Merge "ASoc: msm: qdsp6v2: Add support for compress capture"

sound/soc/msm/msm-dai-fe.c

@@ -2449,6 +2449,63 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
 		.name = "VoiceMMode2",
 		.probe = fe_dai_probe,
 	},
+	{
+		.capture = {
+			.stream_name = "MultiMedia17 Capture",
+			.aif_name = "MM_UL17",
+			.rates = (SNDRV_PCM_RATE_8000_48000|
+					SNDRV_PCM_RATE_KNOT),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE |
+				    SNDRV_PCM_FMTBIT_S24_3LE),
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min =     8000,
+			.rate_max =     48000,
+		},
+		.ops = &msm_fe_Multimedia_dai_ops,
+		.compress_new = snd_soc_new_compress,
+		.name = "MultiMedia17",
+		.probe = fe_dai_probe,
+	},
+	{
+		.capture = {
+			.stream_name = "MultiMedia18 Capture",
+			.aif_name = "MM_UL18",
+			.rates = (SNDRV_PCM_RATE_8000_48000|
+					SNDRV_PCM_RATE_KNOT),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE |
+				    SNDRV_PCM_FMTBIT_S24_3LE),
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min =     8000,
+			.rate_max =     48000,
+		},
+		.ops = &msm_fe_Multimedia_dai_ops,
+		.compress_new = snd_soc_new_compress,
+		.name = "MultiMedia18",
+		.probe = fe_dai_probe,
+	},
+	{
+		.capture = {
+			.stream_name = "MultiMedia19 Capture",
+			.aif_name = "MM_UL19",
+			.rates = (SNDRV_PCM_RATE_8000_48000|
+					SNDRV_PCM_RATE_KNOT),
+			.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+				    SNDRV_PCM_FMTBIT_S24_LE |
+				    SNDRV_PCM_FMTBIT_S24_3LE),
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min =     8000,
+			.rate_max =     48000,
+		},
+		.ops = &msm_fe_Multimedia_dai_ops,
+		.compress_new = snd_soc_new_compress,
+		.name = "MultiMedia19",
+		.probe = fe_dai_probe,
+	},
 };
 
 static int msm_fe_dai_dev_probe(struct platform_device *pdev)

sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c

@@ -127,6 +127,11 @@ struct msm_compr_audio {
 	uint64_t bytes_received; /* from userspace */
 	uint64_t bytes_sent; /* to DSP */
 
+	uint64_t received_total; /* bytes received from DSP */
+	uint64_t bytes_copied; /* to userspace */
+	uint64_t bytes_read; /* from DSP */
+	uint32_t bytes_read_offset; /* bytes read offset*/
+
 	int32_t first_buffer;
 	int32_t last_buffer;
 	int32_t partial_drain_delay;
@@ -362,6 +367,49 @@ static int msm_compr_send_buffer(struct msm_compr_audio *prtd)
 	return 0;
 }
 
+static int msm_compr_read_buffer(struct msm_compr_audio *prtd)
+{
+	int buffer_length;
+	uint64_t bytes_available;
+	uint64_t buffer_sent;
+	struct audio_aio_read_param param;
+	int ret;
+
+	if (!atomic_read(&prtd->start)) {
+		pr_err("%s: stream is not in started state\n", __func__);
+		return -EINVAL;
+	}
+
+	buffer_length = prtd->codec_param.buffer.fragment_size;
+	bytes_available = prtd->received_total - prtd->bytes_copied;
+	buffer_sent = prtd->bytes_read - prtd->bytes_copied;
+	if (buffer_sent + buffer_length > prtd->buffer_size) {
+		pr_debug(" %s : Buffer is Full bytes_available: %llu\n",
+				__func__, bytes_available);
+		return 0;
+	}
+
+	memset(&param, 0x0, sizeof(struct audio_aio_read_param));
+	param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset;
+	param.len = buffer_length;
+	param.uid = buffer_length;
+
+	pr_debug("%s: reading %d bytes from DSP byte_offset = %llu\n",
+			__func__, buffer_length, prtd->bytes_read);
+	ret = q6asm_async_read(prtd->audio_client, &param);
+	if (ret < 0) {
+		pr_err("%s: q6asm_async_read failed - %d\n",
+			__func__, ret);
+		return ret;
+	}
+	prtd->bytes_read += buffer_length;
+	prtd->bytes_read_offset += buffer_length;
+	if (prtd->bytes_read_offset >= prtd->buffer_size)
+		prtd->bytes_read_offset -= prtd->buffer_size;
+
+	return 0;
+}
+
 static void compr_event_handler(uint32_t opcode,
 		uint32_t token, uint32_t *payload, void *priv)
 {
@@ -374,6 +422,7 @@ static void compr_event_handler(uint32_t opcode,
 	int stream_id;
 	uint32_t stream_index;
 	unsigned long flags;
+	uint64_t read_size;
 
