ASoC: qdsp6-dai: add gapless support

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

3 files changed, 275 insertions, 91 deletions

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 4d642bb657cd..e637fe958c23 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -53,7 +53,7 @@ enum stream_state {
 struct q6asm_dai_rtd {
 	struct snd_pcm_substream *substream;
 	struct snd_compr_stream *cstream;
-	struct snd_compr_params codec_param;
+	struct snd_codec codec;
 	struct snd_dma_buffer dma_buffer;
 	spinlock_t lock;
 	phys_addr_t phys;
@@ -67,12 +67,15 @@ struct q6asm_dai_rtd {
 	uint16_t bits_per_sample;
 	uint16_t source; /* Encoding source bit mask */
 	struct audio_client *audio_client;
+	uint32_t next_track_stream_id;
+	bool next_track;
 	/* Active */
 	uint32_t stream_id;
 	uint16_t session_id;
 	enum stream_state state;
 	uint32_t initial_samples_drop;
 	uint32_t trailing_samples_drop;
+	bool notify_on_drain;
 };
 
 struct q6asm_dai_data {
@@ -189,7 +192,7 @@ 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);
+					  prtd->pcm_count, 0, 0, 0);
 		break;
 	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
 		prtd->state = Q6ASM_STREAM_STOPPED;
@@ -199,7 +202,7 @@ 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);
+					  prtd->pcm_count, 0, 0, 0);
 
 		break;
 		}
@@ -509,15 +512,22 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 				   void *payload, void *priv)
 {
 	struct q6asm_dai_rtd *prtd = priv;
-	struct snd_compr_stream *substream = prtd->cstream;
-	unsigned long flags;
+	struct snd_compr_stream *stream = prtd->cstream;
+	unsigned long flags = 0;
+	u32 wflags = 0;
 	uint64_t avail;
-	uint32_t bytes_written;
+	uint32_t bytes_written, bytes_to_write;
+	bool is_last_buffer = false;
 
 	switch (opcode) {
 	case ASM_CLIENT_EVENT_CMD_RUN_DONE:
 		spin_lock_irqsave(&prtd->lock, flags);
 		if (!prtd->bytes_sent) {
+			q6asm_stream_remove_initial_silence(prtd->audio_client,
+						    prtd->stream_id,
+						    prtd->initial_samples_drop);
+			avail = prtd->bytes_received - prtd->bytes_sent;
+
 			q6asm_write_async(prtd->audio_client, prtd->stream_id,
 					  prtd->pcm_count, 0, 0, 0);
 			prtd->bytes_sent += prtd->pcm_count;
@@ -527,7 +537,23 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
 		break;
 
 	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
-		prtd->state = Q6ASM_STREAM_STOPPED;
+		if (prtd->notify_on_drain) {
+			if (stream->partial_drain && prtd->next_track_stream_id) {
+				/* Close old stream and make it stale, switch
+				 * the active stream now! */
+				q6asm_cmd_nowait(prtd->audio_client,
+						 prtd->stream_id,
+						 CMD_CLOSE);
+				prtd->stream_id = prtd->next_track_stream_id;
+				prtd->next_track_stream_id = 0;
+			}
+
+			snd_compr_drain_notify(prtd->cstream);
+			prtd->notify_on_drain = false;
+
+		} else {
+			prtd->state = Q6ASM_STREAM_STOPPED;
+		}
 		break;
 
 	case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
@@ -535,20 +561,41 @@ 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);
 
-		avail = prtd->bytes_received - prtd->bytes_sent;
-		if (prtd->state != Q6ASM_STREAM_RUNNING || avail <= 0) {
+		snd_compr_fragment_elapsed(stream);
+
+		if (prtd->state != Q6ASM_STREAM_RUNNING) {
 			spin_unlock_irqrestore(&prtd->lock, flags);
 			break;
 		}
 
