Adding upstream version 5.10.209.upstream/5.10.209upstream

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

1 files changed, 1760 insertions, 0 deletions

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
new file mode 100644
index 000000000..c547c560c
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -0,0 +1,1760 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/mutex.h>
+#include <linux/wait.h>
+#include <linux/module.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/device.h>
+#include <linux/of_platform.h>
+#include <linux/spinlock.h>
+#include <linux/kref.h>
+#include <linux/of.h>
+#include <uapi/sound/asound.h>
+#include <uapi/sound/compress_params.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/mm.h>
+#include "q6asm.h"
+#include "q6core.h"
+#include "q6dsp-errno.h"
+#include "q6dsp-common.h"
+
+#define ASM_STREAM_CMD_CLOSE			0x00010BCD
+#define ASM_STREAM_CMD_FLUSH			0x00010BCE
+#define ASM_SESSION_CMD_PAUSE			0x00010BD3
+#define ASM_DATA_CMD_EOS			0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS		0x00010C1C
+#define ASM_NULL_POPP_TOPOLOGY			0x00010C68
+#define ASM_STREAM_CMD_FLUSH_READBUFS		0x00010C09
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM		0x00010C10
+#define ASM_STREAM_POSTPROC_TOPO_ID_NONE	0x00010C68
+#define ASM_CMD_SHARED_MEM_MAP_REGIONS		0x00010D92
+#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS	0x00010D93
+#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS	0x00010D94
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2	0x00010D98
+#define ASM_DATA_EVENT_WRITE_DONE_V2		0x00010D99
+#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2	0x00010DA3
+#define ASM_SESSION_CMD_RUN_V2			0x00010DAA
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
+#define ASM_MEDIA_FMT_MP3			0x00010BE9
+#define ASM_MEDIA_FMT_FLAC			0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9			0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10			0x00010DA7
+#define ASM_DATA_CMD_WRITE_V2			0x00010DAB
+#define ASM_DATA_CMD_READ_V2			0x00010DAC
+#define ASM_SESSION_CMD_SUSPEND			0x00010DEC
+#define ASM_STREAM_CMD_OPEN_WRITE_V3		0x00010DB3
+#define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
+#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
+#define ASM_MEDIA_FMT_ALAC			0x00012f31
+#define ASM_MEDIA_FMT_APE			0x00012f32
+#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE	0x00010D67
+#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE	0x00010D68
+
+
+#define ASM_LEGACY_STREAM_SESSION	0
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ              29
+#define ASM_END_POINT_DEVICE_MATRIX	0
+#define ASM_DEFAULT_APP_TYPE		0
+#define ASM_SYNC_IO_MODE		0x0001
+#define ASM_ASYNC_IO_MODE		0x0002
+#define ASM_TUN_READ_IO_MODE		0x0004	/* tunnel read write mode */
+#define ASM_TUN_WRITE_IO_MODE		0x0008	/* tunnel read write mode */
+#define ASM_SHIFT_GAPLESS_MODE_FLAG	31
+#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL	3
+
+struct avs_cmd_shared_mem_map_regions {
+	u16 mem_pool_id;
+	u16 num_regions;
+	u32 property_flag;
+} __packed;
+
+struct avs_shared_map_region_payload {
+	u32 shm_addr_lsw;
+	u32 shm_addr_msw;
+	u32 mem_size_bytes;
+} __packed;
+
+struct avs_cmd_shared_mem_unmap_regions {
+	u32 mem_map_handle;
+} __packed;
+
+struct asm_data_cmd_media_fmt_update_v2 {
+	u32 fmt_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 num_channels;
+	u16 bits_per_sample;
+	u32 sample_rate;
+	u16 is_signed;
+	u16 reserved;
+	u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
+} __packed;
+
+struct asm_flac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 is_stream_info_present;
+	u16 num_channels;
+	u16 min_blk_size;
+	u16 max_blk_size;
+	u16 md5_sum[8];
+	u32 sample_rate;
+	u32 min_frame_size;
+	u32 max_frame_size;
+	u16 sample_size;
+	u16 reserved;
+} __packed;
+
+struct asm_wmastdv9_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          advanced_enc_options1;
+	u32          advanced_enc_options2;
+} __packed;
+
+struct asm_alac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+	u32                  param_id;
+	u32                  param_size;
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+	u32                  frames_per_buf;
+	u32                  enc_cfg_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+	struct asm_stream_cmd_set_encdec_param  encdec;
+	struct asm_enc_cfg_blk_param_v2	encblk;
+	uint16_t  num_channels;
+	uint16_t  bits_per_sample;
+	uint32_t  sample_rate;
+	uint16_t  is_signed;
+	uint16_t  reserved;
+	uint8_t   channel_mapping[8];
+} __packed;
+
+struct asm_data_cmd_read_v2 {
+	u32                  buf_addr_lsw;
+	u32                  buf_addr_msw;
+	u32                  mem_map_handle;
+	u32                  buf_size;
+	u32                  seq_id;
+} __packed;
+
+struct asm_data_cmd_read_v2_done {
+	u32	status;
+	u32	buf_addr_lsw;