 	if (!prtd) {
 		pr_err("%s: prtd is NULL\n", __func__);
@@ -459,6 +508,32 @@ static void compr_event_handler(uint32_t opcode,
 
 		spin_unlock_irqrestore(&prtd->lock, flags);
 		break;
+
+	case ASM_DATA_EVENT_READ_DONE_V2:
+		spin_lock_irqsave(&prtd->lock, flags);
+
+		pr_debug("ASM_DATA_EVENT_READ_DONE_V2 offset %d, length %d\n",
+				 prtd->byte_offset, payload[4]);
+		/* Always assume read_size is same as fragment_size */
+		read_size = prtd->codec_param.buffer.fragment_size;
+		prtd->byte_offset += read_size;
+		prtd->received_total += read_size;
+		if (prtd->byte_offset >= prtd->buffer_size)
+			prtd->byte_offset -= prtd->buffer_size;
+
+		snd_compr_fragment_elapsed(cstream);
+
+		if (!atomic_read(&prtd->start)) {
+			pr_debug("read_done received while not started, treat as xrun");
+			atomic_set(&prtd->xrun, 1);
+			spin_unlock_irqrestore(&prtd->lock, flags);
+			break;
+		}
+		msm_compr_read_buffer(prtd);
+
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		break;
+
 	case ASM_DATA_EVENT_RENDERED_EOS:
 		spin_lock_irqsave(&prtd->lock, flags);
 		pr_debug("%s: ASM_DATA_CMDRSP_EOS token 0x%x,stream id %d\n",
@@ -511,6 +586,14 @@ static void compr_event_handler(uint32_t opcode,
 
 			/* FIXME: A state is a better way, dealing with this*/
 			spin_lock_irqsave(&prtd->lock, flags);
+
+			if (cstream->direction == SND_COMPRESS_CAPTURE) {
+				atomic_set(&prtd->start, 1);
+				msm_compr_read_buffer(prtd);
+				spin_unlock_irqrestore(&prtd->lock, flags);
+				break;
+			}
+
 			if (!prtd->bytes_sent) {
 				bytes_available = prtd->bytes_received - prtd->copied_total;
 				if (bytes_available < cstream->runtime->fragment_size) {
@@ -958,7 +1041,8 @@ static int msm_compr_init_pp_params(struct snd_compr_stream *cstream,
 	return ret;
 }
 
-static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
+static int msm_compr_configure_dsp_for_playback
+			(struct snd_compr_stream *cstream)
 {
 	struct snd_compr_runtime *runtime = cstream->runtime;
 	struct msm_compr_audio *prtd = runtime->private_data;
@@ -1096,7 +1180,103 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
 	return ret;
 }
 
-static int msm_compr_open(struct snd_compr_stream *cstream)
+static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data;
+	uint16_t bits_per_sample;
+	uint16_t sample_word_size;
+	int dir = OUT, ret = 0;
+	struct audio_client *ac = prtd->audio_client;
+	uint32_t stream_index;
+
+	switch (prtd->codec_param.codec.format) {
+	case SNDRV_PCM_FORMAT_S24_LE:
+		bits_per_sample = 24;
+		sample_word_size = 32;
+		break;
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		bits_per_sample = 24;
+		sample_word_size = 24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		bits_per_sample = 32;
+		sample_word_size = 32;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+	default:
+		bits_per_sample = 16;
+		sample_word_size = 16;
+		break;
+	}
+
+	pr_debug("%s: stream_id %d bits_per_sample %d\n",
+			__func__, ac->stream_id, bits_per_sample);
+
+	ret = q6asm_open_read_v3(prtd->audio_client, FORMAT_LINEAR_PCM,
+		bits_per_sample);
+	if (ret < 0) {
+		pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret);
+		return ret;
+	}
+
+	ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+			ac->perf_mode,
+			prtd->session_id,
+			SNDRV_PCM_STREAM_CAPTURE);
+	if (ret) {
+		pr_err("%s: stream reg failed:%d\n", __func__, ret);
+		return ret;
+	}
+
+	ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE));
+	if (ret < 0) {
+		pr_err("%s: Set IO mode failed\n", __func__);
+		return -EINVAL;
+	}
+
+	stream_index = STREAM_ARRAY_INDEX(ac->stream_id);
+	if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) {
+		pr_err("%s: Invalid stream index:%d", __func__, stream_index);
+		return -EINVAL;
+	}
+
+	runtime->fragments = prtd->codec_param.buffer.fragments;
+	runtime->fragment_size = prtd->codec_param.buffer.fragment_size;
+	pr_debug("%s: allocate %d buffers each of size %d\n",
+			__func__, runtime->fragments,
+			runtime->fragment_size);
+	ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac,
+					runtime->fragment_size,
+					runtime->fragments);
+	if (ret < 0) {
+		pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);
+		return -ENOMEM;
+	}
+
+	prtd->byte_offset    = 0;
+	prtd->received_total = 0;
+	prtd->app_pointer    = 0;
+	prtd->bytes_copied   = 0;
+	prtd->bytes_read     = 0;
+	prtd->bytes_read_offset = 0;
+	prtd->buffer         = ac->port[dir].buf[0].data;
+	prtd->buffer_paddr   = ac->port[dir].buf[0].phys;
+	prtd->buffer_size    = runtime->fragments * runtime->fragment_size;
+
+
+	pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n",
+			__func__, prtd->sample_rate, prtd->num_channels,
+					 bits_per_sample, sample_word_size);
+	ret = q6asm_enc_cfg_blk_pcm_format_support_v3(prtd->audio_client,
+					prtd->sample_rate, prtd->num_channels,
+					bits_per_sample, sample_word_size);
+
+	return ret;
+}
+
+static int msm_compr_playback_open(struct snd_compr_stream *cstream)
 {
 	struct snd_compr_runtime *runtime = cstream->runtime;
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
@@ -1181,7 +1361,6 @@ static int msm_compr_open(struct snd_compr_stream *cstream)
 		kfree(prtd);
 		return -ENOMEM;
 	}
-
 	pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
 	prtd->audio_client->perf_mode = false;
 	prtd->session_id = prtd->audio_client->session;
@@ -1189,7 +1368,73 @@ static int msm_compr_open(struct snd_compr_stream *cstream)
 	return 0;
 }
 
