diff --git a/configs/msm8937/audio_output_policy.conf b/configs/msm8937/audio_output_policy.conf
index d983d9a..b19fbb0 100644
--- a/configs/msm8937/audio_output_policy.conf
+++ b/configs/msm8937/audio_output_policy.conf
@@ -18,6 +18,13 @@
     bit_width 16
     app_type 69937
   }
+  voip_rx {
+    flags AUDIO_OUTPUT_FLAG_VOIP_RX|AUDIO_OUTPUT_FLAG_DIRECT
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 8000|16000|32000|48000
+    bit_width 16
+    app_type 69946
+  }
   deep_buffer {
     flags AUDIO_OUTPUT_FLAG_DEEP_BUFFER
     formats AUDIO_FORMAT_PCM_16_BIT
diff --git a/configs/msm8937/audio_platform_info.xml b/configs/msm8937/audio_platform_info.xml
index 3961944..317274c 100644
--- a/configs/msm8937/audio_platform_info.xml
+++ b/configs/msm8937/audio_platform_info.xml
@@ -1,5 +1,5 @@
 <?xml version="1.0" encoding="ISO-8859-1"?>
-<!-- Copyright (c) 2015 - 2016, 2018, The Linux Foundation. All rights reserved.         -->
+<!-- Copyright (c) 2015 - 2016, 2018, 2020, 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 -->
@@ -48,6 +48,11 @@
         <usecase name="USECASE_VOICEMMODE1_CALL" type="out" id="34"/>
         <usecase name="USECASE_VOICEMMODE2_CALL" type="in" id="35"/>
         <usecase name="USECASE_VOICEMMODE2_CALL" type="out" id="35"/>
+        <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="27" />
+        <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="27" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsa881x.0f"/>
@@ -70,6 +75,7 @@
         <device name="SND_DEVICE_IN_HDMI_MIC" acdb_id="4"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO" acdb_id="125"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC" acdb_id="125"/>
+        <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" acdb_id="8"/>
     </acdb_ids>
 
     <backend_names>
@@ -172,6 +178,10 @@
         <device name="SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB" backend="wsa-speaker-and-bt-sco-wb" interface="PRI_MI2S_RX-and-INT_BT_SCO_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" backend="vbat-voice-speaker" interface="PRI_MI2S_RX"/>
         <device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB_NREC" interface="INT_BT_SCO_TX"/>
     </backend_names>
     <!-- below values are for ref purpose to OEM, doesn't contain actual hardware info on MTP -->
     <microphone_characteristics>
diff --git a/configs/msm8937/audio_platform_info_intcodec.xml b/configs/msm8937/audio_platform_info_intcodec.xml
index 9415975..8fa7ae9 100644
--- a/configs/msm8937/audio_platform_info_intcodec.xml
+++ b/configs/msm8937/audio_platform_info_intcodec.xml
@@ -50,6 +50,11 @@
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
         <usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="26"/>
+        <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="27" />
+        <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="27" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsa881x.0f"/>
@@ -72,6 +77,7 @@
         <device name="SND_DEVICE_IN_HDMI_MIC" acdb_id="4"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO" acdb_id="125"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC" acdb_id="125"/>
+        <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" acdb_id="8"/>
     </acdb_ids>
 
     <backend_names>
@@ -174,6 +180,10 @@
         <device name="SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB" backend="wsa-speaker-and-bt-sco-wb" interface="PRI_MI2S_RX-and-INT_BT_SCO_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" backend="vbat-voice-speaker" interface="PRI_MI2S_RX"/>
         <device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB_NREC" interface="INT_BT_SCO_TX"/>
     </backend_names>
     <!-- below values are for ref purpose to OEM, doesn't contain actual hardware info on MTP -->
     <microphone_characteristics>
diff --git a/configs/msm8937/audio_platform_info_mtp.xml b/configs/msm8937/audio_platform_info_mtp.xml
index bc37572..06fe8d7 100644
--- a/configs/msm8937/audio_platform_info_mtp.xml
+++ b/configs/msm8937/audio_platform_info_mtp.xml
@@ -49,6 +49,11 @@
         <usecase name="USECASE_VOICEMMODE2_CALL" type="out" id="35"/>
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="27" />
+        <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="27" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsa881x.0f"/>
@@ -71,6 +76,7 @@
         <device name="SND_DEVICE_IN_HDMI_MIC" acdb_id="4"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO" acdb_id="125"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC" acdb_id="125"/>
+        <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" acdb_id="8"/>
     </acdb_ids>
 
     <backend_names>
@@ -173,6 +179,10 @@
         <device name="SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB" backend="wsa-speaker-and-bt-sco-wb" interface="PRI_MI2S_RX-and-INT_BT_SCO_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" backend="vbat-voice-speaker" interface="PRI_MI2S_RX"/>
         <device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB_NREC" interface="INT_BT_SCO_TX"/>
     </backend_names>
     <!-- below values are for ref purpose to OEM, doesn't contain actual hardware info on MTP -->
     <microphone_characteristics>
diff --git a/configs/msm8937/audio_platform_info_tasha.xml b/configs/msm8937/audio_platform_info_tasha.xml
index ea8826a..2516965 100644
--- a/configs/msm8937/audio_platform_info_tasha.xml
+++ b/configs/msm8937/audio_platform_info_tasha.xml
@@ -52,6 +52,11 @@
         <usecase name="USECASE_QCHAT_CALL" type="out" id="42"/>
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="28" />
+        <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="28" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <acdb_ids>
         <device name="SND_DEVICE_OUT_SPEAKER_EXTERNAL_1" acdb_id="14"/>
@@ -65,6 +70,7 @@
         <device name="SND_DEVICE_IN_HDMI_MIC" acdb_id="4"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO" acdb_id="125"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC" acdb_id="125"/>
+        <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" acdb_id="8"/>
     </acdb_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsatz.11"/>
@@ -95,6 +101,10 @@
         <device name="SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB" backend="wsa-speaker-and-bt-sco-wb" interface="SLIMBUS_0_RX-and-INT_BT_SCO_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" backend="vbat-voice-speaker"/>
         <device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB_NREC" interface="INT_BT_SCO_TX"/>
     </backend_names>
     <!-- below values are for ref purpose to OEM, doesn't contain actual hardware info on MTP -->
     <microphone_characteristics>
diff --git a/configs/msm8937/audio_platform_info_tashalite.xml b/configs/msm8937/audio_platform_info_tashalite.xml
index ea8826a..2516965 100644
--- a/configs/msm8937/audio_platform_info_tashalite.xml
+++ b/configs/msm8937/audio_platform_info_tashalite.xml
@@ -52,6 +52,11 @@
         <usecase name="USECASE_QCHAT_CALL" type="out" id="42"/>
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="28" />
+        <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="28" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <acdb_ids>
         <device name="SND_DEVICE_OUT_SPEAKER_EXTERNAL_1" acdb_id="14"/>
@@ -65,6 +70,7 @@
         <device name="SND_DEVICE_IN_HDMI_MIC" acdb_id="4"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO" acdb_id="125"/>
         <device name="SND_DEVICE_IN_HANDSET_TMIC" acdb_id="125"/>
+        <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" acdb_id="8"/>
     </acdb_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsatz.11"/>
@@ -95,6 +101,10 @@
         <device name="SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB" backend="wsa-speaker-and-bt-sco-wb" interface="SLIMBUS_0_RX-and-INT_BT_SCO_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" backend="vbat-voice-speaker"/>
         <device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="INT_BT_SCO_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB_NREC" interface="INT_BT_SCO_TX"/>
     </backend_names>
     <!-- below values are for ref purpose to OEM, doesn't contain actual hardware info on MTP -->
     <microphone_characteristics>
diff --git a/configs/msm8937/audio_policy_configuration.xml b/configs/msm8937/audio_policy_configuration.xml
index 0b2a31a..50db483 100644
--- a/configs/msm8937/audio_policy_configuration.xml
+++ b/configs/msm8937/audio_policy_configuration.xml
@@ -134,10 +134,10 @@
                 <mixPort name="voip_rx" role="source"
                          flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
 