-		if (avail >= prtd->pcm_count) {
+		avail = prtd->bytes_received - prtd->bytes_sent;
+		if (avail > prtd->pcm_count) {
+			bytes_to_write = prtd->pcm_count;
+		} else {
+			if (stream->partial_drain || prtd->notify_on_drain)
+				is_last_buffer = true;
+			bytes_to_write = avail;
+		}
+
+		if (bytes_to_write) {
+			if (stream->partial_drain && is_last_buffer) {
+				wflags |= ASM_LAST_BUFFER_FLAG;
+				q6asm_stream_remove_trailing_silence(prtd->audio_client,
+						     prtd->stream_id,
+						     prtd->trailing_samples_drop);
+			}
+
 			q6asm_write_async(prtd->audio_client, prtd->stream_id,
-					   prtd->pcm_count, 0, 0, 0);
-			prtd->bytes_sent += prtd->pcm_count;
+					  bytes_to_write, 0, 0, wflags);
+
+			prtd->bytes_sent += bytes_to_write;
 		}
 
+		if (prtd->notify_on_drain && is_last_buffer)
+			q6asm_cmd_nowait(prtd->audio_client,
+					 prtd->stream_id, CMD_EOS);
+
 		spin_unlock_irqrestore(&prtd->lock, flags);
 		break;
 
@@ -606,7 +653,6 @@ static int q6asm_dai_compr_open(struct snd_soc_component *component,
 	else
 		prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
 
-	snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
 	spin_lock_init(&prtd->lock);
 	runtime->private_data = prtd;
 
@@ -628,9 +674,15 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
 	struct snd_soc_pcm_runtime *rtd = stream->private_data;
 
 	if (prtd->audio_client) {
-		if (prtd->state)
+		if (prtd->state) {
 			q6asm_cmd(prtd->audio_client, prtd->stream_id,
 				  CMD_CLOSE);
+			if (prtd->next_track_stream_id) {
+				q6asm_cmd(prtd->audio_client,
+					  prtd->next_track_stream_id,
+					  CMD_CLOSE);
+			}
+		}
 
 		snd_dma_free_pages(&prtd->dma_buffer);
 		q6asm_unmap_memory_regions(stream->direction,
@@ -644,21 +696,20 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
 	return 0;
 }
 
-static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
-				      struct snd_compr_stream *stream,
-				      struct snd_compr_params *params)
+static int __q6asm_dai_compr_set_codec_params(struct snd_compr_stream *stream,
+					      struct snd_codec *codec,
+					      int stream_id)
 {
 	struct snd_compr_runtime *runtime = stream->runtime;
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = stream->private_data;
-	int dir = stream->direction;
-	struct q6asm_dai_data *pdata;
+	struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	struct q6asm_flac_cfg flac_cfg;
 	struct q6asm_wma_cfg wma_cfg;
 	struct q6asm_alac_cfg alac_cfg;
 	struct q6asm_ape_cfg ape_cfg;
 	unsigned int wma_v9 = 0;
-	struct device *dev = component->dev;
+	struct device *dev = c->dev;
 	int ret;
 	union snd_codec_options *codec_options;
 	struct snd_dec_flac *flac;
@@ -666,53 +717,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 	struct snd_dec_alac *alac;
 	struct snd_dec_ape *ape;
 
-	codec_options = &(prtd->codec_param.codec.options);
-
-
-	memcpy(&prtd->codec_param, params, sizeof(*params));
-
-	pdata = snd_soc_component_get_drvdata(component);
-	if (!pdata)
-		return -EINVAL;
-
-	if (!prtd || !prtd->audio_client) {
-		dev_err(dev, "private data null or audio client freed\n");
-		return -EINVAL;
-	}
-
-	prtd->periods = runtime->fragments;
-	prtd->pcm_count = runtime->fragment_size;
-	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
-	prtd->bits_per_sample = 16;
-	if (dir == SND_COMPRESS_PLAYBACK) {
-		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
-				       params->codec.id, params->codec.profile,
-				       prtd->bits_per_sample, true);
+	codec_options = &(prtd->codec.options);
 
-		if (ret < 0) {
-			dev_err(dev, "q6asm_open_write failed\n");
-			q6asm_audio_client_free(prtd->audio_client);
-			prtd->audio_client = NULL;
-			return ret;
-		}
-	}
+	memcpy(&prtd->codec, codec, sizeof(*codec));
 
-	prtd->session_id = q6asm_get_session_id(prtd->audio_client);
-	ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
-			      prtd->session_id, dir);
-	if (ret) {
-		dev_err(dev, "Stream reg failed ret:%d\n", ret);
-		return ret;
-	}
-
-	switch (params->codec.id) {
+	switch (codec->id) {
 	case SND_AUDIOCODEC_FLAC:
 
 		memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
 		flac = &codec_options->flac_d;
 
