diff mbox series

[v2,2/9] ASoC: q6asm: make commands specific to streams

Message ID 20200721165306.29082-3-srinivas.kandagatla@linaro.org (mailing list archive)
State New, archived
Headers show
Series ASoC: qdsp6: add gapless compressed audio support | expand

Commit Message

Srinivas Kandagatla July 21, 2020, 4:52 p.m. UTC 
Each ASM session can have multiple streams attached to it,
current design was to allow only one static stream id 1 per each session.
However for use-case like gapless, we would need 2 streams to open per session.

This patch converts all the q6asm apis to take stream id as argument
to allow multiple streams to open on a single session, This is useful
for gapless playback cases.

Now the dai driver can specify which stream id for each command.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 86 ++++++++++++++++++-----------
 sound/soc/qcom/qdsp6/q6asm.c     | 92 ++++++++++++++++++--------------
 sound/soc/qcom/qdsp6/q6asm.h     | 38 ++++++++-----
 3 files changed, 133 insertions(+), 83 deletions(-)

Comments

Pierre-Louis Bossart July 21, 2020, 7:31 p.m. UTC | #1 
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> index 941f3216399c..fb0488e7beb9 100644
> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -67,6 +67,8 @@ struct q6asm_dai_rtd {
>   	uint16_t bits_per_sample;
>   	uint16_t source; /* Encoding source bit mask */
>   	struct audio_client *audio_client;
> +	/* Active */

nit-pick: what does this 'Active' comment try to say? the stream_id 
seems to be used for RUN/EOS/CLOSE operations.

> +	uint32_t stream_id;
>   	uint16_t session_id;
>   	enum stream_state state;
>   };
> @@ -184,8 +186,8 @@ static void event_handler(uint32_t opcode, uint32_t token,
>   	switch (opcode) {
>   	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
>   		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
> -			q6asm_write_async(prtd->audio_client,
> -				   prtd->pcm_count, 0, 0, NO_TIMESTAMP);
> +			q6asm_write_async(prtd->audio_client, prtd->stream_id,
> +				   prtd->pcm_count, 0, 0, 0);

In the V1 review we discussed this

"
 > sound/soc/qcom/qdsp6/q6asm.h:#define NO_TIMESTAMP    0xFF00
 >
 > is the change on the previous line intentional?

May be not!

Plan is that the users of these apis will send flags directly instead of 
boiler plating this!

This change should go as part of next patch("[PATCH 04/11] ASoC: q6asm: 
use flags directly from asm-dai") which would make it much clear!
"

doesn't look like there was a change here?
Srinivas Kandagatla July 22, 2020, 8:59 a.m. UTC | #2 
Thanks Pierre for review,

On 21/07/2020 20:31, Pierre-Louis Bossart wrote:
> 
> 
> 
>> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c 
>> b/sound/soc/qcom/qdsp6/q6asm-dai.c
>> index 941f3216399c..fb0488e7beb9 100644
>> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c
>> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
>> @@ -67,6 +67,8 @@ struct q6asm_dai_rtd {
>>       uint16_t bits_per_sample;
>>       uint16_t source; /* Encoding source bit mask */
>>       struct audio_client *audio_client;
>> +    /* Active */
> 
> nit-pick: what does this 'Active' comment try to say? the stream_id 
> seems to be used for RUN/EOS/CLOSE operations.

Active mean its the active stream id which is consuming the data at the 
point in time. As we toggle stream ids between 1 and 2. This active 
stream_id is used for every command sent to dsp.

> 
>> +    uint32_t stream_id;
>>       uint16_t session_id;
>>       enum stream_state state;
>>   };
>> @@ -184,8 +186,8 @@ static void event_handler(uint32_t opcode, 
>> uint32_t token,
>>       switch (opcode) {
>>       case ASM_CLIENT_EVENT_CMD_RUN_DONE:
>>           if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
>> -            q6asm_write_async(prtd->audio_client,
>> -                   prtd->pcm_count, 0, 0, NO_TIMESTAMP);
>> +            q6asm_write_async(prtd->audio_client, prtd->stream_id,
>> +                   prtd->pcm_count, 0, 0, 0);
> 
> In the V1 review we discussed this
> 

Sorry, I missed that! will address this in next version!

