diff --git a/hal/Android.mk b/hal/Android.mk
index 6db1f98..72042d1 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -69,6 +69,10 @@
     LOCAL_CFLAGS += -DANC_HEADSET_ENABLED
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_VBAT_MONITOR)),true)
+    LOCAL_CFLAGS += -DVBAT_MONITOR_ENABLED
+endif
+
 ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FLUENCE)),true)
     LOCAL_CFLAGS += -DFLUENCE_ENABLED
 endif
@@ -227,10 +231,6 @@
     LOCAL_SRC_FILES += audio_extn/source_track.c
 endif
 
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUDIOSPHERE)),true)
-    LOCAL_CFLAGS += -DAUDIOSPHERE_ENABLED
-endif
-
 LOCAL_SHARED_LIBRARIES := \
 	liblog \
 	libcutils \
@@ -256,6 +256,14 @@
     LOCAL_SRC_FILES += audio_extn/listen.c
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_EXT_HDMI)),true)
+    LOCAL_CFLAGS += -DAUDIO_EXTERNAL_HDMI_ENABLED
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_PASSTHROUGH)),true)
+    LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-parsers
+    LOCAL_SHARED_LIBRARIES += libaudioparsers
+endif
+endif
+
 ifeq ($(strip $(BOARD_SUPPORTS_SOUND_TRIGGER)),true)
     LOCAL_CFLAGS += -DSOUND_TRIGGER_ENABLED
     LOCAL_CFLAGS += -DSOUND_TRIGGER_PLATFORM_NAME=$(TARGET_BOARD_PLATFORM)
diff --git a/hal/audio_extn/audio_defs.h b/hal/audio_extn/audio_defs.h
index 96a7a19..1c5da54 100644
--- a/hal/audio_extn/audio_defs.h
+++ b/hal/audio_extn/audio_defs.h
@@ -72,6 +72,8 @@
 #define AUDIO_OFFLOAD_CODEC_APE_SAMPLE_RATE "music_offload_ape_sample_rate"
 #define AUDIO_OFFLOAD_CODEC_APE_SEEK_TABLE_PRESENT "music_offload_seek_table_present"
 
+#define AUDIO_OFFLOAD_CODEC_VORBIS_BITSTREAM_FMT "music_offload_vorbis_bitstream_fmt"
+
 /* Query handle fm parameter*/
 #define AUDIO_PARAMETER_KEY_HANDLE_FM "handle_fm"
 
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 67d30ca..39ad4d1 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -62,6 +62,7 @@
     bool custom_stereo_enabled;
     uint32_t proxy_channel_num;
     bool hpx_enabled;
+    bool vbat_enabled;
 };
 
 static struct audio_extn_module aextnmod = {
@@ -70,6 +71,7 @@
     .custom_stereo_enabled = 0,
     .proxy_channel_num = 2,
     .hpx_enabled = 0,
+    .vbat_enabled = 0,
 };
 
 #define AUDIO_PARAMETER_KEY_ANC        "anc_enabled"
@@ -79,8 +81,6 @@
 /* Query offload playback instances count */
 #define AUDIO_PARAMETER_OFFLOAD_NUM_ACTIVE "offload_num_active"
 #define AUDIO_PARAMETER_HPX            "HPX"
-#define AUDIO_PARAMETER_KEY_ASPHERE_ENABLE   "asphere_enable"
-#define AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH "asphere_strength"
 
 #ifndef FM_ENABLED
 #define audio_extn_fm_set_parameters(adev, parms) (0)
@@ -216,6 +216,27 @@
 }
 #endif
 
+#ifdef VBAT_MONITOR_ENABLED
+bool audio_extn_is_vbat_enabled(void)
+{
+    ALOGD("%s: status: %d", __func__, aextnmod.vbat_enabled);
+    return (aextnmod.vbat_enabled ? true: false);
+}
+
+bool audio_extn_can_use_vbat(void)
+{
+    char prop_vbat_enabled[PROPERTY_VALUE_MAX] = "false";
+
+    property_get("persist.audio.vbat.enabled", prop_vbat_enabled, "0");
+    if (!strncmp("true", prop_vbat_enabled, 4)) {
+        aextnmod.vbat_enabled = 1;
+    }
+
+    ALOGD("%s: vbat.enabled property is set to %s", __func__, prop_vbat_enabled);
+    return (aextnmod.vbat_enabled ? true: false);
+}
+#endif
+
 #ifndef ANC_HEADSET_ENABLED
 #define audio_extn_set_anc_parameters(adev, parms)       (0)
 #else
@@ -535,103 +556,6 @@
     return ret;
 }
 
-#ifndef AUDIOSPHERE_ENABLED
-#define audio_extn_asphere_set_parameters(adev, parms)  (0)
-#define audio_extn_asphere_get_parameters(adev, query, reply) (0)
-#else
-int32_t audio_extn_asphere_set_parameters(const struct audio_device *adev,
-                                     struct str_parms *parms)
-{
-    int ret = 0, val[2];
-    char value[32] = {0};
-    int set_enable, set_strength;
-    int enable = -1, strength = -1;
-    struct mixer_ctl *ctl = NULL;
-    const char *mixer_ctl_name = "MSM ASphere Set Param";
-    char propValue[PROPERTY_VALUE_MAX] = {0};
-    bool asphere_prop_enabled = false;
-
-    if (property_get("audio.pp.asphere.enabled", propValue, "false")) {
-        if (!strncmp("true", propValue, 4))
-            asphere_prop_enabled = true;
-    }
-
-    if (!asphere_prop_enabled) {
-        ALOGV("%s: property not set!!! not doing anything", __func__);
-        return ret;
-    }
-
-    set_enable = str_parms_get_str(parms,
-                            AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
-                            value, sizeof(value));
-    if (set_enable > 0)
-        enable = atoi(value);
-
-    set_strength = str_parms_get_str(parms,
-                            AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
-                            value, sizeof(value));
-    if (set_strength > 0)
-        strength = atoi(value);
-
-    if (set_enable >= 0 || set_strength >= 0) {
-        ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
-        if (!ctl) {
-            ALOGE("%s: could not get ctl for mixer cmd - %s",
-                  __func__, mixer_ctl_name);
-            return -EINVAL;
-        }
-        ALOGD("%s: set ctl \"%s:%d,%d\"",
-              __func__, mixer_ctl_name, enable, strength);
-        val[0] = enable;
-        val[1] = strength;
-        ret = mixer_ctl_set_array(ctl, val, sizeof(val)/sizeof(val[0]));
-        if (ret)
-            ALOGE("%s: set ctl failed!!!\"%s:%d,%d\"",
-                  __func__, mixer_ctl_name, enable, strength);
-    }
-    ALOGV("%s: exit ret %d", __func__, ret);
-    return ret;
-}
-
-int32_t audio_extn_asphere_get_parameters(const struct audio_device *adev,
-                                          struct str_parms *query,
-                                          struct str_parms *reply)
-{
-    int ret = 0, val[2] = {-1, -1};
-    char value[32] = {0};
-    int get_enable, get_strength;
-    struct mixer_ctl *ctl = NULL;
-    const char *mixer_ctl_name = "MSM ASphere Set Param";
-
-    get_enable = str_parms_get_str(query,
-                                   AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
-                                   value, sizeof(value));
-    get_strength = str_parms_get_str(query,
-                                     AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
-                                     value, sizeof(value));
-    if (get_enable > 0 || get_strength > 0) {
-        ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
-        if (!ctl) {
-            ALOGE("%s: could not get ctl for mixer cmd - %s",
-                  __func__, mixer_ctl_name);
-            return -EINVAL;
-        }
-        ret = mixer_ctl_get_array(ctl, val, sizeof(val)/sizeof(val[0]));
-        if (ret)
-            ALOGE("%s: got ctl failed!!! \"%s:%d,%d\"",
-                   __func__, mixer_ctl_name, val[0], val[1]);
-        if (get_enable > 0)
-            str_parms_add_int(reply,
-                              AUDIO_PARAMETER_KEY_ASPHERE_ENABLE, val[0]);
-        if (get_strength > 0)
-            str_parms_add_int(reply,
-                              AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH, val[1]);
-    }
-    ALOGV("%s: exit ret %d", __func__, ret);
-    return ret;
-}
-#endif
-
 void audio_extn_set_parameters(struct audio_device *adev,
                                struct str_parms *parms)
 {
@@ -649,7 +573,8 @@
    audio_extn_hpx_set_parameters(adev, parms);
    audio_extn_pm_set_parameters(parms);
    audio_extn_source_track_set_parameters(adev, parms);
-   audio_extn_asphere_set_parameters(adev, parms);
+   if (adev->offload_effects_set_parameters != NULL)
+       adev->offload_effects_set_parameters(parms);
 }
 
 void audio_extn_get_parameters(const struct audio_device *adev,
@@ -663,7 +588,8 @@
     audio_extn_dts_eagle_get_parameters(adev, query, reply);
     audio_extn_hpx_get_parameters(query, reply);
     audio_extn_source_track_get_parameters(adev, query, reply);
-    audio_extn_asphere_get_parameters(adev, query, reply);
+    if (adev->offload_effects_get_parameters != NULL)
+        adev->offload_effects_get_parameters(query, reply);
 
     kv_pairs = str_parms_to_str(reply);
     ALOGD_IF(kv_pairs != NULL, "%s: returns %s", __func__, kv_pairs);
@@ -682,25 +608,23 @@
     if (out->format == AUDIO_FORMAT_FLAC) {
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FLAC_MIN_BLK_SIZE, value, sizeof(value));
         if (ret >= 0) {
-            out->gapless_mdata.min_blk_size =
             out->compr_config.codec->options.flac_dec.min_blk_size = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FLAC_MAX_BLK_SIZE, value, sizeof(value));
         if (ret >= 0) {
-            out->gapless_mdata.max_blk_size =
             out->compr_config.codec->options.flac_dec.max_blk_size = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FLAC_MIN_FRAME_SIZE, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.flac_dec.min_frame_size = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FLAC_MAX_FRAME_SIZE, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.flac_dec.max_frame_size = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ALOGV("FLAC metadata: min_blk_size %d, max_blk_size %d min_frame_size %d max_frame_size %d",
               out->compr_config.codec->options.flac_dec.min_blk_size,
@@ -713,63 +637,63 @@
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_FRAME_LENGTH, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.frame_length = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_COMPATIBLE_VERSION, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.compatible_version = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_BIT_DEPTH, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.bit_depth = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_PB, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.pb = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_MB, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.mb = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
 
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_KB, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.kb = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_NUM_CHANNELS, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.num_channels = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_MAX_RUN, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.max_run = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_MAX_FRAME_BYTES, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.max_frame_bytes = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_AVG_BIT_RATE, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.avg_bit_rate = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_SAMPLING_RATE, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.sample_rate = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_ALAC_CHANNEL_LAYOUT_TAG, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.alac.channel_layout_tag = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ALOGV("ALAC CSD values: frameLength %d bitDepth %d numChannels %d"
                 " maxFrameBytes %d, avgBitRate %d, sampleRate %d",
@@ -785,52 +709,52 @@
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_COMPATIBLE_VERSION, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.compatible_version = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_COMPRESSION_LEVEL, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.compression_level = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_FORMAT_FLAGS, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.format_flags = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_BLOCKS_PER_FRAME, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.blocks_per_frame = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_FINAL_FRAME_BLOCKS, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.final_frame_blocks = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_TOTAL_FRAMES, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.total_frames = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_BITS_PER_SAMPLE, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.bits_per_sample = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_NUM_CHANNELS, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.num_channels = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_SAMPLE_RATE, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.sample_rate = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_APE_SEEK_TABLE_PRESENT, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.ape.seek_table_present = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ALOGV("APE CSD values: compatibleVersion %d compressionLevel %d"
                 " formatFlags %d blocksPerFrame %d finalFrameBlocks %d"
@@ -849,49 +773,59 @@
     }
 
     else if (out->format == AUDIO_FORMAT_VORBIS) {
+        ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_VORBIS_BITSTREAM_FMT, value, sizeof(value));
+        if (ret >= 0) {
         // transcoded bitstream mode
-        out->compr_config.codec->options.vorbis_dec.bit_stream_fmt = 1;
-        out->send_new_metadata = 1;
+            out->compr_config.codec->options.vorbis_dec.bit_stream_fmt = (atoi(value) > 0) ? 1 : 0;
+            out->is_compr_metadata_avail = true;
+        }
     }
 
     else if (out->format == AUDIO_FORMAT_WMA || out->format == AUDIO_FORMAT_WMA_PRO) {
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_FORMAT_TAG, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->format = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
+        }
+        ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE, value, sizeof(value));
+        if (ret >= 0) {
+            out->compr_config.codec->options.wma.avg_bit_rate = atoi(value);
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_BLOCK_ALIGN, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.wma.super_block_align = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_BIT_PER_SAMPLE, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.wma.bits_per_sample = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_CHANNEL_MASK, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.wma.channelmask = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.wma.encodeopt = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION1, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.wma.encodeopt1 = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
         ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION2, value, sizeof(value));
         if (ret >= 0) {
             out->compr_config.codec->options.wma.encodeopt2 = atoi(value);
-            out->send_new_metadata = 1;
+            out->is_compr_metadata_avail = true;
         }
-        ALOGV("WMA params: fmt %x, balgn %x, sr %d, chmsk %x, encop %x, op1 %x, op2 %x",
+        ALOGV("WMA params: fmt %x, bit rate %x, balgn %x, sr %d, chmsk %x"
+                " encop %x, op1 %x, op2 %x",
                 out->compr_config.codec->format,
+                out->compr_config.codec->options.wma.avg_bit_rate,
                 out->compr_config.codec->options.wma.super_block_align,
                 out->compr_config.codec->options.wma.bits_per_sample,
                 out->compr_config.codec->options.wma.channelmask,
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 2bf19fc..3645cdb 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -103,7 +103,8 @@
 #endif
 
 #ifdef AUDIO_EXTN_FORMATS_ENABLED
-#define AUDIO_OUTPUT_BIT_WIDTH (config->offload_info.bit_width)
+#define AUDIO_OUTPUT_BIT_WIDTH ((config->offload_info.bit_width == 32) ? 24\
+                                   :config->offload_info.bit_width)
 #else
 #define AUDIO_OUTPUT_BIT_WIDTH (CODEC_BACKEND_DEFAULT_BIT_WIDTH)
 #endif
@@ -127,6 +128,14 @@
 bool audio_extn_should_use_handset_anc(int in_channels);
 #endif
 
+#ifndef VBAT_MONITOR_ENABLED
+#define audio_extn_is_vbat_enabled()                     (0)
+#define audio_extn_can_use_vbat()                        (0)
+#else
+bool audio_extn_is_vbat_enabled(void);
+bool audio_extn_can_use_vbat(void);
+#endif
+
 #ifndef FLUENCE_ENABLED
 #define audio_extn_set_fluence_parameters(adev, parms) (0)
 #define audio_extn_get_fluence_parameters(adev, query, reply) (0)
@@ -442,7 +451,8 @@
                                   uint32_t sample_rate,
                                   uint32_t bit_width,
                                   struct stream_app_type_cfg *app_type_cfg);
-int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase);
+int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
+                                       struct audio_usecase *usecase);
 void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
                                              struct audio_usecase *usecase);
 #ifdef DS2_DOLBY_DAP_ENABLED
@@ -500,4 +510,10 @@
 void audio_extn_perf_lock_acquire(void);
 void audio_extn_perf_lock_release(void);
 #endif /* KPI_OPTIMIZE_ENABLED */
+
+#ifndef AUDIO_EXTERNAL_HDMI_ENABLED
+#define audio_utils_set_hdmi_channel_status(out, buffer, bytes) (0)
+#else
+void audio_utils_set_hdmi_channel_status(struct stream_out *out, char * buffer, size_t bytes);
+#endif
 #endif /* AUDIO_EXTN_H */
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 78a7956..aab9f74 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -77,6 +77,7 @@
 #define WAIT_TIME_SPKR_CALIB (60 * 1000 * 1000)
 
