Merge "ASoC: Add pseudo port to support downlink playback"

8 files changed, 121 insertions, 8 deletions

diff --git a/asoc/msm-dai-q6-v2.c b/asoc/msm-dai-q6-v2.c
index 2ab7da5e..bbac4053 100644
--- a/asoc/msm-dai-q6-v2.c
+++ b/asoc/msm-dai-q6-v2.c
@@ -1929,6 +1929,7 @@ static int msm_dai_q6_hw_params(struct snd_pcm_substream *substream,
 		rc = msm_dai_q6_afe_rtproxy_hw_params(params, dai);
 		break;
 	case VOICE_PLAYBACK_TX:
+	case VOICE_PLAYBACK_DL_TX:
 	case VOICE2_PLAYBACK_TX:
 	case VOICE_RECORD_RX:
 	case VOICE_RECORD_TX:
@@ -3085,6 +3086,23 @@ static struct snd_soc_dai_driver msm_dai_q6_voc_playback_dai[] = {
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
 	},
+	{
+		.playback = {
+			.stream_name = "Voice Downlink Playback",
+			.aif_name = "VOICE_PLAYBACK_DL_TX",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 2,
+			.rate_min =     8000,
+			.rate_max =     48000,
+		},
+		.ops = &msm_dai_q6_ops,
+		.id = VOICE_PLAYBACK_DL_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+	},
 };
 
 static struct snd_soc_dai_driver msm_dai_q6_incall_record_dai[] = {
@@ -5160,6 +5178,9 @@ register_afe_capture:
 			pr_err("%s: Device not found stream name %s\n",
 			__func__, stream_name);
 		break;
+	case VOICE_PLAYBACK_DL_TX:
+		strlcpy(stream_name, "Voice Downlink Playback", 80);
+		goto register_voice_playback;
 	case VOICE_PLAYBACK_TX:
 		strlcpy(stream_name, "Voice Farend Playback", 80);
 		goto register_voice_playback;
diff --git a/asoc/msm-pcm-routing-v2.c b/asoc/msm-pcm-routing-v2.c
index 8a44b528..6ac7be8b 100644
--- a/asoc/msm-pcm-routing-v2.c
+++ b/asoc/msm-pcm-routing-v2.c
@@ -311,6 +311,8 @@ struct msm_pcm_routing_bdai_data msm_bedais[MSM_BACKEND_DAI_MAX] = {
 	  LPASS_BE_AUXPCM_TX},
 	{ VOICE_PLAYBACK_TX, 0, {0}, {0}, 0, 0, 0, 0, {0},
 	  LPASS_BE_VOICE_PLAYBACK_TX},
+	{ VOICE_PLAYBACK_DL_TX, 0, {0}, {0}, 0, 0, 0, 0, {0},
+	  LPASS_BE_VOICE_PLAYBACK_DL_TX},
 	{ VOICE2_PLAYBACK_TX, 0, {0}, {0}, 0, 0, 0, 0, {0},
 	  LPASS_BE_VOICE2_PLAYBACK_TX},
 	{ VOICE_RECORD_RX, 0, {0}, {0}, 0, 0, 0, 0, {0},
@@ -1656,7 +1658,8 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
 	if (set) {
 		if (!test_bit(val, &msm_bedais[reg].fe_sessions[0]) &&
 			((msm_bedais[reg].port_id == VOICE_PLAYBACK_TX) ||
-			(msm_bedais[reg].port_id == VOICE2_PLAYBACK_TX)))
+			(msm_bedais[reg].port_id == VOICE2_PLAYBACK_TX) ||
+			(msm_bedais[reg].port_id == VOICE_PLAYBACK_DL_TX)))
 			voc_start_playback(set, msm_bedais[reg].port_id);
 
