile, prtd->bits_per_sample,
+				true);
+
+		if (ret < 0) {
+			dev_err(dev, "q6asm_open_write failed\n");
+			q6asm_audio_client_free(prtd->audio_client);
+			prtd->audio_client = NULL;
+			return ret;
+		}
+	}
+
+	prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+	ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
+			      prtd->session_id, dir);
+	if (ret) {
+		dev_err(dev, "Stream reg failed ret:%d\n", ret);
+		return ret;
+	}
+
+	ret = __q6asm_dai_compr_set_codec_params(component, stream,
+						 &params->codec,
+						 prtd->stream_id);
+	if (ret) {
+		dev_err(dev, "codec param setup failed ret:%d\n", ret);
+		return ret;
+	}
+
+	ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
+				       (prtd->pcm_size / prtd->periods),
+				       prtd->periods);
+
+	if (ret < 0) {
+		dev_err(dev, "Buffer Mapping failed ret:%d\n", ret);
+		return -ENOMEM;
+	}
+
+	prtd->state = Q6ASM_STREAM_RUNNING;
+
+	return 0;
+}
+
+static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
+					struct snd_compr_stream *stream,
+					struct snd_compr_metadata *metadata)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	int ret = 0;
+
+	switch (metadata->key) {
+	case SNDRV_COMPRESS_ENCODER_PADDING:
+		prtd->trailing_samples_drop = metadata->value[0];
+		break;
+	case SNDRV_COMPRESS_ENCODER_DELAY:
+		prtd->initial_samples_drop = metadata->value[0];
+		if (prtd->next_track_stream_id) {
+			ret = q6asm_open_write(prtd->audio_client,
+					       prtd->next_track_stream_id,
+					       prtd->codec.id,
+					       prtd->codec.profile,
+					       prtd->bits_per_sample,
+				       true);
+			if (ret < 0) {
+				dev_err(component->dev, "q6asm_open_write failed\n");
+				return ret;
+			}
+			ret = __q6asm_dai_compr_set_codec_params(component, stream,
+								 &prtd->codec,
+								 prtd->next_track_stream_id);
+			if (ret < 0) {
+				dev_err(component->dev, "q6asm_open_write failed\n");
+				return ret;
+			}
+
+			ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
+						    prtd->next_track_stream_id,
+						    prtd->initial_samples_drop);
+			prtd->next_track_stream_id = 0;
+
+		}
+
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
+				   struct snd_compr_stream *stream, int cmd)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+				       0, 0, 0);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		prtd->state = Q6ASM_STREAM_STOPPED;
+		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+				       CMD_EOS);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+				       CMD_PAUSE);
+		break;
+	case SND_COMPR_TRIGGER_NEXT_TRACK:
+		prtd->next_track = true;
+		prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
+		break;
+	case SND_COMPR_TRIGGER_DRAIN:
+	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+		prtd->notify_on_drain = true;
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
+				   struct snd_compr_stream *stream,
+				   struct snd_compr_tstamp *tstamp)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&prtd->lock, flags);
+
+	tstamp->copied_total = prtd->copied_total;
+	tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
+
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	return 0;
+}
+
+static int q6asm_compr_copy(struct snd_soc_component *component,
+			    struct snd_compr_stream *stream, char __user *buf,
+			    size_t count)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	unsigned long flags;
+	u32 wflags = 0;
+	int avail, bytes_in_flight = 0;
+	void *dstn;
+	size_t copy;
+	u32 app_pointer;
+	u32 bytes_received;
+
+	bytes_received = prtd->bytes_received;
+
+	/**
+	 * Make sure that next track data pointer is aligned at 32 bit boundary
+	 * This is a Mandatory requirement from DSP data buffers alignment
+	 */
+	if (prtd->next_track)