 #define MIN_SPKR_IDLE_SEC (60 * 30)
+#define WAKEUP_MIN_IDLE_CHECK 30
 
 /*Once calibration is started sleep for 1 sec to allow
   the calibration to kick off*/
@@ -467,10 +468,16 @@
     uc_info_rx->type = PCM_PLAYBACK;
     uc_info_rx->in_snd_device = SND_DEVICE_NONE;
     uc_info_rx->stream.out = adev->primary_output;
-    uc_info_rx->out_snd_device = SND_DEVICE_OUT_SPEAKER_PROTECTED;
+    if (audio_extn_is_vbat_enabled())
+        uc_info_rx->out_snd_device = SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
+    else
+        uc_info_rx->out_snd_device = SND_DEVICE_OUT_SPEAKER_PROTECTED;
     disable_rx = true;
     list_add_tail(&adev->usecase_list, &uc_info_rx->list);
-    enable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED);
+    if (audio_extn_is_vbat_enabled())
+         enable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
+    else
+         enable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED);
     enable_audio_route(adev, uc_info_rx);
 
     pcm_dev_rx_id = platform_get_pcm_device_id(uc_info_rx->id, PCM_PLAYBACK);
@@ -581,7 +588,7 @@
                 }
                 break;
             } else if (status.status == -EAGAIN) {
-                  ALOGD("%s: spkr_prot_thread try again", __func__);
+                  ALOGV("%s: spkr_prot_thread try again", __func__);
                   usleep(WAIT_FOR_GET_CALIB_STATUS);
             } else {
                 ALOGE("%s: spkr_prot_thread get failed status %d",
@@ -629,7 +636,10 @@
         }
         if (disable_rx) {
             list_remove(&uc_info_rx->list);
-            disable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED);
+            if (audio_extn_is_vbat_enabled())
+                disable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
+            else
+                disable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED);
             disable_audio_route(adev, uc_info_rx);
         }
         if (disable_tx) {
@@ -758,18 +768,22 @@
             if (is_speaker_in_use(&sec)) {
                 ALOGV("%s: WSA Speaker in use retry calibration", __func__);
                 pthread_mutex_unlock(&adev->lock);
+                sleep(WAKEUP_MIN_IDLE_CHECK);
                 continue;
             } else {
+                ALOGD("%s: wsa speaker idle %ld,minimum time %ld", __func__, sec, min_idle_time);
                 if (sec < min_idle_time) {
                     pthread_mutex_unlock(&adev->lock);
+                    sleep(WAKEUP_MIN_IDLE_CHECK);
                     continue;
                }
-               ALOGV("%s: wsa speaker idle %ld min time %ld", __func__, sec, min_idle_time);
                goahead = true;
            }
            if (!list_empty(&adev->usecase_list)) {
                 ALOGD("%s: Usecase active re-try calibration", __func__);
-                goahead = false;
+                pthread_mutex_unlock(&adev->lock);
+                sleep(WAKEUP_MIN_IDLE_CHECK);
+                continue;
            }
            if (goahead) {
                if (spk_1_tzn >= 0) {
@@ -801,15 +815,17 @@
                        if (t0_spk_1 < TZ_TEMP_MIN_THRESHOLD ||
                            t0_spk_1 > TZ_TEMP_MAX_THRESHOLD) {
                            pthread_mutex_unlock(&adev->lock);
+                           sleep(WAKEUP_MIN_IDLE_CHECK);
                            continue;
                        }
+                       ALOGD("%s: temp T0 for spkr1 %d\n", __func__, t0_spk_1);
                        /*Convert temp into q6 format*/
                        t0_spk_1 = (t0_spk_1 * (1 << 6));
-                       ALOGD("%s: temp T0 for spkr1 %d\n", __func__, t0_spk_1);
                    } else {
                        ALOGV("%s: thermal equilibrium failed for spkr1 in %d/%d readings\n",
                                                 __func__, i, NUM_ATTEMPTS);
                        pthread_mutex_unlock(&adev->lock);
+                       sleep(WAKEUP_MIN_IDLE_CHECK);
                        continue;
                    }
                }
@@ -841,15 +857,17 @@
                        if (t0_spk_2 < TZ_TEMP_MIN_THRESHOLD ||
                            t0_spk_2 > TZ_TEMP_MAX_THRESHOLD) {
                            pthread_mutex_unlock(&adev->lock);
+                           sleep(WAKEUP_MIN_IDLE_CHECK);
                            continue;
                        }
+                       ALOGD("%s: temp T0 for spkr2 %d\n", __func__, t0_spk_2);
                        /*Convert temp into q6 format*/
                        t0_spk_2 = (t0_spk_2 * (1 << 6));
-                       ALOGD("%s: temp T0 for spkr2 %d\n", __func__, t0_spk_2);
                    } else {
                        ALOGV("%s: thermal equilibrium failed for spkr2 in %d/%d readings\n",
                                                 __func__, i, NUM_ATTEMPTS);
                        pthread_mutex_unlock(&adev->lock);
+                       sleep(WAKEUP_MIN_IDLE_CHECK);
                        continue;
                    }
                }
@@ -883,19 +901,22 @@
         if (is_speaker_in_use(&sec)) {
             ALOGV("%s: Speaker in use retry calibration", __func__);
             pthread_mutex_unlock(&adev->lock);
+            sleep(WAKEUP_MIN_IDLE_CHECK);
             continue;
         } else {
             if (sec < min_idle_time) {
                 pthread_mutex_unlock(&adev->lock);
+                sleep(WAKEUP_MIN_IDLE_CHECK);
                 continue;
             }
-            ALOGD("%s: speaker idle %ld min time %ld", __func__, sec, min_idle_time);
             goahead = true;
         }
         if (!list_empty(&adev->usecase_list)) {
             ALOGD("%s: Usecase active re-try calibration", __func__);
             goahead = false;
             pthread_mutex_unlock(&adev->lock);
+            sleep(WAKEUP_MIN_IDLE_CHECK);
+            continue;
         }
         if (goahead) {
                 int status;
@@ -1075,6 +1096,12 @@
     case SND_DEVICE_OUT_VOICE_SPEAKER:
         acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
         break;
+    case SND_DEVICE_OUT_SPEAKER_VBAT:
+        acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
+        break;
+    case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
+        acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
+        break;
     default:
         acdb_id = -EINVAL;
         break;
@@ -1092,6 +1119,10 @@
         return SND_DEVICE_OUT_SPEAKER_PROTECTED;
     case SND_DEVICE_OUT_VOICE_SPEAKER:
         return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+    case SND_DEVICE_OUT_SPEAKER_VBAT:
+        return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
+    case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
+        return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
     default:
         return snd_device;
     }
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 24dee64..38dbb46 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -35,6 +35,12 @@
 #include "audio_extn.h"
 #include "voice.h"
 
+#ifdef AUDIO_EXTERNAL_HDMI_ENABLED
+#ifdef HDMI_PASSTHROUGH_ENABLED
+#include "audio_parsers.h"
+#endif
+#endif
+
 #define AUDIO_OUTPUT_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_output_policy.conf"
 
 #define OUTPUTS_TAG "outputs"
@@ -52,6 +58,21 @@
 #define BASE_TABLE_SIZE 64
 #define MAX_BASEINDEX_LEN 256
 
+#ifdef AUDIO_EXTERNAL_HDMI_ENABLED
+#define PROFESSIONAL        (1<<0)      /* 0 = consumer, 1 = professional */
+#define NON_LPCM            (1<<1)      /* 0 = audio, 1 = non-audio */
+#define SR_44100            (0<<0)      /* 44.1kHz */
+#define SR_NOTID            (1<<0)      /* non indicated */
+#define SR_48000            (2<<0)      /* 48kHz */
+#define SR_32000            (3<<0)      /* 32kHz */
+#define SR_22050            (4<<0)      /* 22.05kHz */
+#define SR_24000            (6<<0)      /* 24kHz */
+#define SR_88200            (8<<0)      /* 88.2kHz */
+#define SR_96000            (10<<0)     /* 96kHz */
+#define SR_176400           (12<<0)     /* 176.4kHz */
+#define SR_192000           (14<<0)     /* 192kHz */
+
+#endif
 struct string_to_enum {
     const char *name;
     uint32_t value;
@@ -504,14 +525,11 @@
     app_type_cfg->bit_width = 16;
 }
 
-#define GENERAL_RECORDING_APP_TYPE  0x11132
-int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase)
+int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
+                                       struct audio_usecase *usecase)
 {
     char mixer_ctl_name[MAX_LENGTH_MIXER_CONTROL_IN_INT];
     int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT], len = 0, rc;
-    struct stream_out *out;
-    struct stream_in *in;
-    struct audio_device *adev;
     struct mixer_ctl *ctl;
     int pcm_device_id, acdb_dev_id, snd_device = usecase->out_snd_device;
     int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
@@ -528,21 +546,16 @@
         (usecase->id != USECASE_AUDIO_PLAYBACK_MULTI_CH) &&
         (!is_offload_usecase(usecase->id)) &&
         (usecase->type != PCM_CAPTURE)) {
-        ALOGV("%s: a playback path where app type cfg is not required %d", __func__, usecase->id);
+        ALOGV("%s: a rx/tx path where app type cfg is not required %d", __func__, usecase->id);
         rc = 0;
         goto exit_send_app_type_cfg;
     }
     if (usecase->type == PCM_PLAYBACK) {
-        out = usecase->stream.out;
-        adev = out->dev;
         snd_device = usecase->out_snd_device;
-        pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
-    }
-    if (usecase->type == PCM_CAPTURE) {
-        in = usecase->stream.in;
-        adev = in->dev;
+        pcm_device_id = platform_get_pcm_device_id(usecase->id, PCM_PLAYBACK);
+    } else if (usecase->type == PCM_CAPTURE) {
         snd_device = usecase->in_snd_device;
-        pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
+        pcm_device_id = platform_get_pcm_device_id(usecase->id, PCM_CAPTURE);
     }
 
     snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
@@ -564,7 +577,14 @@
         goto exit_send_app_type_cfg;
     }
 
-    if (usecase->type == PCM_PLAYBACK) {
+    if ((usecase->type == PCM_PLAYBACK) && (usecase->stream.out == NULL)) {
+        sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        app_type_cfg[len++] = platform_get_default_app_type(adev->platform);
+        app_type_cfg[len++] = acdb_dev_id;
+        app_type_cfg[len++] = sample_rate;
+        ALOGI("%s PLAYBACK app_type %d, acdb_dev_id %d, sample_rate %d",
+              __func__, platform_get_default_app_type(adev->platform), acdb_dev_id, sample_rate);
+    } else if (usecase->type == PCM_PLAYBACK) {
         if ((24 == usecase->stream.out->bit_width) &&
             (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
             sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
@@ -573,13 +593,13 @@
             (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
             sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
         } else {
-            sample_rate = out->app_type_cfg.sample_rate;
+            sample_rate = usecase->stream.out->app_type_cfg.sample_rate;
         }
 
-        app_type_cfg[len++] = out->app_type_cfg.app_type;
+        app_type_cfg[len++] = usecase->stream.out->app_type_cfg.app_type;
         app_type_cfg[len++] = acdb_dev_id;
-        if (((out->format == AUDIO_FORMAT_E_AC3) ||
-            (out->format == AUDIO_FORMAT_E_AC3_JOC))
+        if (((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+            (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
 #ifdef HDMI_PASSTHROUGH_ENABLED
             && (out->flags  & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)
 #endif
@@ -588,10 +608,9 @@
         else
             app_type_cfg[len++] = sample_rate;
         ALOGI("%s PLAYBACK app_type %d, acdb_dev_id %d, sample_rate %d",
-              __func__, out->app_type_cfg.app_type, acdb_dev_id, sample_rate);
-    }
-    else if (usecase->type == PCM_CAPTURE) {
-        app_type_cfg[len++] = GENERAL_RECORDING_APP_TYPE;
+              __func__, usecase->stream.out->app_type_cfg.app_type, acdb_dev_id, sample_rate);
+    } else if (usecase->type == PCM_CAPTURE) {
+        app_type_cfg[len++] = platform_get_default_app_type(adev->platform);
         app_type_cfg[len++] = acdb_dev_id;
         app_type_cfg[len++] = sample_rate;
         ALOGI("%s CAPTURE app_type %d, acdb_dev_id %d, sample_rate %d",
@@ -767,3 +786,136 @@
     outp[k] = '\0';
     return k;
 }
+
+#ifdef AUDIO_EXTERNAL_HDMI_ENABLED
+
+void get_default_compressed_channel_status(
+                                  unsigned char *channel_status)
+{
+     int32_t status = 0;
+     unsigned char bit_index;
+     memset(channel_status,0,24);
+
+     /* block start bit in preamble bit 3 */
+     channel_status[0] |= PROFESSIONAL;
+     //compre out
+     channel_status[0] |= NON_LPCM;
+     // sample rate; fixed 48K for default/transcode
+     channel_status[3] |= SR_48000;
+}
+
+#ifdef HDMI_PASSTHROUGH_ENABLED
+int32_t get_compressed_channel_status(void *audio_stream_data,
+                                                   uint32_t audio_frame_size,
+                                                   unsigned char *channel_status,
+                                                   enum audio_parser_code_type codec_type)
+                                                   // codec_type - AUDIO_PARSER_CODEC_AC3
+                                                   //            - AUDIO_PARSER_CODEC_DTS
+{
+     unsigned char *stream;
+     int ret = 0;
+     stream = (unsigned char *)audio_stream_data;
+
+     if (audio_stream_data == NULL || audio_frame_size == 0) {
+         ALOGW("no buffer to get channel status, return default for compress");
+         get_default_compressed_channel_status(channel_status);
+         return ret;
+     }
+
+     memset(channel_status,0,24);
+     if(init_audio_parser(stream, audio_frame_size, codec_type) == -1)
+     {
+         ALOGE("init audio parser failed");
+         return -1;
+     }
+     ret = get_channel_status(channel_status, codec_type);
+     return ret;
+
+}
+
+#endif
+
+void get_lpcm_channel_status(uint32_t sampleRate,
+                                                  unsigned char *channel_status)
+{
+     int32_t status = 0;
+     unsigned char bit_index;
+     memset(channel_status,0,24);
+     /* block start bit in preamble bit 3 */
+     channel_status[0] |= PROFESSIONAL;
+     //LPCM OUT
+     channel_status[0] &= ~NON_LPCM;
+
+     switch (sampleRate) {
+         case 8000:
+         case 11025:
+         case 12000:
+         case 16000:
+         case 22050:
+             channel_status[3] |= SR_NOTID;
+         case 24000:
+             channel_status[3] |= SR_24000;
+             break;
+         case 32000:
+             channel_status[3] |= SR_32000;
+             break;
+         case 44100:
+             channel_status[3] |= SR_44100;
+             break;
+         case 48000:
+             channel_status[3] |= SR_48000;
+             break;
+         case 88200:
+             channel_status[3] |= SR_88200;
+             break;
+         case 96000:
+             channel_status[3] |= SR_96000;
+             break;
+         case 176400:
+             channel_status[3] |= SR_176400;
+             break;
+         case 192000:
+            channel_status[3] |= SR_192000;
+             break;
+         default:
+             ALOGV("Invalid sample_rate %u\n", sampleRate);
+             status = -1;
+             break;
+     }
+}
+
+void audio_utils_set_hdmi_channel_status(struct stream_out *out, char * buffer, size_t bytes)
+{
+    unsigned char channel_status[24]={0};
+    struct snd_aes_iec958 iec958;
+    const char *mixer_ctl_name = "IEC958 Playback PCM Stream";
+    struct mixer_ctl *ctl;
+    int i=0;
+#ifdef HDMI_PASSTHROUGH_ENABLED
+    if (audio_extn_is_dolby_format(out->format) &&
+        /*TODO:Extend code to support DTS passthrough*/
+        /*set compressed channel status bits*/
+        audio_extn_dolby_is_passthrough_stream(out->flags)){
+        get_compressed_channel_status(buffer, bytes, channel_status, AUDIO_PARSER_CODEC_AC3);
+    } else
+#endif
+    {
+        /*set channel status bit for LPCM*/
+        get_lpcm_channel_status(out->sample_rate, channel_status);
+    }
+
+    memcpy(iec958.status, channel_status,sizeof(iec958.status));
+    ctl = mixer_get_ctl_by_name(out->dev->mixer, mixer_ctl_name);
+    if (!ctl) {
+            ALOGE("%s: Could not get ctl for mixer cmd - %s",
+                  __func__, mixer_ctl_name);
+            return;
+    }
+    if (mixer_ctl_set_array(ctl, &iec958, sizeof(iec958)) < 0) {
+        ALOGE("%s: Could not set channel status for ext HDMI ",
+              __func__);
+        return;
+    }
+
+}
+#endif
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 7c83168..b98d71b 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -165,7 +165,8 @@
 const char * const use_case_table[AUDIO_USECASE_MAX] = {
     [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback",
     [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback",
-    [USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback",
+    [USECASE_AUDIO_PLAYBACK_ULL]         = "audio-ull-playback",
+    [USECASE_AUDIO_PLAYBACK_MULTI_CH]    = "multi-channel-playback",
     [USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
 #ifdef MULTIPLE_OFFLOAD_ENABLED
     [USECASE_AUDIO_PLAYBACK_OFFLOAD2] = "compress-offload-playback2",
@@ -453,10 +454,10 @@
     audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_BUSY);
     audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_BUSY);
     audio_extn_utils_send_audio_calibration(adev, usecase);
-    audio_extn_utils_send_app_type_cfg(usecase);
+    audio_extn_utils_send_app_type_cfg(adev, usecase);
     strcpy(mixer_path, use_case_table[usecase->id]);
-    platform_add_backend_name(mixer_path, snd_device);
-    ALOGV("%s: apply mixer and update path: %s", __func__, mixer_path);
+    platform_add_backend_name(mixer_path, snd_device, usecase);
+    ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path);
     audio_route_apply_and_update_path(adev->audio_route, mixer_path);
     ALOGV("%s: exit", __func__);
     return 0;
@@ -477,8 +478,8 @@
     else
         snd_device = usecase->out_snd_device;
     strcpy(mixer_path, use_case_table[usecase->id]);
-    platform_add_backend_name(mixer_path, snd_device);
-    ALOGV("%s: reset and update mixer path: %s", __func__, mixer_path);
+    platform_add_backend_name(mixer_path, snd_device, usecase);
+    ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path);
     audio_route_reset_and_update_path(adev->audio_route, mixer_path);
     audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE);
     audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_FREE);
@@ -521,8 +522,10 @@
        audio_extn_usb_start_capture(adev);
 