 		set_bit(val, &msm_bedais[reg].fe_sessions[0]);
@@ -1755,7 +1758,8 @@ static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
 	} else {
 		if (test_bit(val, &msm_bedais[reg].fe_sessions[0]) &&
 			((msm_bedais[reg].port_id == VOICE_PLAYBACK_TX) ||
-			(msm_bedais[reg].port_id == VOICE2_PLAYBACK_TX)))
+			(msm_bedais[reg].port_id == VOICE2_PLAYBACK_TX) ||
+			(msm_bedais[reg].port_id == VOICE_PLAYBACK_DL_TX)))
 			voc_start_playback(set, msm_bedais[reg].port_id);
 		clear_bit(val, &msm_bedais[reg].fe_sessions[0]);
 		fdai = &fe_dai_map[val][session_type];
@@ -2869,9 +2873,9 @@ static const char *const be_name[] = {
 "ZERO", "PRI_I2S_RX", "PRI_I2S_TX", "SLIM_0_RX",
 "SLIM_0_TX", "HDMI_RX", "INT_BT_SCO_RX", "INT_BT_SCO_TX",
 "INT_FM_RX", "INT_FM_TX", "AFE_PCM_RX", "AFE_PCM_TX",
-"AUXPCM_RX", "AUXPCM_TX", "VOICE_PLAYBACK_TX", "VOICE2_PLAYBACK_TX",
-"INCALL_RECORD_RX", "INCALL_RECORD_TX", "MI2S_RX", "MI2S_TX",
-"SEC_I2S_RX", "SLIM_1_RX", "SLIM_1_TX", "SLIM_2_RX",
+"AUXPCM_RX", "AUXPCM_TX", "VOICE_PLAYBACK_TX", "VOICE_PLAYBACK_DL_TX",
+"VOICE2_PLAYBACK_TX", "INCALL_RECORD_RX", "INCALL_RECORD_TX", "MI2S_RX",
+"MI2S_TX", "SEC_I2S_RX", "SLIM_1_RX", "SLIM_1_TX", "SLIM_2_RX",
 "SLIM_2_TX", "SLIM_3_RX", "SLIM_3_TX", "SLIM_4_RX",
 "SLIM_4_TX", "SLIM_5_RX", "SLIM_5_TX", "SLIM_6_RX",
 "SLIM_6_TX", "SLIM_7_RX", "SLIM_7_TX", "SLIM_8_RX",
@@ -5212,12 +5216,38 @@ static const struct snd_kcontrol_new incall_music_delivery_mixer_controls[] = {
 	MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
 	MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_DOUBLE_EXT("MultiMedia7", SND_SOC_NOPM,
+	MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 	SOC_DOUBLE_EXT("MultiMedia9", SND_SOC_NOPM,
 	MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
 	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
 };
 
+	/* incall music downlink delivery mixer */
+static const struct snd_kcontrol_new incall_music_dl_delivery_mixer_controls[] = {
+	SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia5", MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia7", MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia9", MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
+};
+
 static const struct snd_kcontrol_new incall_music2_delivery_mixer_controls[] = {
 	SOC_DOUBLE_EXT("MultiMedia1", SND_SOC_NOPM,
 	MSM_BACKEND_DAI_VOICE2_PLAYBACK_TX,
@@ -5231,6 +5261,10 @@ static const struct snd_kcontrol_new incall_music2_delivery_mixer_controls[] = {
 	MSM_BACKEND_DAI_VOICE2_PLAYBACK_TX,
 	MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
 	msm_routing_put_audio_mixer),
+	SOC_DOUBLE_EXT("MultiMedia7", SND_SOC_NOPM,
+	MSM_BACKEND_DAI_VOICE2_PLAYBACK_TX,
+	MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+	msm_routing_put_audio_mixer),
 	SOC_DOUBLE_EXT("MultiMedia9", SND_SOC_NOPM,
 	MSM_BACKEND_DAI_VOICE2_PLAYBACK_TX,
 	MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer,
@@ -14276,7 +14310,7 @@ static int msm_voc_session_id_get(struct snd_kcontrol *kcontrol,
 
 static struct snd_kcontrol_new msm_voc_session_controls[] = {
 	SOC_SINGLE_MULTI_EXT("Voc VSID", SND_SOC_NOPM, 0,
-			     0xFFFFFFFF, 0, 1, msm_voc_session_id_get,
+			     0x7FFFFFFF, 0, 1, msm_voc_session_id_get,
 			     msm_voc_session_id_put),
 };
 
