diff --git a/include/sound/q6asm.h b/include/sound/q6asm.h
index 257026e10019..c97521b59e52 100644
--- a/include/sound/q6asm.h
+++ b/include/sound/q6asm.h
@@ -71,10 +71,12 @@
 /* Enable Sample_Rate/Channel_Mode notification event from Decoder */
 #define SR_CM_NOTIFY_ENABLE	0x0004
 
-#define ASYNC_IO_MODE	0x0002
-#define SYNC_IO_MODE	0x0001
-#define NO_TIMESTAMP    0xFF00
-#define SET_TIMESTAMP   0x0000
+#define TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+#define TUN_READ_IO_MODE  0x0004 /* tunnel read write mode */
+#define ASYNC_IO_MODE	  0x0002
+#define SYNC_IO_MODE	  0x0001
+#define NO_TIMESTAMP      0xFF00
+#define SET_TIMESTAMP     0x0000
 
 #define SOFT_PAUSE_ENABLE	1
 #define SOFT_PAUSE_DISABLE	0
diff --git a/sound/soc/msm/msm-pcm-q6.c b/sound/soc/msm/msm-pcm-q6.c
index 62deb63f83de..2a31dd0ece11 100644
--- a/sound/soc/msm/msm-pcm-q6.c
+++ b/sound/soc/msm/msm-pcm-q6.c
@@ -100,6 +100,25 @@ static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
 	.mask = 0,
 };
 
+static void msm_pcm_route_event_handler(enum msm_pcm_routing_event event,
+					void *priv_data)
+{
+	struct msm_audio *prtd = priv_data;
+
+	BUG_ON(!prtd);
+
+	pr_debug("%s: event %x\n", __func__, event);
+
+	switch (event) {
+	case MSM_PCM_RT_EVT_BUF_RECFG:
+		q6asm_cmd(prtd->audio_client, CMD_PAUSE);
+		q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+		q6asm_run(prtd->audio_client, 0, 0, 0);
+	default:
+		break;
+	}
+}
+
 static void event_handler(uint32_t opcode,
 		uint32_t token, uint32_t *payload, void *priv)
 {
@@ -146,18 +165,33 @@ static void event_handler(uint32_t opcode,
 			pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
 		in_frame_info[token][0] = payload[2];
 		in_frame_info[token][1] = payload[3];
-		prtd->pcm_irq_pos += in_frame_info[token][0];
-		pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
-		if (atomic_read(&prtd->start))
-			snd_pcm_period_elapsed(substream);
-		if (atomic_read(&prtd->in_count) <= prtd->periods)
-			atomic_inc(&prtd->in_count);
-		wake_up(&the_locks.read_wait);
-		if (prtd->mmap_flag
-			&& q6asm_is_cpu_buf_avail_nolock(OUT,
+
+		/* assume data size = 0 during flushing */
+		if (in_frame_info[token][0]) {
+			prtd->pcm_irq_pos += in_frame_info[token][0];
+			pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+			if (atomic_read(&prtd->start))
+				snd_pcm_period_elapsed(substream);
+			if (atomic_read(&prtd->in_count) <= prtd->periods)
+				atomic_inc(&prtd->in_count);
+			wake_up(&the_locks.read_wait);
+			if (prtd->mmap_flag &&
+			    q6asm_is_cpu_buf_avail_nolock(OUT,
 				prtd->audio_client,
 				&size, &idx))
-			q6asm_read_nolock(prtd->audio_client);
+				q6asm_read_nolock(prtd->audio_client);
+		} else {
+			pr_debug("%s: reclaim flushed buf in_count %x\n",
+				 __func__, atomic_read(&prtd->in_count));
+			atomic_inc(&prtd->in_count);
+			if (atomic_read(&prtd->in_count) == prtd->periods) {
+				pr_info("%s: reclaimed all bufs\n", __func__);
+				if (atomic_read(&prtd->start))
+					snd_pcm_period_elapsed(substream);
+				wake_up(&the_locks.read_wait);
+			}
+		}
+
 		break;
 	}
 	case APR_BASIC_RSP_RESULT: {
@@ -642,6 +676,7 @@ static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct audio_buffer *buf;
 	int dir, ret;
 	int format = FORMAT_LINEAR_PCM;
+	struct msm_pcm_routing_evt event;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		dir = IN;
@@ -665,10 +700,13 @@ static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
 		pr_debug("%s: session ID %d\n", __func__,
 			prtd->audio_client->session);
 		prtd->session_id = prtd->audio_client->session;
-		msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
-			prtd->audio_client->perf_mode,
-			prtd->session_id, substream->stream);
-		}
+		event.event_func = msm_pcm_route_event_handler;
+		event.priv_data = (void *) prtd;
+		msm_pcm_routing_reg_phy_stream_v2(soc_prtd->dai_link->be_id,
+						  prtd->audio_client->perf_mode,
+						  prtd->session_id,
+						  substream->stream, event);
+	}
 