-static int msm_compr_free(struct snd_compr_stream *cstream)
+static int msm_compr_capture_open(struct snd_compr_stream *cstream)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+	struct msm_compr_audio *prtd;
+	struct msm_compr_pdata *pdata =
+			snd_soc_platform_get_drvdata(rtd->platform);
+
+	pr_debug("%s\n", __func__);
+	prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL);
+	if (prtd == NULL) {
+		pr_err("Failed to allocate memory for msm_compr_audio\n");
+		return -ENOMEM;
+	}
+
+	runtime->private_data = NULL;
+	prtd->cstream = cstream;
+	pdata->cstream[rtd->dai_link->be_id] = cstream;
+
+	prtd->audio_client = q6asm_audio_client_alloc(
+				(app_cb)compr_event_handler, prtd);
+	if (!prtd->audio_client) {
+		pr_err("%s: Could not allocate memory for client\n", __func__);
+		pdata->cstream[rtd->dai_link->be_id] = NULL;
+		kfree(prtd);
+		return -ENOMEM;
+	}
+	pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+	prtd->audio_client->perf_mode = false;
+	prtd->session_id = prtd->audio_client->session;
+	prtd->codec = FORMAT_LINEAR_PCM;
+	prtd->bytes_copied = 0;
+	prtd->bytes_read = 0;
+	prtd->bytes_read_offset = 0;
+	prtd->received_total = 0;
+	prtd->byte_offset = 0;
+	prtd->sample_rate = 48000;
+	prtd->num_channels = 2;
+	prtd->first_buffer = 0;
+
+	spin_lock_init(&prtd->lock);
+
+	atomic_set(&prtd->eos, 0);
+	atomic_set(&prtd->start, 0);
+	atomic_set(&prtd->drain, 0);
+	atomic_set(&prtd->xrun, 0);
+	atomic_set(&prtd->close, 0);
+	atomic_set(&prtd->wait_on_close, 0);
+	atomic_set(&prtd->error, 0);
+
+	runtime->private_data = prtd;
+
+	return 0;
+}
+
+static int msm_compr_open(struct snd_compr_stream *cstream)
+{
+	int ret = 0;
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		ret = msm_compr_playback_open(cstream);
+	else if (cstream->direction == SND_COMPRESS_CAPTURE)
+		ret = msm_compr_capture_open(cstream);
+	return ret;
+}
+
+static int msm_compr_playback_free(struct snd_compr_stream *cstream)
 {
 	struct snd_compr_runtime *runtime;
 	struct msm_compr_audio *prtd;
@@ -1284,6 +1529,76 @@ static int msm_compr_free(struct snd_compr_stream *cstream)
 	return 0;
 }
 