> "
>  > sound/soc/qcom/qdsp6/q6asm.h:#define NO_TIMESTAMP    0xFF00
>  >
>  > is the change on the previous line intentional?
> 
> May be not!
> 
> Plan is that the users of these apis will send flags directly instead of 
> boiler plating this!
> 
> This change should go as part of next patch("[PATCH 04/11] ASoC: q6asm: 
> use flags directly from asm-dai") which would make it much clear!
> "
> 
> doesn't look like there was a change here?
> 
>
diff mbox series

Patch

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 941f3216399c..fb0488e7beb9 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -67,6 +67,8 @@  struct q6asm_dai_rtd {
 	uint16_t bits_per_sample;
 	uint16_t source; /* Encoding source bit mask */
 	struct audio_client *audio_client;
+	/* Active */
+	uint32_t stream_id;
 	uint16_t session_id;
 	enum stream_state state;
 };
@@ -184,8 +186,8 @@  static void event_handler(uint32_t opcode, uint32_t token,
 	switch (opcode) {
 	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			q6asm_write_async(prtd->audio_client,
-				   prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+			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;
@@ -194,8 +196,8 @@  static void event_handler(uint32_t opcode, uint32_t token,
 		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->pcm_count, 0, 0, NO_TIMESTAMP);
+			q6asm_write_async(prtd->audio_client, prtd->stream_id,
+					   prtd->pcm_count, 0, 0, 0);
 
 		break;
 		}
@@ -203,7 +205,7 @@  static void event_handler(uint32_t opcode, uint32_t token,
 		prtd->pcm_irq_pos += prtd->pcm_count;
 		snd_pcm_period_elapsed(substream);
 		if (prtd->state == Q6ASM_STREAM_RUNNING)
-			q6asm_read(prtd->audio_client);
+			q6asm_read(prtd->audio_client, prtd->stream_id);
 
 		break;
 	default:
@@ -236,7 +238,7 @@  static int q6asm_dai_prepare(struct snd_soc_component *component,
 	/* rate and channels are sent to audio driver */
 	if (prtd->state) {
 		/* clear the previous setup if any  */
-		q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+		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,
@@ -255,11 +257,13 @@  static int q6asm_dai_prepare(struct snd_soc_component *component,
 	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
+		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+				       FORMAT_LINEAR_PCM,
 				       0, prtd->bits_per_sample);
 	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
-				       prtd->bits_per_sample);
+		ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
+				      FORMAT_LINEAR_PCM,
+				      prtd->bits_per_sample);
 	}
 
 	if (ret < 0) {
@@ -279,17 +283,19 @@  static int q6asm_dai_prepare(struct snd_soc_component *component,
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		ret = q6asm_media_format_block_multi_ch_pcm(
-				prtd->audio_client, runtime->rate,
-				runtime->channels, NULL,
+				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,
-					runtime->rate, runtime->channels,
-					prtd->bits_per_sample);
+							   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);
+			q6asm_read(prtd->audio_client, prtd->stream_id);
 
 	}
 	if (ret < 0)
@@ -311,15 +317,18 @@  static int q6asm_dai_trigger(struct snd_soc_component *component,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+		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, CMD_EOS);
+		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, CMD_PAUSE);
+		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+				       CMD_PAUSE);
 		break;
 	default:
 		ret = -EINVAL;
@@ -364,6 +373,9 @@  static int q6asm_dai_open(struct snd_soc_component *component,
 		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)
@@ -430,7 +442,8 @@  static int q6asm_dai_close(struct snd_soc_component *component,
 
 	if (prtd->audio_client) {
 		if (prtd->state)
-			q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+			q6asm_cmd(prtd->audio_client, prtd->stream_id,
+				  CMD_CLOSE);
 
 		q6asm_unmap_memory_regions(substream->stream,
 					   prtd->audio_client);
@@ -502,8 +515,8 @@  static void compress_event_handler(uint32_t opcode, uint32_t token,
 	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
 		spin_lock_irqsave(&prtd->lock, flags);
 		if (!prtd->bytes_sent) {
-			q6asm_write_async(prtd->audio_client, prtd->pcm_count,
-					  0, 0, NO_TIMESTAMP);
+			q6asm_write_async(prtd->audio_client, prtd->stream_id,
+					  prtd->pcm_count, 0, 0, 0);
 			prtd->bytes_sent += prtd->pcm_count;
 		}
 
