[PATCH 0/8] ASoC: qcom: q6asm: few fixes and enhancements.

[PATCH 0/8] ASoC: qcom: q6asm: few fixes and enhancements.

[Srinivas Kandagatla]

While trying out gapless playback, I found few issues with the existing code
This patchset has few fixes and some enhancement to the code to handle
multiple streams per asm session and also handle buffers that are not aligned
to period sizes.

I will send Gapless support patches once compressed gapless state machine is fixed.

Thanks,
srini

Srinivas Kandagatla (8):
  ASoC: q6asm: add command opcode to timeout error report
  ASoC: q6asm: handle EOS correctly
  ASoC: q6asm: rename misleading session id variable
  ASoC: q6asm: make commands specific to streams
  ASoC: q6asm: use flags directly from asm-dai
  ASoC: q6asm: add length to write command token
  ASoC: q6asm-dai: check available buffer size before sending
  ASoC: q6asm: allow to specify buffer offset in q6asm_write

 sound/soc/qcom/qdsp6/q6asm-dai.c | 100 ++++++++++++++--------
 sound/soc/qcom/qdsp6/q6asm.c     | 138 ++++++++++++++++++-------------
 sound/soc/qcom/qdsp6/q6asm.h     |  42 ++++++----
 3 files changed, 172 insertions(+), 108 deletions(-)

-- 
2.21.0

[PATCH 1/8] ASoC: q6asm: add command opcode to timeout error report

[Srinivas Kandagatla]

Make the error reporting more useful by adding opcode to it.
Without this its almost impossible to say which command actually
timed out.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 0e0e8f7a460a..0ce70a5c7d9e 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -310,7 +310,7 @@ static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
 					5 * HZ);
 
 	if (!rc) {
-		dev_err(a->dev, "CMD timeout\n");
+		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",
@@ -890,7 +890,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
 	rc = wait_event_timeout(ac->cmd_wait,
 				(ac->result.opcode == hdr->opcode), 5 * HZ);
 	if (!rc) {
-		dev_err(ac->dev, "CMD timeout\n");
+		dev_err(ac->dev, "CMD %x timeout\n", hdr->opcode);
 		rc =  -ETIMEDOUT;
 		goto err;
 	}
-- 
2.21.0

[PATCH 2/8] ASoC: q6asm: handle EOS correctly

[Srinivas Kandagatla]

Successful send of EOS command does not indicate that EOS is actually
finished, correct event to wait EOS is finished is EOS_RENDERED event.
EOS_RENDERED means that the DSP has finished processing all the buffers
for that particular session and stream.

This patch fixes EOS handling!

Fixes: 68fd8480bb7b ("ASoC: qdsp6: q6asm: Add support to audio stream apis")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 0ce70a5c7d9e..e0983970cba9 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -25,6 +25,7 @@
 #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
@@ -622,9 +623,6 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 		case ASM_SESSION_CMD_SUSPEND:
 			client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
 			break;
-		case ASM_DATA_CMD_EOS:
-			client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
-			break;
 		case ASM_STREAM_CMD_FLUSH:
 			client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
 			break;
@@ -727,6 +725,9 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 			spin_unlock_irqrestore(&ac->lock, flags);
 		}
 
+		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
 		break;
 	}
 
-- 
2.21.0

[PATCH 3/8] ASoC: q6asm: rename misleading session id variable

[Srinivas Kandagatla]