-		flac_cfg.ch_cfg = params->codec.ch_in;
-		flac_cfg.sample_rate =  params->codec.sample_rate;
+		flac_cfg.ch_cfg = codec->ch_in;
+		flac_cfg.sample_rate =  codec->sample_rate;
 		flac_cfg.stream_info_present = 1;
 		flac_cfg.sample_size = flac->sample_size;
 		flac_cfg.min_blk_size = flac->min_blk_size;
@@ -721,7 +737,7 @@ 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,
+							   stream_id,
 							   &flac_cfg);
 		if (ret < 0) {
 			dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
@@ -734,10 +750,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 
 		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
 
-		wma_cfg.sample_rate =  params->codec.sample_rate;
-		wma_cfg.num_channels = params->codec.ch_in;
-		wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
-		wma_cfg.block_align = params->codec.align;
+		wma_cfg.sample_rate =  codec->sample_rate;
+		wma_cfg.num_channels = codec->ch_in;
+		wma_cfg.bytes_per_sec = codec->bit_rate / 8;
+		wma_cfg.block_align = codec->align;
 		wma_cfg.bits_per_sample = prtd->bits_per_sample;
 		wma_cfg.enc_options = wma->encoder_option;
 		wma_cfg.adv_enc_options = wma->adv_encoder_option;
@@ -751,7 +767,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 			return -EINVAL;
 
 		/* check the codec profile */
-		switch (params->codec.profile) {
+		switch (codec->profile) {
 		case SND_AUDIOPROFILE_WMA9:
 			wma_cfg.fmtag = 0x161;
 			wma_v9 = 1;
@@ -775,17 +791,17 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 
 		default:
 			dev_err(dev, "Unknown WMA profile:%x\n",
-				params->codec.profile);
+				codec->profile);
 			return -EIO;
 		}
 
 		if (wma_v9)
 			ret = q6asm_stream_media_format_block_wma_v9(
-					prtd->audio_client, prtd->stream_id,
+					prtd->audio_client, stream_id,
 					&wma_cfg);
 		else
 			ret = q6asm_stream_media_format_block_wma_v10(
-					prtd->audio_client, prtd->stream_id,
+					prtd->audio_client, stream_id,
 					&wma_cfg);
 		if (ret < 0) {
 			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
@@ -797,10 +813,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
 		alac = &codec_options->alac_d;
 
-		alac_cfg.sample_rate = params->codec.sample_rate;
-		alac_cfg.avg_bit_rate = params->codec.bit_rate;
+		alac_cfg.sample_rate = codec->sample_rate;
+		alac_cfg.avg_bit_rate = codec->bit_rate;
 		alac_cfg.bit_depth = prtd->bits_per_sample;
-		alac_cfg.num_channels = params->codec.ch_in;
+		alac_cfg.num_channels = codec->ch_in;
 
 		alac_cfg.frame_length = alac->frame_length;
 		alac_cfg.pb = alac->pb;
@@ -810,7 +826,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 		alac_cfg.compatible_version = alac->compatible_version;
 		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
 
-		switch (params->codec.ch_in) {
+		switch (codec->ch_in) {
 		case 1:
 			alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
 			break;
@@ -819,7 +835,7 @@ 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,
+							   stream_id,
 							   &alac_cfg);
 		if (ret < 0) {
 			dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
@@ -831,8 +847,8 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 		memset(&ape_cfg, 0x0, sizeof(ape_cfg));
 		ape = &codec_options->ape_d;
 
-		ape_cfg.sample_rate = params->codec.sample_rate;
-		ape_cfg.num_channels = params->codec.ch_in;
+		ape_cfg.sample_rate = codec->sample_rate;
+		ape_cfg.num_channels = codec->ch_in;
 		ape_cfg.bits_per_sample = prtd->bits_per_sample;
 
 		ape_cfg.compatible_version = ape->compatible_version;
@@ -844,7 +860,7 @@ 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,
+							  stream_id,
 							  &ape_cfg);
 		if (ret < 0) {
 			dev_err(dev, "APE CMD Format block failed:%d\n", ret);
@@ -856,6 +872,98 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 		break;
 	}
 