@@ -528,8 +541,8 @@  static void compress_event_handler(uint32_t opcode, uint32_t token,
 		avail = prtd->bytes_received - prtd->bytes_sent;
 
 		if (avail >= prtd->pcm_count) {
-			q6asm_write_async(prtd->audio_client,
-					   prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+			q6asm_write_async(prtd->audio_client, prtd->stream_id,
+					   prtd->pcm_count, 0, 0, 0);
 			prtd->bytes_sent += prtd->pcm_count;
 		}
 
@@ -563,6 +576,9 @@  static int q6asm_dai_compr_open(struct snd_soc_component *component,
 	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,
@@ -610,7 +626,8 @@  static int q6asm_dai_compr_free(struct snd_soc_component *component,
 
 	if (prtd->audio_client) {
 		if (prtd->state)
-			q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+			q6asm_cmd(prtd->audio_client, prtd->stream_id,
+				  CMD_CLOSE);
 
 		snd_dma_free_pages(&prtd->dma_buffer);
 		q6asm_unmap_memory_regions(stream->direction,
@@ -665,8 +682,9 @@  static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 	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, params->codec.id,
-				params->codec.profile, prtd->bits_per_sample);
+		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+				       params->codec.id, params->codec.profile,
+				       prtd->bits_per_sample);
 
 		if (ret < 0) {
 			dev_err(dev, "q6asm_open_write failed\n");
@@ -700,6 +718,7 @@  static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 		flac_cfg.min_frame_size = flac->min_frame_size;
 
 		ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+							   prtd->stream_id,
 							   &flac_cfg);
 		if (ret < 0) {
 			dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
@@ -759,10 +778,12 @@  static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 
 		if (wma_v9)
 			ret = q6asm_stream_media_format_block_wma_v9(
-					prtd->audio_client, &wma_cfg);
+					prtd->audio_client, prtd->stream_id,
+					&wma_cfg);
 		else
 			ret = q6asm_stream_media_format_block_wma_v10(
-					prtd->audio_client, &wma_cfg);
+					prtd->audio_client, prtd->stream_id,
+					&wma_cfg);
 		if (ret < 0) {
 			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
 			return -EIO;
@@ -795,6 +816,7 @@  static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 			break;
 		}
 		ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+							   prtd->stream_id,
 							   &alac_cfg);
 		if (ret < 0) {
 			dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
@@ -819,6 +841,7 @@  static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 		ape_cfg.seek_table_present = ape->seek_table_present;
 
 		ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+							  prtd->stream_id,
 							  &ape_cfg);
 		if (ret < 0) {
 			dev_err(dev, "APE CMD Format block failed:%d\n", ret);
@@ -855,15 +878,18 @@  static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+		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, CMD_EOS);
+		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, CMD_PAUSE);
+		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+				       CMD_PAUSE);
 		break;
 	default:
 		ret = -EINVAL;
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 51da3717a6a6..f5d1f3c2c1ec 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -270,7 +270,6 @@  struct audio_client {
 	wait_queue_head_t cmd_wait;
 	struct aprv2_ibasic_rsp_result_t result;
 	int perf_mode;
-	int stream_id;
 	struct q6asm *q6asm;
 	struct device *dev;
 };
@@ -862,8 +861,6 @@  struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
 	ac->priv = priv;
 	ac->io_mode = ASM_SYNC_IO_MODE;
 	ac->perf_mode = perf_mode;
-	/* DSP expects stream id from 1 */
-	ac->stream_id = 1;
 	ac->adev = a->adev;
 	kref_init(&ac->refcount);
 