+		bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
+
+	app_pointer = bytes_received/prtd->pcm_size;
+	app_pointer = bytes_received -  (app_pointer * prtd->pcm_size);
+	dstn = prtd->dma_buffer.area + app_pointer;
+
+	if (count < prtd->pcm_size - app_pointer) {
+		if (copy_from_user(dstn, buf, count))
+			return -EFAULT;
+	} else {
+		copy = prtd->pcm_size - app_pointer;
+		if (copy_from_user(dstn, buf, copy))
+			return -EFAULT;
+		if (copy_from_user(prtd->dma_buffer.area, buf + copy,
+				   count - copy))
+			return -EFAULT;
+	}
+
+	spin_lock_irqsave(&prtd->lock, flags);
+
+	bytes_in_flight = prtd->bytes_received - prtd->copied_total;
+
+	if (prtd->next_track) {
+		prtd->next_track = false;
+		prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
+		prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
+	}
+
+	prtd->bytes_received = bytes_received + count;
+
+	/* Kick off the data to dsp if its starving!! */
+	if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
+		uint32_t bytes_to_write = prtd->pcm_count;
+
+		avail = prtd->bytes_received - prtd->bytes_sent;
+
+		if (avail < prtd->pcm_count)
+			bytes_to_write = avail;
+
+		q6asm_write_async(prtd->audio_client, prtd->stream_id,
+				  bytes_to_write, 0, 0, wflags);
+		prtd->bytes_sent += bytes_to_write;
+	}
+
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	return count;
+}
+
+static int q6asm_dai_compr_mmap(struct snd_soc_component *component,
+				struct snd_compr_stream *stream,
+				struct vm_area_struct *vma)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	struct device *dev = component->dev;
+
+	return dma_mmap_coherent(dev, vma,
+			prtd->dma_buffer.area, prtd->dma_buffer.addr,
+			prtd->dma_buffer.bytes);
+}
+
+static int q6asm_dai_compr_get_caps(struct snd_soc_component *component,
+				    struct snd_compr_stream *stream,
+				    struct snd_compr_caps *caps)
+{
+	caps->direction = SND_COMPRESS_PLAYBACK;
+	caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
+	caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
+	caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
+	caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+	caps->num_codecs = 5;
+	caps->codecs[0] = SND_AUDIOCODEC_MP3;
+	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+	caps->codecs[2] = SND_AUDIOCODEC_WMA;
+	caps->codecs[3] = SND_AUDIOCODEC_ALAC;
+	caps->codecs[4] = SND_AUDIOCODEC_APE;
+
+	return 0;
+}
+
+static int q6asm_dai_compr_get_codec_caps(struct snd_soc_component *component,
+					  struct snd_compr_stream *stream,
+					  struct snd_compr_codec_caps *codec)
+{
+	switch (codec->codec) {
+	case SND_AUDIOCODEC_MP3:
+		*codec = q6asm_compr_caps;
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static struct snd_compress_ops q6asm_dai_compress_ops = {
+	.open		= q6asm_dai_compr_open,
+	.free		= q6asm_dai_compr_free,
+	.set_params	= q6asm_dai_compr_set_params,
+	.set_metadata	= q6asm_dai_compr_set_metadata,
+	.pointer	= q6asm_dai_compr_pointer,
+	.trigger	= q6asm_dai_compr_trigger,
+	.get_caps	= q6asm_dai_compr_get_caps,
+	.get_codec_caps	= q6asm_dai_compr_get_codec_caps,
+	.mmap		= q6asm_dai_compr_mmap,
+	.copy		= q6asm_compr_copy,
+};
+
+static int q6asm_dai_pcm_new(struct snd_soc_component *component,
+			     struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm_substream *psubstream, *csubstream;
+	struct snd_pcm *pcm = rtd->pcm;
+	struct device *dev;
+	int size, ret;
+
+	dev = component->dev;
+	size = q6asm_dai_hardware_playback.buffer_bytes_max;
+	psubstream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	if (psubstream) {
+		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+					  &psubstream->dma_buffer);
+		if (ret) {
+			dev_err(dev, "Cannot allocate buffer(s)\n");
+			return ret;
+		}
+	}
+
+	csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+	if (csubstream) {