+	return 0;
+}
+
+static int q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
+					struct snd_compr_stream *stream,
+					struct snd_codec *codec)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	int ret;
+
+	ret = q6asm_open_write(prtd->audio_client, prtd->next_track_stream_id,
+			       codec->id, codec->profile, prtd->bits_per_sample,
+			       true);
+	if (ret < 0) {
+		dev_err(component->dev, "q6asm_open_write failed\n");
+		return ret;
+	}
+
+	ret = __q6asm_dai_compr_set_codec_params(stream, codec,
+						 prtd->next_track_stream_id);
+	if (ret < 0) {
+		dev_err(component->dev, "q6asm_open_write failed\n");
+		return ret;
+	}
+
+	return q6asm_stream_remove_initial_silence(prtd->audio_client,
+						   prtd->next_track_stream_id,
+						   prtd->initial_samples_drop);
+}
+
+static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
+				      struct snd_compr_stream *stream,
+				      struct snd_compr_params *params)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+	struct q6asm_dai_rtd *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+	int dir = stream->direction;
+	struct q6asm_dai_data *pdata;
+	struct device *dev = c->dev;
+	int ret;
+	union snd_codec_options *codec_options;
+
+	codec_options = &(prtd->codec.options);
+
+
+	memcpy(&prtd->codec, &params->codec, sizeof(params->codec));
+
+	pdata = snd_soc_component_get_drvdata(c);
+	if (!pdata)
+		return -EINVAL;
+
+	if (!prtd || !prtd->audio_client) {
+		dev_err(dev, "private data null or audio client freed\n");
+		return -EINVAL;
+	}
+
+	prtd->periods = runtime->fragments;
+	prtd->pcm_count = runtime->fragment_size;
+	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
+	prtd->bits_per_sample = 16;
+
+	if (dir == SND_COMPRESS_PLAYBACK) {
+		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
+				params->codec.profile, prtd->bits_per_sample,
+				true);
+
+		if (ret < 0) {
+			dev_err(dev, "q6asm_open_write failed\n");
+			q6asm_audio_client_free(prtd->audio_client);
+			prtd->audio_client = NULL;
+			return ret;
+		}
+	}
+
+	prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+	ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
+			      prtd->session_id, dir);
+	if (ret) {
+		dev_err(dev, "Stream reg failed ret:%d\n", ret);
+		return ret;
+	}
+
+	ret = __q6asm_dai_compr_set_codec_params(stream, &prtd->codec,
+						 prtd->stream_id);
+	if (ret) {
+		dev_err(dev, "codec param setup failed ret:%d\n", ret);
+		return ret;
+	}
+
 	ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
 				       (prtd->pcm_size / prtd->periods),
 				       prtd->periods);
@@ -870,7 +978,8 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 	return 0;
 }
 
-static int q6asm_dai_compr_set_metadata(struct snd_compr_stream *stream,
+static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
+					struct snd_compr_stream *stream,
 					struct snd_compr_metadata *metadata)
 {
 	struct snd_compr_runtime *runtime = stream->runtime;
@@ -916,6 +1025,17 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
 		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
 				       CMD_PAUSE);
 		break;
+	case SND_COMPR_TRIGGER_NEXT_TRACK:
+		/* Get Next stream id
+		   open it
+		 */
+		prtd->next_track = true;
+		prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
+		break;
+	case SND_COMPR_TRIGGER_DRAIN:
+	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+		prtd->notify_on_drain = true;
+		break;
 	default:
 		ret = -EINVAL;
 		break;
@@ -942,16 +1062,76 @@ static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
 	return 0;
 }
 
-static int q6asm_dai_compr_ack(struct snd_soc_component *component,
-			       struct snd_compr_stream *stream,
-			       size_t count)
+static int q6asm_compr_copy(struct snd_soc_component *component,
+                    struct snd_compr_stream *stream, char __user *buf,
+                    size_t count)
 {
 	struct snd_compr_runtime *runtime = stream->runtime;
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 	unsigned long flags;
+	u32 wflags = 0;
+	int avail, bytes_in_flight = 0;
+	void *dstn;
+	size_t copy;
+	u32 app_pointer;
+	u32 bytes_received;
+
+
+	bytes_received = prtd->bytes_received;
+
+	/**
+	 * Make sure that next track data pointer is aligned at 32 bit boundary
+	 * This is a Mandatory requirement from DSP data buffers alignment
+	 */
+	if (prtd->next_track)
+		bytes_received= ALIGN(prtd->bytes_received, prtd->pcm_count);
+
+	app_pointer = bytes_received/prtd->pcm_size;
+	app_pointer = bytes_received -  (app_pointer * prtd->pcm_size);
+	dstn = prtd->dma_buffer.area + app_pointer;
+
+	if (count < prtd->pcm_size - app_pointer) {
+		if (copy_from_user(dstn, buf, count))
+			return -EFAULT;
+	} else {
+		copy = prtd->pcm_size - app_pointer;
+		if (copy_from_user(dstn, buf, copy))
+			return -EFAULT;
+		if (copy_from_user(prtd->dma_buffer.area, buf + copy,
+				   count - copy))
+			return -EFAULT;
+	}
 