@@ -919,8 +916,9 @@  static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
  *
  * Return: Will be an negative value on error or zero on success
  */
-int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     u32 codec_profile, uint16_t bits_per_sample)
+int q6asm_open_write(struct audio_client *ac, uint32_t stream_id,
+		     uint32_t format, u32 codec_profile,
+		     uint16_t bits_per_sample)
 {
 	struct asm_stream_cmd_open_write_v3 *open;
 	struct apr_pkt *pkt;
@@ -935,7 +933,7 @@  int q6asm_open_write(struct audio_client *ac, uint32_t format,
 
 	pkt = p;
 	open = p + APR_HDR_SIZE;
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
 
 	pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
 	open->mode_flags = 0x00;
@@ -998,8 +996,9 @@  int q6asm_open_write(struct audio_client *ac, uint32_t format,
 }
 EXPORT_SYMBOL_GPL(q6asm_open_write);
 
-static int __q6asm_run(struct audio_client *ac, uint32_t flags,
-	      uint32_t msw_ts, uint32_t lsw_ts, bool wait)
+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;
@@ -1014,7 +1013,7 @@  static int __q6asm_run(struct audio_client *ac, uint32_t flags,
 	pkt = p;
 	run = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
 
 	pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
 	run->flags = flags;
@@ -1042,10 +1041,10 @@  static int __q6asm_run(struct audio_client *ac, uint32_t flags,
  *
  * Return: Will be an negative value on error or zero on success
  */
-int q6asm_run(struct audio_client *ac, uint32_t flags,
+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, flags, msw_ts, lsw_ts, true);
+	return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, true);
 }
 EXPORT_SYMBOL_GPL(q6asm_run);
 
@@ -1053,16 +1052,17 @@  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 flags,
-	      uint32_t msw_ts, uint32_t lsw_ts)
+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, flags, msw_ts, lsw_ts, false);
+	return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, false);
 }
 EXPORT_SYMBOL_GPL(q6asm_run_nowait);
 
@@ -1070,6 +1070,7 @@  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
@@ -1078,6 +1079,7 @@  EXPORT_SYMBOL_GPL(q6asm_run_nowait);
  * 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)
@@ -1096,7 +1098,7 @@  int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 	pkt = p;
 	fmt = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	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);
@@ -1125,8 +1127,8 @@  int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 }
 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;
@@ -1142,7 +1144,7 @@  int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 	pkt = p;
 	fmt = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	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);
@@ -1163,6 +1165,7 @@  int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 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;
@@ -1178,7 +1181,7 @@  int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
 	pkt = p;
 	fmt = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	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);
@@ -1200,6 +1203,7 @@  int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
 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;
@@ -1215,7 +1219,7 @@  int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
 	pkt = p;
 	fmt = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	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);
@@ -1238,6 +1242,7 @@  int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
 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;
@@ -1253,7 +1258,7 @@  int q6asm_stream_media_format_block_alac(struct audio_client *ac,
 	pkt = p;
 	fmt = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	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);
@@ -1279,6 +1284,7 @@  int q6asm_stream_media_format_block_alac(struct audio_client *ac,
 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;
@@ -1294,7 +1300,7 @@  int q6asm_stream_media_format_block_ape(struct audio_client *ac,
 	pkt = p;
 	fmt = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	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);
@@ -1321,6 +1327,7 @@  EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
  * 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
@@ -1328,7 +1335,9 @@  EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
  * 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 rate, uint32_t channels, uint16_t bits_per_sample)
+					 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;
@@ -1344,7 +1353,7 @@  int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
 
 	pkt = p;
 	enc_cfg = p + APR_HDR_SIZE;
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+	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;
@@ -1376,10 +1385,11 @@  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)
+int q6asm_read(struct audio_client *ac, uint32_t stream_id)
 {
 	struct asm_data_cmd_read_v2 *read;
 	struct audio_port_data *port;
@@ -1400,7 +1410,7 @@  int q6asm_read(struct audio_client *ac)
 
 	spin_lock_irqsave(&ac->lock, flags);
 	port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+	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);
@@ -1428,7 +1438,7 @@  int q6asm_read(struct audio_client *ac)
 }
 EXPORT_SYMBOL_GPL(q6asm_read);
 
-static int __q6asm_open_read(struct audio_client *ac,
+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;
@@ -1444,7 +1454,7 @@  static int __q6asm_open_read(struct audio_client *ac,
 	pkt = p;
 	open = p + APR_HDR_SIZE;
 