 	if (dir == OUT) {
 		ret = q6asm_audio_client_buf_alloc_contiguous(dir,
diff --git a/sound/soc/msm/msm-pcm-routing.c b/sound/soc/msm/msm-pcm-routing.c
index 81f8e026f959..27ac47cc6036 100644
--- a/sound/soc/msm/msm-pcm-routing.c
+++ b/sound/soc/msm/msm-pcm-routing.c
@@ -41,6 +41,12 @@ struct msm_pcm_routing_bdai_data {
 	bool perf_mode;
 };
 
+struct msm_pcm_routing_fdai_data {
+	u16 be_srate; /* track prior backend sample rate for flushing purpose */
+	int strm_id; /* ASM stream ID */
+	struct msm_pcm_routing_evt event_info;
+};
+
 #define INVALID_SESSION -1
 #define SESSION_TYPE_RX 0
 #define SESSION_TYPE_TX 1
@@ -194,24 +200,32 @@ static struct msm_pcm_routing_bdai_data msm_bedais[MSM_BACKEND_DAI_MAX] = {
 
 
 /* Track ASM playback & capture sessions of DAI */
-static int fe_dai_map[MSM_FRONTEND_DAI_MM_SIZE][2] = {
+static struct msm_pcm_routing_fdai_data
+	fe_dai_map[MSM_FRONTEND_DAI_MM_SIZE][2] = {
 	/* MULTIMEDIA1 */
-	{INVALID_SESSION, INVALID_SESSION},
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION, {NULL, NULL} } },
 	/* MULTIMEDIA2 */
-	{INVALID_SESSION, INVALID_SESSION},
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION, {NULL, NULL} } },
 	/* MULTIMEDIA3 */
-	{INVALID_SESSION, INVALID_SESSION},
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION, {NULL, NULL} } },
 	/* MULTIMEDIA4 */
-	{INVALID_SESSION, INVALID_SESSION},
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION,  {NULL, NULL} } },
 	/* MULTIMEDIA5 */
-	{INVALID_SESSION, INVALID_SESSION},
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION, {NULL, NULL} } },
 	/* MULTIMEDIA6 */
-	{INVALID_SESSION, INVALID_SESSION},
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION, {NULL, NULL} } },
 	/* MULTIMEDIA7*/
-	{INVALID_SESSION, INVALID_SESSION},
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION, {NULL, NULL} } },
 	/* MULTIMEDIA8 */
-	{INVALID_SESSION, INVALID_SESSION},
-
+	{{0, INVALID_SESSION, {NULL, NULL} },
+	{0, INVALID_SESSION, {NULL, NULL} } },
 };
 
 static uint8_t is_be_dai_extproc(int be_dai)
@@ -272,7 +286,7 @@ void msm_pcm_routing_reg_psthr_stream(int fedai_id, int dspst_id,
 
 	mutex_lock(&routing_lock);
 