 	spin_lock_irqsave(&prtd->lock, flags);
+
+	bytes_in_flight = prtd->bytes_received - prtd->copied_total;
+
+	if (prtd->next_track) {
+		/* Adjust the bytes sent and copied as per new aligment */
+		prtd->next_track = false;
+		prtd->bytes_received= ALIGN(prtd->bytes_received, prtd->pcm_count);
+		prtd->copied_total= ALIGN(prtd->copied_total, prtd->pcm_count);
+		prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
+	}
 	prtd->bytes_received += count;
+
+	/* Kick off the data to dsp if its starving!! */
+	if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
+		uint32_t bytes_to_write = prtd->pcm_count;
+
+		avail = prtd->bytes_received - prtd->bytes_sent;
+
+		if (avail > prtd->pcm_count) {
+			bytes_to_write = prtd->pcm_count;
+		} else {
+			bytes_to_write = avail;
+		}
+
+		q6asm_write_async(prtd->audio_client, prtd->stream_id,
+				  bytes_to_write, 0, 0, wflags);
+		prtd->bytes_sent += bytes_to_write;
+	}
+
 	spin_unlock_irqrestore(&prtd->lock, flags);
 
 	return count;
@@ -1008,13 +1188,14 @@ static struct snd_compress_ops q6asm_dai_compress_ops = {
 	.open		= q6asm_dai_compr_open,
 	.free		= q6asm_dai_compr_free,
 	.set_params	= q6asm_dai_compr_set_params,
+	.set_codec_params = q6asm_dai_compr_set_codec_params,
 	.set_metadata	= q6asm_dai_compr_set_metadata,
 	.pointer	= q6asm_dai_compr_pointer,
 	.trigger	= q6asm_dai_compr_trigger,
 	.get_caps	= q6asm_dai_compr_get_caps,
 	.get_codec_caps	= q6asm_dai_compr_get_codec_caps,
 	.mmap		= q6asm_dai_compr_mmap,
-	.ack		= q6asm_dai_compr_ack,
+	.copy		= q6asm_compr_copy,
 };
 
 static int q6asm_dai_pcm_new(struct snd_soc_component *component,
@@ -1079,7 +1260,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = {
 	.mmap		= q6asm_dai_mmap,
 	.pcm_construct	= q6asm_dai_pcm_new,
 	.pcm_destruct	= q6asm_dai_pcm_free,
-	.compress_ops	= &q6asm_dai_compress_ops,
+	.compress_ops	=&q6asm_dai_compress_ops,
 };
 
 static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 891bfeb74451..c663b9308144 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -654,8 +654,8 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 			}
 			break;
 		default:
-			dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
-				result->opcode);
+			dev_err(ac->dev, "command[0x%x] not expecting rsp status [0x%x]\n",
+				result->opcode, result->status);
 			break;
 		}
 
@@ -675,6 +675,7 @@ 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);
 
@@ -686,12 +687,13 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 				goto done;
 			}
 
-			phys = port->buf[token].phys;
+			ab = &port->buf[token];
+			phys = ab->phys;
 
-			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);
+					&phys);
 				spin_unlock_irqrestore(&ac->lock, flags);
 				ret = -EINVAL;
 				goto done;
@@ -1604,7 +1606,8 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
 
 	write->flags = wflags;
 
-	port->dsp_buf++;
+	if (len)
+		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 287143454a52..34e01826e549 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -32,7 +32,7 @@ enum {
 };
 
 #define MAX_SESSIONS	8
-#define NO_TIMESTAMP    0xFF00
+#define ASM_LAST_BUFFER_FLAG		BIT(30)
 #define FORMAT_LINEAR_PCM   0x0000
 
 struct q6asm_flac_cfg {