+	u32	buf_addr_msw;
+};
+
+struct asm_stream_cmd_open_read_v3 {
+	u32                    mode_flags;
+	u32                    src_endpointype;
+	u32                    preprocopo_id;
+	u32                    enc_cfg_id;
+	u16                    bits_per_sample;
+	u16                    reserved;
+} __packed;
+
+struct asm_data_cmd_write_v2 {
+	u32 buf_addr_lsw;
+	u32 buf_addr_msw;
+	u32 mem_map_handle;
+	u32 buf_size;
+	u32 seq_id;
+	u32 timestamp_lsw;
+	u32 timestamp_msw;
+	u32 flags;
+} __packed;
+
+struct asm_stream_cmd_open_write_v3 {
+	uint32_t mode_flags;
+	uint16_t sink_endpointype;
+	uint16_t bits_per_sample;
+	uint32_t postprocopo_id;
+	uint32_t dec_fmt_id;
+} __packed;
+
+struct asm_session_cmd_run_v2 {
+	u32 flags;
+	u32 time_lsw;
+	u32 time_msw;
+} __packed;
+
+struct audio_buffer {
+	phys_addr_t phys;
+	uint32_t size;		/* size of buffer */
+};
+
+struct audio_port_data {
+	struct audio_buffer *buf;
+	uint32_t num_periods;
+	uint32_t dsp_buf;
+	uint32_t mem_map_handle;
+};
+
+struct q6asm {
+	struct apr_device *adev;
+	struct device *dev;
+	struct q6core_svc_api_info ainfo;
+	wait_queue_head_t mem_wait;
+	spinlock_t slock;
+	struct audio_client *session[MAX_SESSIONS + 1];
+};
+
+struct audio_client {
+	int session;
+	q6asm_cb cb;
+	void *priv;
+	uint32_t io_mode;
+	struct apr_device *adev;
+	struct mutex cmd_lock;
+	spinlock_t lock;
+	struct kref refcount;
+	/* idx:1 out port, 0: in port */
+	struct audio_port_data port[2];
+	wait_queue_head_t cmd_wait;
+	struct aprv2_ibasic_rsp_result_t result;
+	int perf_mode;
+	struct q6asm *q6asm;
+	struct device *dev;
+};
+
+static inline void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+				 uint32_t pkt_size, bool cmd_flg,
+				 uint32_t stream_id)
+{
+	hdr->hdr_field = APR_SEQ_CMD_HDR_FIELD;
+	hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+	hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+	hdr->pkt_size = pkt_size;
+	if (cmd_flg)
+		hdr->token = ac->session;
+}
+
+static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
+				      struct apr_pkt *pkt, uint32_t rsp_opcode)
+{
+	struct apr_hdr *hdr = &pkt->hdr;
+	int rc;
+
+	mutex_lock(&ac->cmd_lock);
+	ac->result.opcode = 0;
+	ac->result.status = 0;
+	rc = apr_send_pkt(a->adev, pkt);
+	if (rc < 0)
+		goto err;
+
+	if (rsp_opcode)
+		rc = wait_event_timeout(a->mem_wait,
+					(ac->result.opcode == hdr->opcode) ||
+					(ac->result.opcode == rsp_opcode),
+					5 * HZ);
+	else
+		rc = wait_event_timeout(a->mem_wait,
+					(ac->result.opcode == hdr->opcode),
+					5 * HZ);
+
+	if (!rc) {
+		dev_err(a->dev, "CMD %x timeout\n", hdr->opcode);
+		rc = -ETIMEDOUT;
+	} else if (ac->result.status > 0) {
+		dev_err(a->dev, "DSP returned error[%x]\n",
+			ac->result.status);
+		rc = -EINVAL;
+	}
+
+err:
+	mutex_unlock(&ac->cmd_lock);
+	return rc;
+}
+
+static int __q6asm_memory_unmap(struct audio_client *ac,
+				phys_addr_t buf_add, int dir)
+{
+	struct avs_cmd_shared_mem_unmap_regions *mem_unmap;
+	struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+	struct apr_pkt *pkt;
+	int rc, pkt_size;
+	void *p;
+
+	if (ac->port[dir].mem_map_handle == 0) {
+		dev_err(ac->dev, "invalid mem handle\n");
+		return -EINVAL;
+	}
+
+	pkt_size = APR_HDR_SIZE + sizeof(*mem_unmap);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	mem_unmap = p + APR_HDR_SIZE;
+
+	pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+	pkt->hdr.src_port = 0;
+	pkt->hdr.dest_port = 0;
+	pkt->hdr.pkt_size = pkt_size;
+	pkt->hdr.token = ((ac->session << 8) | dir);
+
+	pkt->hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
+	mem_unmap->mem_map_handle = ac->port[dir].mem_map_handle;
+
+	rc = q6asm_apr_send_session_pkt(a, ac, pkt, 0);
+	if (rc < 0) {
+		kfree(pkt);
+		return rc;
+	}
+
+	ac->port[dir].mem_map_handle = 0;
+
+	kfree(pkt);
+	return 0;
+}
+
+
+static void q6asm_audio_client_free_buf(struct audio_client *ac,
+					struct audio_port_data *port)
+{
+	unsigned long flags;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port->num_periods = 0;
+	kfree(port->buf);
+	port->buf = NULL;
+	spin_unlock_irqrestore(&ac->lock, flags);
+}
+
+/**
+ * q6asm_unmap_memory_regions() - unmap memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac)
+{
+	struct audio_port_data *port;
+	int cnt = 0;
+	int rc = 0;
+
+	port = &ac->port[dir];
+	if (!port->buf) {
+		rc = -EINVAL;
+		goto err;
+	}
+
+	cnt = port->num_periods - 1;
+	if (cnt >= 0) {
+		rc = __q6asm_memory_unmap(ac, port->buf[dir].phys, dir);
+		if (rc < 0) {
+			dev_err(ac->dev, "%s: Memory_unmap_regions failed %d\n",
+				__func__, rc);
+			goto err;
+		}
+	}
+
+	q6asm_audio_client_free_buf(ac, port);
+
+err:
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_unmap_memory_regions);
+
+static int __q6asm_memory_map_regions(struct audio_client *ac, int dir,
+				      size_t period_sz, unsigned int periods,