-	fe_dai_map[fedai_id][session_type] = dspst_id;
+	fe_dai_map[fedai_id][session_type].strm_id = dspst_id;
 	for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
 		if (!is_be_dai_extproc(i) &&
 		   (afe_get_port_type(msm_bedais[i].port_id) == port_type) &&
@@ -318,7 +332,7 @@ void msm_pcm_routing_reg_phy_stream(int fedai_id, bool perf_mode, int dspst_id,
 	mutex_lock(&routing_lock);
 
 	payload.num_copps = 0; /* only RX needs to use payload */
-	fe_dai_map[fedai_id][session_type] = dspst_id;
+	fe_dai_map[fedai_id][session_type].strm_id = dspst_id;
 	/* re-enable EQ if active */
 	if (eq_data[fedai_id].enable)
 		msm_send_eq_values(fedai_id);
@@ -368,6 +382,19 @@ void msm_pcm_routing_reg_phy_stream(int fedai_id, bool perf_mode, int dspst_id,
 
 	mutex_unlock(&routing_lock);
 }
+void msm_pcm_routing_reg_phy_stream_v2(int fedai_id, bool perf_mode,
+				       int dspst_id, int stream_type,
+				       struct msm_pcm_routing_evt event_info)
+{
+	msm_pcm_routing_reg_phy_stream(fedai_id, perf_mode, dspst_id,
+				       stream_type);
+
+	if (stream_type == SNDRV_PCM_STREAM_PLAYBACK)
+		fe_dai_map[fedai_id][SESSION_TYPE_RX].event_info = event_info;
+	else
+		fe_dai_map[fedai_id][SESSION_TYPE_TX].event_info = event_info;
+
+}
 
 void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type)
 {
@@ -397,8 +424,8 @@ void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type)
 			adm_close(msm_bedais[i].port_id);
 	}
 
-	fe_dai_map[fedai_id][session_type] = INVALID_SESSION;
-
+	fe_dai_map[fedai_id][session_type].strm_id = INVALID_SESSION;
+	fe_dai_map[fedai_id][session_type].be_srate = 0;
 	mutex_unlock(&routing_lock);
 }
 
@@ -423,6 +450,7 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
 {
 	int session_type, path_type;
 	u32 channels;
+	struct msm_pcm_routing_fdai_data *fdai;
 
 	pr_debug("%s: reg %x val %x set %x\n", __func__, reg, val, set);
 
@@ -449,11 +477,24 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
 			voc_start_playback(set);
 
 		set_bit(val, &msm_bedais[reg].fe_sessions);
-		if (msm_bedais[reg].active && fe_dai_map[val][session_type] !=
+		fdai = &fe_dai_map[val][session_type];
+		if (msm_bedais[reg].active && fdai->strm_id !=
 			INVALID_SESSION) {
 
 			channels = msm_bedais[reg].channel;
 
+			if (session_type == SESSION_TYPE_TX && fdai->be_srate &&
+			    (fdai->be_srate != msm_bedais[reg].sample_rate)) {
+				pr_debug("%s: flush strm %d due diff BE rates\n",
+					__func__, fdai->strm_id);
+
+				if (fdai->event_info.event_func)
+					fdai->event_info.event_func(
+						MSM_PCM_RT_EVT_BUF_RECFG,
+						fdai->event_info.priv_data);
+				fdai->be_srate = 0; /* might not need it */
+			}
+
 			if ((session_type == SESSION_TYPE_RX) &&
 				((channels == 1) || (channels == 2))
 				&& msm_bedais[reg].perf_mode) {
@@ -481,7 +522,7 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
 
 
 			msm_pcm_routing_build_matrix(val,
-				fe_dai_map[val][session_type], path_type);
+				fdai->strm_id, path_type);
 			srs_port_id = msm_bedais[reg].port_id;
 			srs_send_params(srs_port_id, 1, 0);
 		}
@@ -490,11 +531,13 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
 			(msm_bedais[reg].port_id == VOICE_PLAYBACK_TX))
 			voc_start_playback(set);
 		clear_bit(val, &msm_bedais[reg].fe_sessions);
-		if (msm_bedais[reg].active && fe_dai_map[val][session_type] !=
+		fdai = &fe_dai_map[val][session_type];
+		if (msm_bedais[reg].active && fdai->strm_id !=
 			INVALID_SESSION) {
+			fdai->be_srate = msm_bedais[reg].sample_rate;
 			adm_close(msm_bedais[reg].port_id);
 			msm_pcm_routing_build_matrix(val,
-				fe_dai_map[val][session_type], path_type);
+				fdai->strm_id, path_type);
 		}
 	}
 	if ((msm_bedais[reg].port_id == VOICE_RECORD_RX)
@@ -970,12 +1013,12 @@ static int msm_routing_set_srs_trumedia_control_HDMI(
 static void msm_send_eq_values(int eq_idx)
 {
 	int result;
-	struct audio_client *ac =
-		q6asm_get_audio_client(fe_dai_map[eq_idx][SESSION_TYPE_RX]);
+	struct audio_client *ac = q6asm_get_audio_client(
+				fe_dai_map[eq_idx][SESSION_TYPE_RX].strm_id);
 
 	if (ac == NULL) {
 		pr_err("%s: Could not get audio client for session: %d\n",
-		      __func__, fe_dai_map[eq_idx][SESSION_TYPE_RX]);
+		      __func__, fe_dai_map[eq_idx][SESSION_TYPE_RX].strm_id);
 		goto done;
 	}
 
@@ -2640,7 +2683,9 @@ static int msm_pcm_routing_close(struct snd_pcm_substream *substream)
 	mutex_lock(&routing_lock);
 
 	for_each_set_bit(i, &bedai->fe_sessions, MSM_FRONTEND_DAI_MM_SIZE) {
-		if (fe_dai_map[i][session_type] != INVALID_SESSION) {
+		if (fe_dai_map[i][session_type].strm_id != INVALID_SESSION) {
+			fe_dai_map[i][session_type].be_srate =
+				bedai->sample_rate;
 			adm_close(bedai->port_id);
 			srs_port_id = -1;
 		}
@@ -2663,6 +2708,7 @@ static int msm_pcm_routing_prepare(struct snd_pcm_substream *substream)
 	struct msm_pcm_routing_bdai_data *bedai;
 	u32 channels;
 	bool playback, capture;
+	struct msm_pcm_routing_fdai_data *fdai;
 
 	if (be_id >= MSM_BACKEND_DAI_MAX) {
 		pr_err("%s: unexpected be_id %d\n", __func__, be_id);
@@ -2694,7 +2740,21 @@ static int msm_pcm_routing_prepare(struct snd_pcm_substream *substream)
 	capture  = substream->stream == SNDRV_PCM_STREAM_CAPTURE;
 
 	for_each_set_bit(i, &bedai->fe_sessions, MSM_FRONTEND_DAI_MM_SIZE) {
-		if (fe_dai_map[i][session_type] != INVALID_SESSION) {
+		fdai = &fe_dai_map[i][session_type];
+		if (fdai->strm_id != INVALID_SESSION) {
+			if (session_type == SESSION_TYPE_TX && fdai->be_srate &&
+			    (fdai->be_srate != bedai->sample_rate)) {
+				pr_debug("%s: flush strm %d due diff BE rates\n",
+					__func__,
+					fdai->strm_id);
+
+				if (fdai->event_info.event_func)
+					fdai->event_info.event_func(
+						MSM_PCM_RT_EVT_BUF_RECFG,
+						fdai->event_info.priv_data);
+				fdai->be_srate = 0; /* might not need it */
+			}
+
 			channels = bedai->channel;
 			if ((playback || capture)
 				&& ((channels == 2) || (channels == 1)) &&
@@ -2721,7 +2781,7 @@ static int msm_pcm_routing_prepare(struct snd_pcm_substream *substream)
 				DEFAULT_COPP_TOPOLOGY);
 
 			msm_pcm_routing_build_matrix(i,
-				fe_dai_map[i][session_type], path_type);
+				fdai->strm_id, path_type);
 			srs_port_id = bedai->port_id;
 			srs_send_params(srs_port_id, 1, 0);
 		}
diff --git a/sound/soc/msm/msm-pcm-routing.h b/sound/soc/msm/msm-pcm-routing.h
index 6be665aaed7f..b9645e703b8d 100644
--- a/sound/soc/msm/msm-pcm-routing.h
+++ b/sound/soc/msm/msm-pcm-routing.h
@@ -107,6 +107,10 @@ enum {
 	MSM_BACKEND_DAI_MAX,
 };
 
+enum msm_pcm_routing_event {
+	MSM_PCM_RT_EVT_BUF_RECFG,
+	MSM_PCM_RT_EVT_MAX,
+};
 /* dai_id: front-end ID,
  * dspst_id:  DSP audio stream ID
  * stream_type: playback or capture
@@ -116,6 +120,15 @@ void msm_pcm_routing_reg_phy_stream(int fedai_id, bool perf_mode,
 void msm_pcm_routing_reg_psthr_stream(int fedai_id, int dspst_id,
 		int stream_type, int enable);
 
+struct msm_pcm_routing_evt {
+	void (*event_func)(enum msm_pcm_routing_event, void *);
+	void *priv_data;
+};
+
+void msm_pcm_routing_reg_phy_stream_v2(int fedai_id, bool perf_mode,
+				       int dspst_id, int stream_type,
+				       struct msm_pcm_routing_evt event_info);
+
 void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type);
 
 int lpa_set_volume(unsigned volume);
diff --git a/sound/soc/msm/qdsp6/q6asm.c b/sound/soc/msm/qdsp6/q6asm.c
index 7ca7f75a3421..2cde92a22240 100644
--- a/sound/soc/msm/qdsp6/q6asm.c
+++ b/sound/soc/msm/qdsp6/q6asm.c
@@ -220,7 +220,7 @@ int q6asm_audio_client_buf_free(unsigned int dir,
 	int rc = 0;
 	pr_debug("%s: Session id %d\n", __func__, ac->session);
 	mutex_lock(&ac->cmd_lock);
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		port = &ac->port[dir];
 		if (!port->buf) {
 			mutex_unlock(&ac->cmd_lock);
@@ -351,7 +351,7 @@ void q6asm_audio_client_free(struct audio_client *ac)
 	if (!ac || !ac->session)
 		return;
 	pr_debug("%s: Session id %d\n", __func__, ac->session);
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
 			port = &ac->port[loopcnt];
 			if (!port->buf)
@@ -386,14 +386,20 @@ int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode)
 		pr_err("%s APR handle NULL\n", __func__);
 		return -EINVAL;
 	}
-	if ((mode == ASYNC_IO_MODE) || (mode == SYNC_IO_MODE)) {
-		ac->io_mode = mode;
-		pr_debug("%s:Set Mode to %d\n", __func__, ac->io_mode);
-		return 0;
+
+	if (mode == ASYNC_IO_MODE) {
+		ac->io_mode &= ~SYNC_IO_MODE;
+		ac->io_mode |= ASYNC_IO_MODE;
+	} else if (mode == SYNC_IO_MODE) {
+		ac->io_mode &= ~ASYNC_IO_MODE;
+		ac->io_mode |= SYNC_IO_MODE;
 	} else {
 		pr_err("%s:Not an valid IO Mode:%d\n", __func__, ac->io_mode);
 		return -EINVAL;
 	}
+
+	pr_debug("%s:Set Mode to %d\n", __func__, ac->io_mode);
+	return 0;
 }
 
 struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv)
@@ -497,7 +503,7 @@ int q6asm_audio_client_buf_alloc(unsigned int dir,
 	if (ac->session <= 0 || ac->session > 8)
 		goto fail;
 