@@ -15505,6 +15539,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 	/* incall */
 	SND_SOC_DAPM_AIF_OUT("VOICE_PLAYBACK_TX", "Voice Farend Playback",
 				0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("VOICE_PLAYBACK_DL_TX", "Voice Downlink Playback",
+				0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("VOICE2_PLAYBACK_TX", "Voice2 Farend Playback",
 				0, 0, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback",
@@ -15783,6 +15819,9 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 	SND_SOC_DAPM_MIXER("Incall_Music Audio Mixer", SND_SOC_NOPM, 0, 0,
 	incall_music_delivery_mixer_controls,
 	ARRAY_SIZE(incall_music_delivery_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Incall_Music_DL Audio Mixer", SND_SOC_NOPM, 0, 0,
+	incall_music_dl_delivery_mixer_controls,
+	ARRAY_SIZE(incall_music_dl_delivery_mixer_controls)),
 	SND_SOC_DAPM_MIXER("Incall_Music_2 Audio Mixer", SND_SOC_NOPM, 0, 0,
 	incall_music2_delivery_mixer_controls,
 	ARRAY_SIZE(incall_music2_delivery_mixer_controls)),
@@ -16271,8 +16310,16 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Incall_Music Audio Mixer", "MultiMedia2", "MM_DL2"},
 	{"Incall_Music Audio Mixer", "MultiMedia4", "MM_DL4"},
 	{"Incall_Music Audio Mixer", "MultiMedia5", "MM_DL5"},
+	{"Incall_Music Audio Mixer", "MultiMedia7", "MM_DL7"},
 	{"Incall_Music Audio Mixer", "MultiMedia9", "MM_DL9"},
 	{"VOICE_PLAYBACK_TX", NULL, "Incall_Music Audio Mixer"},
+	{"Incall_Music_DL Audio Mixer", "MultiMedia1", "MM_DL1"},
+	{"Incall_Music_DL Audio Mixer", "MultiMedia2", "MM_DL2"},
+	{"Incall_Music_DL Audio Mixer", "MultiMedia4", "MM_DL4"},
+	{"Incall_Music_DL Audio Mixer", "MultiMedia5", "MM_DL5"},
+	{"Incall_Music_DL Audio Mixer", "MultiMedia7", "MM_DL7"},
+	{"Incall_Music_DL Audio Mixer", "MultiMedia9", "MM_DL9"},
+	{"VOICE_PLAYBACK_DL_TX", NULL, "Incall_Music_DL Audio Mixer"},
 	{"Incall_Music_2 Audio Mixer", "MultiMedia1", "MM_DL1"},
 	{"Incall_Music_2 Audio Mixer", "MultiMedia2", "MM_DL2"},
 	{"Incall_Music_2 Audio Mixer", "MultiMedia5", "MM_DL5"},
@@ -18805,6 +18852,7 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"BE_OUT", NULL, "PCM_RX"},
 	{"BE_OUT", NULL, "SLIMBUS_3_RX"},
 	{"BE_OUT", NULL, "VOICE_PLAYBACK_TX"},
+	{"BE_OUT", NULL, "VOICE_PLAYBACK_DL_TX"},
 	{"BE_OUT", NULL, "VOICE2_PLAYBACK_TX"},
 	{"BE_OUT", NULL, "PRI_TDM_RX_0"},
 	{"BE_OUT", NULL, "PRI_TDM_RX_1"},
diff --git a/asoc/msm-pcm-routing-v2.h b/asoc/msm-pcm-routing-v2.h
index e25c256e..7ade05d5 100644
--- a/asoc/msm-pcm-routing-v2.h
+++ b/asoc/msm-pcm-routing-v2.h
@@ -43,6 +43,7 @@
 #define LPASS_BE_QUIN_AUXPCM_RX "QUIN_AUX_PCM_RX"
 #define LPASS_BE_QUIN_AUXPCM_TX "QUIN_AUX_PCM_TX"
 #define LPASS_BE_VOICE_PLAYBACK_TX "VOICE_PLAYBACK_TX"
+#define LPASS_BE_VOICE_PLAYBACK_DL_TX "VOICE_PLAYBACK_DL_TX"
 #define LPASS_BE_VOICE2_PLAYBACK_TX "VOICE2_PLAYBACK_TX"
 #define LPASS_BE_INCALL_RECORD_RX "INCALL_RECORD_RX"
 #define LPASS_BE_INCALL_RECORD_TX "INCALL_RECORD_TX"
@@ -254,6 +255,7 @@ enum {
 	MSM_BACKEND_DAI_AUXPCM_RX,
 	MSM_BACKEND_DAI_AUXPCM_TX,
 	MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+	MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
 	MSM_BACKEND_DAI_VOICE2_PLAYBACK_TX,
 	MSM_BACKEND_DAI_INCALL_RECORD_RX,
 	MSM_BACKEND_DAI_INCALL_RECORD_TX,
diff --git a/asoc/sdxpoorwills.c b/asoc/sdxpoorwills.c
index a2dbfd5d..216b6100 100644
--- a/asoc/sdxpoorwills.c
+++ b/asoc/sdxpoorwills.c
@@ -2773,12 +2773,30 @@ static struct snd_soc_dai_link sdx_common_dai_links[] = {
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.codec_name = "snd-soc-dummy",
 	},
+	{
+	/*hw:x,33*/
+		.name = SDX_DAILINK_NAME(Compress2),
+		.stream_name = "COMPR2",
+		.cpu_dai_name = "MultiMedia7",
+		.platform_name = "msm-compress-dsp",
+		.dynamic = 1,
+		.async_ops = ASYNC_DPCM_SND_SOC_HW_PARAMS,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST,
+			SND_SOC_DPCM_TRIGGER_POST},
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.id = MSM_FRONTEND_DAI_MULTIMEDIA7,
+	},
 