+				      bool is_contiguous)
+{
+	struct avs_cmd_shared_mem_map_regions *cmd = NULL;
+	struct avs_shared_map_region_payload *mregions = NULL;
+	struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+	struct audio_port_data *port = NULL;
+	struct audio_buffer *ab = NULL;
+	struct apr_pkt *pkt;
+	void *p;
+	unsigned long flags;
+	uint32_t num_regions, buf_sz;
+	int rc, i, pkt_size;
+
+	if (is_contiguous) {
+		num_regions = 1;
+		buf_sz = period_sz * periods;
+	} else {
+		buf_sz = period_sz;
+		num_regions = periods;
+	}
+
+	/* DSP expects size should be aligned to 4K */
+	buf_sz = ALIGN(buf_sz, 4096);
+
+	pkt_size = APR_HDR_SIZE + sizeof(*cmd) +
+		   (sizeof(*mregions) * num_regions);
+
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	cmd = p + APR_HDR_SIZE;
+	mregions = p + APR_HDR_SIZE +  sizeof(*cmd);
+
+	pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+	pkt->hdr.src_port = 0;
+	pkt->hdr.dest_port = 0;
+	pkt->hdr.pkt_size = pkt_size;
+	pkt->hdr.token = ((ac->session << 8) | dir);
+	pkt->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
+
+	cmd->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
+	cmd->num_regions = num_regions;
+	cmd->property_flag = 0x00;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port = &ac->port[dir];
+
+	for (i = 0; i < num_regions; i++) {
+		ab = &port->buf[i];
+		mregions->shm_addr_lsw = lower_32_bits(ab->phys);
+		mregions->shm_addr_msw = upper_32_bits(ab->phys);
+		mregions->mem_size_bytes = buf_sz;
+		++mregions;
+	}
+	spin_unlock_irqrestore(&ac->lock, flags);
+
+	rc = q6asm_apr_send_session_pkt(a, ac, pkt,
+					ASM_CMDRSP_SHARED_MEM_MAP_REGIONS);
+
+	kfree(pkt);
+
+	return rc;
+}
+
+/**
+ * q6asm_map_memory_regions() - map memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ * @phys: physcial address that needs mapping.
+ * @period_sz: audio period size
+ * @periods: number of periods
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
+			     phys_addr_t phys,
+			     size_t period_sz, unsigned int periods)
+{
+	struct audio_buffer *buf;
+	unsigned long flags;
+	int cnt;
+	int rc;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	if (ac->port[dir].buf) {
+		dev_err(ac->dev, "Buffer already allocated\n");
+		spin_unlock_irqrestore(&ac->lock, flags);
+		return 0;
+	}
+
+	buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC);
+	if (!buf) {
+		spin_unlock_irqrestore(&ac->lock, flags);
+		return -ENOMEM;
+	}
+
+
+	ac->port[dir].buf = buf;
+
+	buf[0].phys = phys;
+	buf[0].size = period_sz;
+
+	for (cnt = 1; cnt < periods; cnt++) {
+		if (period_sz > 0) {
+			buf[cnt].phys = buf[0].phys + (cnt * period_sz);
+			buf[cnt].size = period_sz;
+		}
+	}
+	ac->port[dir].num_periods = periods;
+
+	spin_unlock_irqrestore(&ac->lock, flags);
+
+	rc = __q6asm_memory_map_regions(ac, dir, period_sz, periods, 1);
+	if (rc < 0) {
+		dev_err(ac->dev, "Memory_map_regions failed\n");
+		q6asm_audio_client_free_buf(ac, &ac->port[dir]);
+	}
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_map_memory_regions);
+
+static void q6asm_audio_client_release(struct kref *ref)
+{
+	struct audio_client *ac;
+	struct q6asm *a;
+	unsigned long flags;
+
+	ac = container_of(ref, struct audio_client, refcount);
+	a = ac->q6asm;
+
+	spin_lock_irqsave(&a->slock, flags);
+	a->session[ac->session] = NULL;
+	spin_unlock_irqrestore(&a->slock, flags);
+
+	kfree(ac);
+}
+
+/**
+ * q6asm_audio_client_free() - Freee allocated audio client
+ *
+ * @ac: audio client to free
+ */
+void q6asm_audio_client_free(struct audio_client *ac)
+{
+	kref_put(&ac->refcount, q6asm_audio_client_release);
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_free);
+
+static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
+						   int session_id)
+{
+	struct audio_client *ac = NULL;
+	unsigned long flags;
+
+	spin_lock_irqsave(&a->slock, flags);
+	if ((session_id <= 0) || (session_id > MAX_SESSIONS)) {
+		dev_err(a->dev, "invalid session: %d\n", session_id);
+		goto err;
+	}
+
+	/* check for valid session */
+	if (!a->session[session_id])
+		goto err;
+	else if (a->session[session_id]->session != session_id)
+		goto err;
+
+	ac = a->session[session_id];
+	kref_get(&ac->refcount);
+err:
+	spin_unlock_irqrestore(&a->slock, flags);
+	return ac;
+}
+
+static int32_t q6asm_stream_callback(struct apr_device *adev,
+				     struct apr_resp_pkt *data,
+				     int session_id)
+{
+	struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+	struct aprv2_ibasic_rsp_result_t *result;
+	struct apr_hdr *hdr = &data->hdr;
+	struct audio_port_data *port;
+	struct audio_client *ac;
+	uint32_t client_event = 0;
+	int ret = 0;
+
+	ac = q6asm_get_audio_client(q6asm, session_id);
+	if (!ac)/* Audio client might already be freed by now */
+		return 0;
+
+	result = data->payload;
+
+	switch (hdr->opcode) {
+	case APR_BASIC_RSP_RESULT:
+		switch (result->opcode) {