+static int msm_compr_capture_free(struct snd_compr_stream *cstream)
+{
+	struct snd_compr_runtime *runtime;
+	struct msm_compr_audio *prtd;
+	struct snd_soc_pcm_runtime *soc_prtd;
+	struct msm_compr_pdata *pdata;
+	struct audio_client *ac;
+	int dir = OUT, stream_id;
+	unsigned long flags;
+	uint32_t stream_index;
+
+	if (!cstream) {
+		pr_err("%s cstream is null\n", __func__);
+		return 0;
+	}
+	runtime = cstream->runtime;
+	soc_prtd = cstream->private_data;
+	if (!runtime || !soc_prtd || !(soc_prtd->platform)) {
+		pr_err("%s runtime or soc_prtd or platform is null\n",
+			__func__);
+		return 0;
+	}
+	prtd = runtime->private_data;
+	if (!prtd) {
+		pr_err("%s prtd is null\n", __func__);
+		return 0;
+	}
+	pdata = snd_soc_platform_get_drvdata(soc_prtd->platform);
+	ac = prtd->audio_client;
+	if (!pdata || !ac) {
+		pr_err("%s pdata or ac is null\n", __func__);
+		return 0;
+	}
+
+	spin_lock_irqsave(&prtd->lock, flags);
+	stream_id = ac->stream_id;
+
+	stream_index = STREAM_ARRAY_INDEX(stream_id);
+	if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0)) {
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		pr_debug("close stream %d", stream_id);
+		q6asm_stream_cmd(ac, CMD_CLOSE, stream_id);
+		spin_lock_irqsave(&prtd->lock, flags);
+	}
+	spin_unlock_irqrestore(&prtd->lock, flags);
+
+	pdata->cstream[soc_prtd->dai_link->be_id] = NULL;
+	msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+					SNDRV_PCM_STREAM_CAPTURE);
+
+	q6asm_audio_client_buf_free_contiguous(dir, ac);
+
+	q6asm_audio_client_free(ac);
+
+	kfree(prtd);
+
+	return 0;
+}
+
+static int msm_compr_free(struct snd_compr_stream *cstream)
+{
+	int ret = 0;
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		ret = msm_compr_playback_free(cstream);
+	else if (cstream->direction == SND_COMPRESS_CAPTURE)
+		ret = msm_compr_capture_free(cstream);
+	return ret;
+}
+
 static bool msm_compr_validate_codec_compr(__u32 codec_id)
 {
 	int32_t i;
@@ -1449,7 +1764,10 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
 	prtd->partial_drain_delay =
 		msm_compr_get_partial_drain_delay(frame_sz, prtd->sample_rate);
 
-	ret = msm_compr_configure_dsp(cstream);
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		ret = msm_compr_configure_dsp_for_playback(cstream);
+	else if (cstream->direction == SND_COMPRESS_CAPTURE)
+		ret = msm_compr_configure_dsp_for_capture(cstream);
 
 	return ret;
 }
@@ -1539,11 +1857,6 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 	uint32_t stream_index;
 	uint16_t bits_per_sample = 16;
 
-	if (cstream->direction != SND_COMPRESS_PLAYBACK) {
-		pr_err("%s: Unsupported stream type\n", __func__);
-		return -EINVAL;
-	}
-
 	spin_lock_irqsave(&prtd->lock, flags);
 	if (atomic_read(&prtd->error)) {
 		pr_err("%s Got RESET EVENTS notification, return immediately",
@@ -1566,7 +1879,8 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 		* compress passthrough volume is controlled in
 		* ADM by adm_send_compressed_device_mute()
 		*/
-		if (prtd->compr_passthr == LEGACY_PCM) {
+		if (prtd->compr_passthr == LEGACY_PCM &&
+			cstream->direction == SND_COMPRESS_PLAYBACK) {
 			/* set volume for the stream before RUN */
 			rc = msm_compr_set_volume(cstream,
 				volume[0], volume[1]);
@@ -1578,8 +1892,9 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 			if (rc)
 				pr_err("%s : init PP params failed : %d\n",
 					__func__, rc);
+		} else {
+			msm_compr_read_buffer(prtd);
 		}
-
 		/* issue RUN command for the stream */
 		q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
 		break;
@@ -1589,6 +1904,18 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 					prtd->gapless_state.gapless_transition);
 		stream_id = ac->stream_id;
 		atomic_set(&prtd->start, 0);
+		if (cstream->direction == SND_COMPRESS_CAPTURE) {
+			q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+			atomic_set(&prtd->xrun, 0);
+			prtd->received_total = 0;
+			prtd->bytes_copied = 0;
+			prtd->bytes_read = 0;
+			prtd->bytes_read_offset = 0;
+			prtd->byte_offset  = 0;
+			prtd->app_pointer  = 0;
+			spin_unlock_irqrestore(&prtd->lock, flags);
+			break;
+		}
 		if (prtd->next_stream) {
 			pr_debug("%s: interrupt next track wait queues\n",
 								__func__);
@@ -1979,7 +2306,7 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 }
 