-	q6asm_add_hdr(ac, &pkt->hdr,  pkt_size, true, ac->stream_id);
+	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;
@@ -1475,15 +1485,16 @@  static int __q6asm_open_read(struct audio_client *ac,
  * 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 format,
-			uint16_t bits_per_sample)
+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, format, bits_per_sample);
+	return __q6asm_open_read(ac, stream_id, format, bits_per_sample);
 }
 EXPORT_SYMBOL_GPL(q6asm_open_read);
 
@@ -1491,6 +1502,7 @@  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
@@ -1498,8 +1510,8 @@  EXPORT_SYMBOL_GPL(q6asm_open_read);
  *
  * Return: Will be an negative value on error or zero on success
  */
-int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
-		       uint32_t lsw_ts, uint32_t wflags)
+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;
@@ -1520,7 +1532,7 @@  int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
 
 	spin_lock_irqsave(&ac->lock, flags);
 	port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
-	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
 
 	ab = &port->buf[port->dsp_buf];
 	pkt->hdr.token = port->dsp_buf;
@@ -1567,9 +1579,9 @@  static void q6asm_reset_buf_state(struct audio_client *ac)
 	spin_unlock_irqrestore(&ac->lock, flags);
 }
 
-static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+static int __q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd,
+		       bool wait)
 {
-	int stream_id = ac->stream_id;
 	struct apr_pkt pkt;
 	int rc;
 
@@ -1616,13 +1628,14 @@  static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
  * 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, int cmd)
+int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd)
 {
-	return __q6asm_cmd(ac, cmd, true);
+	return __q6asm_cmd(ac, stream_id, cmd, true);
 }
 EXPORT_SYMBOL_GPL(q6asm_cmd);
 
@@ -1630,13 +1643,14 @@  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, int cmd)
+int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd)
 {
-	return __q6asm_cmd(ac, cmd, false);
+	return __q6asm_cmd(ac, stream_id, cmd, false);
 }
 EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
 
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 38a207d6cd95..ceece124dd3d 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -93,37 +93,47 @@  struct audio_client *q6asm_audio_client_alloc(struct device *dev,
 					      q6asm_cb cb, void *priv,
 					      int session_id, int perf_mode);
 void q6asm_audio_client_free(struct audio_client *ac);
-int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
-		       uint32_t lsw_ts, uint32_t flags);
-int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     u32 codec_profile, uint16_t bits_per_sample);
-
-int q6asm_open_read(struct audio_client *ac, uint32_t format,
+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 flags);
+int q6asm_open_write(struct audio_client *ac, uint32_t stream_id,
+		     uint32_t format, u32 codec_profile,
 		     uint16_t bits_per_sample);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
+		    uint32_t format, uint16_t bits_per_sample);
 int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
-		uint32_t rate, uint32_t channels, uint16_t bits_per_sample);
-int q6asm_read(struct audio_client *ac);
+					 uint32_t stream_id, uint32_t rate,
+					 uint32_t channels,
+					 uint16_t bits_per_sample);
+
+int q6asm_read(struct audio_client *ac, uint32_t stream_id);
 
 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);
 int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+					 uint32_t stream_id,
 					 struct q6asm_flac_cfg *cfg);
 int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   uint32_t stream_id,
 					   struct q6asm_wma_cfg *cfg);
 int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    uint32_t stream_id,
 					    struct q6asm_wma_cfg *cfg);
 int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 uint32_t stream_id,
 					 struct q6asm_alac_cfg *cfg);
 int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					uint32_t stream_id,
 					struct q6asm_ape_cfg *cfg);
-int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
-	      uint32_t lsw_ts);
-int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
-		     uint32_t lsw_ts);
-int q6asm_cmd(struct audio_client *ac, int cmd);
-int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
+int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags,
+	      uint32_t msw_ts, uint32_t lsw_ts);
+int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id,
+		     uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts);
+int q6asm_cmd(struct audio_client *ac, uint32_t stream_id,  int cmd);
+int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id,  int cmd);
 int q6asm_get_session_id(struct audio_client *ac);
 int q6asm_map_memory_regions(unsigned int dir,
 			     struct audio_client *ac,