+		case ASM_SESSION_CMD_PAUSE:
+			client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE;
+			break;
+		case ASM_SESSION_CMD_SUSPEND:
+			client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
+			break;
+		case ASM_STREAM_CMD_FLUSH:
+			client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
+			break;
+		case ASM_SESSION_CMD_RUN_V2:
+			client_event = ASM_CLIENT_EVENT_CMD_RUN_DONE;
+			break;
+		case ASM_STREAM_CMD_CLOSE:
+			client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
+			break;
+		case ASM_STREAM_CMD_FLUSH_READBUFS:
+			client_event = ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE;
+			break;
+		case ASM_STREAM_CMD_OPEN_WRITE_V3:
+		case ASM_STREAM_CMD_OPEN_READ_V3:
+		case ASM_STREAM_CMD_OPEN_READWRITE_V2:
+		case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+		case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+		case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:
+		case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:
+			if (result->status != 0) {
+				dev_err(ac->dev,
+					"cmd = 0x%x returned error = 0x%x\n",
+					result->opcode, result->status);
+				ac->result = *result;
+				wake_up(&ac->cmd_wait);
+				ret = 0;
+				goto done;
+			}
+			break;
+		default:
+			dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
+				result->opcode);
+			break;
+		}
+
+		ac->result = *result;
+		wake_up(&ac->cmd_wait);
+
+		if (ac->cb)
+			ac->cb(client_event, hdr->token,
+			       data->payload, ac->priv);
+
+		ret = 0;
+		goto done;
+
+	case ASM_DATA_EVENT_WRITE_DONE_V2:
+		client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
+		if (ac->io_mode & ASM_SYNC_IO_MODE) {
+			phys_addr_t phys;
+			unsigned long flags;
+			int token = hdr->token & ASM_WRITE_TOKEN_MASK;
+
+			spin_lock_irqsave(&ac->lock, flags);
+
+			port =  &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+			if (!port->buf) {
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = 0;
+				goto done;
+			}
+
+			phys = port->buf[token].phys;
+
+			if (lower_32_bits(phys) != result->opcode ||
+			    upper_32_bits(phys) != result->status) {
+				dev_err(ac->dev, "Expected addr %pa\n",
+					&port->buf[token].phys);
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = -EINVAL;
+				goto done;
+			}
+			spin_unlock_irqrestore(&ac->lock, flags);
+		}
+		break;
+	case ASM_DATA_EVENT_READ_DONE_V2:
+		client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
+		if (ac->io_mode & ASM_SYNC_IO_MODE) {
+			struct asm_data_cmd_read_v2_done *done = data->payload;
+			unsigned long flags;
+			phys_addr_t phys;
+
+			spin_lock_irqsave(&ac->lock, flags);
+			port =  &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+			if (!port->buf) {
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = 0;
+				goto done;
+			}
+
+			phys = port->buf[hdr->token].phys;
+
+			if (upper_32_bits(phys) != done->buf_addr_msw ||
+			    lower_32_bits(phys) != done->buf_addr_lsw) {
+				dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
+					&port->buf[hdr->token].phys,
+					done->buf_addr_lsw,
+					done->buf_addr_msw);
+				spin_unlock_irqrestore(&ac->lock, flags);
+				ret = -EINVAL;
+				goto done;
+			}
+			spin_unlock_irqrestore(&ac->lock, flags);
+		}
+
+		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
+		break;
+	}
+
+	if (ac->cb)
+		ac->cb(client_event, hdr->token, data->payload, ac->priv);
+
+done:
+	kref_put(&ac->refcount, q6asm_audio_client_release);
+	return ret;
+}
+
+static int q6asm_srvc_callback(struct apr_device *adev,
+			       struct apr_resp_pkt *data)
+{
+	struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+	struct aprv2_ibasic_rsp_result_t *result;
+	struct audio_port_data *port;
+	struct audio_client *ac = NULL;
+	struct apr_hdr *hdr = &data->hdr;
+	struct q6asm *a;
+	uint32_t sid = 0;
+	uint32_t dir = 0;
+	int session_id;
+
+	session_id = (hdr->dest_port >> 8) & 0xFF;
+	if (session_id)
+		return q6asm_stream_callback(adev, data, session_id);
+
+	sid = (hdr->token >> 8) & 0x0F;
+	ac = q6asm_get_audio_client(q6asm, sid);
+	if (!ac) {
+		dev_err(&adev->dev, "Audio Client not active\n");
+		return 0;
+	}
+
+	a = dev_get_drvdata(ac->dev->parent);
+	dir = (hdr->token & 0x0F);
+	port = &ac->port[dir];
+	result = data->payload;
+
+	switch (hdr->opcode) {
+	case APR_BASIC_RSP_RESULT:
+		switch (result->opcode) {
+		case ASM_CMD_SHARED_MEM_MAP_REGIONS:
+		case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+			ac->result = *result;
+			wake_up(&a->mem_wait);
+			break;
+		default:
+			dev_err(&adev->dev, "command[0x%x] not expecting rsp\n",
+				 result->opcode);
+			break;
+		}
+		goto done;
+	case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:
+		ac->result.status = 0;
+		ac->result.opcode = hdr->opcode;
+		port->mem_map_handle = result->opcode;
+		wake_up(&a->mem_wait);
+		break;