 static int msm_compr_pointer(struct snd_compr_stream *cstream,
-				struct snd_compr_tstamp *arg)
+					struct snd_compr_tstamp *arg)
 {
 	struct snd_compr_runtime *runtime = cstream->runtime;
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
@@ -1998,7 +2325,10 @@ static int msm_compr_pointer(struct snd_compr_stream *cstream,
 	spin_lock_irqsave(&prtd->lock, flags);
 	tstamp.sampling_rate = prtd->sample_rate;
 	tstamp.byte_offset = prtd->byte_offset;
-	tstamp.copied_total = prtd->copied_total;
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		tstamp.copied_total = prtd->copied_total;
+	else if (cstream->direction == SND_COMPRESS_CAPTURE)
+		tstamp.copied_total = prtd->received_total;
 	first_buffer = prtd->first_buffer;
 	if (atomic_read(&prtd->error)) {
 		pr_err("%s Got RESET EVENTS notification, return error\n",
@@ -2012,37 +2342,39 @@ static int msm_compr_pointer(struct snd_compr_stream *cstream,
 		spin_unlock_irqrestore(&prtd->lock, flags);
 		return -ENETRESET;
 	}
+	if (cstream->direction == SND_COMPRESS_PLAYBACK) {
 
-	gapless_transition = prtd->gapless_state.gapless_transition;
-	spin_unlock_irqrestore(&prtd->lock, flags);
+		gapless_transition = prtd->gapless_state.gapless_transition;
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		if (gapless_transition)
+			pr_debug("%s session time in gapless transition",
+				__func__);
+		/*
+		 *- Do not query if no buffer has been given.
+		 *- Do not query on a gapless transition.
+		 *  Playback for the 2nd stream can start (thus returning time
+		 *  starting from 0) before the driver knows about EOS of first
+		 *  stream.
+		*/
+		if (!first_buffer || gapless_transition) {
 
-	if (gapless_transition)
-		pr_debug("%s session time in gapless transition",
-			 __func__);
-
-	/*
-	 - Do not query if no buffer has been given.
-	 - Do not query on a gapless transition.
-	   Playback for the 2nd stream can start (thus returning time
-	   starting from 0) before the driver knows about EOS of first stream.
-	*/
-
-	if (!first_buffer && !gapless_transition) {
-		if (pdata->use_legacy_api)
-			rc = q6asm_get_session_time_legacy(prtd->audio_client,
-						&prtd->marker_timestamp);
-		else
-			rc = q6asm_get_session_time(prtd->audio_client,
-						&prtd->marker_timestamp);
-
-		if (rc < 0) {
-			pr_err("%s: Get Session Time return value =%lld\n",
-				__func__, timestamp);
-			if (atomic_read(&prtd->error))
-				return -ENETRESET;
+			if (pdata->use_legacy_api)
+				rc = q6asm_get_session_time_legacy(
+				prtd->audio_client, &prtd->marker_timestamp);
 			else
-				return -EAGAIN;
+				rc = q6asm_get_session_time(
+				prtd->audio_client, &prtd->marker_timestamp);
+			if (rc < 0) {
+				pr_err("%s: Get Session Time return =%lld\n",
+					__func__, timestamp);
+				if (atomic_read(&prtd->error))
+					return -ENETRESET;
+				else
+					return -EAGAIN;
+			}
 		}
+	} else {
+		spin_unlock_irqrestore(&prtd->lock, flags);
 	}
 	timestamp = prtd->marker_timestamp;
 
@@ -2103,8 +2435,8 @@ static int msm_compr_ack(struct snd_compr_stream *cstream,
 	return 0;
 }
 
-static int msm_compr_copy(struct snd_compr_stream *cstream,
-			  char __user *buf, size_t count)
+static int msm_compr_playback_copy(struct snd_compr_stream *cstream,
+				  char __user *buf, size_t count)
 {
 	struct snd_compr_runtime *runtime = cstream->runtime;
 	struct msm_compr_audio *prtd = runtime->private_data;
@@ -2166,6 +2498,60 @@ static int msm_compr_copy(struct snd_compr_stream *cstream,
 	return count;
 }
 