     if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
-        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
-        audio_extn_spkr_prot_is_enabled()) {
+         snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
+         snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+         snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
+         audio_extn_spkr_prot_is_enabled()) {
        if (audio_extn_spkr_prot_get_acdb_id(snd_device) < 0) {
            adev->snd_dev_ref_cnt[snd_device]--;
            return -EINVAL;
@@ -534,8 +537,7 @@
             return -EINVAL;
         }
     } else {
-        ALOGV("%s: snd_device(%d: %s)", __func__,
-        snd_device, device_name);
+        ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
         /* due to the possibility of calibration overwrite between listen
             and audio, notify listen hal before audio calibration is sent */
         audio_extn_sound_trigger_update_device_status(snd_device,
@@ -579,8 +581,7 @@
     }
 
     if (adev->snd_dev_ref_cnt[snd_device] == 0) {
-        ALOGV("%s: snd_device(%d: %s)", __func__,
-              snd_device, device_name);
+        ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
         /* exit usb play back thread */
         if(SND_DEVICE_OUT_USB_HEADSET == snd_device ||
            SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET == snd_device)
@@ -591,13 +592,18 @@
             audio_extn_usb_stop_capture();
 
         if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
-            snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
-            audio_extn_spkr_prot_is_enabled()) {
+             snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
+             snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+             snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
+             audio_extn_spkr_prot_is_enabled()) {
             audio_extn_spkr_prot_stop_processing(snd_device);
         } else {
             audio_route_reset_and_update_path(adev->audio_route, device_name);
         }
 
+        if (snd_device == SND_DEVICE_OUT_HDMI)
+            adev->is_channel_status_set = false;
+
         audio_extn_dev_arbi_release(snd_device);
         audio_extn_sound_trigger_update_device_status(snd_device,
                                         ST_EVENT_SND_DEVICE_FREE);
@@ -1155,9 +1161,20 @@
         }
         break;
     }
+
+    ALOGV("%s: pcm_prepare", __func__);
+    ret = pcm_prepare(in->pcm);
+    if (ret < 0) {
+        ALOGE("%s: pcm_prepare returned %d", __func__, ret);
+        pcm_close(in->pcm);
+        in->pcm = NULL;
+        goto error_open;
+    }
+
     audio_extn_perf_lock_release();
 
-    ALOGV("%s: exit", __func__);
+    ALOGD("%s: exit", __func__);
+
     return ret;
 
 error_open:
@@ -1166,11 +1183,30 @@
 
 error_config:
     adev->active_input = NULL;
+    /*
+     * sleep 50ms to allow sufficient time for kernel
+     * drivers to recover incases like SSR.
+     */
+    usleep(50000);
     ALOGD("%s: exit: status(%d)", __func__, ret);
 
     return ret;
 }
 
+void lock_input_stream(struct stream_in *in)
+{
+    pthread_mutex_lock(&in->pre_lock);
+    pthread_mutex_lock(&in->lock);
+    pthread_mutex_unlock(&in->pre_lock);
+}
+
+void lock_output_stream(struct stream_out *out)
+{
+    pthread_mutex_lock(&out->pre_lock);
+    pthread_mutex_lock(&out->lock);
+    pthread_mutex_unlock(&out->pre_lock);
+}
+
 /* must be called with out->lock locked */
 static int send_offload_cmd_l(struct stream_out* out, int command)
 {
@@ -1259,7 +1295,7 @@
     prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
 
     ALOGV("%s", __func__);
-    pthread_mutex_lock(&out->lock);
+    lock_output_stream(out);
     for (;;) {
         struct offload_cmd *cmd = NULL;
         stream_callback_event_t event;
@@ -1318,6 +1354,10 @@
                 ALOGE("%s: Next track returned error %d",__func__, ret);
             if (ret != -ENETRESET) {
                 send_callback = true;
+                pthread_mutex_lock(&out->lock);
+                out->send_new_metadata = 1;
+                out->send_next_track_params = true;
+                pthread_mutex_unlock(&out->lock);
                 event = STREAM_CBK_EVENT_DRAIN_READY;
                 ALOGV("copl(%p):send drain callback, ret %d", out, ret);
             } else
@@ -1334,10 +1374,10 @@
             ALOGE("%s unknown command received: %d", __func__, cmd->cmd);
             break;
         }
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         out->offload_thread_blocked = false;
         pthread_cond_signal(&out->cond);
-        if (send_callback) {
+        if (send_callback && out->offload_callback) {
             ALOGVV("%s: sending offload_callback event %d", __func__, event);
             out->offload_callback(event, NULL, out->offload_cookie);
         }
@@ -1366,7 +1406,7 @@
 
 static int destroy_offload_callback_thread(struct stream_out *out)
 {
-    pthread_mutex_lock(&out->lock);
+    lock_output_stream(out);
     stop_compressed_output_l(out);
     send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
 
@@ -1625,8 +1665,20 @@
             }
             break;
         }
+
         platform_set_stream_channel_map(adev->platform, out->channel_mask,
                                     out->pcm_device_id);
+
+        ALOGV("%s: pcm_prepare", __func__);
+        if (pcm_is_ready(out->pcm)) {
+            ret = pcm_prepare(out->pcm);
+            if (ret < 0) {
+                ALOGE("%s: pcm_prepare returned %d", __func__, ret);
+                pcm_close(out->pcm);
+                out->pcm = NULL;
+                goto error_open;
+            }
+        }
     } else {
         platform_set_stream_channel_map(adev->platform, out->channel_mask,
                                     out->pcm_device_id);
@@ -1641,6 +1693,9 @@
             ret = -EIO;
             goto error_open;
         }
+        /* compress_open sends params of the track, so reset the flag here */
+        out->is_compr_metadata_avail = false;
+
         if (out->offload_callback)
             compress_nonblock(out->compr, out->non_blocking);
 
@@ -1669,11 +1724,18 @@
             audio_extn_check_and_set_dts_hpx_state(adev);
         }
     }
-    ALOGV("%s: exit", __func__);
+
+    ALOGD("%s: exit", __func__);
+
     return 0;
 error_open:
     stop_output_stream(out);
 error_config:
+    /*
+     * sleep 50ms to allow sufficient time for kernel
+     * drivers to recover incases like SSR.
+     */
+    usleep(50000);
     return ret;
 }
 
@@ -1803,8 +1865,11 @@
         return 0;
     }
 
-    pthread_mutex_lock(&out->lock);
+    lock_output_stream(out);
     if (!out->standby) {
+        if (adev->adm_deregister_stream)
+            adev->adm_deregister_stream(adev->adm_data, out->handle);
+
         pthread_mutex_lock(&adev->lock);
         out->standby = true;
         if (!is_offload_usecase(out->usecase)) {
@@ -1815,6 +1880,8 @@
         } else {
             ALOGD("copl(%p):standby", out);
             stop_compressed_output_l(out);
+            out->send_next_track_params = false;
+            out->is_compr_metadata_avail = false;
             out->gapless_mdata.encoder_delay = 0;
             out->gapless_mdata.encoder_padding = 0;
             if (out->compr != NULL) {
@@ -1840,7 +1907,6 @@
 {
     int ret = 0;
     char value[32];
-    bool is_meta_data_params = false;
 
     if (!out || !parms) {
         ALOGE("%s: return invalid ",__func__);
@@ -1849,31 +1915,15 @@
 
     ret = audio_extn_parse_compress_metadata(out, parms);
 
-    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_SAMPLE_RATE, value, sizeof(value));
-    if(ret >= 0)
-        is_meta_data_params = true;
-    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_NUM_CHANNEL, value, sizeof(value));
-    if(ret >= 0)
-        is_meta_data_params = true;
-    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE, value, sizeof(value));
-    if(ret >= 0)
-        is_meta_data_params = true;
     ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
     if (ret >= 0) {
-        is_meta_data_params = true;
         out->gapless_mdata.encoder_delay = atoi(value); //whats a good limit check?
     }
     ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value));
     if (ret >= 0) {
-        is_meta_data_params = true;
         out->gapless_mdata.encoder_padding = atoi(value);
     }
 
-    if(!is_meta_data_params) {
-        ALOGV("%s: Not gapless meta data params", __func__);
-        return 0;
-    }
-    out->send_new_metadata = 1;
     ALOGV("%s new encoder delay %u and padding %u", __func__,
           out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding);
 
@@ -1904,7 +1954,7 @@
     err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
     if (err >= 0) {
         val = atoi(value);
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         pthread_mutex_lock(&adev->lock);
 
         /*
@@ -1968,7 +2018,7 @@
         pthread_mutex_unlock(&adev->lock);
     }
     if (is_offload_usecase(out->usecase)) {
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         parse_compress_metadata(out, parms);
 
         audio_extn_dts_create_state_notifier_node(out->usecase);
@@ -2124,7 +2174,7 @@
     int snd_scard_state = get_snd_card_state(adev);
     ssize_t ret = 0;
 
-    pthread_mutex_lock(&out->lock);
+    lock_output_stream(out);
 
     if (SND_CARD_STATE_OFFLINE == snd_scard_state) {
         // increase written size during SSR to avoid mismatch
@@ -2157,6 +2207,13 @@
             out->standby = true;
             goto exit;
         }
+        if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD && adev->adm_register_output_stream)
+            adev->adm_register_output_stream(adev->adm_data, out->handle, out->flags);
+    }
+
+    if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
+        audio_utils_set_hdmi_channel_status(out, buffer, bytes);
+        adev->is_channel_status_set = true;
     }
 
     if (is_offload_usecase(out->usecase)) {
@@ -2165,6 +2222,12 @@
             ALOGD("copl(%p):send new gapless metadata", out);
             compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
             out->send_new_metadata = 0;
+            if (out->send_next_track_params && out->is_compr_metadata_avail) {
+                ALOGD("copl(%p):send next track params in gapless", out);
+                compress_set_next_track_param(out->compr, &(out->compr_config.codec->options));
+                out->send_next_track_params = false;
+                out->is_compr_metadata_avail = false;
+            }
         }
 
         ret = compress_write(out->compr, buffer, bytes);
@@ -2197,15 +2260,24 @@
         if (out->pcm) {
             if (out->muted)
                 memset((void *)buffer, 0, bytes);
+
             ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
+
+            if (adev->adm_request_focus)
+                adev->adm_request_focus(adev->adm_data, out->handle);
+
             if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY)
                 ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
             else
                 ret = pcm_write(out->pcm, (void *)buffer, bytes);
+
             if (ret < 0)
                 ret = -errno;
             else if (ret == 0)
                 out->written += bytes / (out->config.channels * sizeof(short));
+
+            if (adev->adm_abandon_focus)
+                adev->adm_abandon_focus(adev->adm_data, out->handle);
         }
     }
 
@@ -2247,7 +2319,7 @@
     *dsp_frames = 0;
     if (is_offload_usecase(out->usecase)) {
         ssize_t ret = 0;
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         if (out->compr != NULL) {
             ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
                     &out->sample_rate);
@@ -2306,7 +2378,7 @@
     int ret = -1;
     unsigned long dsp_frames;
 
-    pthread_mutex_lock(&out->lock);
+    lock_output_stream(out);
 
     if (is_offload_usecase(out->usecase)) {
         if (out->compr != NULL) {
@@ -2358,7 +2430,7 @@
     struct stream_out *out = (struct stream_out *)stream;
 
     ALOGV("%s", __func__);
-    pthread_mutex_lock(&out->lock);
+    lock_output_stream(out);
     out->offload_callback = callback;
     out->offload_cookie = cookie;
     pthread_mutex_unlock(&out->lock);
@@ -2372,7 +2444,7 @@
     ALOGV("%s", __func__);
     if (is_offload_usecase(out->usecase)) {
         ALOGD("copl(%p):pause compress driver", out);
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
             struct audio_device *adev = out->dev;
             int snd_scard_state = get_snd_card_state(adev);
@@ -2400,7 +2472,7 @@
     if (is_offload_usecase(out->usecase)) {
         ALOGD("copl(%p):resume compress driver", out);
         status = 0;
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
             struct audio_device *adev = out->dev;
             int snd_scard_state = get_snd_card_state(adev);
@@ -2425,7 +2497,7 @@
     int status = -ENOSYS;
     ALOGV("%s", __func__);
     if (is_offload_usecase(out->usecase)) {
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         if (type == AUDIO_DRAIN_EARLY_NOTIFY)
             status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN);
         else
@@ -2441,7 +2513,7 @@
     ALOGV("%s", __func__);
     if (is_offload_usecase(out->usecase)) {
         ALOGD("copl(%p):calling compress flush", out);
-        pthread_mutex_lock(&out->lock);
+        lock_output_stream(out);
         stop_compressed_output_l(out);
         pthread_mutex_unlock(&out->lock);
         ALOGD("copl(%p):out of compress flush", out);
@@ -2513,7 +2585,7 @@
         return status;
     }
 
-    pthread_mutex_lock(&in->lock);
+    lock_input_stream(in);
     if (!in->standby && in->is_st_session) {
         ALOGD("%s: sound trigger pcm stop lab", __func__);
         audio_extn_sound_trigger_stop_lab(in);
@@ -2521,6 +2593,9 @@
     }
 
     if (!in->standby) {
+        if (adev->adm_deregister_stream)
+            adev->adm_deregister_stream(adev->adm_data, in->capture_handle);
+
         pthread_mutex_lock(&adev->lock);
         in->standby = true;
         if (in->pcm) {
@@ -2555,7 +2630,7 @@
 
     if (!parms)
         goto error;
-    pthread_mutex_lock(&in->lock);
+    lock_input_stream(in);
     pthread_mutex_lock(&adev->lock);
 
     err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
@@ -2639,7 +2714,7 @@
     int i, ret = -1;
     int snd_scard_state = get_snd_card_state(adev);
 