+	case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+		ac->result.opcode = hdr->opcode;
+		ac->result.status = 0;
+		port->mem_map_handle = 0;
+		wake_up(&a->mem_wait);
+		break;
+	default:
+		dev_dbg(&adev->dev, "command[0x%x]success [0x%x]\n",
+			result->opcode, result->status);
+		break;
+	}
+
+	if (ac->cb)
+		ac->cb(hdr->opcode, hdr->token, data->payload, ac->priv);
+
+done:
+	kref_put(&ac->refcount, q6asm_audio_client_release);
+
+	return 0;
+}
+
+/**
+ * q6asm_get_session_id() - get session id for audio client
+ *
+ * @c: audio client pointer
+ *
+ * Return: Will be an session id of the audio client.
+ */
+int q6asm_get_session_id(struct audio_client *c)
+{
+	return c->session;
+}
+EXPORT_SYMBOL_GPL(q6asm_get_session_id);
+
+/**
+ * q6asm_audio_client_alloc() - Allocate a new audio client
+ *
+ * @dev: Pointer to asm child device.
+ * @cb: event callback.
+ * @priv: private data associated with this client.
+ * @session_id: session id
+ * @perf_mode: performace mode for this client
+ *
+ * Return: Will be an error pointer on error or a valid audio client
+ * on success.
+ */
+struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
+					      void *priv, int session_id,
+					      int perf_mode)
+{
+	struct q6asm *a = dev_get_drvdata(dev->parent);
+	struct audio_client *ac;
+	unsigned long flags;
+
+	ac = q6asm_get_audio_client(a, session_id + 1);
+	if (ac) {
+		dev_err(dev, "Audio Client already active\n");
+		return ac;
+	}
+
+	ac = kzalloc(sizeof(*ac), GFP_KERNEL);
+	if (!ac)
+		return ERR_PTR(-ENOMEM);
+
+	spin_lock_irqsave(&a->slock, flags);
+	a->session[session_id + 1] = ac;
+	spin_unlock_irqrestore(&a->slock, flags);
+	ac->session = session_id + 1;
+	ac->cb = cb;
+	ac->dev = dev;
+	ac->q6asm = a;
+	ac->priv = priv;
+	ac->io_mode = ASM_SYNC_IO_MODE;
+	ac->perf_mode = perf_mode;
+	ac->adev = a->adev;
+	kref_init(&ac->refcount);
+
+	init_waitqueue_head(&ac->cmd_wait);
+	mutex_init(&ac->cmd_lock);
+	spin_lock_init(&ac->lock);
+
+	return ac;
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
+
+static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
+{
+	struct apr_hdr *hdr = &pkt->hdr;
+	int rc;
+
+	mutex_lock(&ac->cmd_lock);
+	ac->result.opcode = 0;
+	ac->result.status = 0;
+
+	rc = apr_send_pkt(ac->adev, pkt);
+	if (rc < 0)
+		goto err;
+
+	rc = wait_event_timeout(ac->cmd_wait,
+				(ac->result.opcode == hdr->opcode), 5 * HZ);
+	if (!rc) {
+		dev_err(ac->dev, "CMD %x timeout\n", hdr->opcode);
+		rc =  -ETIMEDOUT;
+		goto err;
+	}
+
+	if (ac->result.status > 0) {
+		dev_err(ac->dev, "DSP returned error[%x]\n",
+			ac->result.status);
+		rc = -EINVAL;
+	} else {
+		rc = 0;
+	}
+
+
+err:
+	mutex_unlock(&ac->cmd_lock);
+	return rc;
+}
+
+/**
+ * q6asm_open_write() - Open audio client for writing
+ * @ac: audio client pointer
+ * @stream_id: stream id of q6asm session
+ * @format: audio sample format
+ * @codec_profile: compressed format profile
+ * @bits_per_sample: bits per sample
+ * @is_gapless: flag to indicate if this is a gapless stream
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_write(struct audio_client *ac, uint32_t stream_id,
+		     uint32_t format, u32 codec_profile,
+		     uint16_t bits_per_sample, bool is_gapless)
+{
+	struct asm_stream_cmd_open_write_v3 *open;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*open);
+
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	open = p + APR_HDR_SIZE;
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
+	open->mode_flags = 0x00;
+	open->mode_flags |= ASM_LEGACY_STREAM_SESSION;
+	if (is_gapless)
+		open->mode_flags |= BIT(ASM_SHIFT_GAPLESS_MODE_FLAG);
+
+	/* source endpoint : matrix */
+	open->sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+	open->bits_per_sample = bits_per_sample;
+	open->postprocopo_id = ASM_NULL_POPP_TOPOLOGY;
+
+	switch (format) {
+	case SND_AUDIOCODEC_MP3:
+		open->dec_fmt_id = ASM_MEDIA_FMT_MP3;
+		break;
+	case FORMAT_LINEAR_PCM:
+		open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+		break;
+	case SND_AUDIOCODEC_FLAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
+		break;
+	case SND_AUDIOCODEC_WMA:
+		switch (codec_profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+			break;
+		case SND_AUDIOPROFILE_WMA10:
+		case SND_AUDIOPROFILE_WMA9_PRO:
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+			break;
+		default:
+			dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+				codec_profile);
+			rc = -EINVAL;
+			goto err;
+		}
+		break;
+	case SND_AUDIOCODEC_ALAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+		break;
+	case SND_AUDIOCODEC_APE:
+		open->dec_fmt_id = ASM_MEDIA_FMT_APE;
+		break;
+	default:
+		dev_err(ac->dev, "Invalid format 0x%x\n", format);
+		rc = -EINVAL;
+		goto err;
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	if (rc < 0)
+		goto err;
+
+	ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