+static int msm_compr_capture_copy(struct snd_compr_stream *cstream,
+					char __user *buf, size_t count)
+{
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct msm_compr_audio *prtd = runtime->private_data;
+	void *source;
+	unsigned long flags;
+
+	pr_debug("%s: count = %zd\n", __func__, count);
+	if (!prtd->buffer) {
+		pr_err("%s: Buffer is not allocated yet ??", __func__);
+		return 0;
+	}
+
+	spin_lock_irqsave(&prtd->lock, flags);
+	if (atomic_read(&prtd->error)) {
+		pr_err("%s Got RESET EVENTS notification", __func__);
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		return -ENETRESET;
+	}
+
+	source = prtd->buffer + prtd->app_pointer;
+	/* check if we have requested amount of data to copy to user*/
+	if (count <= prtd->received_total - prtd->bytes_copied)	{
+		spin_unlock_irqrestore(&prtd->lock, flags);
+		if (copy_to_user(buf, source, count)) {
+			pr_err("copy_to_user failed");
+			return -EFAULT;
+		}
+		spin_lock_irqsave(&prtd->lock, flags);
+		prtd->app_pointer += count;
+		if (prtd->app_pointer >= prtd->buffer_size)
+			prtd->app_pointer -= prtd->buffer_size;
+		prtd->bytes_copied += count;
+	}
+	msm_compr_read_buffer(prtd);
+
+	spin_unlock_irqrestore(&prtd->lock, flags);
+	return count;
+}
+
+static int msm_compr_copy(struct snd_compr_stream *cstream,
+				char __user *buf, size_t count)
+{
+	int ret = 0;
+
+	pr_debug(" In %s\n", __func__);
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		ret = msm_compr_playback_copy(cstream, buf, count);
+	else if (cstream->direction == SND_COMPRESS_CAPTURE)
+		ret = msm_compr_capture_copy(cstream, buf, count);
+	return ret;
+}
+
 static int msm_compr_get_caps(struct snd_compr_stream *cstream,
 				struct snd_compr_caps *arg)
 {

sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c

@@ -548,6 +548,15 @@ static struct msm_pcm_routing_fdai_data
 	/* MULTIMEDIA16 */
 	{{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} },
 	 {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } },
+	/* MULTIMEDIA17 */
+	{{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} },
+	 {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } },
+	/* MULTIMEDIA18 */
+	{{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} },
+	 {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } },
+	/* MULTIMEDIA19 */
+	{{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} },
+	 {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } },
 };
 
 static unsigned long session_copp_map[MSM_FRONTEND_DAI_MM_SIZE][2]
@@ -2288,6 +2297,21 @@ static const struct snd_kcontrol_new ext_ec_ref_mux_ul9 =
 		msm_route_ec_ref_rx_enum[0],
 		msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put);
 
+static const struct snd_kcontrol_new ext_ec_ref_mux_ul17 =
+	SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL17 MUX Mux",
+		msm_route_ec_ref_rx_enum[0],
+		msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put);
+
+static const struct snd_kcontrol_new ext_ec_ref_mux_ul18 =
+	SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL18 MUX Mux",
+		msm_route_ec_ref_rx_enum[0],
+		msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put);
+
+static const struct snd_kcontrol_new ext_ec_ref_mux_ul19 =
+	SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL19 MUX Mux",
+		msm_route_ec_ref_rx_enum[0],
+		msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put);
+
 static int msm_routing_ext_ec_get(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol)
 {
@@ -2417,6 +2441,15 @@ static const struct snd_kcontrol_new pri_i2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_I2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_PRI_I2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_PRI_I2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_PRI_I2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = {
@@ -2468,6 +2501,15 @@ static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_I2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SEC_I2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SEC_I2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SEC_I2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new spdif_rx_mixer_controls[] = {
@@ -2519,6 +2561,16 @@ static const struct snd_kcontrol_new spdif_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SPDIF_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SPDIF_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SPDIF_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SPDIF_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+
 };
 
 static const struct snd_kcontrol_new slimbus_2_rx_mixer_controls[] = {
@@ -2621,6 +2673,15 @@ static const struct snd_kcontrol_new slimbus_5_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SLIMBUS_5_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SLIMBUS_5_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SLIMBUS_5_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SLIMBUS_5_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
@@ -2672,6 +2733,15 @@ static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SLIMBUS_0_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = {
@@ -2723,6 +2793,15 @@ static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_MI2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
@@ -2774,6 +2853,15 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new quinary_mi2s_rx_mixer_controls[] = {
@@ -2825,6 +2913,15 @@ static const struct snd_kcontrol_new quinary_mi2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_QUINARY_MI2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_QUINARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_QUINARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_QUINARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = {
@@ -2870,6 +2967,15 @@ static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERTIARY_MI2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_TERTIARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_TERTIARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_TERTIARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new secondary_mi2s_rx2_mixer_controls[] = {
@@ -2927,6 +3033,15 @@ static const struct snd_kcontrol_new secondary_mi2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SECONDARY_MI2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SECONDARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SECONDARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SECONDARY_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = {
@@ -2978,6 +3093,15 @@ static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_MI2S_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_PRI_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_PRI_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_PRI_MI2S_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new int0_mi2s_rx_mixer_controls[] = {
@@ -3131,6 +3255,15 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_HDMI_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_HDMI_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_HDMI_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_HDMI_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new display_port_mixer_controls[] = {
@@ -3432,6 +3565,15 @@ static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_INT_BT_SCO_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new int_bt_a2dp_rx_mixer_controls[] = {
@@ -3534,6 +3676,15 @@ static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_INT_FM_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_INT_FM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_INT_FM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_INT_FM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = {
@@ -3585,6 +3736,15 @@ static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_AFE_PCM_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_AFE_PCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_AFE_PCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_AFE_PCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
@@ -3636,6 +3796,15 @@ static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_AUXPCM_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_AUXPCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_AUXPCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_AUXPCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = {
@@ -3687,6 +3856,15 @@ static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = {
 	SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_AUXPCM_RX,
 	MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SEC_AUXPCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SEC_AUXPCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SEC_AUXPCM_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 };
 