 };
 
 static struct snd_soc_dai_link sdx_common_misc_fe_dai_links[] = {
 	{
-	/*hw:x,33*/
+	/*hw:x,34*/
 		.name = SDX_DAILINK_NAME(ASM Loopback),
 		.stream_name = "MultiMedia6",
 		.cpu_dai_name = "MultiMedia6",
@@ -2884,6 +2902,21 @@ static struct snd_soc_dai_link sdx_common_be_dai_links[] = {
 		.ignore_suspend = 1,
 		.ignore_pmdown_time = 1,
 	},
+	/* Incall Music BACK END DAI Link */
+	{
+		.name = LPASS_BE_VOICE_PLAYBACK_DL_TX,
+		.stream_name = "Voice Downlink Playback",
+		.cpu_dai_name = "msm-dai-q6-dev.32774",
+		.platform_name = "msm-pcm-routing",
+		.codec_name = "msm-stub-codec.1",
+		.codec_dai_name = "msm-stub-rx",
+		.no_pcm = 1,
+		.dpcm_playback = 1,
+		.id = MSM_BACKEND_DAI_VOICE_PLAYBACK_DL_TX,
+		.be_hw_params_fixup = sdx_be_hw_params_fixup,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+	},
 };
 
 static struct snd_soc_dai_link sdx_mi2s_be_dai_links[] = {
diff --git a/dsp/q6afe.c b/dsp/q6afe.c
index 2d3447a1..ee8f0d6a 100644
--- a/dsp/q6afe.c
+++ b/dsp/q6afe.c
@@ -515,6 +515,7 @@ int afe_get_port_type(u16 port_id)
 	case INT_BT_A2DP_RX:
 	case INT_FM_RX:
 	case VOICE_PLAYBACK_TX:
+	case VOICE_PLAYBACK_DL_TX:
 	case VOICE2_PLAYBACK_TX:
 	case RT_PROXY_PORT_001_RX:
 	case AUDIO_PORT_ID_I2S_RX:
@@ -715,6 +716,7 @@ int afe_sizeof_cfg_cmd(u16 port_id)
 		ret_size = SIZEOF_CFG_CMD(afe_param_id_slimbus_cfg);
 		break;
 	case VOICE_PLAYBACK_TX:
+	case VOICE_PLAYBACK_DL_TX:
 	case VOICE2_PLAYBACK_TX:
 	case VOICE_RECORD_RX:
 	case VOICE_RECORD_TX:
@@ -3491,6 +3493,7 @@ static int __afe_port_start(u16 port_id, union afe_port_config *afe_config,
 		cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
 		break;
 	case VOICE_PLAYBACK_TX:
+	case VOICE_PLAYBACK_DL_TX:
 	case VOICE2_PLAYBACK_TX:
 	case VOICE_RECORD_RX:
 	case VOICE_RECORD_TX:
@@ -3713,6 +3716,7 @@ int afe_get_port_index(u16 port_id)
 	case VOICE_RECORD_RX: return IDX_VOICE_RECORD_RX;
 	case VOICE_RECORD_TX: return IDX_VOICE_RECORD_TX;
 	case VOICE_PLAYBACK_TX: return IDX_VOICE_PLAYBACK_TX;
+	case VOICE_PLAYBACK_DL_TX: return IDX_VOICE_RECORD_RX;
 	case VOICE2_PLAYBACK_TX: return IDX_VOICE2_PLAYBACK_TX;
 	case SLIMBUS_0_RX: return IDX_SLIMBUS_0_RX;
 	case SLIMBUS_0_TX: return IDX_SLIMBUS_0_TX;
@@ -5725,6 +5729,7 @@ int afe_validate_port(u16 port_id)
 	case VOICE_RECORD_RX:
 	case VOICE_RECORD_TX:
 	case VOICE_PLAYBACK_TX:
+	case VOICE_PLAYBACK_DL_TX:
 	case VOICE2_PLAYBACK_TX:
 	case SLIMBUS_0_RX:
 	case SLIMBUS_0_TX:
diff --git a/dsp/q6audio-v2.c b/dsp/q6audio-v2.c
index 088d9b27..e0d09688 100644
--- a/dsp/q6audio-v2.c
+++ b/dsp/q6audio-v2.c
@@ -57,6 +57,7 @@ int q6audio_get_port_index(u16 port_id)
 	case VOICE_RECORD_RX: return IDX_VOICE_RECORD_RX;
 	case VOICE_RECORD_TX: return IDX_VOICE_RECORD_TX;
 	case VOICE_PLAYBACK_TX: return IDX_VOICE_PLAYBACK_TX;
+	case VOICE_PLAYBACK_DL_TX: return IDX_VOICE_RECORD_RX;
 	case VOICE2_PLAYBACK_TX: return IDX_VOICE2_PLAYBACK_TX;
 	case SLIMBUS_0_RX: return IDX_SLIMBUS_0_RX;
 	case SLIMBUS_0_TX: return IDX_SLIMBUS_0_TX;
@@ -344,6 +345,7 @@ int q6audio_get_port_id(u16 port_id)
 	case VOICE_RECORD_RX: return AFE_PORT_ID_VOICE_RECORD_RX;
 	case VOICE_RECORD_TX: return AFE_PORT_ID_VOICE_RECORD_TX;
 	case VOICE_PLAYBACK_TX: return AFE_PORT_ID_VOICE_PLAYBACK_TX;
+	case VOICE_PLAYBACK_DL_TX: return AFE_PORT_ID_VOICE_RECORD_RX;
 	case VOICE2_PLAYBACK_TX: return AFE_PORT_ID_VOICE2_PLAYBACK_TX;
 	case SLIMBUS_0_RX: return AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX;
 	case SLIMBUS_0_TX: return AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX;
@@ -779,6 +781,7 @@ int q6audio_validate_port(u16 port_id)
 	case VOICE_RECORD_RX:
 	case VOICE_RECORD_TX:
 	case VOICE_PLAYBACK_TX:
+	case VOICE_PLAYBACK_DL_TX:
 	case VOICE2_PLAYBACK_TX:
 	case SLIMBUS_0_RX:
 	case SLIMBUS_0_TX:
diff --git a/dsp/q6voice.c b/dsp/q6voice.c
index 1731831c..8b6a4d78 100644
--- a/dsp/q6voice.c
+++ b/dsp/q6voice.c
@@ -6093,7 +6093,7 @@ int voc_start_playback(uint32_t set, uint16_t port_id)
 	voice_itr_init(&itr, ALL_SESSION_VSID);
 	while (voice_itr_get_next_session(&itr, &v)) {
 		if ((v != NULL) &&
-		    (((port_id == VOICE_PLAYBACK_TX) &&
+		    ((((port_id == VOICE_PLAYBACK_TX) || (port_id == VOICE_PLAYBACK_DL_TX))&&
 		       is_sub1_vsid(v->session_id)) ||
 		     ((port_id == VOICE2_PLAYBACK_TX) &&
 		       is_sub2_vsid(v->session_id)))) {
diff --git a/include/dsp/apr_audio-v2.h b/include/dsp/apr_audio-v2.h
index 7e0dfeec..2a925d05 100644
--- a/include/dsp/apr_audio-v2.h
+++ b/include/dsp/apr_audio-v2.h
@@ -946,6 +946,7 @@ struct adm_cmd_connect_afe_port_v5 {
 #define VOICE_RECORD_RX 0x8003
 #define VOICE_RECORD_TX 0x8004
 #define VOICE_PLAYBACK_TX 0x8005
+#define VOICE_PLAYBACK_DL_TX 0x8006
 
 /* Slimbus Multi channel port id pool  */
 #define SLIMBUS_0_RX		0x4000