+
+err:
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_open_write);
+
+static int __q6asm_run(struct audio_client *ac, uint32_t stream_id,
+		       uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts,
+		       bool wait)
+{
+	struct asm_session_cmd_run_v2 *run;
+	struct apr_pkt *pkt;
+	int pkt_size, rc;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*run);
+	p = kzalloc(pkt_size, GFP_ATOMIC);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	run = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+	run->flags = flags;
+	run->time_lsw = lsw_ts;
+	run->time_msw = msw_ts;
+	if (wait) {
+		rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	} else {
+		rc = apr_send_pkt(ac->adev, pkt);
+		if (rc == pkt_size)
+			rc = 0;
+	}
+
+	kfree(pkt);
+	return rc;
+}
+
+/**
+ * q6asm_run() - start the audio client
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id of q6asm session
+ * @flags: flags associated with write
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags,
+	      uint32_t msw_ts, uint32_t lsw_ts)
+{
+	return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, true);
+}
+EXPORT_SYMBOL_GPL(q6asm_run);
+
+/**
+ * q6asm_run_nowait() - start the audio client withou blocking
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ * @flags: flags associated with write
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id,
+		     uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts)
+{
+	return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, false);
+}
+EXPORT_SYMBOL_GPL(q6asm_run_nowait);
+
+/**
+ * q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @channel_map: channel map pointer
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+					  uint32_t stream_id,
+					  uint32_t rate, uint32_t channels,
+					  u8 channel_map[PCM_MAX_NUM_CHANNEL],
+					  uint16_t bits_per_sample)
+{
+	struct asm_multi_channel_pcm_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	u8 *channel_mapping;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->num_channels = channels;
+	fmt->bits_per_sample = bits_per_sample;
+	fmt->sample_rate = rate;
+	fmt->is_signed = 1;
+
+	channel_mapping = fmt->channel_mapping;
+
+	if (channel_map) {
+		memcpy(channel_mapping, channel_map, PCM_MAX_NUM_CHANNEL);
+	} else {
+		if (q6dsp_map_channels(channel_mapping, channels)) {
+			dev_err(ac->dev, " map channels failed %d\n", channels);
+			rc = -EINVAL;
+			goto err;
+		}
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+err:
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
+
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+					 uint32_t stream_id,
+					 struct q6asm_flac_cfg *cfg)
+{
+	struct asm_flac_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->is_stream_info_present = cfg->stream_info_present;
+	fmt->num_channels = cfg->ch_cfg;
+	fmt->min_blk_size = cfg->min_blk_size;
+	fmt->max_blk_size = cfg->max_blk_size;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->min_frame_size = cfg->min_frame_size;
+	fmt->max_frame_size = cfg->max_frame_size;
+	fmt->sample_size = cfg->sample_size;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   uint32_t stream_id,
+					   struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmastdv9_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->reserved = 0;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    uint32_t stream_id,
+					    struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmaprov10_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->advanced_enc_options1 = cfg->adv_enc_options;
+	fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 uint32_t stream_id,
+					 struct q6asm_alac_cfg *cfg)
+{
+	struct asm_alac_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+	fmt->frame_length = cfg->frame_length;
+	fmt->compatible_version = cfg->compatible_version;
+	fmt->bit_depth =  cfg->bit_depth;
+	fmt->num_channels = cfg->num_channels;
+	fmt->max_run = cfg->max_run;
+	fmt->max_frame_bytes = cfg->max_frame_bytes;
+	fmt->avg_bit_rate = cfg->avg_bit_rate;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->channel_layout_tag = cfg->channel_layout_tag;
+	fmt->pb = cfg->pb;
+	fmt->mb = cfg->mb;
+	fmt->kb = cfg->kb;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					uint32_t stream_id,
+					struct q6asm_ape_cfg *cfg)
+{
+	struct asm_ape_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+	fmt->compatible_version = cfg->compatible_version;
+	fmt->compression_level = cfg->compression_level;
+	fmt->format_flags = cfg->format_flags;
+	fmt->blocks_per_frame = cfg->blocks_per_frame;
+	fmt->final_frame_blocks = cfg->final_frame_blocks;
+	fmt->total_frames = cfg->total_frames;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->seek_table_present = cfg->seek_table_present;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