 static const struct snd_kcontrol_new tert_auxpcm_rx_mixer_controls[] = {
@@ -5018,6 +5196,77 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = {
 	msm_routing_put_audio_mixer),
 };
 
+static const struct snd_kcontrol_new mmul17_mixer_controls[] = {
+	SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul18_mixer_controls[] = {
+	SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul19_mixer_controls[] = {
+	SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+};
 static const struct snd_kcontrol_new pri_rx_voice_mixer_controls[] = {
 	SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_PRI_I2S_RX,
 	MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
@@ -8098,6 +8347,9 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("MM_UL9", "MultiMedia9 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL17", "MultiMedia17 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL18", "MultiMedia18 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL19", "MultiMedia19 Capture", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_IN("CS-VOICE_DL1", "CS-VOICE Playback", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("CS-VOICE_UL1", "CS-VOICE Capture", 0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_IN("VOICE2_DL", "Voice2 Playback", 0, 0, 0, 0),
@@ -8796,6 +9048,12 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 	mmul6_mixer_controls, ARRAY_SIZE(mmul6_mixer_controls)),
 	SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,
 	mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),
+	SND_SOC_DAPM_MIXER("MultiMedia17 Mixer", SND_SOC_NOPM, 0, 0,
+	mmul17_mixer_controls, ARRAY_SIZE(mmul17_mixer_controls)),
+	SND_SOC_DAPM_MIXER("MultiMedia18 Mixer", SND_SOC_NOPM, 0, 0,
+	mmul18_mixer_controls, ARRAY_SIZE(mmul18_mixer_controls)),
+	SND_SOC_DAPM_MIXER("MultiMedia19 Mixer", SND_SOC_NOPM, 0, 0,
+	mmul19_mixer_controls, ARRAY_SIZE(mmul19_mixer_controls)),
 	SND_SOC_DAPM_MIXER("AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
 	auxpcm_rx_mixer_controls, ARRAY_SIZE(auxpcm_rx_mixer_controls)),
 	SND_SOC_DAPM_MIXER("SEC_AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
@@ -9060,6 +9318,12 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 		&ext_ec_ref_mux_ul8),
 	SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL9 MUX", SND_SOC_NOPM, 0, 0,
 		&ext_ec_ref_mux_ul9),
+	SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL17 MUX", SND_SOC_NOPM, 0, 0,
+		&ext_ec_ref_mux_ul17),
+	SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL18 MUX", SND_SOC_NOPM, 0, 0,
+		&ext_ec_ref_mux_ul18),
+	SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL19 MUX", SND_SOC_NOPM, 0, 0,
+		&ext_ec_ref_mux_ul19),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
@@ -9287,9 +9551,15 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MultiMedia8 Mixer", "SLIM_6_TX", "SLIMBUS_6_TX"},
 	{"MultiMedia8 Mixer", "SLIM_7_TX", "SLIMBUS_7_TX"},
 	{"MultiMedia4 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+	{"MultiMedia17 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+	{"MultiMedia18 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+	{"MultiMedia19 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
 	{"MultiMedia8 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
 	{"MultiMedia2 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
 	{"MultiMedia4 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+	{"MultiMedia17 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+	{"MultiMedia18 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+	{"MultiMedia19 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
 	{"MultiMedia8 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
 	{"MultiMedia3 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
 	{"MultiMedia5 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
@@ -9884,10 +10154,16 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MultiMedia1 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
 	{"MultiMedia3 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
 	{"MultiMedia4 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+	{"MultiMedia17 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+	{"MultiMedia18 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+	{"MultiMedia19 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
 	{"MultiMedia5 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
 	{"MultiMedia8 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
 	{"MultiMedia1 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
 	{"MultiMedia4 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+	{"MultiMedia17 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+	{"MultiMedia18 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+	{"MultiMedia19 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
 	{"MultiMedia5 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
 	{"MultiMedia6 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
 	{"MultiMedia8 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
@@ -9895,6 +10171,9 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MultiMedia1 Mixer", "AFE_PCM_TX", "PCM_TX"},
 	{"MultiMedia3 Mixer", "AFE_PCM_TX", "PCM_TX"},
 	{"MultiMedia4 Mixer", "AFE_PCM_TX", "PCM_TX"},
+	{"MultiMedia17 Mixer", "AFE_PCM_TX", "PCM_TX"},
+	{"MultiMedia18 Mixer", "AFE_PCM_TX", "PCM_TX"},
+	{"MultiMedia19 Mixer", "AFE_PCM_TX", "PCM_TX"},
 	{"MultiMedia5 Mixer", "AFE_PCM_TX", "PCM_TX"},
 	{"MultiMedia8 Mixer", "AFE_PCM_TX", "PCM_TX"},
 	{"MM_UL1", NULL, "MultiMedia1 Mixer"},
@@ -9905,6 +10184,9 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MM_UL5", NULL, "MultiMedia5 Mixer"},
 	{"MM_UL6", NULL, "MultiMedia6 Mixer"},
 	{"MM_UL8", NULL, "MultiMedia8 Mixer"},
+	{"MM_UL17", NULL, "MultiMedia17 Mixer"},
+	{"MM_UL18", NULL, "MultiMedia18 Mixer"},
+	{"MM_UL19", NULL, "MultiMedia19 Mixer"},
 