-    pthread_mutex_lock(&in->lock);
+    lock_input_stream(in);
 
     if (in->is_st_session) {
         ALOGVV(" %s: reading on st session bytes=%zu", __func__, bytes);
@@ -2666,8 +2741,13 @@
             goto exit;
         }
         in->standby = 0;
+        if (adev->adm_register_input_stream)
+            adev->adm_register_input_stream(adev->adm_data, in->capture_handle, in->flags);
     }
 
+    if (adev->adm_request_focus)
+        adev->adm_request_focus(adev->adm_data, in->capture_handle);
+
     if (in->pcm) {
         if (audio_extn_ssr_get_enabled() &&
                 audio_channel_count_from_in_mask(in->channel_mask) == 6)
@@ -2682,6 +2762,9 @@
             ret = -errno;
     }
 
+    if (adev->adm_abandon_focus)
+        adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+
     /*
      * Instead of writing zeroes here, we could trust the hardware
      * to always provide zeroes when muted.
@@ -2731,7 +2814,7 @@
     if (status != 0)
         return status;
 
-    pthread_mutex_lock(&in->lock);
+    lock_input_stream(in);
     pthread_mutex_lock(&in->dev->lock);
     if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
             in->enable_aec != enable &&
@@ -2799,6 +2882,7 @@
     }
 
     pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
+    pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL);
     pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL);
 
     if (devices == AUDIO_DEVICE_NONE)
@@ -2895,6 +2979,12 @@
         } else {
             ALOGV("%s:: inserting OFFLOAD_USECASE", __func__);
             out->usecase = get_offload_usecase(adev);
+
+            out->stream.set_callback = out_set_callback;
+            out->stream.pause = out_pause;
+            out->stream.resume = out_resume;
+            out->stream.drain = out_drain;
+            out->stream.flush = out_flush;
         }
         if (config->offload_info.channel_mask)
             out->channel_mask = config->offload_info.channel_mask;
@@ -2905,11 +2995,6 @@
         format = out->format = config->offload_info.format;
         out->sample_rate = config->offload_info.sample_rate;
 
-        out->stream.set_callback = out_set_callback;
-        out->stream.pause = out_pause;
-        out->stream.resume = out_resume;
-        out->stream.drain = out_drain;
-        out->stream.flush = out_flush;
         out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
 
         if (audio_extn_is_dolby_format(config->offload_info.format))
@@ -2974,6 +3059,8 @@
         }
 
         out->send_new_metadata = 1;
+        out->send_next_track_params = false;
+        out->is_compr_metadata_avail = false;
         out->offload_state = OFFLOAD_STATE_IDLE;
         out->playback_started = 0;
 
@@ -3014,6 +3101,10 @@
         out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
         out->config = pcm_config_afe_proxy_playback;
         adev->voice_tx_output = out;
+    } else if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
+        out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
+        out->config = pcm_config_low_latency;
+        out->sample_rate = out->config.rate;
     } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
         format = AUDIO_FORMAT_PCM_16_BIT;
         out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
@@ -3390,6 +3481,7 @@
         if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION) &&
                 voice_is_in_call(adev)) {
             voice_stop_call(adev);
+            platform_set_gsm_mode(adev->platform, false);
             adev->current_call_output = NULL;
         }
     }
@@ -3455,6 +3547,7 @@
         devices, &in->stream, handle, source);
 
     pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
+    pthread_mutex_init(&in->pre_lock, (const pthread_mutexattr_t *) NULL);
 
     in->stream.common.get_sample_rate = in_get_sample_rate;
     in->stream.common.set_sample_rate = in_set_sample_rate;
@@ -3478,6 +3571,7 @@
     in->standby = 1;
     in->channel_mask = config->channel_mask;
     in->capture_handle = handle;
+    in->flags = flags;
 
     /* Update config params with the requested sample rate and channels */
     in->usecase = USECASE_AUDIO_RECORD;
@@ -3620,10 +3714,13 @@
         audio_route_free(adev->audio_route);
         free(adev->snd_dev_ref_cnt);
         platform_deinit(adev->platform);
+        if (adev->adm_deinit)
+            adev->adm_deinit(adev->adm_data);
         free(device);
         adev = NULL;
     }
     pthread_mutex_unlock(&adev_init_lock);
+
     return 0;
 }
 
@@ -3635,6 +3732,7 @@
 {
     switch (period_size) {
     case 160:
+    case 192:
     case 240:
     case 320:
     case 480:
@@ -3707,6 +3805,7 @@
     list_init(&adev->usecase_list);
     adev->cur_wfd_channels = 2;
     adev->offload_usecases_state = 0;
+    adev->is_channel_status_set = false;
 
     pthread_mutex_init(&adev->snd_card_status.lock, (const pthread_mutexattr_t *) NULL);
     adev->snd_card_status.state = SND_CARD_STATE_OFFLINE;
@@ -3757,6 +3856,36 @@
             adev->offload_effects_set_hpx_state =
                         (int (*)(bool))dlsym(adev->offload_effects_lib,
                                          "offload_effects_bundle_set_hpx_state");
+            adev->offload_effects_get_parameters =
+                        (void (*)(struct str_parms *, struct str_parms *))
+                                         dlsym(adev->offload_effects_lib,
+                                         "offload_effects_bundle_get_parameters");
+            adev->offload_effects_set_parameters =
+                        (void (*)(struct str_parms *))dlsym(adev->offload_effects_lib,
+                                         "offload_effects_bundle_set_parameters");
+        }
+    }
+
+    if (access(ADM_LIBRARY_PATH, R_OK) == 0) {
+        adev->adm_lib = dlopen(ADM_LIBRARY_PATH, RTLD_NOW);
+        if (adev->adm_lib == NULL) {
+            ALOGE("%s: DLOPEN failed for %s", __func__, ADM_LIBRARY_PATH);
+        } else {
+            ALOGV("%s: DLOPEN successful for %s", __func__, ADM_LIBRARY_PATH);
+            adev->adm_init = (adm_init_t)
+                                    dlsym(adev->adm_lib, "adm_init");
+            adev->adm_deinit = (adm_deinit_t)
+                                    dlsym(adev->adm_lib, "adm_deinit");
+            adev->adm_register_input_stream = (adm_register_input_stream_t)
+                                    dlsym(adev->adm_lib, "adm_register_input_stream");
+            adev->adm_register_output_stream = (adm_register_output_stream_t)
+                                    dlsym(adev->adm_lib, "adm_register_output_stream");
+            adev->adm_deregister_stream = (adm_deregister_stream_t)
+                                    dlsym(adev->adm_lib, "adm_deregister_stream");
+            adev->adm_request_focus = (adm_request_focus_t)
+                                    dlsym(adev->adm_lib, "adm_request_focus");
+            adev->adm_abandon_focus = (adm_abandon_focus_t)
+                                    dlsym(adev->adm_lib, "adm_abandon_focus");
         }
     }
 
@@ -3790,6 +3919,9 @@
 
     pthread_mutex_unlock(&adev_init_lock);
 
+    if (adev->adm_init)
+        adev->adm_data = adev->adm_init();
+
     ALOGV("%s: exit", __func__);
     return 0;
 }
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index c2f37d4..dc0970f 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -50,6 +50,7 @@
 
 #define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
 #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"
+#define ADM_LIBRARY_PATH "/system/vendor/lib/libadm.so"
 
 /* Flags used to initialize acdb_settings variable that goes to ACDB library */
 #define NONE_FLAG            0x00000000
@@ -94,6 +95,7 @@
     USECASE_AUDIO_PLAYBACK_OFFLOAD8,
     USECASE_AUDIO_PLAYBACK_OFFLOAD9,
 #endif
+    USECASE_AUDIO_PLAYBACK_ULL,
 
     USECASE_AUDIO_DIRECT_PCM_OFFLOAD,
 
@@ -182,6 +184,7 @@
 struct stream_out {
     struct audio_stream_out stream;
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
+    pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
     pthread_cond_t  cond;
     struct pcm_config config;
     struct compr_config compr_config;
@@ -216,6 +219,8 @@
     void *offload_cookie;
     struct compr_gapless_mdata gapless_mdata;
     int send_new_metadata;
+    bool send_next_track_params;
+    bool is_compr_metadata_avail;
     unsigned int bit_width;
 
     struct audio_device *dev;
@@ -224,6 +229,7 @@
 struct stream_in {
     struct audio_stream_in stream;
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
+    pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
     struct pcm_config config;
     struct pcm *pcm;
     int standby;
@@ -236,6 +242,7 @@
     bool enable_ns;
     audio_format_t format;
     audio_io_handle_t capture_handle;
+    audio_input_flags_t flags;
     bool is_st_session;
     bool is_st_session_active;
 
@@ -288,6 +295,14 @@
     struct stream_app_type_cfg app_type_cfg;
 };
 
+typedef void* (*adm_init_t)();
+typedef void (*adm_deinit_t)(void *);
+typedef void (*adm_register_output_stream_t)(void *, audio_io_handle_t, audio_output_flags_t);
+typedef void (*adm_register_input_stream_t)(void *, audio_io_handle_t, audio_input_flags_t);
+typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t);
+typedef void (*adm_request_focus_t)(void *, audio_io_handle_t);
+typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t);
+
 struct audio_device {
     struct audio_hw_device device;
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
@@ -314,6 +329,7 @@
     int snd_card;
     unsigned int cur_codec_backend_samplerate;
     unsigned int cur_codec_backend_bit_width;
+    bool is_channel_status_set;
     void *platform;
     unsigned int offload_usecases_state;
     void *visualizer_lib;
@@ -325,6 +341,20 @@
 
     struct sound_card_status snd_card_status;
     int (*offload_effects_set_hpx_state)(bool);
+
+    void *adm_data;
+    void *adm_lib;
+    adm_init_t adm_init;
+    adm_deinit_t adm_deinit;
+    adm_register_input_stream_t adm_register_input_stream;
+    adm_register_output_stream_t adm_register_output_stream;
+    adm_deregister_stream_t adm_deregister_stream;
+    adm_request_focus_t adm_request_focus;
+    adm_abandon_focus_t adm_abandon_focus;
+
+    void (*offload_effects_get_parameters)(struct str_parms *,
+                                           struct str_parms *);
+    void (*offload_effects_set_parameters)(struct str_parms *);
 };
 
 int select_devices(struct audio_device *adev,
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 293b1c6..b5f56c1 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -133,6 +133,7 @@
 char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
         [WCD9XXX_ANC_CAL] = "anc_cal",
         [WCD9XXX_MBHC_CAL] = "mbhc_cal",
+        [WCD9XXX_VBAT_CAL] = "vbat_cal",
 };
 
 #define AUDIO_PARAMETER_KEY_REC_PLAY_CONC "rec_play_conc_on"
@@ -184,6 +185,7 @@
 typedef int (*acdb_set_audio_cal_t) (void *, void *, uint32_t);
 typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
 typedef int (*acdb_send_common_top_t) (void);
+typedef int (*acdb_set_codec_data_t) (void *, char *);
 
 typedef struct codec_backend_cfg {
     uint32_t sample_rate;
@@ -216,6 +218,9 @@
     bool ec_ref_enabled;
     bool is_wsa_speaker;
     bool is_acdb_initialized;
+    /* Vbat monitor related flags */
+    bool is_vbat_speaker;
+    bool gsm_mode_enabled;
     /* Audio calibration related functions */
     void                       *acdb_handle;
     int                        voice_feature_set;
@@ -228,6 +233,7 @@
     acdb_reload_vocvoltable_t  acdb_reload_vocvoltable;
     acdb_get_default_app_type_t acdb_get_default_app_type;
     acdb_send_common_top_t     acdb_send_common_top;
+    acdb_set_codec_data_t      acdb_set_codec_data;
 #ifdef RECORD_PLAY_CONCURRENCY
     bool rec_play_conc_set;
 #endif
@@ -241,11 +247,17 @@
 
 static bool is_external_codec = false;
 static const int pcm_device_table_of_ext_codec[AUDIO_USECASE_MAX][2] = {
-   [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC, QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC}
+   [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC, QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC},
+   [USECASE_VOICEMMODE1_CALL] = {VOICEMMODE1_CALL_PCM_DEVICE_OF_EXT_CODEC,
+                                 VOICEMMODE1_CALL_PCM_DEVICE_OF_EXT_CODEC},
+   [USECASE_VOICEMMODE2_CALL] = {VOICEMMODE2_CALL_PCM_DEVICE_OF_EXT_CODEC,
+                                 VOICEMMODE2_CALL_PCM_DEVICE_OF_EXT_CODEC},
 };
 
 /* List of use cases that has different PCM device ID's for internal and external codecs */
-static const int misc_usecase[AUDIO_USECASE_MAX] = { USECASE_QCHAT_CALL };
+static const int misc_usecase[AUDIO_USECASE_MAX] = {USECASE_QCHAT_CALL,
+                                                    USECASE_VOICEMMODE1_CALL,
+                                                    USECASE_VOICEMMODE2_CALL};
 
 int pcm_device_table[AUDIO_USECASE_MAX][2] = {
     [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {DEEP_BUFFER_PCM_DEVICE,
@@ -286,6 +298,10 @@
     [USECASE_VOLTE_CALL] = {VOLTE_CALL_PCM_DEVICE, VOLTE_CALL_PCM_DEVICE},
     [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE, QCHAT_CALL_PCM_DEVICE},
     [USECASE_VOWLAN_CALL] = {VOWLAN_CALL_PCM_DEVICE, VOWLAN_CALL_PCM_DEVICE},
+    [USECASE_VOICEMMODE1_CALL] = {VOICEMMODE1_CALL_PCM_DEVICE,
+                                  VOICEMMODE1_CALL_PCM_DEVICE},
+    [USECASE_VOICEMMODE2_CALL] = {VOICEMMODE2_CALL_PCM_DEVICE,
+                                  VOICEMMODE2_CALL_PCM_DEVICE},
     [USECASE_COMPRESS_VOIP_CALL] = {COMPRESS_VOIP_CALL_PCM_DEVICE, COMPRESS_VOIP_CALL_PCM_DEVICE},
     [USECASE_INCALL_REC_UPLINK] = {AUDIO_RECORD_PCM_DEVICE,
                                    AUDIO_RECORD_PCM_DEVICE},
@@ -320,6 +336,7 @@
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_1] = "speaker-ext-1",
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = "speaker-ext-2",
     [SND_DEVICE_OUT_SPEAKER_WSA] = "wsa-speaker",
+    [SND_DEVICE_OUT_SPEAKER_VBAT] = "vbat-speaker",
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
     [SND_DEVICE_OUT_HEADPHONES] = "headphones",
     [SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
@@ -329,6 +346,7 @@
     [SND_DEVICE_OUT_VOICE_HANDSET] = "voice-handset",
     [SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
     [SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = "wsa-voice-speaker",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "vbat-voice-speaker",
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
     [SND_DEVICE_OUT_HDMI] = "hdmi",
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
@@ -350,6 +368,8 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+    [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
 #ifdef RECORD_PLAY_CONCURRENCY
     [SND_DEVICE_OUT_VOIP_HANDSET] = "voip-handset",
     [SND_DEVICE_OUT_VOIP_SPEAKER] = "voip-speaker",
@@ -425,6 +445,7 @@
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_1] = 14,
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = 14,
     [SND_DEVICE_OUT_SPEAKER_WSA] = 135,
+    [SND_DEVICE_OUT_SPEAKER_VBAT] = 135,
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
     [SND_DEVICE_OUT_HEADPHONES] = 10,
     [SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
@@ -434,6 +455,7 @@
     [SND_DEVICE_OUT_VOICE_HANDSET] = 7,
     [SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
     [SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = 135,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 135,
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
     [SND_DEVICE_OUT_HDMI] = 18,
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
@@ -455,6 +477,8 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = 103,
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+    [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
 #ifdef RECORD_PLAY_CONCURRENCY
     [SND_DEVICE_OUT_VOIP_HANDSET] = 133,
     [SND_DEVICE_OUT_VOIP_SPEAKER] = 132,
@@ -532,6 +556,7 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_EXTERNAL_1)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_EXTERNAL_2)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
@@ -541,6 +566,7 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_WSA)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
@@ -562,6 +588,8 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
 #ifdef RECORD_PLAY_CONCURRENCY
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_SPEAKER)},
@@ -649,6 +677,8 @@
     {TO_NAME_INDEX(USECASE_VOLTE_CALL)},
     {TO_NAME_INDEX(USECASE_QCHAT_CALL)},
     {TO_NAME_INDEX(USECASE_VOWLAN_CALL)},
+    {TO_NAME_INDEX(USECASE_VOICEMMODE1_CALL)},
+    {TO_NAME_INDEX(USECASE_VOICEMMODE2_CALL)},
     {TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK)},
     {TO_NAME_INDEX(USECASE_INCALL_REC_DOWNLINK)},
     {TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
@@ -735,7 +765,7 @@
      if (!strncmp(snd_card_name, "msm8939-tapan-snd-card",
                   sizeof("msm8939-tapan-snd-card")) ||
          !strncmp(snd_card_name, "msm8939-tapan9302-snd-card",
-                  sizeof("msm8939-tapan9302-snd-card"))||
+                  sizeof("msm8939-tapan9302-snd-card")) ||
          !strncmp(snd_card_name, "msm8939-tomtom9330-snd-card",
                   sizeof("msm8939-tomtom9330-snd-card")) ||
          !strncmp(snd_card_name, "msm8952-tomtom-snd-card",
@@ -866,7 +896,7 @@
         msm_be_id_array_len  =
             sizeof(msm_device_to_be_id_external_codec) / sizeof(msm_device_to_be_id_internal_codec[0]);
     } else if (!strncmp(snd_card_name, "msm8952-snd-card-mtp",
-                 sizeof("msm8952-snd-card-mtpmsm8952-snd-card-mtp"))) {
+                 sizeof("msm8952-snd-card-mtp"))) {
         strlcpy(mixer_xml_path, MIXER_XML_PATH_MTP,
                 sizeof(MIXER_XML_PATH_MTP));
         msm_device_to_be_id = msm_device_to_be_id_internal_codec;
@@ -898,6 +928,24 @@
     }
 }
 