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		if (ac->port[dir].buf) {
 			pr_debug("%s: buffer already allocated\n", __func__);
 			return 0;
@@ -922,7 +928,7 @@ static int32_t q6asm_callback(struct apr_client_data *data, void *priv)
 		pr_debug("%s: Rxed opcode[0x%x] status[0x%x] token[%d]",
 				__func__, payload[0], payload[1],
 				data->token);
-		if (ac->io_mode == SYNC_IO_MODE) {
+		if (ac->io_mode & SYNC_IO_MODE) {
 			if (port->buf == NULL) {
 				pr_err("%s: Unexpected Write Done\n",
 								__func__);
@@ -1004,7 +1010,7 @@ static int32_t q6asm_callback(struct apr_client_data *data, void *priv)
 		if (in_enable_flag)
 			in_cont_index++;
 #endif
-		if (ac->io_mode == SYNC_IO_MODE) {
+		if (ac->io_mode & SYNC_IO_MODE) {
 			if (port->buf == NULL) {
 				pr_err("%s: Unexpected Write Done\n", __func__);
 				return -EINVAL;
@@ -1073,7 +1079,7 @@ void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac, uint32_t *size,
 	if (!ac || ((dir != IN) && (dir != OUT)))
 		return NULL;
 
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		port = &ac->port[dir];
 
 		mutex_lock(&port->lock);
@@ -1167,7 +1173,7 @@ int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac)
 	if (!ac || (dir != OUT))
 		return ret;
 
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		port = &ac->port[dir];
 
 		mutex_lock(&port->lock);
@@ -1298,6 +1304,9 @@ int q6asm_open_read(struct audio_client *ac,
 			rc);
 		goto fail_cmd;
 	}
+
+	ac->io_mode |= TUN_READ_IO_MODE;
+
 	return 0;
 fail_cmd:
 	return -EINVAL;
@@ -1574,6 +1583,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format)
 		pr_err("%s: format = %x not supported\n", __func__, format);
 		goto fail_cmd;
 	}
+
+	ac->io_mode |= TUN_WRITE_IO_MODE;
+
 	return 0;
 fail_cmd:
 	return -EINVAL;
@@ -3322,7 +3334,7 @@ int q6asm_read(struct audio_client *ac)
 		pr_err("APR handle NULL\n");
 		return -EINVAL;
 	}
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		port = &ac->port[OUT];
 
 		q6asm_add_hdr(ac, &read.hdr, sizeof(read), FALSE);
@@ -3374,7 +3386,7 @@ int q6asm_read_nolock(struct audio_client *ac)
 		pr_err("APR handle NULL\n");
 		return -EINVAL;
 	}
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		port = &ac->port[OUT];
 
 		q6asm_add_hdr_async(ac, &read.hdr, sizeof(read), FALSE);
@@ -3398,7 +3410,7 @@ int q6asm_read_nolock(struct audio_client *ac)
 		read.hdr.token = port->dsp_buf;
 
 		port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
-		pr_debug("%s:buf add[0x%x] token[%d] uid[%d]\n", __func__,
+		pr_info("%s:buf add[0x%x] token[%d] uid[%d]\n", __func__,
 					read.buf_add,
 					read.hdr.token,
 					read.uid);
@@ -3559,7 +3571,7 @@ int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
 		return -EINVAL;
 	}
 	pr_debug("%s: session[%d] len=%d", __func__, ac->session, len);
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		port = &ac->port[IN];
 
 		q6asm_add_hdr(ac, &write.hdr, sizeof(write),
@@ -3641,7 +3653,7 @@ int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
 		return -EINVAL;
 	}
 	pr_debug("%s: session[%d] len=%d", __func__, ac->session, len);
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
 		port = &ac->port[IN];
 
 		q6asm_add_hdr_async(ac, &write.hdr, sizeof(write),
@@ -3845,9 +3857,11 @@ static void q6asm_reset_buf_state(struct audio_client *ac)
 {
 	int cnt = 0;
 	int loopcnt = 0;
+	int used;
 	struct audio_port_data *port = NULL;
 
-	if (ac->io_mode == SYNC_IO_MODE) {
+	if (ac->io_mode & SYNC_IO_MODE) {
+		used = (ac->io_mode & TUN_WRITE_IO_MODE ? 1 : 0);
 		mutex_lock(&ac->cmd_lock);
 		for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
 			port = &ac->port[loopcnt];
@@ -3857,7 +3871,7 @@ static void q6asm_reset_buf_state(struct audio_client *ac)
 			while (cnt >= 0) {
 				if (!port->buf)
 					continue;
-				port->buf[cnt].used = 1;
+				port->buf[cnt].used = used;
 				cnt--;
 			}
 		}