+static int q6asm_stream_remove_silence(struct audio_client *ac, uint32_t stream_id,
+				       uint32_t cmd,
+				       uint32_t num_samples)
+{
+	uint32_t *samples;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(uint32_t);
+	p = kzalloc(pkt_size, GFP_ATOMIC);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	samples = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = cmd;
+	*samples = num_samples;
+	rc = apr_send_pkt(ac->adev, pkt);
+	if (rc == pkt_size)
+		rc = 0;
+
+	kfree(pkt);
+
+	return rc;
+}
+
+int q6asm_stream_remove_initial_silence(struct audio_client *ac,
+					uint32_t stream_id,
+					uint32_t initial_samples)
+{
+	return q6asm_stream_remove_silence(ac, stream_id,
+					   ASM_DATA_CMD_REMOVE_INITIAL_SILENCE,
+					   initial_samples);
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_remove_initial_silence);
+
+int q6asm_stream_remove_trailing_silence(struct audio_client *ac, uint32_t stream_id,
+					 uint32_t trailing_samples)
+{
+	return q6asm_stream_remove_silence(ac, stream_id,
+				   ASM_DATA_CMD_REMOVE_TRAILING_SILENCE,
+				   trailing_samples);
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_remove_trailing_silence);
+
+/**
+ * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+					 uint32_t stream_id, uint32_t rate,
+					 uint32_t channels,
+					 uint16_t bits_per_sample)
+{
+	struct asm_multi_channel_pcm_enc_cfg_v2  *enc_cfg;
+	struct apr_pkt *pkt;
+	u8 *channel_mapping;
+	u32 frames_per_buf = 0;
+	int pkt_size, rc;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*enc_cfg);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	enc_cfg = p + APR_HDR_SIZE;
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+	pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+	enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+	enc_cfg->encdec.param_size = sizeof(*enc_cfg) - sizeof(enc_cfg->encdec);
+	enc_cfg->encblk.frames_per_buf = frames_per_buf;
+	enc_cfg->encblk.enc_cfg_blk_size  = enc_cfg->encdec.param_size -
+					sizeof(struct asm_enc_cfg_blk_param_v2);
+
+	enc_cfg->num_channels = channels;
+	enc_cfg->bits_per_sample = bits_per_sample;
+	enc_cfg->sample_rate = rate;
+	enc_cfg->is_signed = 1;
+	channel_mapping = enc_cfg->channel_mapping;
+
+	if (q6dsp_map_channels(channel_mapping, channels)) {
+		rc = -EINVAL;
+		goto err;
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+err:
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
+
+
+/**
+ * q6asm_read() - read data of period size from audio client
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_read(struct audio_client *ac, uint32_t stream_id)
+{
+	struct asm_data_cmd_read_v2 *read;
+	struct audio_port_data *port;
+	struct audio_buffer *ab;
+	struct apr_pkt *pkt;
+	unsigned long flags;
+	int pkt_size;
+	int rc = 0;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*read);
+	p = kzalloc(pkt_size, GFP_ATOMIC);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	read = p + APR_HDR_SIZE;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
+	ab = &port->buf[port->dsp_buf];
+	pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
+	read->buf_addr_lsw = lower_32_bits(ab->phys);
+	read->buf_addr_msw = upper_32_bits(ab->phys);
+	read->mem_map_handle = port->mem_map_handle;
+
+	read->buf_size = ab->size;
+	read->seq_id = port->dsp_buf;
+	pkt->hdr.token = port->dsp_buf;
+
+	port->dsp_buf++;
+
+	if (port->dsp_buf >= port->num_periods)
+		port->dsp_buf = 0;
+
+	spin_unlock_irqrestore(&ac->lock, flags);
+	rc = apr_send_pkt(ac->adev, pkt);
+	if (rc == pkt_size)
+		rc = 0;
+	else
+		pr_err("read op[0x%x]rc[%d]\n", pkt->hdr.opcode, rc);
+
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_read);
+
+static int __q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
+		uint32_t format, uint16_t bits_per_sample)
+{
+	struct asm_stream_cmd_open_read_v3 *open;
+	struct apr_pkt *pkt;
+	int pkt_size, rc;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*open);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	open = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr,  pkt_size, true, stream_id);
+	pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
+	/* Stream prio : High, provide meta info with encoded frames */
+	open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+
+	open->preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE;
+	open->bits_per_sample = bits_per_sample;
+	open->mode_flags = 0x0;
+
+	open->mode_flags |= ASM_LEGACY_STREAM_SESSION <<
+				ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
+
+	switch (format) {
+	case FORMAT_LINEAR_PCM:
+		open->mode_flags |= 0x00;
+		open->enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+		break;
+	default:
+		pr_err("Invalid format[%d]\n", format);
+	}
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+	kfree(pkt);
+	return rc;
+}
+
+/**
+ * q6asm_open_read() - Open audio client for reading