+void platform_set_gsm_mode(void *platform, bool enable)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+
+    if (my_data->gsm_mode_enabled) {
+        my_data->gsm_mode_enabled = false;
+        ALOGV("%s: disabling gsm mode", __func__);
+        audio_route_reset_and_update_path(adev->audio_route, "gsm-mode");
+    }
+
+    if (enable) {
+         my_data->gsm_mode_enabled = true;
+         ALOGD("%s: enabling gsm mode", __func__);
+         audio_route_apply_and_update_path(adev->audio_route, "gsm-mode");
+    }
+}
+
 void platform_set_echo_reference(struct audio_device *adev, bool enable, audio_devices_t out_device)
 {
     struct platform_data *my_data = (struct platform_data *)adev->platform;
@@ -908,20 +956,27 @@
 
     if (strcmp(my_data->ec_ref_mixer_path, "")) {
         ALOGV("%s: disabling %s", __func__, my_data->ec_ref_mixer_path);
-        audio_route_reset_and_update_path(adev->audio_route, my_data->ec_ref_mixer_path);
+        audio_route_reset_and_update_path(adev->audio_route,
+                                          my_data->ec_ref_mixer_path);
     }
 
     if (enable) {
-        strlcpy(my_data->ec_ref_mixer_path, "echo-reference", sizeof(my_data->ec_ref_mixer_path));
         snd_device = platform_get_output_snd_device(adev->platform, &out);
+
         /*
          * If native audio device reference count > 0, then apply codec EC otherwise
-         * fallback to headphones if so or default
+         * fallback to Speakers with VBat if enabled or default
          */
         if (adev->snd_dev_ref_cnt[SND_DEVICE_OUT_HEADPHONES_44_1] > 0)
-            platform_add_backend_name(my_data->ec_ref_mixer_path, SND_DEVICE_OUT_HEADPHONES_44_1);
+            strlcpy(my_data->ec_ref_mixer_path, "echo-reference headphones-44.1",
+                    sizeof(my_data->ec_ref_mixer_path));
+        else if ((snd_device == SND_DEVICE_OUT_SPEAKER_VBAT) ||
+                 (snd_device == SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT))
+            strlcpy(my_data->ec_ref_mixer_path, "vbat-speaker echo-reference",
+                    sizeof(my_data->ec_ref_mixer_path));
         else
-            platform_add_backend_name(my_data->ec_ref_mixer_path, snd_device);
+            strlcpy(my_data->ec_ref_mixer_path, "echo-reference",
+                    sizeof(my_data->ec_ref_mixer_path));
 
         ALOGD("%s: enabling %s", __func__, my_data->ec_ref_mixer_path);
         audio_route_apply_and_update_path(adev->audio_route, my_data->ec_ref_mixer_path);
@@ -1103,6 +1158,7 @@
     backend_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
     backend_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
     backend_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
+    backend_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("vbat-voice-speaker");
 }
 
 void get_cvd_version(char *cvd_version, struct audio_device *adev)
@@ -1158,7 +1214,38 @@
     void   *buff;
 };
 
-static int send_codec_cal(acdb_loader_get_calibration_t acdb_loader_get_calibration, int fd)
+static int send_vbat_adc_data_to_acdb(struct platform_data *plat_data, char *cal_type)
+{
+    int ret = 0;
+    struct mixer_ctl *ctl;
+    uint16_t vbat_adc_data[2];
+    struct platform_data *my_data = plat_data;
+    struct audio_device *adev = my_data->adev;
+
+    const char *mixer_ctl_name = "Vbat ADC data";
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer ctl name - %s",
+               __func__, mixer_ctl_name);
+        ret = -EINVAL;
+        goto done;
+    }
+
+    vbat_adc_data[0] = mixer_ctl_get_value(ctl, 0);
+    vbat_adc_data[1] = mixer_ctl_get_value(ctl, 1);
+
+    ALOGD("%s: Vbat ADC output values: Dcp1: %d , Dcp2: %d",
+           __func__, vbat_adc_data[0], vbat_adc_data[1]);
+
+    ret = my_data->acdb_set_codec_data(&vbat_adc_data[0], cal_type);
+
+done:
+    return ret;
+}
+
+static int send_codec_cal(acdb_loader_get_calibration_t acdb_loader_get_calibration,
+                          struct platform_data *plat_data, int fd)
 {
     int ret = 0, type;
 
@@ -1170,6 +1257,12 @@
         if (type == WCD9XXX_MAD_CAL)
             continue;
 
+        if((plat_data->is_vbat_speaker) && (WCD9XXX_VBAT_CAL == type)) {
+           ret = send_vbat_adc_data_to_acdb(plat_data, cal_name_info[type]);
+           if (ret < 0)
+               ALOGE("%s error in sending vbat adc data to acdb", __func__);
+        }
+
         calib.get_size = 1;
         ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
                                                                  &calib);
@@ -1223,7 +1316,7 @@
            dlerror());
         return;
     }
-    if (send_codec_cal(acdb_loader_get_calibration, fd) < 0)
+    if (send_codec_cal(acdb_loader_get_calibration, plat_data, fd) < 0)
         ALOGE("%s: Could not send anc cal", __FUNCTION__);
 }
 
@@ -1410,6 +1503,12 @@
         acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 131;
     }
 
+    /* Check if Vbat speaker enabled property is set, this should be done before acdb init */
+    bool ret = false;
+    ret = audio_extn_can_use_vbat();
+    if (ret)
+        my_data->is_vbat_speaker = true;
+
     my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
     my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
     if (my_data->acdb_handle == NULL) {
@@ -1466,6 +1565,14 @@
             ALOGE("%s: Could not find the symbol acdb_get_default_app_type from %s",
                   __func__, LIB_ACDB_LOADER);
 
+        my_data->acdb_set_codec_data = (acdb_set_codec_data_t)dlsym(
+                                                    my_data->acdb_handle,
+                                                    "acdb_loader_set_codec_data");
+        if (!my_data->acdb_set_codec_data)
+            ALOGE("%s: Could not find the symbol acdb_get_default_app_type from %s",
+                  __func__, LIB_ACDB_LOADER);
+
+
         my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
                                                     "acdb_loader_init_v2");
         if (my_data->acdb_init == NULL) {
@@ -1517,7 +1624,6 @@
     audio_extn_dap_hal_init(adev->snd_card);
 
     audio_extn_dolby_set_license(adev);
-    audio_hwdep_send_cal(my_data);
 
     /* init audio device arbitration */
     audio_extn_dev_arbi_init();
@@ -1542,6 +1648,26 @@
     my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
         strdup("SLIM_5_RX SampleRate");
 
+
+    if (platform_get_native_support()) {
+
+        if (!strncmp(snd_card_name, "msm8976-tasha-snd-card",
+            sizeof("msm8976-tasha-snd-card")) ||
+            !strncmp(snd_card_name, "msm8976-tashalite-snd-card",
+            sizeof("msm8976-tashalite-snd-card"))) {
+            /* do nothing */
+        }
+        else {
+            platform_set_native_support(false);
+        }
+    }
+
+    ALOGD("native: native audio: %s for sound card %s",
+        (platform_get_native_support() ? "enabled" : "disabled"),
+        snd_card_name);
+
+
+
     my_data->edid_info = NULL;
     return my_data;
 }
@@ -1625,13 +1751,20 @@
     return 0;
 }
 
-void platform_add_backend_name(char *mixer_path, snd_device_t snd_device)
+void platform_add_backend_name(char *mixer_path, snd_device_t snd_device,
+                               struct audio_usecase *usecase)
 {
     if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
         ALOGE("%s: Invalid snd_device = %d", __func__, snd_device);
         return;
     }
 
+    if((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+        !(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
+        ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
+        return;
+    }
+
     const char * suffix = backend_table[snd_device];
 
     if (suffix != NULL) {
@@ -1824,7 +1957,7 @@
 {
     na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled
         = codec_support;
-    ALOGD("%s: na_props.platform_na_prop_enabled: %d", __func__,
+    ALOGV("%s: na_props.platform_na_prop_enabled: %d", __func__,
            na_props.platform_na_prop_enabled);
     return 0;
 }
@@ -1995,10 +2128,14 @@
     if (my_data->csd == NULL)
         return ret;
 
-    if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
-        audio_extn_spkr_prot_is_enabled())
-        acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
-    else
+    if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         audio_extn_spkr_prot_is_enabled()) {
+        if (my_data->is_vbat_speaker)
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
+        else
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
+    } else
         acdb_rx_id = acdb_device_table[out_snd_device];
 
     acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2055,10 +2192,14 @@
     if (my_data->csd == NULL)
         return ret;
 
-    if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
-        audio_extn_spkr_prot_is_enabled())
-        acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
-    else
+    if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         audio_extn_spkr_prot_is_enabled()) {
+        if (my_data->is_vbat_speaker)
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
+         else
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
+    } else
         acdb_rx_id = acdb_device_table[out_snd_device];
 
     acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2320,7 +2461,9 @@
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
         } else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
-                if (my_data->is_wsa_speaker)
+                if (my_data->is_vbat_speaker)
+                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+                else if (my_data->is_wsa_speaker)
                     snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
                 else
                     snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
@@ -2381,7 +2524,9 @@
                 snd_device = SND_DEVICE_OUT_SPEAKER_REVERSE;
             else
             {
-                if (my_data->is_wsa_speaker)
+                if (my_data->is_vbat_speaker)
+                    snd_device = SND_DEVICE_OUT_SPEAKER_VBAT;
+                else if (my_data->is_wsa_speaker)
                     snd_device = SND_DEVICE_OUT_SPEAKER_WSA;
                 else
                     snd_device = SND_DEVICE_OUT_SPEAKER;
@@ -3433,6 +3578,8 @@
     *  USECASE_VOLTE_CALL:
     *  USECASE_QCHAT_CALL:
     *  USECASE_VOWLAN_CALL:
+    *  USECASE_VOICEMMODE1_CALL:
+    *  USECASE_VOICEMMODE2_CALL:
     *  USECASE_COMPRESS_VOIP_CALL:
     *  USECASE_AUDIO_RECORD_FM_VIRTUAL:
     *  USECASE_INCALL_REC_UPLINK:
@@ -3495,6 +3642,8 @@
     *  USECASE_VOLTE_CALL:
     *  USECASE_QCHAT_CALL:
     *  USECASE_VOWLAN_CALL:
+    *  USECASE_VOICEMMODE1_CALL:
+    *  USECASE_VOICEMMODE2_CALL:
     *  USECASE_COMPRESS_VOIP_CALL:
     *  USECASE_AUDIO_RECORD_FM_VIRTUAL:
     *  USECASE_INCALL_REC_UPLINK:
@@ -4041,15 +4190,12 @@
     }
     edid_data[0] = count;
     memcpy(&edid_data[1], block, count);
-
-#ifdef AUDIO_FEATURE_ENABLED_HDMI_EDID
     if (!edid_get_sink_caps(info, edid_data)) {
         ALOGE("%s: Failed to get HDMI sink capabilities", __func__);
         goto fail;
     }
     my_data->edid_valid = true;
     return 0;
-#endif
 fail:
     if (my_data->edid_info) {
         free(my_data->edid_info);
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 9710147..a5b97ac 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -72,6 +72,7 @@
     SND_DEVICE_OUT_SPEAKER_EXTERNAL_2,
     SND_DEVICE_OUT_SPEAKER_REVERSE,
     SND_DEVICE_OUT_SPEAKER_WSA,
+    SND_DEVICE_OUT_SPEAKER_VBAT,
     SND_DEVICE_OUT_HEADPHONES,
     SND_DEVICE_OUT_HEADPHONES_44_1,
     SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
@@ -80,6 +81,7 @@
     SND_DEVICE_OUT_VOICE_HANDSET,
     SND_DEVICE_OUT_VOICE_SPEAKER,
     SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
+    SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
     SND_DEVICE_OUT_VOICE_HEADPHONES,
     SND_DEVICE_OUT_HDMI,
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
@@ -101,6 +103,8 @@
     SND_DEVICE_OUT_ANC_HANDSET,
     SND_DEVICE_OUT_SPEAKER_PROTECTED,
     SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+    SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
+    SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
 #ifdef RECORD_PLAY_CONCURRENCY
     SND_DEVICE_OUT_VOIP_HANDSET,
     SND_DEVICE_OUT_VOIP_SPEAKER,
@@ -260,6 +264,12 @@
 #define QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC 28
 #define VOWLAN_CALL_PCM_DEVICE 16
 
+#define VOICEMMODE1_CALL_PCM_DEVICE 26
+#define VOICEMMODE2_CALL_PCM_DEVICE 27
+
+#define VOICEMMODE1_CALL_PCM_DEVICE_OF_EXT_CODEC 29
+#define VOICEMMODE2_CALL_PCM_DEVICE_OF_EXT_CODEC 30
+
 #define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
 #define AFE_PROXY_RECORD_PCM_DEVICE 8
 
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 6d2d478..b72cae9 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -373,7 +373,8 @@
     return 0;
 }
 
-void platform_add_backend_name(char *mixer_path, snd_device_t snd_device)
+void platform_add_backend_name(char *mixer_path, snd_device_t snd_device,
+                               struct audio_usecase *usecase)
 {
     if (snd_device == SND_DEVICE_IN_BT_SCO_MIC)
         strlcat(mixer_path, " bt-sco", MIXER_PATH_MAX_LENGTH);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index aba47fd..723bf5b 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -130,6 +130,7 @@
 char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
         [WCD9XXX_ANC_CAL] = "anc_cal",
         [WCD9XXX_MBHC_CAL] = "mbhc_cal",
+        [WCD9XXX_VBAT_CAL] = "vbat_cal",
 };
 
 #define  AUDIO_PARAMETER_IS_HW_DECODER_SESSION_ALLOWED  "is_hw_dec_session_allowed"
@@ -186,6 +187,7 @@
 typedef int (*acdb_set_audio_cal_t) (void *, void *, uint32_t);
 typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
 typedef int (*acdb_send_common_top_t) (void);
+typedef int (*acdb_set_codec_data_t) (void *, char *);
 
 typedef struct codec_backend_cfg {
     uint32_t sample_rate;
@@ -218,6 +220,9 @@
     bool ec_ref_enabled;
     bool is_i2s_ext_modem;
     bool is_acdb_initialized;
+    /* Vbat monitor related flags */
+    bool is_vbat_speaker;
+    bool gsm_mode_enabled;
     /* Audio calibration related functions */
     void                       *acdb_handle;
     int                        voice_feature_set;
@@ -230,6 +235,7 @@
     acdb_reload_vocvoltable_t  acdb_reload_vocvoltable;
     acdb_get_default_app_type_t acdb_get_default_app_type;
     acdb_send_common_top_t     acdb_send_common_top;
+    acdb_set_codec_data_t      acdb_set_codec_data;
 
     void *hw_info;
     acdb_send_gain_dep_cal_t   acdb_send_gain_dep_cal;
@@ -245,8 +251,10 @@
                                             DEEP_BUFFER_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
                                            LOWLATENCY_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_ULL]         = {MULTIMEDIA3_PCM_DEVICE,
+                                            MULTIMEDIA3_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_MULTI_CH] = {MULTIMEDIA2_PCM_DEVICE,
-                                        MULTIMEDIA2_PCM_DEVICE},
+                                         MULTIMEDIA2_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_OFFLOAD] =
                      {PLAYBACK_OFFLOAD_DEVICE, PLAYBACK_OFFLOAD_DEVICE},
 #ifdef MULTIPLE_OFFLOAD_ENABLED
@@ -324,6 +332,7 @@
     [SND_DEVICE_OUT_SPEAKER] = "speaker",
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_1] = "speaker-ext-1",
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = "speaker-ext-2",
+    [SND_DEVICE_OUT_SPEAKER_VBAT] = "speaker-vbat",
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
     [SND_DEVICE_OUT_HEADPHONES] = "headphones",
     [SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
@@ -332,6 +341,7 @@
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = "speaker-and-headphones-ext-2",
     [SND_DEVICE_OUT_VOICE_HANDSET] = "voice-handset",
     [SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "voice-speaker-vbat",
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
     [SND_DEVICE_OUT_HDMI] = "hdmi",
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
@@ -353,6 +363,8 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+    [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
 
     /* Capture sound devices */
     [SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -422,6 +434,7 @@
     [SND_DEVICE_OUT_SPEAKER] = 14,
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_1] = 130,
     [SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = 130,
+    [SND_DEVICE_OUT_SPEAKER_VBAT] = 14,
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
     [SND_DEVICE_OUT_HEADPHONES] = 10,
     [SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
@@ -430,6 +443,7 @@
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = 130,
     [SND_DEVICE_OUT_VOICE_HANDSET] = 7,
     [SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 14,
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
     [SND_DEVICE_OUT_HDMI] = 18,
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
@@ -451,6 +465,8 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = 103,
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+    [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
 
     [SND_DEVICE_IN_HANDSET_MIC] = 4,
     [SND_DEVICE_IN_HANDSET_MIC_EXTERNAL] = 4,
@@ -522,6 +538,7 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_EXTERNAL_1)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_EXTERNAL_2)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
@@ -530,6 +547,7 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
@@ -550,6 +568,8 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_EXTERNAL)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC)},
@@ -610,6 +630,7 @@
 static struct name_to_index usecase_name_index[AUDIO_USECASE_MAX] = {
     {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_DEEP_BUFFER)},
     {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_LOW_LATENCY)},
+    {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_ULL)},
     {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_MULTI_CH)},
     {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_OFFLOAD)},
 #ifdef MULTIPLE_OFFLOAD_ENABLED
@@ -780,7 +801,26 @@
     return ret_val;
 }
 
-void platform_set_echo_reference(struct audio_device *adev, bool enable, audio_devices_t out_device)
+void platform_set_gsm_mode(void *platform, bool enable)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+
+    if (my_data->gsm_mode_enabled) {
+        my_data->gsm_mode_enabled = false;
+        ALOGV("%s: disabling gsm mode", __func__);
+        audio_route_reset_and_update_path(adev->audio_route, "gsm-mode");
+    }
+
+    if (enable) {
+         my_data->gsm_mode_enabled = true;
+         ALOGD("%s: enabling gsm mode", __func__);
+         audio_route_apply_and_update_path(adev->audio_route, "gsm-mode");
+    }
+}
+
+void platform_set_echo_reference(struct audio_device *adev, bool enable,
+                                 audio_devices_t out_device)
 {
     struct platform_data *my_data = (struct platform_data *)adev->platform;
     snd_device_t snd_device = SND_DEVICE_NONE;
@@ -790,20 +830,27 @@
 
     if (strcmp(my_data->ec_ref_mixer_path, "")) {
         ALOGV("%s: disabling %s", __func__, my_data->ec_ref_mixer_path);
-        audio_route_reset_and_update_path(adev->audio_route, my_data->ec_ref_mixer_path);
+        audio_route_reset_and_update_path(adev->audio_route,
+                                          my_data->ec_ref_mixer_path);
     }
 
     if (enable) {
-        strlcpy(my_data->ec_ref_mixer_path, "echo-reference", sizeof(my_data->ec_ref_mixer_path));
         snd_device = platform_get_output_snd_device(adev->platform, &out);
+
         /*
          * If native audio device reference count > 0, then apply codec EC otherwise
-         * fallback to headphones if so or default
+         * fallback to Speakers with VBat if enabled or default
          */
         if (adev->snd_dev_ref_cnt[SND_DEVICE_OUT_HEADPHONES_44_1] > 0)
-            platform_add_backend_name(my_data->ec_ref_mixer_path, SND_DEVICE_OUT_HEADPHONES_44_1);
+            strlcpy(my_data->ec_ref_mixer_path, "echo-reference headphones-44.1",
+                    sizeof(my_data->ec_ref_mixer_path));
+        else if ((snd_device == SND_DEVICE_OUT_SPEAKER_VBAT) ||
+                 (snd_device == SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT))
+            strlcpy(my_data->ec_ref_mixer_path, "echo-reference speaker-vbat",
+                    sizeof(my_data->ec_ref_mixer_path));
         else
-            platform_add_backend_name(my_data->ec_ref_mixer_path, snd_device);
+            strlcpy(my_data->ec_ref_mixer_path, "echo-reference",
+                    sizeof(my_data->ec_ref_mixer_path));
 
         ALOGD("%s: enabling %s", __func__, my_data->ec_ref_mixer_path);
         audio_route_apply_and_update_path(adev->audio_route, my_data->ec_ref_mixer_path);
@@ -1011,6 +1058,7 @@
     backend_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
     backend_table[SND_DEVICE_OUT_HEADPHONES] = strdup("headphones");
     backend_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
+    backend_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
 }
 
 void get_cvd_version(char *cvd_version, struct audio_device *adev)
@@ -1066,7 +1114,38 @@
     void   *buff;
 };
 
-static void send_codec_cal(acdb_loader_get_calibration_t acdb_loader_get_calibration, int fd)
+static int send_vbat_adc_data_to_acdb(struct platform_data *plat_data, char *cal_type)
+{
+    int ret = 0;
+    struct mixer_ctl *ctl;
+    uint16_t vbat_adc_data[2];
+    struct platform_data *my_data = plat_data;
+    struct audio_device *adev = my_data->adev;
+
+    const char *mixer_ctl_name = "Vbat ADC data";
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer ctl name - %s",
+               __func__, mixer_ctl_name);
+        ret = -EINVAL;
+        goto done;
+    }
+
+    vbat_adc_data[0] = mixer_ctl_get_value(ctl, 0);
+    vbat_adc_data[1] = mixer_ctl_get_value(ctl, 1);
+
+    ALOGD("%s: Vbat ADC output values: Dcp1: %d , Dcp2: %d",
+           __func__, vbat_adc_data[0], vbat_adc_data[1]);
+
+    ret = my_data->acdb_set_codec_data(&vbat_adc_data[0], cal_type);
+
+done:
+    return ret;
+}
+
+static void send_codec_cal(acdb_loader_get_calibration_t acdb_loader_get_calibration,
+                           struct platform_data *plat_data, int fd)
 {
     int type;
 
@@ -1080,6 +1159,13 @@
             continue;
 
         ret = 0;
+
+        if((plat_data->is_vbat_speaker) && (WCD9XXX_VBAT_CAL == type)) {
+           ret = send_vbat_adc_data_to_acdb(plat_data, cal_name_info[type]);
+           if (ret < 0)
+               ALOGE("%s error in sending vbat adc data to acdb", __func__);
+        }
+
         calib.get_size = 1;
         ret = acdb_loader_get_calibration(cal_name_info[type],
                                           sizeof(struct param_data),
@@ -1140,7 +1226,7 @@
         return;
     }
 
-    send_codec_cal(acdb_loader_get_calibration, fd);
+    send_codec_cal(acdb_loader_get_calibration, plat_data, fd);
 }
 
 static int platform_acdb_init(void *platform)
@@ -1180,7 +1266,7 @@
     char baseband[PROPERTY_VALUE_MAX];
     char value[PROPERTY_VALUE_MAX];
     struct platform_data *my_data = NULL;
-    int retry_num = 0, snd_card_num = 0, key = 0, ret = 0;
+    int retry_num = 0, snd_card_num = 0, key = 0;
     const char *snd_card_name;
     char *cvd_version = NULL;
     char *snd_internal_name = NULL;
@@ -1212,7 +1298,12 @@
             continue;
         }
 
-        snd_card_name = mixer_get_name(adev->mixer);
+        snd_card_name = strdup(mixer_get_name(adev->mixer));
+        if (!snd_card_name) {
+            ALOGE("failed to allocate memory for snd_card_name\n");
+            free(my_data);
+            return NULL;
+        }
         ALOGV("%s: snd_card_name: %s", __func__, snd_card_name);
 
         my_data->hw_info = hw_info_init(snd_card_name);
@@ -1274,6 +1365,7 @@
                 ALOGE("%s: Failed to init audio route controls, aborting.",
                        __func__);
                 free(my_data);
+                free(snd_card_name);
                 return NULL;
             }
             adev->snd_card = snd_card_num;
@@ -1287,6 +1379,7 @@
     if (snd_card_num >= MAX_SND_CARD) {
         ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
         free(my_data);
+        free(snd_card_name);
         return NULL;
     }
 
@@ -1340,6 +1433,12 @@
         }
     }
 
+    /* Check if Vbat speaker enabled property is set, this should be done before acdb init */
+    bool ret = false;
+    ret = audio_extn_can_use_vbat();
+    if (ret)
+        my_data->is_vbat_speaker = true;
+
     my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
     my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
     if (my_data->acdb_handle == NULL) {
@@ -1402,6 +1501,14 @@
             ALOGE("%s: Could not find the symbol acdb_get_default_app_type from %s",
                   __func__, LIB_ACDB_LOADER);
 
+        my_data->acdb_set_codec_data = (acdb_set_codec_data_t)dlsym(
+                                                    my_data->acdb_handle,
+                                                    "acdb_loader_set_codec_data");
+        if (!my_data->acdb_set_codec_data)
+            ALOGE("%s: Could not find the symbol acdb_get_default_app_type from %s",
+                  __func__, LIB_ACDB_LOADER);
+
+
         my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
                                                     "acdb_loader_init_v2");
         if (my_data->acdb_init == NULL) {
@@ -1448,7 +1555,6 @@
     audio_extn_spkr_prot_init(adev);
 
     audio_extn_dolby_set_license(adev);
-    audio_hwdep_send_cal(my_data);
 
     /* init audio device arbitration */
     audio_extn_dev_arbi_init();
@@ -1472,6 +1578,7 @@
         strdup("SLIM_5_RX SampleRate");
 
     my_data->edid_info = NULL;
+    free(snd_card_name);
     return my_data;
 }
 
@@ -1554,13 +1661,20 @@
     return 0;
 }
 
-void platform_add_backend_name(char *mixer_path, snd_device_t snd_device)
+void platform_add_backend_name(char *mixer_path, snd_device_t snd_device,
+                               struct audio_usecase *usecase)
 {
     if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
         ALOGE("%s: Invalid snd_device = %d", __func__, snd_device);
         return;
     }
 
+    if((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+        !(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
+        ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
+        return;
+    }
+
     const char * suffix = backend_table[snd_device];
 
     if (suffix != NULL) {
@@ -1914,10 +2028,14 @@
     if (my_data->csd == NULL)
         return ret;
 
-    if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
-        audio_extn_spkr_prot_is_enabled())
-        acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
-    else
+    if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         audio_extn_spkr_prot_is_enabled()) {
+        if (my_data->is_vbat_speaker)
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
+        else
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
+    } else
         acdb_rx_id = acdb_device_table[out_snd_device];
 
     acdb_tx_id = acdb_device_table[in_snd_device];
@@ -1974,10 +2092,14 @@
     if (my_data->csd == NULL)
         return ret;
 
-    if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
-        audio_extn_spkr_prot_is_enabled())
-        acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
-    else
+    if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         audio_extn_spkr_prot_is_enabled()) {
+        if (my_data->is_vbat_speaker)
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
+         else
+            acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
+    } else
         acdb_rx_id = acdb_device_table[out_snd_device];
 
     acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2236,7 +2358,10 @@
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
         } else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
-            snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+                if (my_data->is_vbat_speaker)
+                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+                else
+                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
         } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
                    devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
             snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -2275,6 +2400,8 @@
             snd_device = SND_DEVICE_OUT_SPEAKER_EXTERNAL_2;
         else if (adev->speaker_lr_swap)
             snd_device = SND_DEVICE_OUT_SPEAKER_REVERSE;
+        else if (my_data->is_vbat_speaker)
+            snd_device = SND_DEVICE_OUT_SPEAKER_VBAT;
         else
             snd_device = SND_DEVICE_OUT_SPEAKER;
     } else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index b8f8cee..54406b6 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -69,6 +69,7 @@
     SND_DEVICE_OUT_SPEAKER_EXTERNAL_1,
     SND_DEVICE_OUT_SPEAKER_EXTERNAL_2,
     SND_DEVICE_OUT_SPEAKER_REVERSE,
+    SND_DEVICE_OUT_SPEAKER_VBAT,
     SND_DEVICE_OUT_HEADPHONES,
     SND_DEVICE_OUT_HEADPHONES_44_1,
     SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
@@ -76,6 +77,7 @@
     SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2,
     SND_DEVICE_OUT_VOICE_HANDSET,
     SND_DEVICE_OUT_VOICE_SPEAKER,
+    SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
     SND_DEVICE_OUT_VOICE_HEADPHONES,
     SND_DEVICE_OUT_HDMI,
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
@@ -97,6 +99,8 @@
     SND_DEVICE_OUT_ANC_HANDSET,
     SND_DEVICE_OUT_SPEAKER_PROTECTED,
     SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+    SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
+    SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
     SND_DEVICE_OUT_END,
 
     /*
@@ -217,6 +221,7 @@
 #define DEEP_BUFFER_PCM_DEVICE 0
 #define AUDIO_RECORD_PCM_DEVICE 0
 #define MULTIMEDIA2_PCM_DEVICE 1
+#define MULTIMEDIA3_PCM_DEVICE 4
 #define FM_PLAYBACK_PCM_DEVICE 5
 #define FM_CAPTURE_PCM_DEVICE  6
 #define HFP_PCM_RX 5
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 516f2d7..3ab5191 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -29,7 +29,8 @@
 const char *platform_get_snd_device_name(snd_device_t snd_device);
 int platform_get_snd_device_name_extn(void *platform, snd_device_t snd_device,
                                       char *device_name);
-void platform_add_backend_name(char *mixer_path, snd_device_t snd_device);
+void platform_add_backend_name(char *mixer_path, snd_device_t snd_device,
+                               struct audio_usecase *usecase);
 bool platform_send_gain_dep_cal(void *platform, int level);
 int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type);
 int platform_get_snd_device_index(char *snd_device_index_name);
@@ -119,4 +120,5 @@
 int platform_set_device_params(struct stream_out *out, int param, int value);
 int platform_set_audio_device_interface(const char * device_name, const char *intf_name,
                                         const char * codec_type);
+void platform_set_gsm_mode(void *platform, bool enable);
 #endif // AUDIO_PLATFORM_API_H
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 960b46e..3cf370d 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -42,6 +42,7 @@
 #define AUDIO_PARAMETER_KEY_DEVICE_MUTE         "device_mute"
 #define AUDIO_PARAMETER_KEY_DIRECTION           "direction"
 #define AUDIO_PARAMETER_KEY_IN_CALL             "in_call"
+#define AUDIO_PARAMETER_KEY_CALL_TYPE           "call_type"
 
 #define VOICE_EXTN_PARAMETER_VALUE_MAX_LEN 256
 
@@ -515,6 +516,20 @@
         }
     }
 
+    err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_CALL_TYPE, str_value,
+                            sizeof(str_value));
+    if (err >= 0) {
+          str_parms_del(parms, AUDIO_PARAMETER_KEY_CALL_TYPE);
+          ALOGD("%s: call type is %s",__func__,str_value);
+
+           /* Expected call types are CDMA/GSM/WCDMA/LTE/TDSDMA/WLAN/UNKNOWN */
+           if (!strncmp("GSM", str_value, sizeof("GSM"))) {
+               platform_set_gsm_mode(adev->platform, true);
+           } else {
+               platform_set_gsm_mode(adev->platform, false);
+           }
+    }
+
 done:
     ALOGV("%s: exit with code(%d)", __func__, ret);
     free(kv_pairs);
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
index d5a362a..52ed11d 100644
--- a/policy_hal/Android.mk
+++ b/policy_hal/Android.mk
@@ -51,6 +51,14 @@
     LOCAL_CFLAGS += -DAAC_ADTS_OFFLOAD_ENABLED
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_SPK)),true)
+    LOCAL_CFLAGS += -DAUDIO_EXTN_HDMI_SPK_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PROXY_DEVICE)),true)
+    LOCAL_CFLAGS += -DAUDIO_EXTN_AFE_PROXY_ENABLED
+endif
+
 LOCAL_MODULE := libaudiopolicymanager
 
 include $(BUILD_SHARED_LIBRARY)
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index f105e15..9fd6838 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -122,6 +122,15 @@
         // handle output device connection
         case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
             if (index >= 0) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+                if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                   if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                        mHdmiAudioDisabled = false;
+                    } else {
+                        mHdmiAudioEvent = true;
+                    }
+                }
+#endif
                 ALOGW("setDeviceConnectionState() device already connected: %x", device);
                 return INVALID_OPERATION;
             }
@@ -129,6 +138,20 @@
 
             // register new device as available
             index = mAvailableOutputDevices.add(devDesc);
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+            if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                    mHdmiAudioDisabled = false;
+                } else {
+                    mHdmiAudioEvent = true;
+                }
+                if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
+                    mAvailableOutputDevices.remove(devDesc);
+                    ALOGW("HDMI sink not connected, do not route audio to HDMI out");
+                    return INVALID_OPERATION;
+                }
+            }
+#endif
             if (index >= 0) {
                 sp<HwModule> module = mHwModules.getModuleForDevice(device);
                 if (module == 0) {
@@ -164,6 +187,15 @@
         // handle output device disconnection
         case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
             if (index < 0) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+                if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                    if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                        mHdmiAudioDisabled = true;
+                    } else {
+                        mHdmiAudioEvent = false;
+                    }
+                }
+#endif
                 ALOGW("setDeviceConnectionState() device not connected: %x", device);
                 return INVALID_OPERATION;
             }
@@ -177,7 +209,15 @@
 
             // remove device from available output devices
             mAvailableOutputDevices.remove(devDesc);
-
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+            if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                    mHdmiAudioDisabled = true;
+                } else {
+                    mHdmiAudioEvent = false;
+                }
+            }
+#endif
             checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
 
             // Propagate device availability to Engine
@@ -524,9 +564,8 @@
                 (isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION)
                 && (!isStrategyActive(mPrimaryOutput,STRATEGY_MEDIA)))) {
         device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
-    } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)||
-                (isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION_RESPECTFUL)
-                && (!isStrategyActive(mPrimaryOutput, STRATEGY_MEDIA)))) {
+    } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL) ||
+                isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION_RESPECTFUL)) {
         device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
     } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
         device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
@@ -547,6 +586,7 @@
 {
     ALOGV("setPhoneState() state %d", state);
     // store previous phone state for management of sonification strategy below
+    audio_devices_t newDevice = AUDIO_DEVICE_NONE;
     int oldState = mEngine->getPhoneState();
 
     if (mEngine->setPhoneState(state) != NO_ERROR) {
@@ -564,8 +604,7 @@
                 if (stream == AUDIO_STREAM_PATCH) {
                     continue;
                 }
-
-            handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
+                handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
             }
         }
 
@@ -837,7 +876,14 @@
             setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
         }
     }
-
+    //update device for all non-primary outputs
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        audio_io_handle_t output = mOutputs.keyAt(i);
+        if (output != mPrimaryOutput->mIoHandle) {
+            newDevice = getNewOutputDevice(mOutputs.valueFor(output), false /*fromCache*/);
+            setOutputDevice(mOutputs.valueFor(output), newDevice, (newDevice != AUDIO_DEVICE_NONE));
+        }
+    }
     // if entering in call state, handle special case of active streams
     // pertaining to sonification strategy see handleIncallSonification()
     if (isStateInCall(state)) {
@@ -848,7 +894,7 @@
                if (stream == AUDIO_STREAM_PATCH) {
                     continue;
                 }
-            handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
+                handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
            }
         }
     }
@@ -861,6 +907,53 @@
         mLimitRingtoneVolume = false;
     }
 }
+
+void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
+                                         audio_policy_forced_cfg_t config)
+{
+    ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
+
+    if (mEngine->setForceUse(usage, config) != NO_ERROR) {
+        ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
+        return;
+    }
+    bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
+            (usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
+            (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
+
+    // check for device and output changes triggered by new force usage
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+    if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+        audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+        updateCallRouting(newDevice);
+    }
+    // Use reverse loop to make sure any low latency usecases (generally tones)
+    // are not routed before non LL usecases (generally music).
+    // We can safely assume that LL output would always have lower index,
+    // and use this work-around to avoid routing of output with music stream
+    // from the context of short lived LL output.
+    // Note: in case output's share backend(HAL sharing is implicit) all outputs
+    //       gets routing update while processing first output itself.
+    for (size_t i = mOutputs.size(); i > 0; i--) {
+        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
+        audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+        if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || outputDesc != mPrimaryOutput) {
+            setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+        }
+        if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+            applyStreamVolumes(outputDesc, newDevice, 0, true);
+        }
+    }
+
+    audio_io_handle_t activeInput = mInputs.getActiveInput();
+    if (activeInput != 0) {
+        setInputDevice(activeInput, getNewInputDevice(activeInput));
+    }
+
+}
+
 status_t AudioPolicyManagerCustom::stopSource(sp<SwAudioOutputDescriptor> outputDesc,
                                             audio_stream_type_t stream,
                                             bool forceDeviceUpdate)
@@ -869,10 +962,10 @@
     handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
 
     // handle special case for sonification while in call
-    if (isInCall()) {
+    if (isInCall() && (outputDesc->mRefCount[stream] == 1)) {
         if (outputDesc->isDuplicated()) {
-            handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
-            handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
+            handleIncallSonification(stream, false, false, outputDesc->mOutput1->mIoHandle);
+            handleIncallSonification(stream, false, false, outputDesc->mOutput2->mIoHandle);
         }
         handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
     }
@@ -884,6 +977,7 @@
         // store time at which the stream was stopped - see isStreamActive()
         if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) {
             outputDesc->mStopTime[stream] = systemTime();
+            audio_devices_t prevDevice = outputDesc->device();
             audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
             // delay the device switch by twice the latency because stopOutput() is executed when
             // the track stop() command is received and at that time the audio track buffer can
@@ -901,10 +995,17 @@
                         desc->isActive() &&
                         outputDesc->sharesHwModuleWith(desc) &&
                         (newDevice != desc->device())) {
-                    setOutputDevice(desc,
-                                    getNewOutputDevice(desc, false /*fromCache*/),
+                        audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
+                        uint32_t delayMs;
+                        if (dev == prevDevice) {
+                            delayMs = 0;
+                        } else {
+                            delayMs = outputDesc->mLatency*2;
+                        }
+                        setOutputDevice(desc,
+                                    dev,
                                     true,
-                                    outputDesc->latency()*2);
+                                    delayMs);
                 }
             }
             // update the outputs if stopping one with a stream that can affect notification routing
@@ -1361,6 +1462,7 @@
     }
 #endif
 
+#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
     /*
     * WFD audio routes back to target speaker when starting a ringtone playback.
     * This is because primary output is reused for ringtone, so output device is
@@ -1380,6 +1482,8 @@
         else //route every thing else to ULL path
             flags = AUDIO_OUTPUT_FLAG_FAST;
     }
+#endif
+
     // open a direct output if required by specified parameters
     // force direct flag if offload flag is set: offloading implies a direct output stream
     // and all common behaviors are driven by checking only the direct flag
@@ -1720,7 +1824,7 @@
         // Move tracks associated to this strategy from previous output to new output
         for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
             // Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder)
-            if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE && (i != AUDIO_STREAM_PATCH)) {
+            if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE && (i != AUDIO_STREAM_PATCH))) {
                ALOGD("Invalidate on releaseInput for stream :: %d ", i);
                //FIXME see fixme on name change
                mpClientInterface->invalidateStream((audio_stream_type_t)i);
@@ -1818,8 +1922,42 @@
 }
 
 AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
-    : AudioPolicyManager(clientInterface)
+    : AudioPolicyManager(clientInterface),
+      mHdmiAudioDisabled(false),
+      mHdmiAudioEvent(false),
+      mPrevPhoneState(0)
 {
+    char ssr_enabled[PROPERTY_VALUE_MAX] = {0};
+    bool prop_ssr_enabled = false;
+
+    if (property_get("ro.qc.sdk.audio.ssr", ssr_enabled, NULL)) {
+        prop_ssr_enabled = atoi(ssr_enabled) || !strncmp("true", ssr_enabled, 4);
+    }
+
+    for (size_t i = 0; i < mHwModules.size(); i++) {
+        ALOGV("Hw module %d", i);
+        for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
+            const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
+            ALOGV("Input profile ", j);
+            for (size_t k = 0; k  < inProfile->mChannelMasks.size(); k++) {
+                audio_channel_mask_t channelMask =
+                    inProfile->mChannelMasks.itemAt(k);
+                ALOGV("Channel Mask %x size %d", channelMask,
+                    inProfile->mChannelMasks.size());
+                if (AUDIO_CHANNEL_IN_5POINT1 == channelMask) {
+                    if (!prop_ssr_enabled) {
+                        ALOGI("removing AUDIO_CHANNEL_IN_5POINT1 from"
+                            " input profile as SSR(surround sound record)"
+                            " is not supported on this chipset variant");
+                        inProfile->mChannelMasks.removeItemsAt(k, 1);
+                        ALOGV("Channel Mask size now %d",
+                            inProfile->mChannelMasks.size());
+                    }
+                }
+            }
+        }
+    }
+
 #ifdef RECORD_PLAY_CONCURRENCY
     mIsInputRequestOnProgress = false;
 #endif
@@ -1829,31 +1967,4 @@
     mFallBackflag = getFallBackPath();
 #endif
 }
-audio_devices_t AudioPolicyManagerCustom::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
-{
-    audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
-    audio_devices_t device = AUDIO_DEVICE_NONE;
-    switch (strategy) {
-        case STRATEGY_SONIFICATION:
-        case STRATEGY_ENFORCED_AUDIBLE:
-        case STRATEGY_ACCESSIBILITY:
-        case STRATEGY_REROUTING:
-        case STRATEGY_MEDIA:
-            if (strategy != STRATEGY_SONIFICATION){
-                // no sonification on WFD sink
-                device |= availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY;
-                if (device != AUDIO_DEVICE_NONE) {
-                    ALOGV("Found proxy for strategy %d", strategy);
-                    return device;
-                }
-            }
-            break;
-        default:
-            ALOGV("getDeviceForStrategy() unknown strategy: %d", strategy);
-            break;
-    }
-    device = AudioPolicyManager::getDeviceForStrategy(strategy, fromCache);
-    return device;
-}
-
 }
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index b5a4461..dfaad65 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -36,7 +36,7 @@
 #define AUDIO_FORMAT_AAC_ADTS 0x1E000000UL
 #endif
 
-#ifndef AFE_PROXY_ENABLED
+#ifndef AUDIO_EXTN_AFE_PROXY_ENABLED
 #define AUDIO_DEVICE_OUT_PROXY 0x1000000
 #endif
 // ----------------------------------------------------------------------------
@@ -54,7 +54,8 @@
                                           const char *device_address,
                                           const char *device_name);
         virtual void setPhoneState(audio_mode_t state);
-
+        virtual void setForceUse(audio_policy_force_use_t usage,
+                                 audio_policy_forced_cfg_t config);
 
         virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
 
@@ -74,7 +75,6 @@
         // indicates to the audio policy manager that the input stops being used.
         virtual status_t stopInput(audio_io_handle_t input,
                                    audio_session_t session);
-        virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, bool fromCache);
 protected:
 
          status_t checkAndSetVolume(audio_stream_type_t stream,
@@ -101,9 +101,9 @@
          status_t stopSource(sp<SwAudioOutputDescriptor> outputDesc,
                             audio_stream_type_t stream,
                             bool forceDeviceUpdate);
-        // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON   313
-        // returns 0 if no mute/unmute event happened, the largest latency of the device where   314
-        //   the mute/unmute happened 315
+        // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
+        // returns 0 if no mute/unmute event happened, the largest latency of the device where
+        //   the mute/unmute happened
         uint32_t handleEventForBeacon(int){return 0;}
         uint32_t setBeaconMute(bool){return 0;}
 #ifdef VOICE_CONCURRENCY
@@ -119,14 +119,9 @@
         //parameter indicates if HDMI plug in/out detected
         bool mHdmiAudioEvent;
 private:
-        static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
-                int indexInUi);
         // updates device caching and output for streams that can influence the
         //    routing of notifications
         void handleNotificationRoutingForStream(audio_stream_type_t stream);
-        static bool isVirtualInputDevice(audio_devices_t device);
-        static bool deviceDistinguishesOnAddress(audio_devices_t device);
-        uint32_t nextUniqueId();
         // internal method to return the output handle for the given device and format
         audio_io_handle_t getOutputForDevice(
                 audio_devices_t device,
@@ -156,8 +151,6 @@
         // Used for record + playback concurrency
         bool mIsInputRequestOnProgress;
 #endif
-
-
 };
 