-                <mixPort name="primary input" role="sink" maxOpenCount="2" maxActiveCount="2">
+                <mixPort name="primary input" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
@@ -268,7 +268,7 @@
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
                 <route type="mix" sink="primary input"
-                       sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
+                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic,FM Tuner"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="voice_rx"
diff --git a/configs/msm8937/audio_policy_configuration_common.xml b/configs/msm8937/audio_policy_configuration_common.xml
new file mode 100644
index 0000000..7019df3
--- /dev/null
+++ b/configs/msm8937/audio_policy_configuration_common.xml
@@ -0,0 +1,181 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (c) 2016-2017, 2019-2020, The Linux Foundation. All rights reserved
+     Not a Contribution.
+-->
+<!-- Copyright (C) 2015 The Android Open Source Project
+
+     Licensed under the Apache License, Version 2.0 (the "License");
+     you may not use this file except in compliance with the License.
+     You may obtain a copy of the License at
+
+          http://www.apache.org/licenses/LICENSE-2.0
+
+     Unless required by applicable law or agreed to in writing, software
+     distributed under the License is distributed on an "AS IS" BASIS,
+     WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+     See the License for the specific language governing permissions and
+     limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+    <!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” -->
+
+    <!-- Global configuration Decalaration -->
+    <globalConfiguration speaker_drc_enabled="true"/>
+
+
+    <!-- Modules section:
+        There is one section per audio HW module present on the platform.
+        Each module section will contains two mandatory tags for audio HAL “halVersion” and “name”.
+        The module names are the same as in current .conf file:
+                “primary”, “A2DP”, “remote_submix”, “USB”
+        Each module will contain the following sections:
+        “devicePorts”: a list of device descriptors for all input and output devices accessible via this
+        module.
+        This contains both permanently attached devices and removable devices.
+        “mixPorts”: listing all output and input streams exposed by the audio HAL
+        “routes”: list of possible connections between input and output devices or between stream and
+        devices.
+            "route": is defined by an attribute:
+                -"type": <mux|mix> means all sources are mutual exclusive (mux) or can be mixed (mix)
+                -"sink": the sink involved in this route
+                -"sources": all the sources than can be connected to the sink via vis route
+        “attachedDevices”: permanently attached devices.
+        The attachedDevices section is a list of devices names. The names correspond to device names
+        defined in <devicePorts> section.
+        “defaultOutputDevice”: device to be used by default when no policy rule applies
+    -->
+    <modules>
+        <module name="primary" halVersion="2.0">
+            <attachedDevices>
+                <item>Speaker</item>
+                <item>Speaker Safe</item>
+                <item>Earpiece</item>
+                <item>Telephony Tx</item>
+                <item>Built-In Mic</item>
+                <item>Built-In Back Mic</item>
+                <item>Telephony Rx</item>
+            </attachedDevices>
+            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <mixPorts>
+                <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY|AUDIO_OUTPUT_FLAG_FAST">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="raw" role="source" flags="AUDIO_OUTPUT_FLAG_RAW|AUDIO_OUTPUT_FLAG_FAST">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="deep_buffer" role="source"
+                        flags="AUDIO_OUTPUT_FLAG_DEEP_BUFFER">
+                    <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+                             samplingRates="44100,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="compressed_offload" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+                    <profile name="" format="AUDIO_FORMAT_MP3"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_LC"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_HE_V1"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_HE_V2"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                </mixPort>
+                <mixPort name="voice_tx" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="voip_rx" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_VOIP_RX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                           samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="incall_music_uplink" role="source"
+                        flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="primary input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3"/>
+                </mixPort>
+                <mixPort name="voice_rx" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
+                </devicePort>
+                <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+                </devicePort>
+                <devicePort tagName="Speaker Safe" type="AUDIO_DEVICE_OUT_SPEAKER_SAFE" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+                </devicePort>
+                <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
+                </devicePort>
+                <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+                </devicePort>
+                <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC" role="source">
+                </devicePort>
+                <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+                </devicePort>
+                <devicePort tagName="Telephony Rx" type="AUDIO_DEVICE_IN_TELEPHONY_RX" role="source">
+                </devicePort>
+                <!-- TODO: Enable multi-channel recording -->
+            </devicePorts>
+            <routes>
+                <route type="mix" sink="Earpiece"
+                       sources="primary output,raw,deep_buffer"/>
+                <route type="mix" sink="Speaker"
+                       sources="primary output,raw,deep_buffer,compressed_offload"/>
+                <route type="mix" sink="Speaker Safe"
+                       sources="primary output,raw,deep_buffer,compressed_offload"/>
+                <route type="mix" sink="BT SCO"
+                       sources="primary output,raw,deep_buffer,voip_rx"/>
+                <route type="mix" sink="BT SCO Headset"
+                       sources="primary output,raw,deep_buffer,voip_rx"/>
+                <route type="mix" sink="BT SCO Car Kit"
+                       sources="primary output,raw,deep_buffer,voip_rx"/>
+                <route type="mix" sink="Telephony Tx"
+                       sources="voice_tx,incall_music_uplink"/>
+                <route type="mix" sink="primary input"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic"/>
+                <route type="mix" sink="voice_rx"
+                       sources="Telephony Rx"/>
+            </routes>
+        </module>
+
+        <!-- A2dp Audio HAL -->
+        <xi:include href="/vendor/etc/a2dp_audio_policy_configuration.xml"/>
+
+        <!-- Usb Audio HAL -->
+        <xi:include href="/vendor/etc/usb_audio_policy_configuration.xml"/>
+
+        <!-- Remote Submix Audio HAL -->
+        <xi:include href="r_submix_audio_policy_configuration.xml"/>
+
+    </modules>
+    <!-- End of Modules section -->
+
+    <!-- Volume section -->
+
+    <xi:include href="audio_policy_volumes.xml"/>
+    <xi:include href="default_volume_tables.xml"/>
+
+    <!-- End of Volume section -->
+
+</audioPolicyConfiguration>
diff --git a/configs/msm8937/mixer_paths.xml b/configs/msm8937/mixer_paths.xml
index ae482b0..2baa523 100644
--- a/configs/msm8937/mixer_paths.xml
+++ b/configs/msm8937/mixer_paths.xml
@@ -168,6 +168,15 @@
     <ctl name="Voip_Tx Mixer INTERNAL_BT_SCO_TX_Voip" value="0" />
     <!-- Voip end -->
 
+    <!-- VoIP over audio -->
+    <ctl name="MultiMedia10 Mixer TERT_MI2S_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+    <ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="0" />
+    <!-- VoIP over audio -->
+
     <!-- fm -->
     <ctl name="Internal FM RX Volume" value="0" />
     <ctl name="PRI_MI2S_RX Port Mixer INTERNAL_FM_TX" value="0" />
@@ -184,8 +193,8 @@
     <!-- RT Proxy Cal end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording end -->
 
     <!-- Incall Music -->
@@ -890,7 +899,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -930,7 +939,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
@@ -1442,4 +1451,51 @@
           <path name="speaker-and-headphones" />
     </path>
 
+    <!-- VoIP Rx settings -->
+    <path name="audio-playback-voip">
+        <ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip headphones">
+        <path name="audio-playback-voip" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco">
+        <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco-wb">
+        <path name="audio-playback-voip bt-sco" />
+    </path>
+
+    <path name="audio-playback-voip afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headphones">
+        <path name="audio-playback-voip afe-proxy" />
+    </path>
+
+    <path name="audio-playback-voip speaker-and-usb-headphones">
+        <path name="audio-playback-voip usb-headphones" />
+        <path name="audio-playback-voip" />
+    </path>
+
+    <!-- VoIP Tx settings -->
+    <path name="audio-record-voip">
+        <ctl name="MultiMedia10 Mixer TERT_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco">
+        <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco-wb">
+        <path name="audio-record-voip bt-sco" />
+    </path>
+
+    <path name="audio-record-voip usb-headset-mic">
+        <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/msm8937/mixer_paths_mtp.xml b/configs/msm8937/mixer_paths_mtp.xml
index a8be96c..8ded630 100644
--- a/configs/msm8937/mixer_paths_mtp.xml
+++ b/configs/msm8937/mixer_paths_mtp.xml
@@ -195,6 +195,15 @@
     <ctl name="Voip_Tx Mixer INTERNAL_BT_SCO_TX_Voip" value="0" />
     <!-- Voip end -->
 
+    <!-- VoIP over audio -->
+    <ctl name="MultiMedia10 Mixer TERT_MI2S_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+    <ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="0" />
+    <!-- VoIP over audio -->
+
     <!-- fm -->
     <ctl name="Internal FM RX Volume" value="0" />
     <ctl name="PRI_MI2S_RX Port Mixer INTERNAL_FM_TX" value="0" />
@@ -211,8 +220,8 @@
     <!-- RT Proxy Cal end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording end -->
 
     <!-- Incall Music -->
@@ -1015,7 +1024,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1055,7 +1064,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
@@ -1588,4 +1597,51 @@
           <path name="speaker-and-headphones" />
     </path>
 