+		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+					  &csubstream->dma_buffer);
+		if (ret) {
+			dev_err(dev, "Cannot allocate buffer(s)\n");
+			if (psubstream)
+				snd_dma_free_pages(&psubstream->dma_buffer);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static void q6asm_dai_pcm_free(struct snd_soc_component *component,
+			       struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+		substream = pcm->streams[i].substream;
+		if (substream) {
+			snd_dma_free_pages(&substream->dma_buffer);
+			substream->dma_buffer.area = NULL;
+			substream->dma_buffer.addr = 0;
+		}
+	}
+}
+
+static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_component_driver q6asm_fe_dai_component = {
+	.name		= DRV_NAME,
+	.open		= q6asm_dai_open,
+	.hw_params	= q6asm_dai_hw_params,
+	.close		= q6asm_dai_close,
+	.prepare	= q6asm_dai_prepare,
+	.trigger	= q6asm_dai_trigger,
+	.pointer	= q6asm_dai_pointer,
+	.mmap		= q6asm_dai_mmap,
+	.pcm_construct	= q6asm_dai_pcm_new,
+	.pcm_destruct	= q6asm_dai_pcm_free,
+	.compress_ops	= &q6asm_dai_compress_ops,
+	.dapm_widgets	= q6asm_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets),
+};
+
+static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
+	Q6ASM_FEDAI_DRIVER(1),
+	Q6ASM_FEDAI_DRIVER(2),
+	Q6ASM_FEDAI_DRIVER(3),
+	Q6ASM_FEDAI_DRIVER(4),
+	Q6ASM_FEDAI_DRIVER(5),
+	Q6ASM_FEDAI_DRIVER(6),
+	Q6ASM_FEDAI_DRIVER(7),
+	Q6ASM_FEDAI_DRIVER(8),
+};
+
+static int of_q6asm_parse_dai_data(struct device *dev,
+				    struct q6asm_dai_data *pdata)
+{
+	struct snd_soc_dai_driver *dai_drv;
+	struct snd_soc_pcm_stream empty_stream;
+	struct device_node *node;
+	int ret, id, dir, idx = 0;
+
+
+	pdata->num_dais = of_get_child_count(dev->of_node);
+	if (!pdata->num_dais) {
+		dev_err(dev, "No dais found in DT\n");
+		return -EINVAL;
+	}
+
+	pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
+				   GFP_KERNEL);
+	if (!pdata->dais)
+		return -ENOMEM;
+
+	memset(&empty_stream, 0, sizeof(empty_stream));
+
+	for_each_child_of_node(dev->of_node, node) {
+		ret = of_property_read_u32(node, "reg", &id);
+		if (ret || id >= MAX_SESSIONS || id < 0) {
+			dev_err(dev, "valid dai id not found:%d\n", ret);
+			continue;
+		}
+
+		dai_drv = &pdata->dais[idx++];
+		*dai_drv = q6asm_fe_dais_template[id];
+
+		ret = of_property_read_u32(node, "direction", &dir);
+		if (ret)
+			continue;
+
+		if (dir == Q6ASM_DAI_RX)
+			dai_drv->capture = empty_stream;
+		else if (dir == Q6ASM_DAI_TX)
+			dai_drv->playback = empty_stream;
+
+		if (of_property_read_bool(node, "is-compress-dai"))
+			dai_drv->compress_new = snd_soc_new_compress;
+	}
+
+	return 0;
+}
+
+static int q6asm_dai_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct device_node *node = dev->of_node;
+	struct of_phandle_args args;
+	struct q6asm_dai_data *pdata;
+	int rc;
+
+	pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
+	if (!pdata)
+		return -ENOMEM;
+
+	rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
+	if (rc < 0)
+		pdata->sid = -1;
+	else
+		pdata->sid = args.args[0] & SID_MASK_DEFAULT;
+
+	dev_set_drvdata(dev, pdata);
+
+	rc = of_q6asm_parse_dai_data(dev, pdata);
+	if (rc)
+		return rc;
+
+	return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
+					       pdata->dais, pdata->num_dais);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id q6asm_dai_device_id[] = {
+	{ .compatible = "qcom,q6asm-dais" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
+#endif
+
+static struct platform_driver q6asm_dai_platform_driver = {
+	.driver = {
+		.name = "q6asm-dai",
+		.of_match_table = of_match_ptr(q6asm_dai_device_id),
+	},
+	.probe = q6asm_dai_probe,
+};
+module_platform_driver(q6asm_dai_platform_driver);
+
+MODULE_DESCRIPTION("Q6ASM dai driver");
+MODULE_LICENSE("GPL v2");