Each q6asm session can have multiple streams, mixing usage of these
names in variable are bit misleading to reader, so rename them accordingly.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e0983970cba9..51da3717a6a6 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -828,21 +828,21 @@ EXPORT_SYMBOL_GPL(q6asm_get_session_id);
  * @dev: Pointer to asm child device.
  * @cb: event callback.
  * @priv: private data associated with this client.
- * @stream_id: stream id
+ * @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 stream_id,
+					      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, stream_id + 1);
+	ac = q6asm_get_audio_client(a, session_id + 1);
 	if (ac) {
 		dev_err(dev, "Audio Client already active\n");
 		return ac;
@@ -853,9 +853,9 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
 		return ERR_PTR(-ENOMEM);
 
 	spin_lock_irqsave(&a->slock, flags);
-	a->session[stream_id + 1] = ac;
+	a->session[session_id + 1] = ac;
 	spin_unlock_irqrestore(&a->slock, flags);
-	ac->session = stream_id + 1;
+	ac->session = session_id + 1;
 	ac->cb = cb;
 	ac->dev = dev;
 	ac->q6asm = a;
-- 
2.21.0

[PATCH 4/8] ASoC: q6asm: make commands specific to streams

[Srinivas Kandagatla]

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 asm 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(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 125af00bba53..57fb19f6c306 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:
@@ -235,7 +237,7 @@ 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,
@@ -254,11 +256,13 @@ 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) {
@@ -278,17 +282,19 @@ 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)
@@ -310,15 +316,18 @@ 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;
@@ -363,6 +372,9 @@ 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)
@@ -429,7 +441,8 @@ 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);
@@ -501,8 +514,8 @@ 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;
 		}
 
@@ -527,8 +540,8 @@ 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;
 		}
 
@@ -562,6 +575,9 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
 	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,
@@ -608,7 +624,8 @@ static int q6asm_dai_compr_free(struct snd_compr_stream *stream)
 
 	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,
@@ -663,8 +680,9 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 	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");
@@ -698,6 +716,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 		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);
@@ -757,10 +776,12 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 
 		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;
@@ -793,6 +814,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 			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);
@@ -817,6 +839,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
 		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);
@@ -852,15 +875,18 @@ static int q6asm_dai_compr_trigger(struct snd_compr_stream *stream, int cmd)
 	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,
-- 
2.21.0

[PATCH 5/8] ASoC: q6asm: use flags directly from asm-dai

[Srinivas Kandagatla]

use flags set by asm-dais directly!

This will be useful gapless case where write needs a special flag to indicate
that last buffer.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index f5d1f3c2c1ec..d6728304ce6a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -1546,10 +1546,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
 	write->mem_map_handle =
 	    ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
 
-	if (wflags == NO_TIMESTAMP)
-		write->flags = (wflags & 0x800000FF);
-	else
-		write->flags = (0x80000000 | wflags);
+	write->flags = wflags;
 
 	port->dsp_buf++;
 
-- 
2.21.0

[PATCH 6/8] ASoC: q6asm: add length to write command token

[Srinivas Kandagatla]

Add length to write command packet token so that we can track exactly
how many bytes are consumed by DSP in the command reply.

This is useful in some use-cases where the end of the file/stream
is not aligned with period size.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +++-
 sound/soc/qcom/qdsp6/q6asm.c     | 7 ++++---
 sound/soc/qcom/qdsp6/q6asm.h     | 3 +++
 3 files changed, 10 insertions(+), 4 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 57fb19f6c306..892411f81613 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -509,6 +509,7 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 	struct snd_compr_stream *substream = prtd->cstream;
 	unsigned long flags;
 	uint64_t avail;
+	uint32_t bytes_written;
 
 	switch (opcode) {
 	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
@@ -529,7 +530,8 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 	case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
 		spin_lock_irqsave(&prtd->lock, flags);
 
-		prtd->copied_total += prtd->pcm_count;
+		bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
+		prtd->copied_total += bytes_written;
 		snd_compr_fragment_elapsed(substream);
 
 		if (prtd->state != Q6ASM_STREAM_RUNNING) {
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index d6728304ce6a..205453d1c1fc 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -670,6 +670,7 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 		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);
 
@@ -681,12 +682,12 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 				goto done;
 			}
 