+    <!-- VoIP Rx settings -->
+    <path name="audio-playback-voip">
+        <ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip headphones">
+        <path name="audio-playback-voip" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco">
+        <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco-wb">
+        <path name="audio-playback-voip bt-sco" />
+    </path>
+
+    <path name="audio-playback-voip afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headphones">
+        <path name="audio-playback-voip afe-proxy" />
+    </path>
+
+    <path name="audio-playback-voip speaker-and-usb-headphones">
+        <path name="audio-playback-voip usb-headphones" />
+        <path name="audio-playback-voip" />
+    </path>
+
+    <!-- VoIP Tx settings -->
+    <path name="audio-record-voip">
+        <ctl name="MultiMedia10 Mixer TERT_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco">
+        <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco-wb">
+        <path name="audio-record-voip bt-sco" />
+    </path>
+
+    <path name="audio-record-voip usb-headset-mic">
+        <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/msm8937/mixer_paths_sku1.xml b/configs/msm8937/mixer_paths_sku1.xml
index 992dffd..9190d78 100644
--- a/configs/msm8937/mixer_paths_sku1.xml
+++ b/configs/msm8937/mixer_paths_sku1.xml
@@ -195,6 +195,15 @@
     <ctl name="Voip_Tx Mixer INTERNAL_BT_SCO_TX_Voip" value="0" />
     <!-- Voip end -->
 
+    <!-- VoIP over audio -->
+    <ctl name="MultiMedia10 Mixer TERT_MI2S_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+    <ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="0" />
+    <!-- VoIP over audio -->
+
     <!-- fm -->
     <ctl name="Internal FM RX Volume" value="0" />
     <ctl name="PRI_MI2S_RX Port Mixer INTERNAL_FM_TX" value="0" />
@@ -211,8 +220,8 @@
     <!-- RT Proxy Cal end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording end -->
 
     <!-- Incall Music -->
@@ -963,7 +972,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1003,7 +1012,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
@@ -1346,6 +1355,11 @@
         <path name="headphones" />
     </path>
 
+    <path name="wsa-speaker-and-headphones">
+        <path name="wsa-speaker" />
+        <path name="headphones" />
+    </path>
+
     <path name="usb-headphones">
     </path>
 
@@ -1523,4 +1537,51 @@
           <path name="speaker-and-headphones" />
     </path>
 
+    <!-- VoIP Rx settings -->
+    <path name="audio-playback-voip">
+        <ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip headphones">
+        <path name="audio-playback-voip" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco">
+        <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco-wb">
+        <path name="audio-playback-voip bt-sco" />
+    </path>
+
+    <path name="audio-playback-voip afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headphones">
+        <path name="audio-playback-voip afe-proxy" />
+    </path>
+
+    <path name="audio-playback-voip speaker-and-usb-headphones">
+        <path name="audio-playback-voip usb-headphones" />
+        <path name="audio-playback-voip" />
+    </path>
+
+    <!-- VoIP Tx settings -->
+    <path name="audio-record-voip">
+        <ctl name="MultiMedia10 Mixer TERT_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco">
+        <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco-wb">
+        <path name="audio-record-voip bt-sco" />
+    </path>
+
+    <path name="audio-record-voip usb-headset-mic">
+        <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/msm8937/mixer_paths_tasha.xml b/configs/msm8937/mixer_paths_tasha.xml
index 931c0b8..24b5c1e 100644
--- a/configs/msm8937/mixer_paths_tasha.xml
+++ b/configs/msm8937/mixer_paths_tasha.xml
@@ -439,8 +439,8 @@
     <!-- Voice external ec. reference end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording End -->
@@ -483,6 +483,12 @@
     <ctl name="VoWLAN_Tx Mixer AFE_PCM_TX_VoWLAN" value="0" />
     <!-- VoWLAN end-->
 
+    <!-- VoIP over audio -->
+    <ctl name="MultiMedia10 Mixer SLIM_0_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+    <!-- VoIP over audio -->
+
     <!-- Audio BTSCO -->
     <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia4" value="0" />
@@ -1586,7 +1592,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1630,7 +1636,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL"  value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
@@ -2907,5 +2913,51 @@
     <path name="call-proxy-in">
     </path>
 
+    <!-- VoIP Rx settings -->
+    <path name="audio-playback-voip">
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco">
+        <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco-wb">
+        <path name="audio-playback-voip bt-sco" />
+    </path>
+
+    <path name="audio-playback-voip afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headphones">
+        <path name="audio-playback-voip afe-proxy" />
+    </path>
+
+    <path name="audio-playback-voip speaker-and-usb-headphones">
+        <path name="audio-playback-voip usb-headphones" />
+        <path name="audio-playback-voip" />
+    </path>
+
+    <!-- VoIP Tx settings -->
+    <path name="audio-record-voip">
+        <ctl name="MultiMedia10 Mixer SLIM_0_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco">
+        <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco-wb">
+        <path name="audio-record-voip bt-sco" />
+    </path>
+
+    <path name="audio-record-voip usb-headset-mic">
+        <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="1" />
+    </path>
 
 </mixer>
diff --git a/configs/msm8937/mixer_paths_tashalite.xml b/configs/msm8937/mixer_paths_tashalite.xml
index dcb9eb0..00b2d74 100644
--- a/configs/msm8937/mixer_paths_tashalite.xml
+++ b/configs/msm8937/mixer_paths_tashalite.xml
@@ -434,8 +434,8 @@
     <!-- Multimode Voice2 end-->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording End -->
@@ -478,6 +478,12 @@
     <ctl name="VoWLAN_Tx Mixer AFE_PCM_TX_VoWLAN" value="0" />
     <!-- VoWLAN end-->
 
+    <!-- VoIP over audio -->
+    <ctl name="MultiMedia10 Mixer SLIM_0_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+    <!-- VoIP over audio -->
+
     <!-- Audio BTSCO -->
     <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia4" value="0" />
@@ -1567,7 +1573,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1611,7 +1617,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL"  value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
@@ -2720,5 +2726,51 @@
     <path name="call-proxy-in">
     </path>
 
+    <!-- VoIP Rx settings -->
+    <path name="audio-playback-voip">
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco">
+        <ctl name="INTERNAL_BT_SCO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco-wb">
+        <path name="audio-playback-voip bt-sco" />
+    </path>
+
+    <path name="audio-playback-voip afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headphones">
+        <path name="audio-playback-voip afe-proxy" />
+    </path>
+
+    <path name="audio-playback-voip speaker-and-usb-headphones">
+        <path name="audio-playback-voip usb-headphones" />
+        <path name="audio-playback-voip" />
+    </path>
+
+    <!-- VoIP Tx settings -->
+    <path name="audio-record-voip">
+        <ctl name="MultiMedia10 Mixer SLIM_0_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco">
+        <ctl name="MultiMedia10 Mixer INTERNAL_BT_SCO_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco-wb">
+        <path name="audio-record-voip bt-sco" />
+    </path>
+
+    <path name="audio-record-voip usb-headset-mic">
+        <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="1" />
+    </path>
 
