ASoC: msm: qdsp6v2: Add ASM low latency loopback support

Add ASM low latency loopback support.
ICC (in-car communication) use case needs the low latency
ASM loopback to satisfy the latency requirement.

CRs-fixed: 998118
Change-Id: If225e809072f4296bc22028da29e589137e5799d
Signed-off-by: Derek Chen <chenche@codeaurora.org>

Documentation/devicetree/bindings/sound/qcom-audio-dev.txt

@@ -177,6 +177,11 @@ Required properties:
 
  - compatible : "qcom,msm-pcm-loopback"
 
+Optional properties:
+
+ - qcom,msm-pcm-loopback-low-latency : Flag indicating whether
+   the device node is of type low latency.
+
 * msm-dai-q6
 
 [First Level Nodes]
@@ -415,6 +420,11 @@ Example:
 		qcom,msm-pcm-low-latency;
 	};
 
+	qcom,msm-pcm-loopback-low-latency {
+		compatible = "qcom,msm-pcm-loopback";
+		qcom,msm-pcm-loopback-low-latency;
+	};
+
         qcom,msm-pcm-routing {
                 compatible = "qcom,msm-pcm-routing";
         };
@@ -2116,13 +2126,15 @@ Example:
 
 		asoc-platform = <&pcm0>, <&pcm1>, <&pcm2>, <&voip>, <&voice>,
 				<&loopback>, <&compress>, <&hostless>,
-				<&afe>, <&lsm>, <&routing>, <&compr>;
+				<&afe>, <&lsm>, <&routing>, <&compr>,
+				<&loopback1>;
 		asoc-platform-names = "msm-pcm-dsp.0", "msm-pcm-dsp.1",
 				"msm-pcm-dsp.2", "msm-voip-dsp",
 				"msm-pcm-voice", "msm-pcm-loopback",
 				"msm-compress-dsp", "msm-pcm-hostless",
 				"msm-pcm-afe", "msm-lsm-client",
-				"msm-pcm-routing", "msm-compr-dsp";
+				"msm-pcm-routing", "msm-compr-dsp",
+				"msm-pcm-loopback.1";
 		asoc-cpu = <&dai_pri_auxpcm>, <&dai_sec_auxpcm>, <&dai_hdmi>,
 				<&dai_mi2s>, <&dai_mi2s_quat>,
 				<&afe_pcm_rx>, <&afe_pcm_tx>,

include/sound/apr_audio-v2.h

@@ -5812,6 +5812,138 @@ struct asm_stream_cmd_open_loopback_v2 {
 /* Reserved for future use. This field must be set to zero. */
 } __packed;
 