-			phys = port->buf[hdr->token].phys;
+			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[hdr->token].phys);
+					&port->buf[token].phys);
 				spin_unlock_irqrestore(&ac->lock, flags);
 				ret = -EINVAL;
 				goto done;
@@ -1535,7 +1536,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
 	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
 
 	ab = &port->buf[port->dsp_buf];
-	pkt->hdr.token = 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);
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index ceece124dd3d..0379580f0742 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -20,6 +20,9 @@
 #define ASM_CLIENT_EVENT_CMD_RUN_DONE		0x1008
 #define ASM_CLIENT_EVENT_DATA_WRITE_DONE	0x1009
 #define ASM_CLIENT_EVENT_DATA_READ_DONE		0x100a
+#define ASM_WRITE_TOKEN_MASK			GENMASK(15, 0)
+#define ASM_WRITE_TOKEN_LEN_MASK		GENMASK(31, 16)
+#define ASM_WRITE_TOKEN_LEN_SHIFT		16
 
 enum {
 	LEGACY_PCM_MODE = 0,
-- 
2.21.0

[PATCH 7/8] ASoC: q6asm-dai: check available buffer size before sending

[Srinivas Kandagatla]

This patch add simple check before sending data buffer to dsp which will
ensure that we are not sending any buffers which are not already available.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 892411f81613..7aba3c07810b 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -534,13 +534,12 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 		prtd->copied_total += bytes_written;
 		snd_compr_fragment_elapsed(substream);
 
-		if (prtd->state != Q6ASM_STREAM_RUNNING) {
+		avail = prtd->bytes_received - prtd->bytes_sent;
+		if (prtd->state != Q6ASM_STREAM_RUNNING || avail <= 0) {
 			spin_unlock_irqrestore(&prtd->lock, flags);
 			break;
 		}
 
-		avail = prtd->bytes_received - prtd->bytes_sent;
-
 		if (avail >= prtd->pcm_count) {
 			q6asm_write_async(prtd->audio_client, prtd->stream_id,
 					   prtd->pcm_count, 0, 0, 0);
-- 
2.21.0

[PATCH 8/8] ASoC: q6asm: allow to specify buffer offset in q6asm_write

[Srinivas Kandagatla]

In usecase like gapless, its possible that next stream/track
can start an offset where previous track ends, and this offset
may not be at period boundary.

So adding offset parameter to write function will provide ablity
to send buffers with an offset inside period boundary.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 15 +++++++++------
 sound/soc/qcom/qdsp6/q6asm.c     | 21 ++++++++++++++-------
 sound/soc/qcom/qdsp6/q6asm.h     |  3 ++-
 3 files changed, 25 insertions(+), 14 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 7aba3c07810b..113929e79407 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -187,7 +187,7 @@ static void event_handler(uint32_t opcode, uint32_t token,
 	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			q6asm_write_async(prtd->audio_client, prtd->stream_id,
-				   prtd->pcm_count, 0, 0, 0);
+				   0, prtd->pcm_count, 0, 0, 0);
 		break;
 	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
 		prtd->state = Q6ASM_STREAM_STOPPED;
@@ -197,7 +197,7 @@ static void event_handler(uint32_t opcode, uint32_t token,
 		snd_pcm_period_elapsed(substream);
 		if (prtd->state == Q6ASM_STREAM_RUNNING)
 			q6asm_write_async(prtd->audio_client, prtd->stream_id,
-					   prtd->pcm_count, 0, 0, 0);
+					  0, prtd->pcm_count, 0, 0, 0);
 
 		break;
 		}
@@ -509,14 +509,14 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 	struct snd_compr_stream *substream = prtd->cstream;
 	unsigned long flags;
 	uint64_t avail;
-	uint32_t bytes_written;
+	uint32_t bytes_written, offset;
 
 	switch (opcode) {
 	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
 		spin_lock_irqsave(&prtd->lock, flags);
 		if (!prtd->bytes_sent) {
 			q6asm_write_async(prtd->audio_client, prtd->stream_id,
-					  prtd->pcm_count, 0, 0, 0);
+					  0, prtd->pcm_count, 0, 0, 0);
 			prtd->bytes_sent += prtd->pcm_count;
 		}
 