 </mixer>
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 6d0d756..3cb92d7 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -9,7 +9,7 @@
 #AUDIO_FEATURE_ENABLED_VOICE_CONCURRENCY := true
 AUDIO_FEATURE_ENABLED_AAC_ADTS_OFFLOAD := true
 AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
-AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := true
+AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := false
 AUDIO_FEATURE_ENABLED_EXTN_FORMATS := true
 AUDIO_FEATURE_ENABLED_EXTN_FLAC_DECODER := true
 AUDIO_FEATURE_ENABLED_EXTN_RESAMPLER := true
@@ -109,7 +109,7 @@
     $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8937/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio/audio_policy_configuration.xml
 endif
 PRODUCT_COPY_FILES += \
-    $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/common/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
+    $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8937/audio_policy_configuration_common.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
     $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/common/bluetooth_qti_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/a2dp_audio_policy_configuration.xml \
     $(TOPDIR)frameworks/av/services/audiopolicy/config/audio_policy_volumes.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_volumes.xml \
     $(TOPDIR)frameworks/av/services/audiopolicy/config/default_volume_tables.xml:$(TARGET_COPY_OUT_VENDOR)/etc/default_volume_tables.xml \
@@ -257,7 +257,7 @@
 vendor.audio.feature.compr_cap.enable=false \
 vendor.audio.feature.compress_in.enable=false \
 vendor.audio.feature.compress_meta_data.enable=true \
-vendor.audio.feature.compr_voip.enable=true \
+vendor.audio.feature.compr_voip.enable=false \
 vendor.audio.feature.concurrent_capture.enable=false \
 vendor.audio.feature.custom_stereo.enable=true \
 vendor.audio.feature.display_port.enable=false \
diff --git a/configs/msm8953/audio_platform_info_intcodec.xml b/configs/msm8953/audio_platform_info_intcodec.xml
index 339456e..157d998 100644
--- a/configs/msm8953/audio_platform_info_intcodec.xml
+++ b/configs/msm8953/audio_platform_info_intcodec.xml
@@ -50,6 +50,9 @@
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
         <usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="26"/>
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsa881x.0f"/>
diff --git a/configs/msm8953/audio_platform_info_sku3_tasha.xml b/configs/msm8953/audio_platform_info_sku3_tasha.xml
index 1d30160..d3cf620 100644
--- a/configs/msm8953/audio_platform_info_sku3_tasha.xml
+++ b/configs/msm8953/audio_platform_info_sku3_tasha.xml
@@ -59,6 +59,9 @@
         <usecase name="USECASE_QCHAT_CALL" type="out" id="42"/>
         <usecase name="USECASE_AUDIO_EC_REF_LOOPBACK" type="in" id="14"/>
         <usecase name="USECASE_AUDIO_PLAYBACK_SILENCE" type="out" id="14"/>
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="native_audio_mode" value="src"/>
diff --git a/configs/msm8953/audio_platform_info_sku4.xml b/configs/msm8953/audio_platform_info_sku4.xml
index a071157..8b211a7 100644
--- a/configs/msm8953/audio_platform_info_sku4.xml
+++ b/configs/msm8953/audio_platform_info_sku4.xml
@@ -49,6 +49,9 @@
         <usecase name="USECASE_VOICEMMODE2_CALL" type="out" id="35"/>
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsa881x.0f"/>
diff --git a/configs/msm8953/audio_platform_info_tasha.xml b/configs/msm8953/audio_platform_info_tasha.xml
index 96a34d0..b9ae049 100644
--- a/configs/msm8953/audio_platform_info_tasha.xml
+++ b/configs/msm8953/audio_platform_info_tasha.xml
@@ -61,6 +61,9 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_SILENCE" type="out" id="14"/>
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="native_audio_mode" value="src"/>
diff --git a/configs/msm8953/audio_platform_info_tashalite.xml b/configs/msm8953/audio_platform_info_tashalite.xml
index 671095b..a358144 100644
--- a/configs/msm8953/audio_platform_info_tashalite.xml
+++ b/configs/msm8953/audio_platform_info_tashalite.xml
@@ -61,6 +61,9 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_SILENCE" type="out" id="14"/>
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="1" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="35" />
+        <usecase name="USECASE_INCALL_REC_UPLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_DOWNLINK" type="in" id="14" />
+        <usecase name="USECASE_INCALL_REC_UPLINK_AND_DOWNLINK" type="in" id="14" />
     </pcm_ids>
     <config_params>
         <param key="native_audio_mode" value="src"/>
diff --git a/configs/msm8953/audio_policy_configuration.xml b/configs/msm8953/audio_policy_configuration.xml
index 0b8b7f3..00c2f31 100644
--- a/configs/msm8953/audio_policy_configuration.xml
+++ b/configs/msm8953/audio_policy_configuration.xml
@@ -137,7 +137,7 @@
                              samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
 
-                <mixPort name="primary input" role="sink" maxOpenCount="2" maxActiveCount="2">
+                <mixPort name="primary input" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
@@ -279,7 +279,7 @@
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
                 <route type="mix" sink="primary input"
-                       sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
+                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic,FM Tuner"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/msm8953/audio_policy_configuration_common.xml b/configs/msm8953/audio_policy_configuration_common.xml
new file mode 100644
index 0000000..7019df3
--- /dev/null
+++ b/configs/msm8953/audio_policy_configuration_common.xml
@@ -0,0 +1,181 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (c) 2016-2017, 2019-2020, The Linux Foundation. All rights reserved
+     Not a Contribution.
+-->
+<!-- Copyright (C) 2015 The Android Open Source Project
+
+     Licensed under the Apache License, Version 2.0 (the "License");
+     you may not use this file except in compliance with the License.
+     You may obtain a copy of the License at
+
+          http://www.apache.org/licenses/LICENSE-2.0
+
+     Unless required by applicable law or agreed to in writing, software
+     distributed under the License is distributed on an "AS IS" BASIS,
+     WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+     See the License for the specific language governing permissions and
+     limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+    <!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” -->
+
+    <!-- Global configuration Decalaration -->
+    <globalConfiguration speaker_drc_enabled="true"/>
+
+
+    <!-- Modules section:
+        There is one section per audio HW module present on the platform.
+        Each module section will contains two mandatory tags for audio HAL “halVersion” and “name”.
+        The module names are the same as in current .conf file:
+                “primary”, “A2DP”, “remote_submix”, “USB”
+        Each module will contain the following sections:
+        “devicePorts”: a list of device descriptors for all input and output devices accessible via this
+        module.
+        This contains both permanently attached devices and removable devices.
+        “mixPorts”: listing all output and input streams exposed by the audio HAL
+        “routes”: list of possible connections between input and output devices or between stream and
+        devices.
+            "route": is defined by an attribute:
+                -"type": <mux|mix> means all sources are mutual exclusive (mux) or can be mixed (mix)
+                -"sink": the sink involved in this route
+                -"sources": all the sources than can be connected to the sink via vis route
+        “attachedDevices”: permanently attached devices.
+        The attachedDevices section is a list of devices names. The names correspond to device names
+        defined in <devicePorts> section.
+        “defaultOutputDevice”: device to be used by default when no policy rule applies
+    -->
+    <modules>
+        <module name="primary" halVersion="2.0">
+            <attachedDevices>
+                <item>Speaker</item>
+                <item>Speaker Safe</item>
+                <item>Earpiece</item>
+                <item>Telephony Tx</item>
+                <item>Built-In Mic</item>
+                <item>Built-In Back Mic</item>
+                <item>Telephony Rx</item>
+            </attachedDevices>
+            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <mixPorts>
+                <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY|AUDIO_OUTPUT_FLAG_FAST">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="raw" role="source" flags="AUDIO_OUTPUT_FLAG_RAW|AUDIO_OUTPUT_FLAG_FAST">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="deep_buffer" role="source"
+                        flags="AUDIO_OUTPUT_FLAG_DEEP_BUFFER">
+                    <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+                             samplingRates="44100,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="compressed_offload" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+                    <profile name="" format="AUDIO_FORMAT_MP3"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_LC"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_HE_V1"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_HE_V2"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                </mixPort>
+                <mixPort name="voice_tx" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="voip_rx" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_VOIP_RX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                           samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="incall_music_uplink" role="source"
+                        flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="primary input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3"/>
+                </mixPort>
+                <mixPort name="voice_rx" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
+                </devicePort>
+                <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+                </devicePort>
+                <devicePort tagName="Speaker Safe" type="AUDIO_DEVICE_OUT_SPEAKER_SAFE" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+                </devicePort>
+                <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+                </devicePort>
+                <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
+                </devicePort>
+                <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+                </devicePort>
+                <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC" role="source">
+                </devicePort>
+                <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+                </devicePort>
+                <devicePort tagName="Telephony Rx" type="AUDIO_DEVICE_IN_TELEPHONY_RX" role="source">
+                </devicePort>
+                <!-- TODO: Enable multi-channel recording -->
+            </devicePorts>
+            <routes>
+                <route type="mix" sink="Earpiece"
+                       sources="primary output,raw,deep_buffer"/>
+                <route type="mix" sink="Speaker"
+                       sources="primary output,raw,deep_buffer,compressed_offload"/>
+                <route type="mix" sink="Speaker Safe"
+                       sources="primary output,raw,deep_buffer,compressed_offload"/>
+                <route type="mix" sink="BT SCO"
+                       sources="primary output,raw,deep_buffer,voip_rx"/>
+                <route type="mix" sink="BT SCO Headset"
+                       sources="primary output,raw,deep_buffer,voip_rx"/>
+                <route type="mix" sink="BT SCO Car Kit"
+                       sources="primary output,raw,deep_buffer,voip_rx"/>
+                <route type="mix" sink="Telephony Tx"
+                       sources="voice_tx,incall_music_uplink"/>
+                <route type="mix" sink="primary input"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic"/>
+                <route type="mix" sink="voice_rx"
+                       sources="Telephony Rx"/>
+            </routes>
+        </module>
+
+        <!-- A2dp Audio HAL -->
+        <xi:include href="/vendor/etc/a2dp_audio_policy_configuration.xml"/>
+
+        <!-- Usb Audio HAL -->
+        <xi:include href="/vendor/etc/usb_audio_policy_configuration.xml"/>
+
+        <!-- Remote Submix Audio HAL -->
+        <xi:include href="r_submix_audio_policy_configuration.xml"/>
+
+    </modules>
+    <!-- End of Modules section -->
+
+    <!-- Volume section -->
+
+    <xi:include href="audio_policy_volumes.xml"/>
+    <xi:include href="default_volume_tables.xml"/>
+
+    <!-- End of Volume section -->
+
+</audioPolicyConfiguration>
diff --git a/configs/msm8953/mixer_paths.xml b/configs/msm8953/mixer_paths.xml
index 8845c5c..4241202 100644
--- a/configs/msm8953/mixer_paths.xml
+++ b/configs/msm8953/mixer_paths.xml
@@ -186,8 +186,8 @@
     <!-- RT Proxy Cal end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording end -->
 