 };
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 192320a..929991c 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -21,6 +21,11 @@
     LOCAL_SRC_FILES += hw_accelerator.c
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUDIOSPHERE)),true)
+    LOCAL_CFLAGS += -DAUDIOSPHERE_ENABLED
+    LOCAL_SRC_FILES += asphere.c
+endif
+
 LOCAL_CFLAGS+= -O2 -fvisibility=hidden
 
 ifneq ($(strip $(AUDIO_FEATURE_DISABLED_DTS_EAGLE)),true)
diff --git a/post_proc/asphere.c b/post_proc/asphere.c
new file mode 100644
index 0000000..bbf1056
--- /dev/null
+++ b/post_proc/asphere.c
@@ -0,0 +1,301 @@
+/* Copyright (c) 2015, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ *     * Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *     * Redistributions in binary form must reproduce the above
+ *       copyright notice, this list of conditions and the following
+ *       disclaimer in the documentation and/or other materials provided
+ *       with the distribution.
+ *     * Neither the name of The Linux Foundation nor the names of its
+ *       contributors may be used to endorse or promote products derived
+ *       from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#define LOG_TAG "audio_pp_asphere"
+/*#define LOG_NDEBUG 0*/
+
+#include <errno.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdbool.h>
+#include <sys/stat.h>
+#include <cutils/log.h>
+#include <cutils/list.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+#include <hardware/audio_effect.h>
+#include "bundle.h"
+#include "equalizer.h"
+#include "bass_boost.h"
+#include "virtualizer.h"
+#include "reverb.h"
+#include "asphere.h"
+
+#define ASPHERE_MIXER_NAME  "MSM ASphere Set Param"
+
+#define AUDIO_PARAMETER_KEY_ASPHERE_STATUS  "asphere_status"
+#define AUDIO_PARAMETER_KEY_ASPHERE_ENABLE   "asphere_enable"
+#define AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH "asphere_strength"
+
+#define AUDIO_ASPHERE_EVENT_NODE "/data/misc/audio_pp/event_node"
+
+enum {
+    ASPHERE_ACTIVE = 0,
+    ASPHERE_SUSPENDED,
+    ASPHERE_ERROR
+};
+
+struct asphere_module {
+    bool enabled;
+    int status;
+    int strength;
+    pthread_mutex_t lock;
+    int init_status;
+};
+
+static struct asphere_module asphere;
+pthread_once_t asphere_once = PTHREAD_ONCE_INIT;
+
+static int asphere_create_app_notification_node(void)
+{
+    int fd;
+    if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_CREAT|O_TRUNC|O_WRONLY,
+                            S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH)) < 0) {
+        ALOGE("creating notification node failed %d", errno);
+        return -EINVAL;
+    }
+    chmod(AUDIO_ASPHERE_EVENT_NODE, S_IRWXU|S_IRGRP|S_IXGRP|S_IROTH);
+    close(fd);
+    ALOGD("%s: successfully created notification node %s",
+                               __func__, AUDIO_ASPHERE_EVENT_NODE);
+    return 0;
+}
+
+static int asphere_notify_app(void)
+{
+    int fd;
+    if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_TRUNC|O_WRONLY)) < 0) {
+        ALOGE("opening notification node failed %d", errno);
+        return -EINVAL;
+    }
+    close(fd);
+    ALOGD("%s: successfully opened notification node", __func__);
+    return 0;
+}
+
+static int asphere_get_values_from_mixer(void)
+{
+    int ret = 0, val[2] = {-1, -1};
+    struct mixer_ctl *ctl = NULL;
+    struct mixer *mixer = mixer_open(MIXER_CARD);
+    if (mixer)
+        ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
+    if (!ctl) {
+        ALOGE("%s: could not get ctl for mixer cmd - %s",
+              __func__, ASPHERE_MIXER_NAME);
+        return -EINVAL;
+    }
+    ret = mixer_ctl_get_array(ctl, val, sizeof(val)/sizeof(val[0]));
+    if (!ret) {
+        asphere.enabled = (val[0] == 0) ? false : true;
+        asphere.strength = val[1];
+    }
+    ALOGD("%s: returned %d, enabled:%d, strength:%d",
+          __func__, ret, val[0], val[1]);
+
+    return ret;
+}
+
+static int asphere_set_values_to_mixer(void)
+{
+    int ret = 0, val[2] = {-1, -1};
+    struct mixer_ctl *ctl = NULL;
+    struct mixer *mixer = mixer_open(MIXER_CARD);
+    if (mixer)
+        ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
+    if (!ctl) {
+        ALOGE("%s: could not get ctl for mixer cmd - %s",
+              __func__, ASPHERE_MIXER_NAME);
+        return -EINVAL;
+    }
+    val[0] = ((asphere.status == ASPHERE_ACTIVE) && asphere.enabled) ? 1 : 0;
+    val[1] = asphere.strength;
+
+    ret = mixer_ctl_set_array(ctl, val, sizeof(val)/sizeof(val[0]));
+    ALOGD("%s: returned %d, enabled:%d, strength:%d",
+          __func__, ret, val[0], val[1]);
+
+    return ret;
+}
+
+static void asphere_init_once() {
+    ALOGD("%s", __func__);
+    pthread_mutex_init(&asphere.lock, NULL);
+    asphere.init_status = 1;
+    asphere_get_values_from_mixer();
+    asphere_create_app_notification_node();
+}
+
+static int asphere_init() {
+    pthread_once(&asphere_once, asphere_init_once);
+    return asphere.init_status;
+}
+
+void asphere_set_parameters(struct str_parms *parms)
+{
+    int ret = 0;
+    bool enable = false;
+    int strength = -1;
+    char value[32] = {0};
+    char propValue[PROPERTY_VALUE_MAX] = {0};
+    bool set_enable = false, set_strength = false;
+
+    if (!property_get("audio.pp.asphere.enabled", propValue, "false") ||
+        (strncmp("true", propValue, 4) != 0)) {
+        ALOGV("%s: property not set!!! not doing anything", __func__);
+        return;
+    }
+    if (asphere_init() != 1) {
+        ALOGW("%s: init check failed!!!", __func__);
+        return;
+    }
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
+                                                  value, sizeof(value));
+    if (ret > 0) {
+        enable = (atoi(value) == 1) ? true : false;
+        set_enable = true;
+    }
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
+                                                   value, sizeof(value));
+    if (ret > 0) {
+        strength = atoi(value);
+        if (strength >= 0 && strength <= 1000)
+            set_strength = true;
+    }
+
+    if (set_enable || set_strength) {
+        pthread_mutex_lock(&asphere.lock);
+        asphere.enabled = set_enable ? enable : asphere.enabled;
+        asphere.strength = set_strength ? strength : asphere.strength;
+        ret = asphere_set_values_to_mixer();
+        pthread_mutex_unlock(&asphere.lock);
+        ALOGV("%s: exit ret %d", __func__, ret);
+    }
+}
+
+void asphere_get_parameters(struct str_parms *query,
+                                      struct str_parms *reply)
+{
+    char value[32] = {0};
+    char propValue[PROPERTY_VALUE_MAX] = {0};
+    int get_status, get_enable, get_strength, ret;
+
+    if (!property_get("audio.pp.asphere.enabled", propValue, "false") ||
+        (strncmp("true", propValue, 4) != 0)) {
+        ALOGV("%s: property not set!!! not doing anything", __func__);
+        return;
+    }
+    if (asphere_init() != 1) {
+        ALOGW("%s: init check failed!!!", __func__);
+        return;
+    }
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
+                                                 value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
+                                                     asphere.status);
+    }
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
+                                                 value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
+                                              asphere.enabled ? 1 : 0);
+    }
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
+                                                  value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
+                                                     asphere.strength);
+    }
+}
+
+static bool effect_needs_asphere_concurrency_handling(effect_context_t *context)
+{
+    if (memcmp(&context->desc->type,
+                &equalizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+        memcmp(&context->desc->type,
+                &bassboost_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+        memcmp(&context->desc->type,
+                &virtualizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+        memcmp(&context->desc->type,
+                &ins_preset_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+        memcmp(&context->desc->type,
+                &ins_env_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0)
+        return true;
+
+    return false;
+}
+
+void handle_asphere_on_effect_enabled(bool enable,
+                                      effect_context_t *context,
+                                      struct listnode *created_effects)
+{
+    struct listnode *node;
+    char propValue[PROPERTY_VALUE_MAX] = {0};
+
+    ALOGV("%s: effect %0x", __func__, context->desc->type.timeLow);
+    if (!property_get("audio.pp.asphere.enabled", propValue, "false") ||
+        (strncmp("true", propValue, 4) != 0)) {
+        ALOGV("%s: property not set!!! not doing anything", __func__);
+        return;
+    }
+    if (asphere_init() != 1) {
+        ALOGW("%s: init check failed!!!", __func__);
+        return;
+    }
+
+    if (!effect_needs_asphere_concurrency_handling(context)) {
+        ALOGV("%s: effect %0x, do not need concurrency handling",
+                                 __func__, context->desc->type.timeLow);
+        return;
+    }
+
+    list_for_each(node, created_effects) {
+        effect_context_t *fx_ctxt = node_to_item(node,
+                                                 effect_context_t,
+                                                 effects_list_node);
+        if (fx_ctxt != NULL &&
+            effect_needs_asphere_concurrency_handling(fx_ctxt) == true &&
+            fx_ctxt != context && effect_is_active(fx_ctxt) == true) {
+            ALOGV("%s: found another effect %0x, skip processing %0x", __func__,
+                      fx_ctxt->desc->type.timeLow, context->desc->type.timeLow);
+            return;
+        }
+    }
+    pthread_mutex_lock(&asphere.lock);
+    asphere.status = enable ? ASPHERE_SUSPENDED : ASPHERE_ACTIVE;
+    asphere_set_values_to_mixer();
+    asphere_notify_app();
+    pthread_mutex_unlock(&asphere.lock);
+}
diff --git a/post_proc/asphere.h b/post_proc/asphere.h
new file mode 100644
index 0000000..d0e6830
--- /dev/null
+++ b/post_proc/asphere.h
@@ -0,0 +1,50 @@
+/* Copyright (c) 2015, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ *     * Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *     * Redistributions in binary form must reproduce the above
+ *       copyright notice, this list of conditions and the following
+ *       disclaimer in the documentation and/or other materials provided
+ *       with the distribution.
+ *     * Neither the name of The Linux Foundation nor the names of its
+ *       contributors may be used to endorse or promote products derived
+ *       from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+
+#ifndef OFFLOAD_ASPHERE_H_
+#define OFFLOAD_ASPHERE_H_
+
+#include <cutils/str_parms.h>
+#include <cutils/list.h>
+#include "bundle.h"
+
+#ifdef AUDIOSPHERE_ENABLED
+void asphere_get_parameters(struct str_parms *query,
+                            struct str_parms *reply);
+void asphere_set_parameters(struct str_parms *reply);
+void handle_asphere_on_effect_enabled(bool enable,
+                                      effect_context_t *context,
+                                      struct listnode *created_effects);
+#else
+#define asphere_get_parameters(query, reply) (0)
+#define asphere_set_parameters(parms)  (0)
+#define handle_asphere_on_effect_enabled(enable, context, created_effects) (0)
+#endif /* AUDIOSPHERE_ENABLED */
+
+#endif /* OFFLOAD_ASPHERE_H_ */
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index 6f1bdfa..1449eb4 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -51,6 +51,7 @@
 #include "bass_boost.h"
 #include "virtualizer.h"
 #include "reverb.h"
+#include "asphere.h"
 
 #ifdef DTS_EAGLE
 #include "effect_util.h"
@@ -456,6 +457,24 @@
 }
 
 /*
+ * Effect Bundle Set and get param operations.
+ * currently only handles audio sphere scenario,
+ * but the interface itself can be utilized for any effect.
+ */
+__attribute__ ((visibility ("default")))
+void offload_effects_bundle_get_parameters(struct str_parms *query,
+                                           struct str_parms *reply)
+{
+    asphere_get_parameters(query, reply);
+}
+
+__attribute__ ((visibility ("default")))
+void offload_effects_bundle_set_parameters(struct str_parms *parms)
+{
+    asphere_set_parameters(parms);
+}
+
+/*
  * Effect operations
  */
 int set_config(effect_context_t *context, effect_config_t *config)
@@ -811,6 +830,7 @@
             status = -ENOSYS;
             goto exit;
         }
+        handle_asphere_on_effect_enabled(true, context, &created_effects_list);
         context->state = EFFECT_STATE_ACTIVE;
         if (context->ops.enable)
             context->ops.enable(context);
@@ -825,6 +845,7 @@
             status = -ENOSYS;
             goto exit;
         }
+        handle_asphere_on_effect_enabled(false, context, &created_effects_list);
         context->state = EFFECT_STATE_INITIALIZED;
         if (context->ops.disable)
             context->ops.disable(context);
diff --git a/post_proc/volume_listener.c b/post_proc/volume_listener.c
index 51f8803..d4f418a 100644
--- a/post_proc/volume_listener.c
+++ b/post_proc/volume_listener.c
@@ -344,7 +344,7 @@
     if (in_buffer->raw != out_buffer->raw) {
         memcpy(out_buffer->raw, in_buffer->raw, out_buffer->frameCount * 2 * sizeof(int16_t));
     } else {
-        ALOGW("%s: something wrong, didn't handle in_buffer and out_buffer same address case",
+        ALOGV("%s: something wrong, didn't handle in_buffer and out_buffer same address case",
               __func__);
     }
 
@@ -681,6 +681,11 @@
     bool recompute_flag = false;
     int active_stream_count = 0;
     ALOGV("%s context %p", __func__, handle);
+    if (recv_contex == NULL || recv_contex->desc == NULL) {
+        ALOGE("%s: Got invalid handle while release, DO NOTHING ", __func__);
+        return status;
+    }
+
     pthread_mutex_lock(&vol_listner_init_lock);
 
     // check if the handle/context provided is valid
@@ -690,13 +695,14 @@
             && (context->session_id == recv_contex->session_id)
             && (context->stream_type == recv_contex->stream_type)) {
             ALOGV("--- Found something to remove ---");
-            list_remove(&context->effect_list_node);
             PRINT_STREAM_TYPE(context->stream_type);
             if (context->dev_id && AUDIO_DEVICE_OUT_SPEAKER) {
                 recompute_flag = true;
             }
+            list_remove(&context->effect_list_node);
             free(context);
             status = 0;
+            break;
         } else {
             ++active_stream_count;
         }
diff --git a/visualizer/offload_visualizer.c b/visualizer/offload_visualizer.c
index d363b77..b2f0952 100644
--- a/visualizer/offload_visualizer.c
+++ b/visualizer/offload_visualizer.c
@@ -300,20 +300,40 @@
     return false;
 }
 
-int configure_proxy_capture(struct mixer *mixer, int value) {
-    const char *proxy_ctl_name = "AFE_PCM_RX Audio Mixer MultiMedia4";
+int set_control(const char* name, struct mixer *mixer, int value) {
     struct mixer_ctl *ctl;
 
+    ctl = mixer_get_ctl_by_name(mixer, name);
+    if (ctl == NULL) {
+        ALOGW("%s: could not get %s ctl", __func__, name);
+        return -EINVAL;
+    }
+    if (mixer_ctl_set_value(ctl, 0, value) != 0) {
+        ALOGW("%s: error setting value %d on %s ", __func__, value, name);
+        return -EINVAL;
+    }
+
+    return 0;
+}
+
+int configure_proxy_capture(struct mixer *mixer, int value) {
+    int retval = 0;
+
     if (value && acdb_send_audio_cal)
         acdb_send_audio_cal(AFE_PROXY_ACDB_ID, ACDB_DEV_TYPE_OUT);
 
-    ctl = mixer_get_ctl_by_name(mixer, proxy_ctl_name);
-    if (ctl == NULL) {
-        ALOGW("%s: could not get %s ctl", __func__, proxy_ctl_name);
-        return -EINVAL;
-    }
-    if (mixer_ctl_set_value(ctl, 0, value) != 0)
-        ALOGW("%s: error setting value %d on %s ", __func__, value, proxy_ctl_name);
+    retval = set_control("AFE_PCM_RX Audio Mixer MultiMedia4", mixer, value);
+
+    if (retval != 0)
+        return retval;
+
+    // Extending visualizer to capture for compress2 path as well.
+    // for extending it to multiple offload either this needs to be extended
+    // or need to find better solution to enable only active offload sessions
+
+    retval = set_control("AFE_PCM_RX Audio Mixer MultiMedia7", mixer, value);
+    if (retval != 0)
+        return retval;
 
     return 0;
 }