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ * @format: audio sample format
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
+		    uint32_t format, uint16_t bits_per_sample)
+{
+	return __q6asm_open_read(ac, stream_id, format, bits_per_sample);
+}
+EXPORT_SYMBOL_GPL(q6asm_open_read);
+
+/**
+ * q6asm_write_async() - non blocking write
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ * @len: length in bytes
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ * @wflags: flags associated with write
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
+		      uint32_t msw_ts, uint32_t lsw_ts, uint32_t wflags)
+{
+	struct asm_data_cmd_write_v2 *write;
+	struct audio_port_data *port;
+	struct audio_buffer *ab;
+	unsigned long flags;
+	struct apr_pkt *pkt;
+	int pkt_size;
+	int rc = 0;
+	void *p;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*write);
+	p = kzalloc(pkt_size, GFP_ATOMIC);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	write = p + APR_HDR_SIZE;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
+
+	ab = &port->buf[port->dsp_buf];
+	pkt->hdr.token = port->dsp_buf | (len << ASM_WRITE_TOKEN_LEN_SHIFT);
+	pkt->hdr.opcode = ASM_DATA_CMD_WRITE_V2;
+	write->buf_addr_lsw = lower_32_bits(ab->phys);
+	write->buf_addr_msw = upper_32_bits(ab->phys);
+	write->buf_size = len;
+	write->seq_id = port->dsp_buf;
+	write->timestamp_lsw = lsw_ts;
+	write->timestamp_msw = msw_ts;
+	write->mem_map_handle =
+	    ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
+
+	write->flags = wflags;
+
+	port->dsp_buf++;
+
+	if (port->dsp_buf >= port->num_periods)
+		port->dsp_buf = 0;
+
+	spin_unlock_irqrestore(&ac->lock, flags);
+	rc = apr_send_pkt(ac->adev, pkt);
+	if (rc == pkt_size)
+		rc = 0;
+
+	kfree(pkt);
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_write_async);
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+	struct audio_port_data *port = NULL;
+	unsigned long flags;
+
+	spin_lock_irqsave(&ac->lock, flags);
+	port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+	port->dsp_buf = 0;
+	port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+	port->dsp_buf = 0;
+	spin_unlock_irqrestore(&ac->lock, flags);
+}
+
+static int __q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd,
+		       bool wait)
+{
+	struct apr_pkt pkt;
+	int rc;
+
+	q6asm_add_hdr(ac, &pkt.hdr, APR_HDR_SIZE, true, stream_id);
+
+	switch (cmd) {
+	case CMD_PAUSE:
+		pkt.hdr.opcode = ASM_SESSION_CMD_PAUSE;
+		break;
+	case CMD_SUSPEND:
+		pkt.hdr.opcode = ASM_SESSION_CMD_SUSPEND;
+		break;
+	case CMD_FLUSH:
+		pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH;
+		break;
+	case CMD_OUT_FLUSH:
+		pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+		break;
+	case CMD_EOS:
+		pkt.hdr.opcode = ASM_DATA_CMD_EOS;
+		break;
+	case CMD_CLOSE:
+		pkt.hdr.opcode = ASM_STREAM_CMD_CLOSE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (wait)
+		rc = q6asm_ac_send_cmd_sync(ac, &pkt);
+	else
+		return apr_send_pkt(ac->adev, &pkt);
+
+	if (rc < 0)
+		return rc;
+
+	if (cmd == CMD_FLUSH)
+		q6asm_reset_buf_state(ac);
+
+	return 0;
+}
+
+/**
+ * q6asm_cmd() - run cmd on audio client
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ * @cmd: command to run on audio client.
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd)
+{
+	return __q6asm_cmd(ac, stream_id, cmd, true);
+}
+EXPORT_SYMBOL_GPL(q6asm_cmd);
+
+/**
+ * q6asm_cmd_nowait() - non blocking, run cmd on audio client
+ *
+ * @ac: audio client pointer
+ * @stream_id: stream id
+ * @cmd: command to run on audio client.
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd)
+{
+	return __q6asm_cmd(ac, stream_id, cmd, false);
+}
+EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
+
+static int q6asm_probe(struct apr_device *adev)
+{
+	struct device *dev = &adev->dev;
+	struct q6asm *q6asm;
+
+	q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
+	if (!q6asm)
+		return -ENOMEM;
+
+	q6core_get_svc_api_info(adev->svc_id, &q6asm->ainfo);
+
+	q6asm->dev = dev;
+	q6asm->adev = adev;
+	init_waitqueue_head(&q6asm->mem_wait);
+	spin_lock_init(&q6asm->slock);
+	dev_set_drvdata(dev, q6asm);
+
+	return of_platform_populate(dev->of_node, NULL, NULL, dev);
+}
+
+static int q6asm_remove(struct apr_device *adev)
+{
+	of_platform_depopulate(&adev->dev);
+
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id q6asm_device_id[]  = {
+	{ .compatible = "qcom,q6asm" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, q6asm_device_id);
+#endif
+
+static struct apr_driver qcom_q6asm_driver = {
+	.probe = q6asm_probe,
+	.remove = q6asm_remove,
+	.callback = q6asm_srvc_callback,
+	.driver = {
+		.name = "qcom-q6asm",
+		.of_match_table = of_match_ptr(q6asm_device_id),
+	},
+};
+
+module_apr_driver(qcom_q6asm_driver);
+MODULE_DESCRIPTION("Q6 Audio Stream Manager driver");
+MODULE_LICENSE("GPL v2");