     <!-- Incall Music -->
@@ -896,7 +896,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -936,7 +936,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
diff --git a/configs/msm8953/mixer_paths_mtp.xml b/configs/msm8953/mixer_paths_mtp.xml
index 710556c..fea32be 100644
--- a/configs/msm8953/mixer_paths_mtp.xml
+++ b/configs/msm8953/mixer_paths_mtp.xml
@@ -213,8 +213,8 @@
     <!-- RT Proxy Cal end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording end -->
 
     <!-- Incall Music -->
@@ -1026,7 +1026,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1066,7 +1066,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
diff --git a/configs/msm8953/mixer_paths_sku3_tasha.xml b/configs/msm8953/mixer_paths_sku3_tasha.xml
index c5a8b04..7efefd2 100644
--- a/configs/msm8953/mixer_paths_sku3_tasha.xml
+++ b/configs/msm8953/mixer_paths_sku3_tasha.xml
@@ -433,8 +433,8 @@
     <!-- Multimode Voice2 end-->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording End -->
@@ -1563,7 +1563,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1607,7 +1607,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL"  value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
diff --git a/configs/msm8953/mixer_paths_sku4.xml b/configs/msm8953/mixer_paths_sku4.xml
index 9baf756..91a4d1a 100644
--- a/configs/msm8953/mixer_paths_sku4.xml
+++ b/configs/msm8953/mixer_paths_sku4.xml
@@ -213,8 +213,8 @@
     <!-- RT Proxy Cal end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording end -->
 
     <!-- Incall Music -->
@@ -1025,7 +1025,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1065,7 +1065,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
diff --git a/configs/msm8953/mixer_paths_tasha.xml b/configs/msm8953/mixer_paths_tasha.xml
index 8b80c4c..1a2987c 100644
--- a/configs/msm8953/mixer_paths_tasha.xml
+++ b/configs/msm8953/mixer_paths_tasha.xml
@@ -441,8 +441,8 @@
     <!-- Voice external ec. reference end -->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording End -->
@@ -1593,7 +1593,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1637,7 +1637,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL"  value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
diff --git a/configs/msm8953/mixer_paths_tashalite.xml b/configs/msm8953/mixer_paths_tashalite.xml
index 59e348e..5efe6d3 100644
--- a/configs/msm8953/mixer_paths_tashalite.xml
+++ b/configs/msm8953/mixer_paths_tashalite.xml
@@ -437,8 +437,8 @@
     <!-- Multimode Voice2 end-->
 
     <!-- Incall Recording -->
-    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
-    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia9 Mixer VOC_REC_DL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
     <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
     <!-- Incall Recording End -->
@@ -1570,7 +1570,7 @@
     </path>
 
     <path name="incall-rec-uplink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_UL" value="1" />
     </path>
 
     <path name="incall-rec-uplink bt-sco">
@@ -1614,7 +1614,7 @@
     </path>
 
     <path name="incall-rec-downlink">
-        <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
+        <ctl name="MultiMedia9 Mixer VOC_REC_DL"  value="1" />
     </path>
 
     <path name="incall-rec-downlink bt-sco">
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index 4365abf..a72a102 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -108,7 +108,7 @@
     $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8953/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio/audio_policy_configuration.xml
 endif
 PRODUCT_COPY_FILES += \
-    $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/common/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
+    $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8953/audio_policy_configuration_common.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
     $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/common/bluetooth_qti_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/a2dp_audio_policy_configuration.xml \
     $(TOPDIR)frameworks/av/services/audiopolicy/config/audio_policy_volumes.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_volumes.xml \
     $(TOPDIR)frameworks/av/services/audiopolicy/config/default_volume_tables.xml:$(TARGET_COPY_OUT_VENDOR)/etc/default_volume_tables.xml \
diff --git a/configs/sdm660/audio_policy_configuration.xml b/configs/sdm660/audio_policy_configuration.xml
index c5872bb..aff987d 100644
--- a/configs/sdm660/audio_policy_configuration.xml
+++ b/configs/sdm660/audio_policy_configuration.xml
@@ -155,7 +155,7 @@
                 <mixPort name="voip_rx" role="source"
                          flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
 
                 <mixPort name="primary input" role="sink">
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
index 105c88d..d0170db 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -255,7 +255,7 @@
 
 // --- external function dependency ---
 fp_platform_get_pcm_device_id_t fp_platform_get_pcm_device_id;
-fp_check_a2dp_restore_t fp_check_a2dp_restore;
+fp_check_a2dp_restore_t fp_check_a2dp_restore_l;
 
 /* PCM config for ABR Feedback hostless front end */
 static struct pcm_config pcm_config_abr = {
@@ -2817,141 +2817,138 @@
 
 int a2dp_set_parameters(struct str_parms *parms, bool *reconfig)
 {
-     int ret = 0, val, status = 0;
-     char value[32]={0};
-     struct audio_usecase *uc_info;
-     struct listnode *node;
+    int ret = 0, val, status = 0;
+    char value[32] = {0};
+    struct audio_usecase *uc_info;
+    struct listnode *node;
 
-     if (a2dp.is_a2dp_offload_supported == false) {
+    if (a2dp.is_a2dp_offload_supported == false) {
         ALOGV("no supported encoders identified,ignoring a2dp setparam");
         status = -EINVAL;
         goto param_handled;
-     }
+    }
 
-     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value,
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value,
                             sizeof(value));
-     if (ret >= 0) {
-         val = atoi(value);
-         if (audio_is_a2dp_out_device(val)) {
-             ALOGV("Received device connect request for A2DP source");
-             open_a2dp_source();
-         }
-         goto param_handled;
-     }
+    if (ret >= 0) {
+        val = atoi(value);
+        if (audio_is_a2dp_out_device(val)) {
+            ALOGV("Received device connect request for A2DP source");
+            open_a2dp_source();
+        }
+        goto param_handled;
+    }
 
-     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value,
-                         sizeof(value));
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value,
+                            sizeof(value));
 
-     if (ret >= 0) {
-         val = atoi(value);
-         if (audio_is_a2dp_out_device(val)) {
-             ALOGV("Received source device dis- connect request");
-             close_a2dp_output();
-             reset_a2dp_enc_config_params();
-             reset_a2dp_source_dec_config_params();
-             a2dp_reset_backend_cfg(SOURCE);
-         } else if (audio_is_a2dp_in_device(val)) {
-             ALOGV("Received sink device dis- connect request");
-             close_a2dp_input();
-             reset_a2dp_sink_dec_config_params();
-             a2dp_reset_backend_cfg(SINK);
-         }
-         goto param_handled;
-     }
+    if (ret >= 0) {
+        val = atoi(value);
+        if (audio_is_a2dp_out_device(val)) {
+            ALOGV("Received source device dis- connect request");
+            close_a2dp_output();
+            reset_a2dp_enc_config_params();
+            reset_a2dp_source_dec_config_params();
+            a2dp_reset_backend_cfg(SOURCE);
+        } else if (audio_is_a2dp_in_device(val)) {
+            ALOGV("Received sink device dis- connect request");
+            close_a2dp_input();
+            reset_a2dp_sink_dec_config_params();
+            a2dp_reset_backend_cfg(SINK);
+        }
+        goto param_handled;
+    }
 #ifndef LINUX_ENABLED
-     ret = str_parms_get_str(parms, "TwsChannelConfig", value, sizeof(value));
-     if (ret>=0) {
-         ALOGD("Setting tws channel mode to %s",value);
-         if (!(strncmp(value,"mono",strlen(value))))
+    ret = str_parms_get_str(parms, "TwsChannelConfig", value, sizeof(value));
+    if (ret >= 0) {
+        ALOGD("Setting tws channel mode to %s",value);
+        if (!(strncmp(value, "mono", strlen(value))))
             a2dp.is_tws_mono_mode_on = true;
-         else if (!(strncmp(value,"dual-mono",strlen(value))))
+        else if (!(strncmp(value, "dual-mono", strlen(value))))
             a2dp.is_tws_mono_mode_on = false;
-         audio_a2dp_update_tws_channel_mode();
-     goto param_handled;
-     }
+        audio_a2dp_update_tws_channel_mode();
+        goto param_handled;
+    }
 #endif
-     ret = str_parms_get_str(parms, "A2dpSuspended", value, sizeof(value));
-     if (ret >= 0) {
-         if (a2dp.bt_lib_source_handle) {
-             if ((!strncmp(value,"true",sizeof(value)))) {
-                if (a2dp.a2dp_source_suspended) {
-                    ALOGD("%s: A2DP is already suspended", __func__);
-                    goto param_handled;
+    ret = str_parms_get_str(parms, "A2dpSuspended", value, sizeof(value));
+    if (ret >= 0) {
+        if (a2dp.bt_lib_source_handle == NULL)
+            goto param_handled;
+
+        if ((!strncmp(value, "true", sizeof(value)))) {
+            if (a2dp.a2dp_source_suspended) {
+                ALOGD("%s: A2DP is already suspended", __func__);
+                goto param_handled;
+            }
+            ALOGD("Setting a2dp to suspend state");
+            a2dp.a2dp_source_suspended = true;
+            if (a2dp.bt_state_source == A2DP_STATE_DISCONNECTED)
+                goto param_handled;
+            list_for_each(node, &a2dp.adev->usecase_list) {
+                uc_info = node_to_item(node, struct audio_usecase, list);
+                if (uc_info->type == PCM_PLAYBACK &&
+                    (uc_info->out_snd_device == SND_DEVICE_OUT_BT_A2DP ||
+                     uc_info->out_snd_device == SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP ||
+                     uc_info->out_snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP)) {
+                    fp_check_a2dp_restore_l(a2dp.adev, uc_info->stream.out, false);
                 }
-                ALOGD("Setting a2dp to suspend state");
-                a2dp.a2dp_source_suspended = true;
-                if (a2dp.bt_state_source == A2DP_STATE_DISCONNECTED)
-                    goto param_handled;
-                list_for_each(node, &a2dp.adev->usecase_list) {
-                    uc_info = node_to_item(node, struct audio_usecase, list);
-                    if (uc_info->type == PCM_PLAYBACK &&
-                        (uc_info->out_snd_device == SND_DEVICE_OUT_BT_A2DP ||
-                         uc_info->out_snd_device == SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP ||
-                         uc_info->out_snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP)) {
-                        pthread_mutex_unlock(&a2dp.adev->lock);
-                        fp_check_a2dp_restore(a2dp.adev, uc_info->stream.out, false);
-                        pthread_mutex_lock(&a2dp.adev->lock);
+            }
+            if (!a2dp.swb_configured)
+                reset_codec_config();
+            if (a2dp.audio_source_suspend)
+                a2dp.audio_source_suspend();
+        } else if (a2dp.a2dp_source_suspended == true) {
+            ALOGD("Resetting a2dp suspend state");
+            struct audio_usecase *uc_info;
+            struct listnode *node;
+            if (a2dp.clear_source_a2dpsuspend_flag)
+                a2dp.clear_source_a2dpsuspend_flag();
+            a2dp.a2dp_source_suspended = false;
+            /*
+             * It is possible that before suspend,a2dp sessions can be active
+             * for example during music + voice activation concurrency
+             * a2dp suspend will be called & BT will change to sco mode
+             * though music is paused as a part of voice activation
+             * compress session close happens only after pause timeout(10secs)
+             * so if resume request comes before pause timeout as a2dp session
+             * is already active IPC start will not be called from APM/audio_hw
+             * Fix is to call a2dp start for IPC library post suspend
+             * based on number of active session count
+             */
+            if (a2dp.a2dp_source_total_active_session_requests > 0) {
+                ALOGD(" Calling IPC lib start post suspend state");
+                if (a2dp.audio_source_start) {
+                    ret =  a2dp.audio_source_start();
+                    if (ret != 0) {
+                        ALOGE("BT controller start failed");
+                        a2dp.a2dp_source_started = false;
                     }
                 }
-                if (!a2dp.swb_configured)
-                    reset_codec_config();
-                if (a2dp.audio_source_suspend)
-                   a2dp.audio_source_suspend();
-            } else if (a2dp.a2dp_source_suspended == true) {
-                ALOGD("Resetting a2dp suspend state");
-                struct audio_usecase *uc_info;
-                struct listnode *node;
-                if (a2dp.clear_source_a2dpsuspend_flag)
-                    a2dp.clear_source_a2dpsuspend_flag();
-                a2dp.a2dp_source_suspended = false;
-                /*
-                 * It is possible that before suspend,a2dp sessions can be active
-                 * for example during music + voice activation concurrency
-                 * a2dp suspend will be called & BT will change to sco mode
-                 * though music is paused as a part of voice activation
-                 * compress session close happens only after pause timeout(10secs)
-                 * so if resume request comes before pause timeout as a2dp session
-                 * is already active IPC start will not be called from APM/audio_hw
-                 * Fix is to call a2dp start for IPC library post suspend
-                 * based on number of active session count
-                 */
-                if (a2dp.a2dp_source_total_active_session_requests > 0) {
-                    ALOGD(" Calling IPC lib start post suspend state");
-                    if (a2dp.audio_source_start) {
-                        ret =  a2dp.audio_source_start();
-                        if (ret != 0) {
-                            ALOGE("BT controller start failed");
-                            a2dp.a2dp_source_started = false;
-                        }
-                    }
-                }
-                list_for_each(node, &a2dp.adev->usecase_list) {
-                    uc_info = node_to_item(node, struct audio_usecase, list);
-                    if (uc_info->stream.out && uc_info->type == PCM_PLAYBACK &&
-                        is_a2dp_out_device_type(&uc_info->stream.out->device_list)) {
-                        pthread_mutex_unlock(&a2dp.adev->lock);
-                        fp_check_a2dp_restore(a2dp.adev, uc_info->stream.out, true);
-                        pthread_mutex_lock(&a2dp.adev->lock);
-                    }
+            }
+            list_for_each(node, &a2dp.adev->usecase_list) {
+                uc_info = node_to_item(node, struct audio_usecase, list);
+                if (uc_info->stream.out && uc_info->type == PCM_PLAYBACK &&
+                    is_a2dp_out_device_type(&uc_info->stream.out->device_list)) {
+                    fp_check_a2dp_restore_l(a2dp.adev, uc_info->stream.out, true);
                 }
             }
         }
         goto param_handled;
-     }
+    }
 
-     ret = str_parms_get_str(parms, AUDIO_PARAMETER_RECONFIG_A2DP, value,
-                         sizeof(value));
-     if (ret >= 0) {
-         if (a2dp.is_a2dp_offload_supported &&
-                a2dp.bt_state_source != A2DP_STATE_DISCONNECTED) {
-             *reconfig = true;
-         }
-         goto param_handled;
-     }
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_RECONFIG_A2DP, value,
+                            sizeof(value));
+    if (ret >= 0) {
+        if (a2dp.is_a2dp_offload_supported &&
+            a2dp.bt_state_source != A2DP_STATE_DISCONNECTED) {
+            *reconfig = true;
+        }
+        goto param_handled;
+    }
 
 param_handled:
-     ALOGV("end of a2dp setparam");
-     return status;
+    ALOGV("end of a2dp setparam");
+    return status;
 }
 
 void a2dp_set_handoff_mode(bool is_on)
@@ -3022,7 +3019,7 @@
   // init function pointers
   fp_platform_get_pcm_device_id =
               init_config.fp_platform_get_pcm_device_id;
-  fp_check_a2dp_restore = init_config.fp_check_a2dp_restore;
+  fp_check_a2dp_restore_l = init_config.fp_check_a2dp_restore_l;
 
   reset_a2dp_enc_config_params();
   reset_a2dp_source_dec_config_params();
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index ed10568..04e7a55 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -4750,7 +4750,7 @@
     if (a2dp_init) {
         a2dp_offload_init_config_t a2dp_init_config;
         a2dp_init_config.fp_platform_get_pcm_device_id = platform_get_pcm_device_id;
-        a2dp_init_config.fp_check_a2dp_restore = check_a2dp_restore;
+        a2dp_init_config.fp_check_a2dp_restore_l = check_a2dp_restore_l;
 
         a2dp_init(adev, a2dp_init_config);
     }
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 853b034..da986ad 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -337,7 +337,7 @@
                                        struct stream_out *, bool);
 struct a2dp_offload_init_config {
     fp_platform_get_pcm_device_id_t fp_platform_get_pcm_device_id;
-    fp_check_a2dp_restore_t fp_check_a2dp_restore;
+    fp_check_a2dp_restore_t fp_check_a2dp_restore_l;
 };
 typedef struct a2dp_offload_init_config a2dp_offload_init_config_t;
 // END: A2DP_OFFLOAD FEATURE ====================================================
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index df4bbec..4516448 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -515,7 +515,6 @@
 //cache last MBDRC cal step level
 static int last_known_cal_step = -1 ;
 
-static int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore);
 static int out_set_compr_volume(struct audio_stream_out *stream, float left, float right);
 static int out_set_mmap_volume(struct audio_stream_out *stream, float left, float right);
 static int out_set_voip_volume(struct audio_stream_out *stream, float left, float right);
@@ -3372,7 +3371,7 @@
     return 0;
 }
 
-/* must be called iwth out->lock locked */
+/* must be called with out->lock and latch lock */
 static void stop_compressed_output_l(struct stream_out *out)
 {
     out->offload_state = OFFLOAD_STATE_IDLE;
@@ -3633,9 +3632,11 @@
 static int destroy_offload_callback_thread(struct stream_out *out)
 {
     lock_output_stream(out);
+    pthread_mutex_lock(&out->latch_lock);
     stop_compressed_output_l(out);
     send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
 
+    pthread_mutex_unlock(&out->latch_lock);
     pthread_mutex_unlock(&out->lock);
     pthread_join(out->offload_thread, (void **) NULL);
     pthread_cond_destroy(&out->offload_cond);
@@ -3708,6 +3709,9 @@
 
     list_remove(&uc_info->list);
     out->started = 0;
+    pthread_mutex_lock(&out->latch_lock);
+    out->muted = false;
+    pthread_mutex_unlock(&out->latch_lock);
     if (is_offload_usecase(out->usecase) &&
         (audio_extn_passthru_is_passthrough_stream(out))) {
         ALOGV("Disable passthrough , reset mixer to pcm");
@@ -4527,8 +4531,11 @@
         if (adev->adm_deregister_stream)
             adev->adm_deregister_stream(adev->adm_data, out->handle);
 
-        if (is_offload_usecase(out->usecase))
+        if (is_offload_usecase(out->usecase)) {
+            pthread_mutex_lock(&out->latch_lock);
             stop_compressed_output_l(out);
+            pthread_mutex_unlock(&out->latch_lock);
+        }
 
         pthread_mutex_lock(&adev->lock);
         out->standby = true;
@@ -4602,7 +4609,9 @@
     // is needed e.g. when SSR happens within compress_open
     // since the stream is active, offload_callback_thread is also active.
     if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+        pthread_mutex_lock(&out->latch_lock);
         stop_compressed_output_l(out);
+        pthread_mutex_unlock(&out->latch_lock);
     }
     pthread_mutex_unlock(&out->lock);
 
@@ -4624,8 +4633,8 @@
 }
 
 /*
- *standby implementation without locks, assumes that the callee already
- *has taken adev and out lock.
+ * standby implementation without locks, assumes that the callee already
+ * has taken adev and out lock.
  */
 int out_standby_l(struct audio_stream *stream)
 {
@@ -4640,8 +4649,11 @@
         if (adev->adm_deregister_stream)
             adev->adm_deregister_stream(adev->adm_data, out->handle);
 
-        if (is_offload_usecase(out->usecase))
+        if (is_offload_usecase(out->usecase)) {
+            pthread_mutex_lock(&out->latch_lock);
             stop_compressed_output_l(out);
+            pthread_mutex_unlock(&out->latch_lock);
+        }
 
         out->standby = true;
         if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
@@ -4988,12 +5000,13 @@
                 audio_extn_perf_lock_release(&adev->perf_lock_handle);
             }
             if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
-                 out->a2dp_compress_mute &&
                  (!is_a2dp_out_device_type(&out->device_list) || audio_extn_a2dp_source_is_ready())) {
-                pthread_mutex_lock(&out->compr_mute_lock);
-                out->a2dp_compress_mute = false;
-                out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
-                pthread_mutex_unlock(&out->compr_mute_lock);
+                pthread_mutex_lock(&out->latch_lock);
+                if (out->a2dp_compress_mute) {
+                    out->a2dp_compress_mute = false;
+                    out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
+                }
+                pthread_mutex_unlock(&out->latch_lock);
             } else if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP) {
                 out_set_voip_volume(&out->stream, out->volume_l, out->volume_r);
             }
@@ -5548,7 +5561,9 @@
     ALOGD("%s: called with left_vol=%f, right_vol=%f", __func__, left, right);
     if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
         /* only take left channel into account: the API is for stereo anyway */
+        pthread_mutex_lock(&out->latch_lock);
         out->muted = (left == 0.0f);
+        pthread_mutex_unlock(&out->latch_lock);
         return 0;
     } else if (is_offload_usecase(out->usecase)) {
         if (audio_extn_passthru_is_passthrough_stream(out)) {
@@ -5585,18 +5600,20 @@
                     out->volume_r = out_ctxt->output->volume_r;
                 }
             }
+            pthread_mutex_lock(&out->latch_lock);
             if (!out->a2dp_compress_mute) {
                 ret = out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
             }
+            pthread_mutex_unlock(&out->latch_lock);
             return ret;
         } else {
-            pthread_mutex_lock(&out->compr_mute_lock);
+            pthread_mutex_lock(&out->latch_lock);
             ALOGV("%s: compress mute %d", __func__, out->a2dp_compress_mute);
             if (!out->a2dp_compress_mute)
                 ret = out_set_compr_volume(stream, left, right);
             out->volume_l = left;
             out->volume_r = right;
-            pthread_mutex_unlock(&out->compr_mute_lock);
+            pthread_mutex_unlock(&out->latch_lock);
             return ret;
         }
     } else if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP) {
@@ -6007,7 +6024,9 @@
             }
             audio_extn_dts_eagle_fade(adev, true, out);
             out->playback_started = 1;
+            pthread_mutex_lock(&out->latch_lock);
             out->offload_state = OFFLOAD_STATE_PLAYING;
+            pthread_mutex_unlock(&out->latch_lock);
 
             audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
                                                      popcount(out->channel_mask),
@@ -6019,8 +6038,10 @@
     } else {
         if (out->pcm) {
             size_t bytes_to_write = bytes;
+            pthread_mutex_lock(&out->latch_lock);
             if (out->muted)
                 memset((void *)buffer, 0, bytes);
+            pthread_mutex_unlock(&out->latch_lock);
             ALOGV("%s: frames=%zu, frame_size=%zu, bytes_to_write=%zu",
                      __func__, frames, frame_size, bytes_to_write);
 
@@ -6394,6 +6415,7 @@
         ALOGD("copl(%p):pause compress driver", out);
         status = -ENODATA;
         lock_output_stream(out);
+        pthread_mutex_lock(&out->latch_lock);
         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
             if (out->card_status != CARD_STATUS_OFFLINE)
                 status = compress_pause(out->compr);
@@ -6410,6 +6432,7 @@
                                                  out->sample_rate, popcount(out->channel_mask),
                                                  0);
         }
+        pthread_mutex_unlock(&out->latch_lock);
         pthread_mutex_unlock(&out->lock);
     }
     return status;
@@ -6424,6 +6447,7 @@
         ALOGD("copl(%p):resume compress driver", out);
         status = -ENODATA;
         lock_output_stream(out);
+        pthread_mutex_lock(&out->latch_lock);
         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
             if (out->card_status != CARD_STATUS_OFFLINE) {
                 status = compress_resume(out->compr);
@@ -6435,6 +6459,7 @@
             audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
                                                      popcount(out->channel_mask), 1);
         }
+        pthread_mutex_unlock(&out->latch_lock);
         pthread_mutex_unlock(&out->lock);
     }
     return status;
@@ -6463,12 +6488,14 @@
     if (is_offload_usecase(out->usecase)) {
         ALOGD("copl(%p):calling compress flush", out);
         lock_output_stream(out);
+        pthread_mutex_lock(&out->latch_lock);
         if (out->offload_state == OFFLOAD_STATE_PAUSED) {
             stop_compressed_output_l(out);
         } else {
             ALOGW("%s called in invalid state %d", __func__, out->offload_state);
         }
         out->written = 0;
+        pthread_mutex_unlock(&out->latch_lock);
         pthread_mutex_unlock(&out->lock);
         ALOGD("copl(%p):out of compress flush", out);
         return 0;
@@ -7743,7 +7770,7 @@
 
     pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
     pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL);
-    pthread_mutex_init(&out->compr_mute_lock, (const pthread_mutexattr_t *) NULL);
+    pthread_mutex_init(&out->latch_lock, (const pthread_mutexattr_t *) NULL);
     pthread_mutex_init(&out->position_query_lock, (const pthread_mutexattr_t *) NULL);
     pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL);
 
@@ -8658,6 +8685,9 @@
 
     pthread_cond_destroy(&out->cond);
     pthread_mutex_destroy(&out->lock);
+    pthread_mutex_destroy(&out->pre_lock);
+    pthread_mutex_destroy(&out->latch_lock);
+    pthread_mutex_destroy(&out->position_query_lock);
 
     pthread_mutex_lock(&adev->lock);
     streams_output_ctxt_t *out_ctxt = out_get_stream(adev, out->handle);
@@ -8720,7 +8750,7 @@
     }
 
     ret = str_parms_get_str(parms, "A2dpSuspended", value, sizeof(value));
-    if (ret>=0) {
+    if (ret >= 0) {
         if (!strncmp(value, "false", 5) &&
             audio_extn_a2dp_source_is_suspended()) {
             struct audio_usecase *usecase;
@@ -8881,30 +8911,30 @@
 
             if (is_a2dp_out_device_type(&usecase->device_list)) {
                 ALOGD("reconfigure a2dp... forcing device switch");
-                pthread_mutex_unlock(&adev->lock);
-                lock_output_stream(usecase->stream.out);
-                pthread_mutex_lock(&adev->lock);
                 audio_extn_a2dp_set_handoff_mode(true);
                 ALOGD("Switching to speaker and muting the stream before select_devices");
                 check_a2dp_restore_l(adev, usecase->stream.out, false);
                 //force device switch to re configure encoder
                 select_devices(adev, usecase->id);
                 ALOGD("Unmuting the stream after select_devices");
+                pthread_mutex_lock(&usecase->stream.out->latch_lock);
                 usecase->stream.out->a2dp_compress_mute = false;
-                out_set_compr_volume(&usecase->stream.out->stream, usecase->stream.out->volume_l, usecase->stream.out->volume_r);
+                out_set_compr_volume(&usecase->stream.out->stream,
+                                     usecase->stream.out->volume_l,
+                                     usecase->stream.out->volume_r);
+                pthread_mutex_unlock(&usecase->stream.out->latch_lock);
                 audio_extn_a2dp_set_handoff_mode(false);
-                pthread_mutex_unlock(&usecase->stream.out->lock);
                 break;
-            } else if ((usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
-                        usecase->stream.out->a2dp_compress_mute) {
-                pthread_mutex_unlock(&adev->lock);
-                lock_output_stream(usecase->stream.out);
-                pthread_mutex_lock(&adev->lock);
-                reassign_device_list(&usecase->stream.out->device_list,
-                                     AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "");
-                check_a2dp_restore_l(adev, usecase->stream.out, true);
-                pthread_mutex_unlock(&usecase->stream.out->lock);
-                break;
+            } else if (usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+                pthread_mutex_lock(&usecase->stream.out->latch_lock);
+                if (usecase->stream.out->a2dp_compress_mute) {
+                    pthread_mutex_unlock(&usecase->stream.out->latch_lock);
+                    reassign_device_list(&usecase->stream.out->device_list,
+                                         AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "");
+                    check_a2dp_restore_l(adev, usecase->stream.out, true);
+                    break;
+                }
+                pthread_mutex_unlock(&usecase->stream.out->latch_lock);
             }
         }
     }
@@ -9080,7 +9110,7 @@
         ALOGD("%s: mode %d , prev_mode %d \n", __func__, mode , adev->mode);
         adev->prev_mode = adev->mode; /* prev_mode is kept to handle voip concurrency*/
         adev->mode = mode;
-        if( mode == AUDIO_MODE_CALL_SCREEN ){
+        if (mode == AUDIO_MODE_CALL_SCREEN) {
             adev->current_call_output = adev->primary_output;
             voice_start_call(adev);
         } else if (voice_is_in_call_or_call_screen(adev) &&
@@ -9724,6 +9754,9 @@
     } else
         in_standby(&stream->common);
 
+    pthread_mutex_destroy(&in->lock);
+    pthread_mutex_destroy(&in->pre_lock);
+
     pthread_mutex_lock(&adev->lock);
     if (in->usecase == USECASE_AUDIO_RECORD) {
         adev->pcm_record_uc_state = 0;
@@ -10328,8 +10361,8 @@
     return;
 }
 
-/* out and adev lock held */
-static int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore)
+/* adev lock held */
+int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore)
 {
     struct audio_usecase *uc_info;
     float left_p;
@@ -10348,23 +10381,26 @@
           out->usecase, use_case_table[out->usecase]);
 
     if (restore) {
+        pthread_mutex_lock(&out->latch_lock);
         // restore A2DP device for active usecases and unmute if required
         if (is_a2dp_out_device_type(&out->device_list)) {
             ALOGD("%s: restoring A2dp and unmuting stream", __func__);
             if (uc_info->out_snd_device != SND_DEVICE_OUT_BT_A2DP)
                 select_devices(adev, uc_info->id);
-            pthread_mutex_lock(&out->compr_mute_lock);
             if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
-                (out->a2dp_compress_mute) && (uc_info->out_snd_device == SND_DEVICE_OUT_BT_A2DP)) {
-                out->a2dp_compress_mute = false;
-                out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
+                (uc_info->out_snd_device == SND_DEVICE_OUT_BT_A2DP)) {
+                if (out->a2dp_compress_mute) {
+                    out->a2dp_compress_mute = false;
+                    out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
+                }
             }
-            pthread_mutex_unlock(&out->compr_mute_lock);
         }
+        out->muted = false;
+        pthread_mutex_unlock(&out->latch_lock);
     } else {
+        pthread_mutex_lock(&out->latch_lock);
         if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
             // mute compress stream if suspended
-            pthread_mutex_lock(&out->compr_mute_lock);
             if (!out->a2dp_compress_mute && !out->standby) {
                 ALOGD("%s: selecting speaker and muting stream", __func__);
                 assign_devices(&devices, &out->device_list);
@@ -10382,31 +10418,18 @@
                 out->volume_l = left_p;
                 out->volume_r = right_p;
             }
-            pthread_mutex_unlock(&out->compr_mute_lock);
         } else {
-            // tear down a2dp path for non offloaded streams
-            if (audio_extn_a2dp_source_is_suspended())
-                out_standby_l(&out->stream.common);
+            // mute for non offloaded streams
+            if (audio_extn_a2dp_source_is_suspended()) {
+                out->muted = true;
+            }
         }
+        pthread_mutex_unlock(&out->latch_lock);
     }
     ALOGV("%s: exit", __func__);
     return 0;
 }
 
-int check_a2dp_restore(struct audio_device *adev, struct stream_out *out, bool restore)
-{
-    int ret = 0;
-
-    lock_output_stream(out);
-    pthread_mutex_lock(&adev->lock);
-
-    ret = check_a2dp_restore_l(adev, out, restore);
-
-    pthread_mutex_unlock(&adev->lock);
-    pthread_mutex_unlock(&out->lock);
-    return ret;
-}
-
 void adev_on_battery_status_changed(bool charging)
 {
     pthread_mutex_lock(&adev->lock);
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 6931d8d..f1e8672 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -362,13 +362,18 @@
     stream_callback_t client_callback;
     void *client_cookie;
 };
+
 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_mutex_t compr_mute_lock; /* acquire before setting compress volume */
-    pthread_mutex_t position_query_lock; /* acquire before updating/getting position of track offload*/
     pthread_cond_t  cond;
+    /* stream_out->lock is of large granularity, and can only be held before device lock
+     * latch is a supplemetary lock to protect certain fields of out stream and
+     * it can be held after device lock
+     */
+    pthread_mutex_t latch_lock;
+    pthread_mutex_t position_query_lock; /* sychronize frame written */
     struct pcm_config config;
     struct compr_config compr_config;
     struct pcm *pcm;
@@ -778,7 +783,7 @@
 audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
                                                  usecase_type_t type);
 
-int check_a2dp_restore(struct audio_device *adev, struct stream_out *out, bool restore);
+int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore);
 
 int adev_open_output_stream(struct audio_hw_device *dev,
                             audio_io_handle_t handle,
diff --git a/hal/voice.c b/hal/voice.c
index 72c3372..586247f 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -231,6 +231,11 @@
         return -EINVAL;
     }
 
+    if (!adev->current_call_output) {
+        ALOGE("start_call: invalid current call output");
+        return -EINVAL;
+    }
+
     uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
     if (!uc_info) {
         ALOGE("start_call: couldn't allocate mem for audio_usecase");