+
+#define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK    0x00010DBA
+
+/* Bitmask for the stream's Performance mode. */
+#define ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK \
+	(0x70000000UL)
+
+/* Bit shift for the stream's Performance mode. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK    28
+
+/* Bitmask for the decoder converter enable flag. */
+#define ASM_BIT_MASK_DECODER_CONVERTER_FLAG    (0x00000078UL)
+
+/* Shift value for the decoder converter enable flag. */
+#define ASM_SHIFT_DECODER_CONVERTER_FLAG                              3
+
+/* Converter mode is None (Default). */
+#define ASM_CONVERTER_MODE_NONE                                       0
+
+/* Converter mode is DDP-to-DD. */
+#define ASM_DDP_DD_CONVERTER_MODE                                     1
+
+/*  Identifies a special converter mode where source and sink formats
+ *  are the same but postprocessing must applied. Therefore, Decode
+ *  @rarrow Re-encode is necessary.
+ */
+#define ASM_POST_PROCESS_CONVERTER_MODE                               2
+
+
+struct asm_stream_cmd_open_transcode_loopback_t {
+	struct apr_hdr         hdr;
+	u32                    mode_flags;
+/* Mode Flags specifies the performance mode in which this stream
+ * is to be opened.
+ * Supported values{for bits 30 to 28}(stream_perf_mode flag)
+ *
+ * #ASM_LEGACY_STREAM_SESSION -- This mode ensures backward
+ *       compatibility to the original behavior
+ *       of ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK
+ *
+ * #ASM_LOW_LATENCY_STREAM_SESSION -- Opens a loopback session by using
+ *  shortened buffers in low latency POPP
+ *  - Recommendation: Do not enable high latency algorithms. They might
+ *    negate the benefits of opening a low latency stream, and they
+ *    might also suffer quality degradation from unexpected jitter.
+ *  - This Low Latency mode is supported only for PCM In and PCM Out
+ *    loopbacks. An error is returned if Low Latency mode is opened for
+ *    other transcode loopback modes.
+ *  - To configure this subfield, use
+ *     ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK and
+ *     ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK.
+ *
+ * Supported values{for bits 6 to 3} (decoder-converter compatibility)
+ * #ASM_CONVERTER_MODE_NONE (0x0) -- Default
+ * #ASM_DDP_DD_CONVERTER_MODE (0x1)
+ * #ASM_POST_PROCESS_CONVERTER_MODE (0x2)
+ * 0x3-0xF -- Reserved for future use
+ * - Use #ASM_BIT_MASK_DECODER_CONVERTER_FLAG and
+ *        ASM_SHIFT_DECODER_CONVERTER_FLAG to set this bit
+ * All other bits are reserved; clients must set them to 0.
+ */
+
+	u32                    src_format_id;
+/* Specifies the media format of the input audio stream.
+ *
+ * Supported values
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
+ * - #ASM_MEDIA_FMT_DTS
+ * - #ASM_MEDIA_FMT_EAC3_DEC
+ * - #ASM_MEDIA_FMT_EAC3
+ * - #ASM_MEDIA_FMT_AC3_DEC
+ * - #ASM_MEDIA_FMT_AC3
+ */
+	u32                    sink_format_id;
+/* Specifies the media format of the output stream.
+ *
+ * Supported values
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3
+ * - #ASM_MEDIA_FMT_DTS (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_EAC3_DEC (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_EAC3 (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_AC3_DEC (not supported in Low Latency mode)
+ * - #ASM_MEDIA_FMT_AC3 (not supported in Low Latency mode)
+ */
+
+	u32                    audproc_topo_id;
+/* Postprocessing topology ID, which specifies the topology (order of
+ *        processing) of postprocessing algorithms.
+ *
+ * Supported values
+ *    - #ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT
+ *    - #ASM_STREAM_POSTPROC_TOPO_ID_PEAKMETER
+ *    - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL
+ *    - #ASM_STREAM_POSTPROC_TOPO_ID_NONE
+ *  Topologies can be added through #ASM_CMD_ADD_TOPOLOGIES.
+ *  This field is ignored for the Converter mode, in which no
+ *  postprocessing is performed.
+ */
+
+	u16                    src_endpoint_type;
+/* Specifies the source endpoint that provides the input samples.
+ *
+ * Supported values
+ *  - 0 -- Tx device matrix or stream router (gateway to the hardware
+ *    ports)
+ *  - All other values are reserved
+ *  Clients must set this field to 0. Otherwise, an error is returned.
+ */
+
+	u16                    sink_endpoint_type;
+/*  Specifies the sink endpoint type.
+ *
+ *  Supported values
+ *  - 0 -- Rx device matrix or stream router (gateway to the hardware
+ *    ports)
+ *  - All other values are reserved
+ *   Clients must set this field to 0. Otherwise, an error is returned.
+ */
+
+	u16                    bits_per_sample;
+/*   Number of bits per sample processed by the ASM modules.
+ *   Supported values 16, 24
+ */
+
+	u16                    reserved;
+/*   This field must be set to 0.
+ */
+} __packed;
+
+
 #define ASM_STREAM_CMD_CLOSE             0x00010BCD
 #define ASM_STREAM_CMD_FLUSH             0x00010BCE
 

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

@@ -25,6 +25,7 @@
 #include <sound/control.h>
 #include <sound/tlv.h>
 #include <asm/dma.h>
+#include <sound/q6audio-v2.h>
 
 #include "msm-pcm-routing-v2.h"
 
@@ -67,6 +68,10 @@ static struct fe_dai_session_map session_map[LOOPBACK_SESSION_MAX] = {
 
 static u32 hfp_tx_mute;
 
+struct msm_pcm_pdata {
+	int perf_mode;
+};
+
 static void stop_pcm(struct msm_pcm_loopback *pcm);
 static int msm_pcm_loopback_get_session(struct snd_soc_pcm_runtime *rtd,
 					struct msm_pcm_loopback **pcm);
@@ -244,6 +249,7 @@ static int msm_pcm_open(struct snd_pcm_substream *substream)
 	struct msm_pcm_routing_evt event;
 	struct asm_session_mtmx_strtr_param_window_v2_t asm_mtmx_strtr_window;
 	uint32_t param_id;
+	struct msm_pcm_pdata *pdata;
 
 	ret =  msm_pcm_loopback_get_session(rtd, &pcm);
 	if (ret)
@@ -269,6 +275,15 @@ static int msm_pcm_open(struct snd_pcm_substream *substream)
 		if (pcm->audio_client != NULL)
 			stop_pcm(pcm);
 
+		pdata = (struct msm_pcm_pdata *)
+			dev_get_drvdata(rtd->platform->dev);
+		if (!pdata) {
+			dev_err(rtd->platform->dev,
+				"%s: platform data not populated\n", __func__);
+			mutex_unlock(&pcm->lock);
+			return -EINVAL;
+		}
+
 		pcm->audio_client = q6asm_audio_client_alloc(
 				(app_cb)msm_pcm_loopback_event_handler, pcm);
 		if (!pcm->audio_client) {
@@ -278,7 +293,7 @@ static int msm_pcm_open(struct snd_pcm_substream *substream)
 			return -ENOMEM;
 		}
 		pcm->session_id = pcm->audio_client->session;
-		pcm->audio_client->perf_mode = false;
+		pcm->audio_client->perf_mode = pdata->perf_mode;
 		ret = q6asm_open_loopback_v2(pcm->audio_client,
 					     bits_per_sample);
 		if (ret < 0) {
@@ -745,9 +760,23 @@ static struct snd_soc_platform_driver msm_soc_platform = {
 
 static int msm_pcm_probe(struct platform_device *pdev)
 {
+	struct msm_pcm_pdata *pdata;
+
 	dev_dbg(&pdev->dev, "%s: dev name %s\n",
 		__func__, dev_name(&pdev->dev));
 
+	pdata = kzalloc(sizeof(struct msm_pcm_pdata), GFP_KERNEL);
+	if (!pdata)
+		return -ENOMEM;
+
+	if (of_property_read_bool(pdev->dev.of_node,
+				"qcom,msm-pcm-loopback-low-latency"))
+		pdata->perf_mode = LOW_LATENCY_PCM_MODE;
+	else
+		pdata->perf_mode = LEGACY_PCM_MODE;
+
+	dev_set_drvdata(&pdev->dev, pdata);
+
 	return snd_soc_register_platform(&pdev->dev,
 				   &msm_soc_platform);
 }

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

@@ -5242,6 +5242,57 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = {
 	msm_routing_put_audio_mixer),
 };
 
+static const struct snd_kcontrol_new mmul9_mixer_controls[] = {
+	SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 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_MULTIMEDIA9, 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_MULTIMEDIA9, 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_MULTIMEDIA9, 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_MULTIMEDIA9, 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_MULTIMEDIA9, 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_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("SLIM_6_TX", MSM_BACKEND_DAI_SLIMBUS_6_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_TERT_TDM_TX_0,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("TERT_TDM_TX_1", MSM_BACKEND_DAI_TERT_TDM_TX_1,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("TERT_TDM_TX_2", MSM_BACKEND_DAI_TERT_TDM_TX_2,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("TERT_TDM_TX_3", MSM_BACKEND_DAI_TERT_TDM_TX_3,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("QUAT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_TX_0,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("QUAT_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_TX_1,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("QUAT_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_TX_2,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("QUAT_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_TX_3,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	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,
@@ -9210,6 +9261,8 @@ 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("MultiMedia9 Mixer", SND_SOC_NOPM, 0, 0,
+	mmul9_mixer_controls, ARRAY_SIZE(mmul9_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,
@@ -10251,6 +10304,15 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MultiMedia8 Mixer", "QUAT_TDM_TX_2", "QUAT_TDM_TX_2"},
 	{"MultiMedia8 Mixer", "QUAT_TDM_TX_3", "QUAT_TDM_TX_3"},
 
+	{"MultiMedia9 Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"},
+	{"MultiMedia9 Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"},
+	{"MultiMedia9 Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"},
+	{"MultiMedia9 Mixer", "TERT_TDM_TX_3", "TERT_TDM_TX_3"},
+	{"MultiMedia9 Mixer", "QUAT_TDM_TX_0", "QUAT_TDM_TX_0"},
+	{"MultiMedia9 Mixer", "QUAT_TDM_TX_1", "QUAT_TDM_TX_1"},
+	{"MultiMedia9 Mixer", "QUAT_TDM_TX_2", "QUAT_TDM_TX_2"},
+	{"MultiMedia9 Mixer", "QUAT_TDM_TX_3", "QUAT_TDM_TX_3"},
+
 	{"MultiMedia1 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"},
 	{"MultiMedia2 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"},
 	{"MultiMedia4 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"},
@@ -10365,6 +10427,7 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MM_UL5", NULL, "MultiMedia5 Mixer"},
 	{"MM_UL6", NULL, "MultiMedia6 Mixer"},
 	{"MM_UL8", NULL, "MultiMedia8 Mixer"},
+	{"MM_UL9", NULL, "MultiMedia9 Mixer"},
 	{"MM_UL17", NULL, "MultiMedia17 Mixer"},
 	{"MM_UL18", NULL, "MultiMedia18 Mixer"},
 	{"MM_UL19", NULL, "MultiMedia19 Mixer"},

sound/soc/msm/qdsp6v2/q6asm.c

@@ -1710,6 +1710,7 @@ static int32_t q6asm_callback(struct apr_client_data *data, void *priv)
 		case ASM_STREAM_CMD_OPEN_PUSH_MODE_READ:
 		case ASM_STREAM_CMD_OPEN_READWRITE_V2:
 		case ASM_STREAM_CMD_OPEN_LOOPBACK_V2:
+		case ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK:
 		case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
 		case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
 		case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:
@@ -2982,7 +2983,6 @@ int q6asm_open_read_write_v2(struct audio_client *ac, uint32_t rd_format,
 int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample)
 {
 	int rc = 0x00;
-	struct asm_stream_cmd_open_loopback_v2 open;
 
 	if (ac == NULL) {
 		pr_err("%s: APR handle NULL\n", __func__);
@@ -2994,29 +2994,67 @@ int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample)
 	}
 	pr_debug("%s: session[%d]\n", __func__, ac->session);
 
-	q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
-	atomic_set(&ac->cmd_state, -1);
-	open.hdr.opcode = ASM_STREAM_CMD_OPEN_LOOPBACK_V2;
+	if (ac->perf_mode == LOW_LATENCY_PCM_MODE) {
+		struct asm_stream_cmd_open_transcode_loopback_t open;
 
-	open.mode_flags = 0;
-	open.src_endpointype = 0;
-	open.sink_endpointype = 0;
-	/* source endpoint : matrix */
-	open.postprocopo_id = q6asm_get_asm_topology_cal();
+		q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+		atomic_set(&ac->cmd_state, -1);
+		open.hdr.opcode = ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK;
 
-	ac->app_type = q6asm_get_asm_app_type_cal();
-	ac->topology = open.postprocopo_id;
-	open.bits_per_sample = bits_per_sample;
-	open.reserved = 0;
+		open.mode_flags = 0;
+		open.src_endpoint_type = 0;
+		open.sink_endpoint_type = 0;
+		open.src_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+		open.sink_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+		/* source endpoint : matrix */
+		open.audproc_topo_id = q6asm_get_asm_topology_cal();
 
-	rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
-	if (rc < 0) {
-		pr_err("%s: open failed op[0x%x]rc[%d]\n", __func__,
-				open.hdr.opcode, rc);
-		rc = -EINVAL;
-		goto fail_cmd;
+		ac->app_type = q6asm_get_asm_app_type_cal();
+		if (ac->perf_mode == LOW_LATENCY_PCM_MODE)
+			open.mode_flags |= ASM_LOW_LATENCY_STREAM_SESSION;
+		else
+			open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
+		ac->topology = open.audproc_topo_id;
+		open.bits_per_sample = bits_per_sample;
+		open.reserved = 0;
+		pr_debug("%s: opening a transcode_loopback with mode_flags =[%d] session[%d]\n",
+				__func__, open.mode_flags, ac->session);
+
+		rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+		if (rc < 0) {
+			pr_err("%s: open failed op[0x%x]rc[%d]\n",
+					__func__, open.hdr.opcode, rc);
+			rc = -EINVAL;
+			goto fail_cmd;
+		}
+	} else {/*if(ac->perf_mode == LEGACY_PCM_MODE)*/
+		struct asm_stream_cmd_open_loopback_v2 open;
+
+		q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+		atomic_set(&ac->cmd_state, -1);
+		open.hdr.opcode = ASM_STREAM_CMD_OPEN_LOOPBACK_V2;
+
+		open.mode_flags = 0;
+		open.src_endpointype = 0;
+		open.sink_endpointype = 0;
+		/* source endpoint : matrix */
+		open.postprocopo_id = q6asm_get_asm_topology_cal();
+
+		ac->app_type = q6asm_get_asm_app_type_cal();
+		ac->topology = open.postprocopo_id;
+		open.bits_per_sample = bits_per_sample;
+		open.reserved = 0;
+		pr_debug("%s: opening a loopback_v2 with mode_flags =[%d] session[%d]\n",
+				__func__, open.mode_flags, ac->session);
+
+		rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+		if (rc < 0) {
+			pr_err("%s: open failed op[0x%x]rc[%d]\n",
+					__func__, open.hdr.opcode, rc);
+			rc = -EINVAL;
+			goto fail_cmd;
+		}
 	}
-
 	rc = wait_event_timeout(ac->cmd_wait,
 			(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
 	if (!rc) {