 	{"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
 	{"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
@@ -10284,6 +10566,21 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"AUDIO_REF_EC_UL9 MUX", "TERT_MI2S_TX" , "TERT_MI2S_TX"},
 	{"AUDIO_REF_EC_UL9 MUX", "QUAT_MI2S_TX" , "QUAT_MI2S_TX"},
 
+	{"AUDIO_REF_EC_UL17 MUX", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+	{"AUDIO_REF_EC_UL17 MUX", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+	{"AUDIO_REF_EC_UL17 MUX", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+	{"AUDIO_REF_EC_UL17 MUX", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+
+	{"AUDIO_REF_EC_UL18 MUX", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+	{"AUDIO_REF_EC_UL18 MUX", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+	{"AUDIO_REF_EC_UL18 MUX", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+	{"AUDIO_REF_EC_UL18 MUX", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+
+	{"AUDIO_REF_EC_UL19 MUX", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+	{"AUDIO_REF_EC_UL19 MUX", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+	{"AUDIO_REF_EC_UL19 MUX", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+	{"AUDIO_REF_EC_UL19 MUX", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+
 	{"MM_UL1", NULL, "AUDIO_REF_EC_UL1 MUX"},
 	{"MM_UL2", NULL, "AUDIO_REF_EC_UL2 MUX"},
 	{"MM_UL3", NULL, "AUDIO_REF_EC_UL3 MUX"},
@@ -10292,6 +10589,9 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MM_UL6", NULL, "AUDIO_REF_EC_UL6 MUX"},
 	{"MM_UL8", NULL, "AUDIO_REF_EC_UL8 MUX"},
 	{"MM_UL9", NULL, "AUDIO_REF_EC_UL9 MUX"},
+	{"MM_UL17", NULL, "AUDIO_REF_EC_UL17 MUX"},
+	{"MM_UL18", NULL, "AUDIO_REF_EC_UL18 MUX"},
+	{"MM_UL19", NULL, "AUDIO_REF_EC_UL19 MUX"},
 
 	{"Voice_Tx Mixer", "PRI_TX_Voice", "PRI_I2S_TX"},
 	{"Voice_Tx Mixer", "PRI_MI2S_TX_Voice", "PRI_MI2S_TX"},

sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h

@@ -182,6 +182,9 @@ enum {
 	MSM_FRONTEND_DAI_MULTIMEDIA14,
 	MSM_FRONTEND_DAI_MULTIMEDIA15,
 	MSM_FRONTEND_DAI_MULTIMEDIA16,
+	MSM_FRONTEND_DAI_MULTIMEDIA17,
+	MSM_FRONTEND_DAI_MULTIMEDIA18,
+	MSM_FRONTEND_DAI_MULTIMEDIA19,
 	MSM_FRONTEND_DAI_CS_VOICE,
 	MSM_FRONTEND_DAI_VOIP,
 	MSM_FRONTEND_DAI_AFE_RX,
@@ -207,8 +210,8 @@ enum {
 	MSM_FRONTEND_DAI_MAX,
 };
 
-#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA16 + 1)
-#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA16
+#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA19 + 1)
+#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA19
 
 enum {
 	MSM_BACKEND_DAI_PRI_I2S_RX = 0,