@@ -532,7 +532,10 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 
 		bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
 		prtd->copied_total += bytes_written;
-		snd_compr_fragment_elapsed(substream);
+		offset = prtd->copied_total % prtd->pcm_count;
+
+		if (bytes_written && !offset)
+			snd_compr_fragment_elapsed(substream);
 
 		avail = prtd->bytes_received - prtd->bytes_sent;
 		if (prtd->state != Q6ASM_STREAM_RUNNING || avail <= 0) {
@@ -542,7 +545,7 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 
 		if (avail >= prtd->pcm_count) {
 			q6asm_write_async(prtd->audio_client, prtd->stream_id,
-					   prtd->pcm_count, 0, 0, 0);
+					  0, prtd->pcm_count, 0, 0, 0);
 			prtd->bytes_sent += prtd->pcm_count;
 		}
 
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 205453d1c1fc..8e3746ce4fa4 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -238,6 +238,7 @@ struct asm_session_cmd_run_v2 {
 struct audio_buffer {
 	phys_addr_t phys;
 	uint32_t size;		/* size of buffer */
+	uint32_t offset;
 };
 
 struct audio_port_data {
@@ -671,6 +672,7 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 			phys_addr_t phys;
 			unsigned long flags;
 			int token = hdr->token & ASM_WRITE_TOKEN_MASK;
+			struct audio_buffer *ab;
 
 			spin_lock_irqsave(&ac->lock, flags);
 
@@ -682,12 +684,13 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 				goto done;
 			}
 
-			phys = port->buf[token].phys;
+			ab = &port->buf[token];
+			phys = ab->phys + ab->offset;
 
-			if (lower_32_bits(phys) != result->opcode ||
+			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);
+					&ab->phys + ab->offset);
 				spin_unlock_irqrestore(&ac->lock, flags);
 				ret = -EINVAL;
 				goto done;
@@ -1504,6 +1507,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read);
  *
  * @ac: audio client pointer
  * @stream_id: stream id
+ * @offset: offset in the period buffer
  * @len: length in bytes
  * @msw_ts: timestamp msw
  * @lsw_ts: timestamp lsw
@@ -1511,8 +1515,9 @@ 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 stream_id, 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 offset, 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;
@@ -1538,7 +1543,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
 	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_lsw = lower_32_bits(ab->phys + offset);
 	write->buf_addr_msw = upper_32_bits(ab->phys);
 	write->buf_size = len;
 	write->seq_id = port->dsp_buf;
@@ -1549,7 +1554,9 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
 
 	write->flags = wflags;
 
-	port->dsp_buf++;
+	ab->offset = offset;
+	if ((offset + len) == ab->size)
+		port->dsp_buf++;
 
 	if (port->dsp_buf >= port->num_periods)
 		port->dsp_buf = 0;
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 0379580f0742..ad7e2f6f122f 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -96,7 +96,8 @@ 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 stream_id, uint32_t len,
+int q6asm_write_async(struct audio_client *ac, uint32_t stream_id,
+		      uint32_t offset, 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,
-- 
2.21.0

Re: [PATCH 0/8] ASoC: qcom: q6asm: few fixes and enhancements.

[Mark Brown]

On Thu, 11 Jun 2020 13:41:51 +0100, Srinivas Kandagatla wrote:
> While trying out gapless playback, I found few issues with the existing code
> This patchset has few fixes and some enhancement to the code to handle
> multiple streams per asm session and also handle buffers that are not aligned
> to period sizes.
> 
> I will send Gapless support patches once compressed gapless state machine is fixed.
> 
> [...]

Applied to

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next

Thanks!

[1/1] ASoC: q6asm: handle EOS correctly
      commit: 6476b60f32866be49d05e2e0163f337374c55b06

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark