diff --git a/Android.mk b/Android.mk
index 00332c7..6db5a30 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,5 +1,5 @@
 ifneq ($(AUDIO_USE_STUB_HAL), true)
-ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916 msm8994 msm8992 msm8909 msm8996 msm8952 msm8937 thorium msm8953 msmgold msm8998 sdm660 sdm845 sdm710 apq8098_latv qcs605 msmnile kona $(MSMSTEPPE) $(TRINKET) atoll lito,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916 msm8994 msm8992 msm8909 msm8996 msm8952 msm8937 thorium msm8953 msmgold msm8998 sdm660 sdm845 sdm710 apq8098_latv qcs605 sdmshrike msmnile kona $(MSMSTEPPE) $(TRINKET) atoll lito,$(TARGET_BOARD_PLATFORM)),)
 
 MY_LOCAL_PATH := $(call my-dir)
 
diff --git a/configs/atoll/atoll.mk b/configs/atoll/atoll.mk
index 3517085..fc25ebf 100755
--- a/configs/atoll/atoll.mk
+++ b/configs/atoll/atoll.mk
@@ -248,6 +248,10 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.audio.fluence.voicecomm=true
 
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
 # for HIDL related packages
 PRODUCT_PACKAGES += \
     android.hardware.audio@2.0-service \
diff --git a/configs/atoll/audio_policy_configuration.xml b/configs/atoll/audio_policy_configuration.xml
index a6d7eef..b65a9b6 100644
--- a/configs/atoll/audio_policy_configuration.xml
+++ b/configs/atoll/audio_policy_configuration.xml
@@ -176,6 +176,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -266,20 +271,17 @@
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
@@ -361,6 +363,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="usb_surround_sound"
diff --git a/configs/common_au/audio_policy_configuration.xml b/configs/common_au/audio_policy_configuration.xml
new file mode 100644
index 0000000..ce3632a
--- /dev/null
+++ b/configs/common_au/audio_policy_configuration.xml
@@ -0,0 +1,272 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (c) 2016-2017, 2019, 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.
+            "gain": constraints applied to the millibel values:
+                - maxValueMB >= minValueMB
+                - defaultValueMB >= minValueMB && defaultValueMB <= maxValueMB
+                - (maxValueMB - minValueMB) % stepValueMB == 0
+                - (defaultValueMB - minValueMB) % stepValueMB == 0
+        “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>
+        <!-- Primary Audio HAL -->
+        <module name="primary" halVersion="3.0">
+            <attachedDevices>
+                <item>Media Bus</item>
+                <item>Sys Notification Bus</item>
+                <item>Nav Guidance Bus</item>
+                <item>Phone Bus</item>
+                <item>Built-In Mic</item>
+                <item>Built-In Back Mic</item>
+            </attachedDevices>
+            <defaultOutputDevice>Media Bus</defaultOutputDevice>
+            <mixPorts>
+                <mixPort name="media" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="sys_notification" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="nav_guidance" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="phone" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="primary output" role="source"
+                        flags="AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY">
+                    <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_16_BIT"
+                             samplingRates="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="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AAC_ADTS_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_ADTS_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_ADTS_HE_V2"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
+                             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" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </mixPort>
+                <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"/>
+                </mixPort>
+                <mixPort name="voice_rx" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+                <devicePort tagName="Media Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS00_MEDIA">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
+                <devicePort tagName="Sys Notification Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS01_SYS_NOTIFICATION">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
+                <devicePort tagName="Nav Guidance Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS02_NAV_GUIDANCE">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
+                <devicePort tagName="Phone Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS03_PHONE">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
+                <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
+                   <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                            samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </devicePort>
+                <devicePort tagName="Speaker" role="sink" type="AUDIO_DEVICE_OUT_SPEAKER">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="Wired Headset" type="AUDIO_DEVICE_OUT_WIRED_HEADSET" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="Wired Headphones" type="AUDIO_DEVICE_OUT_WIRED_HEADPHONE" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </devicePort>
+                <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </devicePort>
+                <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </devicePort>
+                <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </devicePort>
+                <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+                    <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"/>
+                </devicePort>
+                <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC" role="source">
+                    <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"/>
+                </devicePort>
+                <devicePort tagName="Wired Headset Mic" type="AUDIO_DEVICE_IN_WIRED_HEADSET" role="source">
+                    <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"/>
+                </devicePort>
+                <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </devicePort>
+                <devicePort tagName="Telephony Rx" type="AUDIO_DEVICE_IN_TELEPHONY_RX" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </devicePort>
+            </devicePorts>
+            <!-- route declaration, i.e. list all available sources for a given sink -->
+            <routes>
+                <route type="mix" sink="Media Bus"
+                       sources="media,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+                <route type="mix" sink="Sys Notification Bus"
+                       sources="sys_notification"/>
+                <route type="mix" sink="Nav Guidance Bus"
+                       sources="nav_guidance"/>
+                <route type="mix" sink="Phone Bus"
+                       sources="phone"/>
+                <route type="mix" sink="Earpiece"
+                       sources="primary output,deep_buffer,BT SCO Headset Mic"/>
+                <route type="mix" sink="Speaker"
+                       sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+                <route type="mix" sink="Wired Headset"
+                       sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+                <route type="mix" sink="Wired Headphones"
+                       sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+                <route type="mix" sink="Telephony Tx"
+                       sources="voice_tx"/>
+                <route type="mix" sink="primary input"
+                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
+                <route type="mix" sink="voice_rx"
+                       sources="Telephony Rx"/>
+            </routes>
+
+        </module>
+
+        <!-- A2dp Audio HAL -->
+        <xi:include href="a2dp_audio_policy_configuration.xml"/>
+
+        <!-- Usb Audio HAL -->
+        <xi:include href="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/common_au/overlay/frameworks/base/core/res/res/values/config.xml b/configs/common_au/overlay/frameworks/base/core/res/res/values/config.xml
new file mode 100644
index 0000000..acde6c2
--- /dev/null
+++ b/configs/common_au/overlay/frameworks/base/core/res/res/values/config.xml
@@ -0,0 +1,38 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+    Copyright (c) 2016, The Linux Foundation. All rights reserved.
+
+    Redistribution and use in source and binary forms, with or without
+    modification, are permitted provided that the following conditions are
+    met:
+        * Redistributions of source code must retain the above copyright
+          notice, this list of conditions and the following disclaimer.
+        * Redistributions in binary form must reproduce the above
+          copyright notice, this list of conditions and the following
+          disclaimer in the documentation and/or other materials provided
+          with the distribution.
+        * Neither the name of The Linux Foundation nor the names of its
+          contributors may be used to endorse or promote products derived
+          from this software without specific prior written permission.
+
+    THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+    WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+    BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+    BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+    WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+    OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+    IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE
+-->
+<resources>
+
+   <!-- This file contains only audio specific overrides for overlays -->
+
+   <!-- When true use the linux /dev/input/event subsystem to detect the switch changes
+   on the headphone/microphone jack. When false use the older uevent framework -->
+   <bool name="config_useDevInputEventForAudioJack">true</bool>
+
+</resources>
diff --git a/configs/kona/audio_platform_info.xml b/configs/kona/audio_platform_info.xml
index 346041f..6afca97 100644
--- a/configs/kona/audio_platform_info.xml
+++ b/configs/kona/audio_platform_info.xml
@@ -110,6 +110,7 @@
         <usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="36" />
         <usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="out" id="36" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="23" />
+        <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="23" />
         <usecase name="USECASE_AUDIO_RECORD_COMPRESS2" type="in" id="37" />
     </pcm_ids>
     <config_params>
diff --git a/configs/kona/audio_policy_configuration.xml b/configs/kona/audio_policy_configuration.xml
index 8bb3328..1e4e338 100644
--- a/configs/kona/audio_policy_configuration.xml
+++ b/configs/kona/audio_policy_configuration.xml
@@ -173,6 +173,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="usb_surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
@@ -350,6 +355,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="usb_surround_sound"
                        sources="USB Device In,USB Headset In"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/kona/mixer_paths.xml b/configs/kona/mixer_paths.xml
index 3949c7c..e987687 100644
--- a/configs/kona/mixer_paths.xml
+++ b/configs/kona/mixer_paths.xml
@@ -236,6 +236,7 @@
     <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
     <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="0" />
+    <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="0" />
     <ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
     <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
@@ -1740,6 +1741,10 @@
         <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
     </path>
 
+    <path name="compress-voip-call headset">
+        <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="1" />
+    </path>
 
     <path name="compress-voip-call bt-sco">
         <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
@@ -1887,6 +1892,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="audio-playback-voip headset">
+        <path name="audio-playback-voip headphones" />
+    </path>
+
     <path name="audio-playback-voip bt-sco">
         <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -2517,7 +2526,12 @@
     </path>
 
     <path name="voice-tty-hco-headset-mic">
-        <path name="voice-tty-full-headset-mic" />
+        <ctl name="TX DEC6 MUX" value="SWR_MIC" />
+        <ctl name="TX SMIC MUX6" value="ADC1" />
+        <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+        <ctl name="TX_AIF1_CAP Mixer DEC6" value="1" />
+        <ctl name="ADC2_MIXER Switch" value="1" />
+        <ctl name="ADC2 MUX" value="INP2" />
     </path>
 
     <path name="voice-tty-vco-handset-mic">
@@ -2703,6 +2717,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="mmap-playback headset">
+        <path name="mmap-playback headphones" />
+    </path>
+
     <path name="mmap-playback speaker-and-headphones">
         <path name="mmap-playback" />
         <path name="mmap-playback headphones" />
@@ -2891,4 +2909,80 @@
     <path name="incall_music_uplink afe-proxy">
         <path name="incall_music_uplink" />
     </path>
+
+    <path name="incall_music_uplink2">
+        <ctl name="Incall_Music_2 Audio Mixer MultiMedia9" value="1" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 handset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 handset-hac">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 display-port">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-sco">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-sco-wb">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-display-port">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 afe-proxy">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 usb-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 usb-headset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-usb-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-bt-sco">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 voice-tty-hco-handset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-bt-a2dp">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-a2dp">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 afe-proxy">
+        <path name="incall_music_uplink2" />
+    </path>
 </mixer>
diff --git a/configs/kona/mixer_paths_cdp.xml b/configs/kona/mixer_paths_cdp.xml
index 67041e6..ed9bfc0 100644
--- a/configs/kona/mixer_paths_cdp.xml
+++ b/configs/kona/mixer_paths_cdp.xml
@@ -237,6 +237,7 @@
     <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
     <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="0" />
+    <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="0" />
     <ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
     <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
@@ -1746,6 +1747,10 @@
         <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
     </path>
 
+    <path name="compress-voip-call headset">
+        <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="1" />
+    </path>
 
     <path name="compress-voip-call bt-sco">
         <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
@@ -1893,6 +1898,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="audio-playback-voip headset">
+        <path name="audio-playback-voip headphones" />
+    </path>
+
     <path name="audio-playback-voip bt-sco">
         <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -2725,6 +2734,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="mmap-playback headset">
+        <path name="mmap-playback headphones" />
+    </path>
+
     <path name="mmap-playback speaker-and-headphones">
         <path name="mmap-playback" />
         <path name="mmap-playback headphones" />
diff --git a/configs/kona/mixer_paths_qrd.xml b/configs/kona/mixer_paths_qrd.xml
index 08176d5..3cf38a1 100644
--- a/configs/kona/mixer_paths_qrd.xml
+++ b/configs/kona/mixer_paths_qrd.xml
@@ -236,6 +236,7 @@
     <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
     <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="0" />
+    <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="0" />
     <ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
     <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
@@ -1757,6 +1758,11 @@
         <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
     </path>
 
+    <path name="compress-voip-call headset">
+        <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="1" />
+    </path>
+
     <path name="compress-voip-call bt-sco">
         <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
         <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="1" />
@@ -1917,6 +1923,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="audio-playback-voip headset">
+        <path name="audio-playback-voip headphones" />
+    </path>
+
     <path name="audio-playback-voip bt-sco">
         <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -2110,6 +2120,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="mmap-playback headset">
+        <path name="mmap-playback headphones" />
+    </path>
+
     <path name="mmap-playback speaker-and-headphones">
         <path name="mmap-playback" />
         <path name="mmap-playback headphones" />
@@ -2299,6 +2313,85 @@
         <path name="incall_music_uplink" />
     </path>
 
+    <path name="incall_music_uplink2">
+        <ctl name="Incall_Music_2 Audio Mixer MultiMedia9" value="1" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 handset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 handset-hac">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 display-port">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-sco">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-sco-wb">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-sco-swb">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-display-port">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 afe-proxy">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 usb-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 usb-headset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-usb-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-bt-sco">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 voice-tty-hco-handset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-bt-a2dp">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-a2dp">
+        <path name="incall_music_uplink2" />
+    </path>
+    <path name="incall_music_uplink2 afe-proxy">
+        <path name="incall_music_uplink2" />
+    </path>
+
    <path name="spkr-rx-calib">
         <ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch"  value="1" />
     </path>
diff --git a/configs/kona/sound_trigger_platform_info.xml b/configs/kona/sound_trigger_platform_info.xml
index 7c8c25f..2a44adc 100644
--- a/configs/kona/sound_trigger_platform_info.xml
+++ b/configs/kona/sound_trigger_platform_info.xml
@@ -98,6 +98,14 @@
             <param bit_wdith="16"/>
             <param channel_count="1"/>
         </arm_ss_usecase>
+        <arm_ss_usecase>
+            <param sm_detection_type= "KEYWORD_DETECTION" />
+            <param sm_id="0x8" />
+            <param module_lib="libcapiv2svarnn.so"/>
+            <param sample_rate="16000"/>
+            <param bit_wdith="16"/>
+            <param channel_count="1"/>
+        </arm_ss_usecase>
         <!-- Module and param ids with which the algorithm is integrated
             in non-graphite firmware (note these must come after gcs params)
             Extends flexibility to have different ids based on execution type.
diff --git a/configs/lito/audio_platform_info.xml b/configs/lito/audio_platform_info.xml
index 774bfe3..93d47c5 100644
--- a/configs/lito/audio_platform_info.xml
+++ b/configs/lito/audio_platform_info.xml
@@ -218,8 +218,8 @@
         <device name="SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE" interface="TX_CDC_DMA_TX_3"/>
         <device name="SND_DEVICE_IN_AANC_HANDSET_MIC" interface="TX_CDC_DMA_TX_3"/>
         <device name="SND_DEVICE_IN_QUAD_MIC" interface="TX_CDC_DMA_TX_3"/>
-        <device name="SND_DEVICE_IN_HANDSET_STEREO_DMIC" interface="TX_CDC_DMA_TX_3"/>
-        <device name="SND_DEVICE_IN_SPEAKER_STEREO_DMIC" interface="TX_CDC_DMA_TX_3"/>
+        <device name="SND_DEVICE_IN_HANDSET_DMIC_STEREO" interface="TX_CDC_DMA_TX_3"/>
+        <device name="SND_DEVICE_IN_SPEAKER_DMIC_STEREO" interface="TX_CDC_DMA_TX_3"/>
         <device name="SND_DEVICE_IN_CAPTURE_VI_FEEDBACK" interface="WSA_CDC_DMA_TX_0"/>
         <device name="SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1" interface="WSA_CDC_DMA_TX_0"/>
         <device name="SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2" interface="WSA_CDC_DMA_TX_0"/>
@@ -432,13 +432,13 @@
                         <mic_info mic_device_id="builtin_mic_4"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                     </snd_dev>
-                    <snd_dev in_snd_device="SND_DEVICE_IN_HANDSET_STEREO_DMIC">
+                    <snd_dev in_snd_device="SND_DEVICE_IN_HANDSET_DMIC_STEREO">
                         <mic_info mic_device_id="builtin_mic_1"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                         <mic_info mic_device_id="builtin_mic_2"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                     </snd_dev>
-                    <snd_dev in_snd_device="SND_DEVICE_IN_SPEAKER_STEREO_DMIC">
+                    <snd_dev in_snd_device="SND_DEVICE_IN_SPEAKER_DMIC_STEREO">
                         <mic_info mic_device_id="builtin_mic_1"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                         <mic_info mic_device_id="builtin_mic_2"
diff --git a/configs/lito/audio_policy_configuration.xml b/configs/lito/audio_policy_configuration.xml
index 9849a0c..c6365ff 100644
--- a/configs/lito/audio_policy_configuration.xml
+++ b/configs/lito/audio_policy_configuration.xml
@@ -173,6 +173,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -263,20 +268,17 @@
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
@@ -358,6 +360,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="usb_surround_sound"
diff --git a/configs/lito/lito.mk b/configs/lito/lito.mk
index d6e51bc..e969a17 100644
--- a/configs/lito/lito.mk
+++ b/configs/lito/lito.mk
@@ -380,6 +380,10 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.vendor.bt.aac_frm_ctl.enabled=true
 
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
 #add dynamic feature flags here
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.feature.a2dp_offload.enable=true \
diff --git a/configs/lito/sound_trigger_platform_info.xml b/configs/lito/sound_trigger_platform_info.xml
index 1337e9f..3f5f5df 100644
--- a/configs/lito/sound_trigger_platform_info.xml
+++ b/configs/lito/sound_trigger_platform_info.xml
@@ -72,14 +72,14 @@
     <sound_model_config>
         <param vendor_uuid="68ab2d40-e860-11e3-95ef-0002a5d5c51b" />
         <param execution_type="ADSP" />
-        <param library="libsmwrapper.so" />
+        <param merge_first_stage_sound_models="false" />
         <param max_ape_phrases="20" />
         <param max_ape_users="10" />
         <!-- Profile specific data which the algorithm can support -->
         <param sample_rate="16000" />
         <param bit_width="16" />
         <param out_channels="1"/> <!-- Module output channels -->
-
+        <param dam_token_id="1"/>
         <arm_ss_usecase>
             <!-- Options are "KEYWORD_DETECTION", "USER_VERIFICATION", "CUSTOM_DETECTION"  -->
             <param sm_detection_type= "KEYWORD_DETECTION" />
@@ -113,7 +113,7 @@
             <param execution_mode="ADSP" />
             <param app_type="2" /> <!-- app type used in ACDB -->
             <param in_channels="5"/> <!-- Module input channels -->
-            <param in_channels_lpi="3"/>
+            <param in_channels_lpi="1"/>
             <param load_sound_model_ids="0x00012C1C, 0x0, 0x00012C14" />
             <param unload_sound_model_ids="0x00012C1C, 0x0, 0x00012C15" />
             <param confidence_levels_ids="0x00012C1C, 0x0, 0x00012C07" />
@@ -121,6 +121,7 @@
             <param polling_enable_ids="0x00012C1C, 0x0, 0x00012C1B" />
             <param custom_config_ids="0x00012C1C, 0x0, 0x00012C20" />
             <param det_event_type_ids="0x00012C1C, 0x0, 0x00012C2C" />
+            <param lab_dam_cfg_ids="0x00012C08, 0x0, 0x000102C4" />
         </lsm_usecase>
         <lsm_usecase>
             <param capture_device="HEADSET" />
@@ -136,6 +137,7 @@
             <param polling_enable_ids="0x00012C1C, 0x0, 0x00012C1B" />
             <param custom_config_ids="0x00012C1C, 0x0, 0x00012C20" />
             <param det_event_type_ids="0x00012C1C, 0x0, 0x00012C2C" />
+            <param lab_dam_cfg_ids="0x00012C08, 0x0, 0x000102C4" />
         </lsm_usecase>
 
         <!-- format: "ADPCM_packet" or "PCM_packet" !-->
@@ -150,7 +152,6 @@
     <sound_model_config>
         <param vendor_uuid="876c1b46-9d4d-40cc-a4fd-4d5ec7a80e47" />
         <param execution_type="ADSP" />
-        <param library="libsmwrapper.so" />
         <param max_ape_phrases="1" />
         <param max_ape_users="1" />
         <!-- Profile specific data which the algorithm can support -->
@@ -205,7 +206,6 @@
     <sound_model_config>
         <param vendor_uuid="7038ddc8-30f2-11e6-b0ac-40a8f03d3f15" />
         <param execution_type="ADSP" />
-        <param library="none" />
         <param max_ape_phrases="1" />
         <param max_ape_users="1" />
         <!-- Profile specific data which the algorithm can support -->
@@ -258,7 +258,6 @@
     <sound_model_config>
         <param vendor_uuid="9f6ad62a-1f0b-11e7-87c5-40a8f03d3f15" />
         <param execution_type="ADSP" />
-        <param library="none" />
         <param max_ape_phrases="1" />
         <param max_ape_users="1" />
         <!-- Profile specific data which the algorithm can support -->
diff --git a/configs/msm8998/audio_policy_configuration.xml b/configs/msm8998/audio_policy_configuration.xml
index 505a205..62e75c2 100644
--- a/configs/msm8998/audio_policy_configuration.xml
+++ b/configs/msm8998/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -299,16 +304,8 @@
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
                 </devicePort>
                 <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
 
             </devicePorts>
@@ -346,6 +343,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/msm8998/msm8998.mk b/configs/msm8998/msm8998.mk
index 6b77f69..81077c1 100644
--- a/configs/msm8998/msm8998.mk
+++ b/configs/msm8998/msm8998.mk
@@ -99,7 +99,9 @@
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/sound_trigger_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_platform_info.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/audio_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_platform_info.xml \
-    vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+    vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
+    frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+    frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
 
 #XML Audio configuration files
 ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -250,6 +252,10 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.audio.fluence.voicecomm=true
 
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
 #add dynamic feature flags here
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.feature.a2dp_offload.enable=true \
diff --git a/configs/msmnile/audio_platform_info.xml b/configs/msmnile/audio_platform_info.xml
index 80924e2..042f081 100644
--- a/configs/msmnile/audio_platform_info.xml
+++ b/configs/msmnile/audio_platform_info.xml
@@ -101,6 +101,7 @@
         <usecase name="USECASE_AUDIO_RECORD_MMAP" type="in" id="33" />
         <usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="40" />
         <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="27" />
+        <usecase name="USECASE_INCALL_MUSIC_UPLINK2" type="out" id="27" />
         <usecase name="USECASE_AUDIO_RECORD_COMPRESS2" type="in" id="41" />
     </pcm_ids>
     <config_params>
diff --git a/configs/msmnile/audio_policy_configuration.xml b/configs/msmnile/audio_policy_configuration.xml
index 5c05206..1e4e338 100644
--- a/configs/msmnile/audio_policy_configuration.xml
+++ b/configs/msmnile/audio_policy_configuration.xml
@@ -173,6 +173,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="usb_surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
@@ -184,7 +189,7 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
                              channelMasks="AUDIO_CHANNEL_IN_5POINT1,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
                 </mixPort>
-                <mixPort name="record_24" role="sink">
+                <mixPort name="record_24" role="sink" maxOpenCount="2" maxActiveCount="2">
                     <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3,AUDIO_CHANNEL_INDEX_MASK_4"/>
@@ -350,6 +355,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="usb_surround_sound"
                        sources="USB Device In,USB Headset In"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/msmnile/mixer_paths_tavil.xml b/configs/msmnile/mixer_paths_tavil.xml
index fb315bf..d4ee6fd 100644
--- a/configs/msmnile/mixer_paths_tavil.xml
+++ b/configs/msmnile/mixer_paths_tavil.xml
@@ -3025,7 +3025,12 @@
     </path>
 
     <path name="voice-tty-hco-headset-mic">
-        <path name="voice-tty-full-headset-mic" />
+        <ctl name="AIF1_CAP Mixer SLIM TX0" value="1"/>
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+        <ctl name="ADC MUX0" value="AMIC" />
+        <ctl name="AMIC MUX0" value="ADC2" />
+        <ctl name="IIR0 INP0 MUX" value="DEC0" />
     </path>
 
     <path name="voice-tty-vco-handset-mic">
@@ -3448,4 +3453,80 @@
     <path name="incall_music_uplink afe-proxy">
         <path name="incall_music_uplink" />
     </path>
+
+    <path name="incall_music_uplink2">
+        <ctl name="Incall_Music_2 Audio Mixer MultiMedia9" value="1" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 handset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 handset-hac">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 display-port">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-sco">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-sco-wb">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-display-port">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 afe-proxy">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 usb-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 usb-headset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-usb-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-headphones">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-bt-sco">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 voice-tty-hco-handset">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 speaker-and-bt-a2dp">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 bt-a2dp">
+        <path name="incall_music_uplink2" />
+    </path>
+
+    <path name="incall_music_uplink2 afe-proxy">
+        <path name="incall_music_uplink2" />
+    </path>
 </mixer>
diff --git a/configs/msmnile_au/audio_output_policy.conf b/configs/msmnile_au/audio_io_policy.conf
similarity index 72%
copy from configs/msmnile_au/audio_output_policy.conf
copy to configs/msmnile_au/audio_io_policy.conf
index 67febfa..3362dd9 100644
--- a/configs/msmnile_au/audio_output_policy.conf
+++ b/configs/msmnile_au/audio_io_policy.conf
@@ -12,11 +12,11 @@
 
 outputs {
   default {
-    flags AUDIO_OUTPUT_FLAG_PRIMARY
+    flags AUDIO_OUTPUT_FLAG_PRIMARY|AUDIO_OUTPUT_FLAG_MEDIA
     formats AUDIO_FORMAT_PCM_16_BIT
     sampling_rates 48000
     bit_width 16
-    app_type 69937
+    app_type 69936
   }
   proaudio {
     flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW
@@ -81,4 +81,53 @@
     bit_width 24
     app_type 69940
   }
+  sys_notification {
+    flags AUDIO_OUTPUT_FLAG_SYS_NOTIFICATION
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69937
+  }
+  nav_guidance {
+    flags AUDIO_OUTPUT_FLAG_NAV_GUIDANCE
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69937
+  }
+  phone {
+    flags AUDIO_OUTPUT_FLAG_PHONE
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69936
+  }
 }
+
+inputs {
+  primary {
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 8000|16000|32000|44100|48000|88200|96000|176400|192000
+    bit_width 16
+    app_type 69938
+  }
+  record_24bit {
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_24_BIT
+    sampling_rates 44100|48000|88200|96000|176400|192000
+    bit_width 24
+    app_type 69948
+  }
+  record_32bit {
+    formats AUDIO_FORMAT_PCM_32_BIT|AUDIO_FORMAT_PCM_FLOAT
+    sampling_rates 44100|48000|88200|96000|176400|192000
+    bit_width 32
+    app_type 69949
+  }
+  record_unprocessed {
+    profile record_unprocessed
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED
+    sampling_rates 16000|48000
+    bit_width 24
+    app_type 69942
+  }
+}
\ No newline at end of file
diff --git a/configs/msmnile_au/audio_platform_info.xml b/configs/msmnile_au/audio_platform_info.xml
index e02397c..678e354 100644
--- a/configs/msmnile_au/audio_platform_info.xml
+++ b/configs/msmnile_au/audio_platform_info.xml
@@ -22,28 +22,32 @@
 <!-- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF   -->
 <!-- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR        -->
 <!-- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,  -->
+<!-- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE   -->
 <!-- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
 <!-- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                          -->
 <audio_platform_info>
     <acdb_ids>
-        <device name="SND_DEVICE_OUT_HANDSET" acdb_id="78"/>
-        <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="78"/>
-        <device name="SND_DEVICE_OUT_HEADPHONES" acdb_id="78"/>
-        <device name="SND_DEVICE_OUT_BUS_MEDIA" acdb_id="78"/>
-        <device name="SND_DEVICE_OUT_BUS_SYS" acdb_id="78"/>
+        <device name="SND_DEVICE_OUT_HANDSET" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_HEADPHONES" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_BUS_MEDIA" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_BUS_SYS" acdb_id="60"/>
         <device name="SND_DEVICE_OUT_BUS_NAV" acdb_id="14"/>
         <device name="SND_DEVICE_OUT_BUS_PHN" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_BT_SCO" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="94"/>
-        <device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="78"/>
+        <device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="60"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" acdb_id="94"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_HFP" acdb_id="94"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" acdb_id="94"/>
         <device name="SND_DEVICE_IN_HANDSET_MIC" acdb_id="11"/>
         <device name="SND_DEVICE_IN_SPEAKER_MIC" acdb_id="11"/>
         <device name="SND_DEVICE_IN_BUS" acdb_id="11"/>
         <device name="SND_DEVICE_IN_HEADSET_MIC" acdb_id="11"/>
-        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" acdb_id="95"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" acdb_id="95"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" acdb_id="95"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" acdb_id="80"/>
@@ -140,11 +144,14 @@
         <device name="SND_DEVICE_OUT_BT_A2DP" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="TERT_TDM_RX_2"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="TERT_TDM_RX_2"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_HFP" interface="TERT_TDM_RX_2"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="TERT_TDM_RX_2"/>
         <device name="SND_DEVICE_IN_HANDSET_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_HEADSET_MIC" interface="TERT_TDM_TX_0"/>
-        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" interface="TERT_TDM_TX_0"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" interface="TERT_TDM_TX_0"/>
diff --git a/configs/msmnile_au/audio_policy_configuration.xml b/configs/msmnile_au/audio_policy_configuration.xml
index b00e62f..1bbb52e 100644
--- a/configs/msmnile_au/audio_policy_configuration.xml
+++ b/configs/msmnile_au/audio_policy_configuration.xml
@@ -391,7 +391,7 @@
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
                 <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,Telephony Rx"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/msmnile_au/mixer_paths_adp.xml b/configs/msmnile_au/mixer_paths_adp.xml
index 63012be..a2a1eb3 100644
--- a/configs/msmnile_au/mixer_paths_adp.xml
+++ b/configs/msmnile_au/mixer_paths_adp.xml
@@ -286,10 +286,6 @@
     <ctl name="SEC_TDM_RX_0 Audio Mixer MultiMedia9" value="0" />
     <ctl name="MultiMedia9 Mixer TERT_TDM_TX_0" value="0" />
 
-    <path name="dummy-hostless">
-        <ctl name="SEC_TDM_RX_7 Port Mixer TERT_TDM_TX_7" value="1" />
-    </path>
-
     <!-- These are audio route (FE to BE) specific mixer settings -->
 
     <!-- EC Reference -->
@@ -329,8 +325,6 @@
     <path name="deep-buffer-playback">
         <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
         <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
-        <ctl name="QUAT_TDM_RX_0 Channels" value="Six" />
-        <ctl name="QUAT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
     </path>
 
     <path name="deep-buffer-playback speaker-protected">
@@ -529,6 +523,8 @@
         <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia4" value="1" />
         <ctl name="QUAT_TDM_RX_0 Channels" value="Six" />
         <ctl name="QUAT_TDM_RX_0 Audio Mixer MultiMedia4" value="1" />
+        <ctl name="QUIN_TDM_RX_0 Channels" value="Two" />
+        <ctl name="QUIN_TDM_RX_0 Audio Mixer MultiMedia4" value="1" />
     </path>
 
     <path name="compress-offload-playback speaker-protected">
@@ -606,6 +602,8 @@
         <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia7" value="1" />
         <ctl name="QUAT_TDM_RX_0 Channels" value="Six" />
         <ctl name="QUAT_TDM_RX_0 Audio Mixer MultiMedia7" value="1" />
+        <ctl name="QUIN_TDM_RX_0 Channels" value="Two" />
+        <ctl name="QUIN_TDM_RX_0 Audio Mixer MultiMedia7" value="1" />
     </path>
 
     <path name="compress-offload-playback2 display-port">
@@ -1144,6 +1142,10 @@
     <path name="media-playback">
         <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
         <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+        <ctl name="QUAT_TDM_RX_0 Channels" value="Six" />
+        <ctl name="QUAT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+        <ctl name="QUIN_TDM_RX_0 Channels" value="Two" />
+        <ctl name="QUIN_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
     </path>
 
     <path name="sys-notification-playback">
diff --git a/configs/msmnile_au/msmnile_au.mk b/configs/msmnile_au/msmnile_au.mk
index 394dfea..db33c9a 100644
--- a/configs/msmnile_au/msmnile_au.mk
+++ b/configs/msmnile_au/msmnile_au.mk
@@ -62,7 +62,7 @@
 AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
 BOARD_SUPPORTS_QAHW := false
 AUDIO_FEATURE_ENABLED_RAS := true
-AUDIO_FEATURE_ENABLED_SND_MONITOR := true
+AUDIO_FEATURE_ENABLED_SND_MONITOR := false
 AUDIO_FEATURE_ENABLED_DLKM := true
 AUDIO_FEATURE_ENABLED_USB_BURST_MODE := false
 AUDIO_FEATURE_ENABLED_SVA_MULTI_STAGE := false
@@ -72,18 +72,22 @@
 AUDIO_FEATURE_ENABLED_AUTO_HAL := true
 AUDIO_FEATURE_ENABLED_EXT_HW_PLUGIN := true
 AUDIO_FEATURE_ENABLED_AUDIO_CONTROL_HAL := true
+ifneq ($(ENABLE_HYP),true)
+AUDIO_FEATURE_ENABLED_AUTO_AUDIOD := true
+endif
+AUDIO_FEATURE_ENABLED_FM_TUNER_EXT := true
 ##AUTOMOTIVE_AUDIO_FEATURE_FLAGS
 
 ifneq ($(strip $(TARGET_USES_RRO)), true)
 #Audio Specific device overlays
-DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
+DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common_au/overlay
 endif
 
 #Automotive audio specific device overlays
-DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmnile_au/overlay
+DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/overlay
 
 PRODUCT_COPY_FILES += \
-    vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_output_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_output_policy.conf \
+    vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_io_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_io_policy.conf \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_effects.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.conf \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_effects.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/mixer_paths_adp.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_adp.xml \
@@ -102,7 +106,7 @@
     $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_policy_configuration.xml:$(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 \
+    $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/common_au/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
     $(TOPDIR)frameworks/av/services/audiopolicy/config/a2dp_audio_policy_configuration.xml:$(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/msmsteppe/audio_policy_configuration.xml b/configs/msmsteppe/audio_policy_configuration.xml
index b092687..99f03bf 100644
--- a/configs/msmsteppe/audio_policy_configuration.xml
+++ b/configs/msmsteppe/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="incall_music_uplink" role="source"
                         flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -350,6 +355,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="usb_surround_sound"
                        sources="USB Device In,USB Headset In"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/msmnile_au/audio_output_policy.conf b/configs/msmsteppe_au/audio_io_policy.conf
similarity index 75%
rename from configs/msmnile_au/audio_output_policy.conf
rename to configs/msmsteppe_au/audio_io_policy.conf
index 67febfa..56cb909 100644
--- a/configs/msmnile_au/audio_output_policy.conf
+++ b/configs/msmsteppe_au/audio_io_policy.conf
@@ -12,11 +12,11 @@
 
 outputs {
   default {
-    flags AUDIO_OUTPUT_FLAG_PRIMARY
+    flags AUDIO_OUTPUT_FLAG_PRIMARY|AUDIO_OUTPUT_FLAG_MEDIA
     formats AUDIO_FORMAT_PCM_16_BIT
     sampling_rates 48000
     bit_width 16
-    app_type 69937
+    app_type 69936
   }
   proaudio {
     flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW
@@ -81,4 +81,46 @@
     bit_width 24
     app_type 69940
   }
+  sys_notification {
+    flags AUDIO_OUTPUT_FLAG_SYS_NOTIFICATION
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69937
+  }
+  nav_guidance {
+    flags AUDIO_OUTPUT_FLAG_NAV_GUIDANCE
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69937
+  }
+  phone {
+    flags AUDIO_OUTPUT_FLAG_PHONE
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69936
+  }
+}
+
+inputs {
+  record_16bit {
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 8000|16000|32000|44100|48000|88200|96000|176400|192000
+    bit_width 16
+    app_type 69938
+  }
+  record_24bit {
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_24_BIT
+    sampling_rates 44100|48000|88200|96000|176400|192000
+    bit_width 24
+    app_type 69948
+  }
+  record_32bit {
+    formats AUDIO_FORMAT_PCM_32_BIT|AUDIO_FORMAT_PCM_FLOAT
+    sampling_rates 44100|48000|88200|96000|176400|192000
+    bit_width 32
+    app_type 69949
+  }
 }
diff --git a/configs/msmsteppe_au/audio_output_policy.conf b/configs/msmsteppe_au/audio_output_policy.conf
deleted file mode 100644
index 67febfa..0000000
--- a/configs/msmsteppe_au/audio_output_policy.conf
+++ /dev/null
@@ -1,84 +0,0 @@
-# List of profiles for the output device session where stream is routed.
-# A stream opened with the inputs attributes which match the "flags" and
-# "formats" as specified in the profile is routed to a device at
-# sample rate specified under "sampling_rates" and bit width under
-# "bit_width" and the topology extracted from the acdb data against
-# the "app_type".
-#
-# the flags and formats are specified using the strings corresponding to
-# enums in audio.h and audio_policy.h. They are concatenated with "|"
-# without space or "\n".
-# the flags and formats should match the ones in "audio_policy.conf"
-
-outputs {
-  default {
-    flags AUDIO_OUTPUT_FLAG_PRIMARY
-    formats AUDIO_FORMAT_PCM_16_BIT
-    sampling_rates 48000
-    bit_width 16
-    app_type 69937
-  }
-  proaudio {
-    flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW
-    formats AUDIO_FORMAT_PCM_16_BIT
-    sampling_rates 48000
-    bit_width 16
-    app_type 69943
-  }
-  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
-    sampling_rates 48000
-    bit_width 16
-    app_type 69936
-  }
-  direct_pcm_16 {
-    flags AUDIO_OUTPUT_FLAG_DIRECT
-    formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT|AUDIO_FORMAT_PCM_32_BIT
-    sampling_rates 44100|48000|88200|96000|176400|192000
-    bit_width 16
-    app_type 69936
-  }
-  direct_pcm_24 {
-    flags AUDIO_OUTPUT_FLAG_DIRECT
-    formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT|AUDIO_FORMAT_PCM_32_BIT
-    sampling_rates 44100|48000|88200|96000|176400|192000|352800|384000
-    bit_width 24
-    app_type 69940
-  }
-  direct_pcm_32 {
-    flags AUDIO_OUTPUT_FLAG_DIRECT
-    formats AUDIO_FORMAT_PCM_32_BIT
-    sampling_rates 44100|48000|88200|96000|176400|192000|352800|384000
-    bit_width 32
-    app_type 69942
-  }
-  compress_passthrough {
-    flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING|AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
-    formats AUDIO_FORMAT_DTS|AUDIO_FORMAT_DTS_HD|AUDIO_FORMAT_DSD
-    sampling_rates 32000|44100|48000|88200|96000|176400|192000|352800
-    bit_width 16
-    app_type 69941
-  }
-  compress_offload_16 {
-    flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
-    sampling_rates 44100|48000|88200|96000|176400|192000
-    bit_width 16
-    app_type 69936
-  }
-  compress_offload_24 {
-    flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO
-    sampling_rates 44100|48000|88200|96000|176400|192000
-    bit_width 24
-    app_type 69940
-  }
-}
diff --git a/configs/msmsteppe_au/audio_platform_info.xml b/configs/msmsteppe_au/audio_platform_info.xml
index a33ae3f..678e354 100644
--- a/configs/msmsteppe_au/audio_platform_info.xml
+++ b/configs/msmsteppe_au/audio_platform_info.xml
@@ -27,19 +27,27 @@
 <!-- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                          -->
 <audio_platform_info>
     <acdb_ids>
-        <device name="SND_DEVICE_OUT_HANDSET" acdb_id="78"/>
-        <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="78"/>
-        <device name="SND_DEVICE_OUT_HEADPHONES" acdb_id="78"/>
+        <device name="SND_DEVICE_OUT_HANDSET" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_HEADPHONES" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_BUS_MEDIA" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_BUS_SYS" acdb_id="60"/>
+        <device name="SND_DEVICE_OUT_BUS_NAV" acdb_id="14"/>
+        <device name="SND_DEVICE_OUT_BUS_PHN" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_BT_SCO" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="94"/>
-        <device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="78"/>
+        <device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="60"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" acdb_id="94"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_HFP" acdb_id="94"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" acdb_id="94"/>
         <device name="SND_DEVICE_IN_HANDSET_MIC" acdb_id="11"/>
         <device name="SND_DEVICE_IN_SPEAKER_MIC" acdb_id="11"/>
+        <device name="SND_DEVICE_IN_BUS" acdb_id="11"/>
         <device name="SND_DEVICE_IN_HEADSET_MIC" acdb_id="11"/>
-        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" acdb_id="95"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" acdb_id="95"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" acdb_id="95"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" acdb_id="80"/>
@@ -136,11 +144,14 @@
         <device name="SND_DEVICE_OUT_BT_A2DP" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="TERT_TDM_RX_2"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="TERT_TDM_RX_2"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_HFP" interface="TERT_TDM_RX_2"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="TERT_TDM_RX_2"/>
         <device name="SND_DEVICE_IN_HANDSET_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_HEADSET_MIC" interface="TERT_TDM_TX_0"/>
-        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" interface="TERT_TDM_TX_0"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" interface="TERT_TDM_TX_0"/>
diff --git a/configs/msmsteppe_au/audio_policy_configuration.xml b/configs/msmsteppe_au/audio_policy_configuration.xml
index 4d9340d..9fe1345 100644
--- a/configs/msmsteppe_au/audio_policy_configuration.xml
+++ b/configs/msmsteppe_au/audio_policy_configuration.xml
@@ -33,6 +33,11 @@
         “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.
+            "gain": constraints applied to the millibel values:
+                - maxValueMB >= minValueMB
+                - defaultValueMB >= minValueMB && defaultValueMB <= maxValueMB
+                - (maxValueMB - minValueMB) % stepValueMB == 0
+                - (defaultValueMB - minValueMB) % stepValueMB == 0
         “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.
@@ -47,18 +52,37 @@
     -->
     <modules>
         <!-- Primary Audio HAL -->
-        <module name="primary" halVersion="2.0">
+        <module name="primary" halVersion="3.0">
             <attachedDevices>
-                <item>Earpiece</item>
-                <item>Speaker</item>
+                <item>Media Bus</item>
+                <item>Sys Notification Bus</item>
+                <item>Nav Guidance Bus</item>
+                <item>Phone Bus</item>
                 <item>Telephony Tx</item>
                 <item>Built-In Mic</item>
                 <item>Built-In Back Mic</item>
                 <item>FM Tuner</item>
                 <item>Telephony Rx</item>
             </attachedDevices>
-            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <defaultOutputDevice>Media Bus</defaultOutputDevice>
             <mixPorts>
+                <mixPort name="media" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="sys_notification" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="nav_guidance" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="phone" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
                 <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
@@ -203,6 +227,42 @@
 
             <devicePorts>
                 <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+                <devicePort tagName="Media Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS00_MEDIA">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
+                <devicePort tagName="Sys Notification Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS01_SYS_NOTIFICATION">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
+                <devicePort tagName="Nav Guidance Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS02_NAV_GUIDANCE">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
+                <devicePort tagName="Phone Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+                        address="BUS03_PHONE">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                    <gains>
+                        <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+                                minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+                    </gains>
+                </devicePort>
                 <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
@@ -301,6 +361,14 @@
             </devicePorts>
             <!-- route declaration, i.e. list all available sources for a given sink -->
             <routes>
+                <route type="mix" sink="Media Bus"
+                       sources="media,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
+                <route type="mix" sink="Sys Notification Bus"
+                       sources="sys_notification"/>
+                <route type="mix" sink="Nav Guidance Bus"
+                       sources="nav_guidance"/>
+                <route type="mix" sink="Phone Bus"
+                       sources="phone"/>
                 <route type="mix" sink="Earpiece"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="Speaker"
@@ -324,7 +392,7 @@
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
                 <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,Telephony Rx"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/msmsteppe_au/mixer_paths_adp.xml b/configs/msmsteppe_au/mixer_paths_adp.xml
index e2de539..7386e48 100644
--- a/configs/msmsteppe_au/mixer_paths_adp.xml
+++ b/configs/msmsteppe_au/mixer_paths_adp.xml
@@ -287,10 +287,6 @@
     <ctl name="SEC_TDM_RX_0 Audio Mixer MultiMedia9" value="0" />
     <ctl name="MultiMedia9 Mixer TERT_TDM_TX_0" value="0" />
 
-    <path name="dummy-hostless">
-        <ctl name="SEC_TDM_RX_7 Port Mixer TERT_TDM_TX_7" value="1" />
-    </path>
-
     <!-- These are audio route (FE to BE) specific mixer settings -->
 
     <!-- EC Reference -->
@@ -755,6 +751,11 @@
         <path name="compress-offload-playback4" />
     </path>
 
+    <path name="voice-call">
+        <!-- Added AFE loopback ctrl path for CS-Voice call-->
+        <ctl name="TERT_TDM_RX_0 Port Mixer AUX_PCM_UL_TX" value="1" />
+        <ctl name="AUX_PCM_RX Port Mixer TERT_TDM_TX_0" value="1" />
+    </path>
 
     <path name="compress-offload-playback4 afe-proxy">
         <ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="1" />
@@ -1135,6 +1136,27 @@
         <path name="compress-offload-playback9" />
     </path>
 
+    <!-- The following use cases are used for car streams  -->
+    <path name="media-playback">
+        <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+        <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="sys-notification-playback">
+        <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+        <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="nav-guidance-playback">
+        <ctl name="TERT_TDM_RX_1 Channels" value="One" />
+        <ctl name="TERT_TDM_RX_1 Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="phone-playback">
+        <ctl name="TERT_TDM_RX_2 Channels" value="One" />
+        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia10" value="1" />
+    </path>
+
     <path name="audio-record">
         <ctl name="TERT_TDM_TX_0 Channels" value="One" />
         <ctl name="MultiMedia1 Mixer TERT_TDM_TX_0" value="1" />
@@ -1622,9 +1644,15 @@
     </path>
 
     <path name="speaker-adp">
+        <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
     </path>
 
     <path name="speaker-custom">
+        <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
+    </path>
+
+    <path name="bus-speaker">
+        <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
     </path>
 
    <path name="sidetone-iir">
diff --git a/configs/msmsteppe_au/mixer_paths_custom.xml b/configs/msmsteppe_au/mixer_paths_custom.xml
index 3de6f0f..5175d9f 100644
--- a/configs/msmsteppe_au/mixer_paths_custom.xml
+++ b/configs/msmsteppe_au/mixer_paths_custom.xml
@@ -1131,6 +1131,27 @@
         <path name="compress-offload-playback9" />
     </path>
 
+    <!-- The following use cases are used for car streams  -->
+    <path name="media-playback">
+        <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+        <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="sys-notification-playback">
+        <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+        <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="nav-guidance-playback">
+        <ctl name="TERT_TDM_RX_1 Channels" value="One" />
+        <ctl name="TERT_TDM_RX_1 Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="phone-playback">
+        <ctl name="TERT_TDM_RX_2 Channels" value="One" />
+        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia10" value="1" />
+    </path>
+
     <path name="audio-record">
         <ctl name="TERT_TDM_TX_0 Channels" value="One" />
         <ctl name="MultiMedia1 Mixer TERT_TDM_TX_0" value="1" />
@@ -1618,9 +1639,15 @@
     </path>
 
     <path name="speaker-adp">
+        <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
     </path>
 
     <path name="speaker-custom">
+        <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
+    </path>
+
+    <path name="bus-speaker">
+        <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
     </path>
 
    <path name="sidetone-iir">
diff --git a/configs/msmsteppe_au/msmsteppe_au.mk b/configs/msmsteppe_au/msmsteppe_au.mk
index 51829bd..9fac734 100644
--- a/configs/msmsteppe_au/msmsteppe_au.mk
+++ b/configs/msmsteppe_au/msmsteppe_au.mk
@@ -62,23 +62,32 @@
 AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
 BOARD_SUPPORTS_QAHW := false
 AUDIO_FEATURE_ENABLED_RAS := true
-AUDIO_FEATURE_ENABLED_SND_MONITOR := true
+AUDIO_FEATURE_ENABLED_SND_MONITOR := false
 AUDIO_FEATURE_ENABLED_DLKM := true
 AUDIO_FEATURE_ENABLED_USB_BURST_MODE := false
 AUDIO_FEATURE_ENABLED_SVA_MULTI_STAGE := false
 AUDIO_FEATURE_ENABLED_BATTERY_LISTENER := false
 ##AUDIO_FEATURE_FLAGS
 
+AUDIO_FEATURE_ENABLED_AUTO_HAL := true
+AUDIO_FEATURE_ENABLED_EXT_HW_PLUGIN := true
+AUDIO_FEATURE_ENABLED_AUDIO_CONTROL_HAL := true
+ifneq ($(ENABLE_HYP),true)
+AUDIO_FEATURE_ENABLED_AUTO_AUDIOD := true
+endif
+AUDIO_FEATURE_ENABLED_FM_TUNER_EXT := true
+##AUTOMOTIVE_AUDIO_FEATURE_FLAGS
+
 ifneq ($(strip $(TARGET_USES_RRO)), true)
 #Audio Specific device overlays
-DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
+DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common_au/overlay
 endif
 
 #Automotive audio specific device overlays
-DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmsteppe_au/overlay
+DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/overlay
 
 PRODUCT_COPY_FILES += \
-    vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_output_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_output_policy.conf \
+    vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_io_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_io_policy.conf \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_effects.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.conf \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_effects.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/mixer_paths_adp.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_adp.xml \
@@ -97,7 +106,7 @@
     $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_policy_configuration.xml:$(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 \
+    $(TOPDIR)vendor/qcom/opensource/audio-hal/primary-hal/configs/common_au/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
     $(TOPDIR)frameworks/av/services/audiopolicy/config/a2dp_audio_policy_configuration.xml:$(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 \
@@ -109,6 +118,10 @@
 PRODUCT_COPY_FILES += \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/listen_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/listen_platform_info.xml
 
+#Audio HAL version
+PRODUCT_PROPERTY_OVERRIDES += \
+vendor.audio.hal.maj.version=3
+
 # Reduce client buffer size for fast audio output tracks
 PRODUCT_PROPERTY_OVERRIDES += \
     af.fast_track_multiplier=1
@@ -261,13 +274,20 @@
     android.hardware.audio.effect@4.0 \
     android.hardware.audio.effect@4.0-impl
 
+# for HIDL related audiocontrol packages
+PRODUCT_PACKAGES += \
+    vendor.qti.hardware.automotive.audiocontrol@1.0-service \
+    android.hardware.automotive.audiocontrol@1.0
+
+ifeq ($(ENABLE_HYP),true)
 PRODUCT_PROPERTY_OVERRIDES += \
-persist.audio.calfile0=/vendor/etc/acdbdata/adsp_avs_config.acdb\
-persist.audio.calfile1=/vendor/etc/acdbdata/ADP/Bluetooth_cal.acdb\
-persist.audio.calfile2=/vendor/etc/acdbdata/ADP/Codec_cal.acdb\
-persist.audio.calfile3=/vendor/etc/acdbdata/ADP/General_cal.acdb\
-persist.audio.calfile4=/vendor/etc/acdbdata/ADP/Global_cal.acdb\
-persist.audio.calfile5=/vendor/etc/acdbdata/ADP/Handset_cal.acdb\
-persist.audio.calfile6=/vendor/etc/acdbdata/ADP/Hdmi_cal.acdb\
-persist.audio.calfile7=/vendor/etc/acdbdata/ADP/Headset_cal.acdb\
-persist.audio.calfile8=/vendor/etc/acdbdata/ADP/Speaker_cal.acdb
+persist.vendor.audio.calfile0=/vendor/etc/acdbdata/adsp_avs_config.acdb\
+persist.vendor.audio.calfile1=/vendor/etc/acdbdata/ADP/Bluetooth_cal.acdb\
+persist.vendor.audio.calfile2=/vendor/etc/acdbdata/ADP/Codec_cal.acdb\
+persist.vendor.audio.calfile3=/vendor/etc/acdbdata/ADP/General_cal.acdb\
+persist.vendor.audio.calfile4=/vendor/etc/acdbdata/ADP/Global_cal.acdb\
+persist.vendor.audio.calfile5=/vendor/etc/acdbdata/ADP/Handset_cal.acdb\
+persist.vendor.audio.calfile6=/vendor/etc/acdbdata/ADP/Hdmi_cal.acdb\
+persist.vendor.audio.calfile7=/vendor/etc/acdbdata/ADP/Headset_cal.acdb\
+persist.vendor.audio.calfile8=/vendor/etc/acdbdata/ADP/Speaker_cal.acdb
+endif
diff --git a/configs/sdm660/audio_policy_configuration.xml b/configs/sdm660/audio_policy_configuration.xml
index 662764f..5ec3c56 100644
--- a/configs/sdm660/audio_policy_configuration.xml
+++ b/configs/sdm660/audio_policy_configuration.xml
@@ -163,6 +163,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -290,16 +295,8 @@
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
                 </devicePort>
                 <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
 
             </devicePorts>
@@ -337,6 +334,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/sdm660/sdm660.mk b/configs/sdm660/sdm660.mk
index 5695851..b598a2c 100644
--- a/configs/sdm660/sdm660.mk
+++ b/configs/sdm660/sdm660.mk
@@ -106,7 +106,9 @@
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/sound_trigger_mixer_paths_wcd9340.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_wcd9340.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/sound_trigger_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_platform_info.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
-    vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+    vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
+    frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+    frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
 
 #XML Audio configuration files
 ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -261,6 +263,10 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.volume.headset.gain.depcal=true
 
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
 #add dynamic feature flags here
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.feature.a2dp_offload.enable=true \
diff --git a/configs/sdm710/audio_policy_configuration.xml b/configs/sdm710/audio_policy_configuration.xml
index a7f4869..3f17b95 100644
--- a/configs/sdm710/audio_policy_configuration.xml
+++ b/configs/sdm710/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -299,16 +304,8 @@
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
                 </devicePort>
                 <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
 
             </devicePorts>
@@ -346,6 +343,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index fb01728..5cb6a5a 100644
--- a/configs/sdm710/sdm710.mk
+++ b/configs/sdm710/sdm710.mk
@@ -170,7 +170,9 @@
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_wcd9340.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_wcd9340.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
-    vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+    vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
+    frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+    frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
 
 #XML Audio configuration files
 ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -360,6 +362,10 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.audio.fluence.voicecomm=true
 
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
 #add dynamic feature flags here
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.feature.a2dp_offload.enable=true \
diff --git a/configs/sdm845/audio_policy_configuration.xml b/configs/sdm845/audio_policy_configuration.xml
index 76f5236..65b503c 100644
--- a/configs/sdm845/audio_policy_configuration.xml
+++ b/configs/sdm845/audio_policy_configuration.xml
@@ -161,12 +161,16 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
                 </mixPort>
-
                 <mixPort name="primary input" role="sink" maxOpenCount="2" maxActiveCount="2">
                     <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"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="record_24" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
@@ -294,16 +298,8 @@
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
                 </devicePort>
                 <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
 
             </devicePorts>
@@ -341,6 +337,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="record_24"
                        sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
                 <route type="mix" sink="mmap_no_irq_in"
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index c3c3578..77da76c 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -124,7 +124,9 @@
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/sound_trigger_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_platform_info.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/audio_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_platform_info.xml \
-    vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+    vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
+    frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+    frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
 
 #XML Audio configuration files
 ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -309,6 +311,11 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.audio.fluence.voicecomm=true
 
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
+
 #add dynamic feature flags here
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.feature.a2dp_offload.enable=true \
diff --git a/configs/trinket/audio_platform_info.xml b/configs/trinket/audio_platform_info.xml
index ff03dd2..8aec518 100644
--- a/configs/trinket/audio_platform_info.xml
+++ b/configs/trinket/audio_platform_info.xml
@@ -329,13 +329,13 @@
                         <mic_info mic_device_id="builtin_mic_4"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                     </snd_dev>
-                    <snd_dev in_snd_device="SND_DEVICE_IN_HANDSET_STEREO_DMIC">
+                    <snd_dev in_snd_device="SND_DEVICE_IN_HANDSET_DMIC_STEREO">
                         <mic_info mic_device_id="builtin_mic_1"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                         <mic_info mic_device_id="builtin_mic_2"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                     </snd_dev>
-                    <snd_dev in_snd_device="SND_DEVICE_IN_SPEAKER_STEREO_DMIC">
+                    <snd_dev in_snd_device="SND_DEVICE_IN_SPEAKER_DMIC_STEREO">
                         <mic_info mic_device_id="builtin_mic_1"
                             channel_mapping="AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED"/>
                         <mic_info mic_device_id="builtin_mic_2"
diff --git a/configs/trinket/audio_policy_configuration.xml b/configs/trinket/audio_policy_configuration.xml
index 8015afa..a5d7f88 100644
--- a/configs/trinket/audio_policy_configuration.xml
+++ b/configs/trinket/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
+                <mixPort name="voip_tx" role="sink"
+                         flags="AUDIO_INPUT_FLAG_VOIP_TX">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
                 <mixPort name="incall_music_uplink" role="source"
                         flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -263,20 +268,17 @@
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
-                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
-SP">
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TWSP">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
@@ -358,6 +360,8 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+                <route type="mix" sink="voip_tx"
+                       sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="usb_surround_sound"
diff --git a/configs/trinket/trinket.mk b/configs/trinket/trinket.mk
index 5176889..56ef3a8 100644
--- a/configs/trinket/trinket.mk
+++ b/configs/trinket/trinket.mk
@@ -30,8 +30,8 @@
 MM_AUDIO_ENABLED_SAFX := true
 AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS := false
 AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
-AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
-AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
+AUDIO_FEATURE_ENABLED_USB_TUNNEL := true
+AUDIO_FEATURE_ENABLED_A2DP_OFFLOAD := true
 AUDIO_FEATURE_ENABLED_3D_AUDIO := false
 DOLBY_ENABLE := false
 endif
@@ -40,7 +40,7 @@
 AUDIO_FEATURE_ENABLED_DLKM := true
 BOARD_SUPPORTS_SOUND_TRIGGER := true
 AUDIO_FEATURE_ENABLED_INSTANCE_ID := true
-AUDIO_USE_LL_AS_PRIMARY_OUTPUT := true
+AUDIO_USE_DEEP_AS_PRIMARY_OUTPUT := false
 AUDIO_FEATURE_ENABLED_VBAT_MONITOR := true
 AUDIO_FEATURE_ENABLED_ANC_HEADSET := true
 AUDIO_FEATURE_ENABLED_CUSTOMSTEREO := true
@@ -71,7 +71,7 @@
 ##AUDIO_FEATURE_FLAGS
 
 #Audio Specific device overlays
-DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/common/overlay
+DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
 
 PRODUCT_COPY_FILES += \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/trinket/audio_io_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_io_policy.conf \
@@ -201,6 +201,18 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.vendor.bt.a2dp_offload_cap=sbc-aptx-aptxtws-aptxhd-aac
 
+# A2DP offload support
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.bluetooth.a2dp_offload.supported=true
+
+# Disable A2DP offload
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.bluetooth.a2dp_offload.disabled=false
+
+# A2DP offload DSP supported encoder list
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.bluetooth.a2dp_offload.cap=sbc-aac-aptx-aptxhd-ldac
+
 #enable software decoders for ALAC and APE
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.use.sw.alac.decoder=true
@@ -239,6 +251,14 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.vendor.bt.aac_frm_ctl.enabled=true
 
+#enable headset calibration
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.volume.headset.gain.depcal=true
+
+#enable dualmic fluence for voice communication
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.audio.fluence.voicecomm=true
+
 #add dynamic feature flags here
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.feature.a2dp_offload.enable=true \
diff --git a/hal/Android.mk b/hal/Android.mk
index 1a0c2e2..6e47039 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -9,7 +9,7 @@
 
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter msm8974 msm8226 msm8084 msm8610 apq8084 msm8994 msm8992 msm8996 msm8998 apq8098_latv sdm845 sdm710 qcs605 msmnile kona sdm660 msm8937 $(MSMSTEPPE) $(TRINKET) lito atoll,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8610 apq8084 msm8994 msm8992 msm8996 msm8998 apq8098_latv sdm845 sdm710 qcs605 sdmshrike msmnile kona sdm660 msm8937 $(MSMSTEPPE) $(TRINKET) lito atoll,$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM = msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -65,7 +65,7 @@
 ifneq ($(filter qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_CFLAGS := -DPLATFORM_QCS605
 endif
-ifneq ($(filter msmnile,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msmnile sdmshrike,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_CFLAGS := -DPLATFORM_MSMNILE
   LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
   LOCAL_CFLAGS += -DINCALL_MUSIC_ENABLED
@@ -87,6 +87,7 @@
 ifneq ($(filter lito,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_CFLAGS := -DPLATFORM_LITO
   LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
+  LOCAL_CFLAGS += -DINCALL_STEREO_CAPTURE_ENABLED
 endif
 ifneq ($(filter atoll,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_CFLAGS := -DPLATFORM_ATOLL
@@ -354,6 +355,10 @@
 LOCAL_SHARED_LIBRARIES += libbase libhidlbase libhwbinder libutils android.hardware.power@1.2 liblog
 LOCAL_SRC_FILES += audio_perf.cpp
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FM_TUNER_EXT)),true)
+    LOCAL_CFLAGS += -DFM_TUNER_EXT_ENABLED
+endif
+
 LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
 
 LOCAL_MODULE_RELATIVE_PATH := hw
diff --git a/hal/audio_extn/Android.mk b/hal/audio_extn/Android.mk
index fee6977..2aba6e1 100644
--- a/hal/audio_extn/Android.mk
+++ b/hal/audio_extn/Android.mk
@@ -63,7 +63,7 @@
 
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 qcs605 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 qcs605 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -121,7 +121,7 @@
 
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -182,7 +182,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
 endif
@@ -234,7 +234,7 @@
 #--------------------------------------------
 include $(CLEAR_VARS)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
 endif
@@ -289,7 +289,7 @@
 
 include $(CLEAR_VARS)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
 endif
@@ -350,7 +350,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -410,7 +410,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -469,7 +469,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -528,7 +528,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -590,7 +590,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -660,7 +660,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM := msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -719,7 +719,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona sdm660 msm8937 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona sdm660 msm8937 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM = msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -776,7 +776,7 @@
 PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter sdm845 sdm710 msmnile kona sdm660 msm8937 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 sdmshrike msmnile kona sdm660 msm8937 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM = msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index f9f33d1..8e65471 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -475,7 +475,7 @@
     int num_devices = 0, pcm_device_id = -1, i = 0, ret = 0;
     snd_device_t new_snd_devices[SND_DEVICE_OUT_END] = {0};
     struct audio_backend_cfg backend_cfg = {0};
-    uint32_t feature_id = 0;
+    uint32_t feature_id = 0, idx = 0;
 
     switch(usecase->type) {
     case PCM_PLAYBACK:
@@ -520,7 +520,7 @@
      * if features like dual_mono is enabled and overrides the default(i.e. 0).
      */
     info.id = feature_id;
-    info.usecase_id = usecase->id;
+    info.usecase_id[0] = usecase->id;
     for (i = 0, ret = 0; i < num_devices; i++) {
         info.snd_device = new_snd_devices[i];
         platform_get_codec_backend_cfg(adev, info.snd_device, &backend_cfg);
@@ -533,7 +533,7 @@
             info.op_channels = audio_channel_count_from_in_mask(
                                    usecase->stream.in->channel_mask);
         }
-        params = platform_get_custom_mtmx_params(adev->platform, &info);
+        params = platform_get_custom_mtmx_params(adev->platform, &info, &idx);
         if (params) {
             if (enable)
                 ret = update_custom_mtmx_coefficients_v2(adev, params,
@@ -678,7 +678,8 @@
                                            struct audio_custom_mtmx_in_params *in_params,
                                            int pcm_device_id,
                                            usecase_type_t type,
-                                           bool enable)
+                                           bool enable,
+                                           uint32_t idx)
 {
     struct mixer_ctl *ctl = NULL;
     char mixer_ctl_name[128] = {0};
@@ -692,13 +693,13 @@
           __func__, pinfo->ip_channels, pinfo->op_channels, pcm_device_id,
           type, enable);
 
-    if (!strcmp(pinfo->fe_name, "")) {
+    if (pinfo->fe_id[idx] == 0) {
         ALOGE("%s: Error. no front end defined", __func__);
         return -EINVAL;
     }
 
-    strlcpy(mixer_name_prefix, pinfo->fe_name, sizeof(mixer_name_prefix));
-
+    snprintf(mixer_name_prefix, sizeof(mixer_name_prefix), "%s%d",
+             "MultiMedia", pinfo->fe_id[idx]);
     /*
      * Enable/Disable channel mixer.
      * If enable, use params and in_params to configure mixer.
@@ -840,7 +841,7 @@
     struct audio_custom_mtmx_in_params_info in_info = {0};
     struct audio_custom_mtmx_in_params *in_params = NULL;
     int pcm_device_id = -1, ret = 0;
-    uint32_t feature_id = 0;
+    uint32_t feature_id = 0, idx = 0;
 
     switch(usecase->type) {
     case PCM_CAPTURE:
@@ -862,26 +863,26 @@
 
     ALOGD("%s: snd device %d", __func__, info.snd_device);
     info.id = feature_id;
-    info.usecase_id = usecase->id;
+    info.usecase_id[0] = usecase->id;
     info.op_channels = audio_channel_count_from_in_mask(
                                 usecase->stream.in->channel_mask);
 
-    in_info.usecase_id = info.usecase_id;
+    in_info.usecase_id[0] = info.usecase_id[0];
     in_info.op_channels = info.op_channels;
     in_params = platform_get_custom_mtmx_in_params(adev->platform, &in_info);
     if (!in_params) {
         ALOGE("%s: Could not get in params for usecase %d, channels %d",
-               __func__, in_info.usecase_id, in_info.op_channels);
+               __func__, in_info.usecase_id[0], in_info.op_channels);
         return;
     }
 
     info.ip_channels = in_params->ip_channels;
     ALOGD("%s: ip channels %d, op channels %d", __func__, info.ip_channels, info.op_channels);
 
-    params = platform_get_custom_mtmx_params(adev->platform, &info);
+    params = platform_get_custom_mtmx_params(adev->platform, &info, &idx);
     if (params) {
         ret = update_custom_mtmx_coefficients_v1(adev, params, in_params,
-                             pcm_device_id, usecase->type, enable);
+                             pcm_device_id, usecase->type, enable, idx);
         if (ret < 0)
             ALOGE("%s: error updating mtmx coeffs err:%d", __func__, ret);
     }
@@ -900,12 +901,12 @@
         return snd_device;
     }
 
-    in_info.usecase_id = usecase->id;
+    in_info.usecase_id[0] = usecase->id;
     in_info.op_channels = channel_count;
     in_params = platform_get_custom_mtmx_in_params(adev->platform, &in_info);
     if (!in_params) {
         ALOGE("%s: Could not get in params for usecase %d, channels %d",
-               __func__, in_info.usecase_id, in_info.op_channels);
+               __func__, in_info.usecase_id[0], in_info.op_channels);
         return snd_device;
     }
 
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 18bfc2a..581b802 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -859,6 +859,7 @@
 int b64encode(uint8_t *inp, int ilen, char* outp);
 int read_line_from_file(const char *path, char *buf, size_t count);
 int audio_extn_utils_get_codec_version(const char *snd_card_name, int card_num, char *codec_version);
+int audio_extn_utils_get_codec_variant(int card_num, char *codec_variant);
 audio_format_t alsa_format_to_hal(uint32_t alsa_format);
 uint32_t hal_format_to_alsa(audio_format_t hal_format);
 audio_format_t pcm_format_to_hal(uint32_t pcm_format);
@@ -1275,8 +1276,6 @@
 #ifndef AUDIO_EXTN_AUTO_HAL_ENABLED
 #define audio_extn_auto_hal_init(adev)                (0)
 #define audio_extn_auto_hal_deinit()                  (0)
-#define audio_extn_auto_hal_enable_hostless()         (0)
-#define audio_extn_auto_hal_disable_hostless()        (0)
 #define audio_extn_auto_hal_create_audio_patch(dev, num_sources,\
     sources, num_sinks, sinks, handle) (0)
 #define audio_extn_auto_hal_release_audio_patch(dev, handle) (0)
@@ -1288,10 +1287,12 @@
 #define audio_extn_auto_hal_set_audio_port_config(dev, config) (0)
 #define audio_extn_auto_hal_set_parameters(adev, parms) (0)
 #else
+#define AUDIO_OUTPUT_FLAG_MEDIA 0x100000
+#define AUDIO_OUTPUT_FLAG_SYS_NOTIFICATION 0x200000
+#define AUDIO_OUTPUT_FLAG_NAV_GUIDANCE 0x400000
+#define AUDIO_OUTPUT_FLAG_PHONE 0x800000
 int32_t audio_extn_auto_hal_init(struct audio_device *adev);
 void audio_extn_auto_hal_deinit(void);
-int32_t audio_extn_auto_hal_enable_hostless(void);
-void audio_extn_auto_hal_disable_hostless(void);
 int audio_extn_auto_hal_create_audio_patch(struct audio_hw_device *dev,
                                 unsigned int num_sources,
                                 const struct audio_port_config *sources,
diff --git a/hal/audio_extn/auto_hal.c b/hal/audio_extn/auto_hal.c
index ad5e331..304c117 100644
--- a/hal/audio_extn/auto_hal.c
+++ b/hal/audio_extn/auto_hal.c
@@ -47,15 +47,9 @@
 
 #ifdef AUDIO_EXTN_AUTO_HAL_ENABLED
 
-struct hostless_config {
-    struct pcm *pcm_tx;
-    struct pcm *pcm_rx;
-};
-
 typedef struct auto_hal_module {
     struct audio_device *adev;
     card_status_t card_status;
-    struct hostless_config hostless;
 } auto_hal_module_t;
 
 /* Auto hal module struct */
@@ -71,102 +65,17 @@
     USECASE_AUDIO_PLAYBACK_PHONE,
 };
 
-/* Note: Due to ADP H/W design, SoC TERT/SEC TDM CLK and FSYNC lines are
- * both connected with CODEC and a single master is needed to provide
- * consistent CLK and FSYNC to slaves, hence configuring SoC TERT TDM as
- * single master and bring up a dummy hostless from TERT to SEC to ensure
- * both slave SoC SEC TDM and CODEC are driven upon system boot. */
-int32_t audio_extn_auto_hal_enable_hostless(void)
+static struct audio_patch_record *get_patch_from_list(struct audio_device *adev,
+                                                    audio_patch_handle_t patch_id)
 {
-    int32_t ret = 0;
-    char mixer_path[MIXER_PATH_MAX_LENGTH];
-
-    ALOGD("%s: Enable TERT -> SEC Hostless", __func__);
-
-    if (auto_hal == NULL) {
-        ALOGE("%s: Invalid device", __func__);
-        return -EINVAL;
+    struct audio_patch_record *patch;
+    struct listnode *node;
+    list_for_each(node, &adev->audio_patch_record_list) {
+        patch = node_to_item(node, struct audio_patch_record, list);
+        if (patch->handle == patch_id)
+            return patch;
     }
-
-    strlcpy(mixer_path, "dummy-hostless", MIXER_PATH_MAX_LENGTH);
-    ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path);
-    if (audio_route_apply_and_update_path(auto_hal->adev->audio_route,
-            mixer_path)) {
-        ALOGD("%s: %s not supported, continue", __func__, mixer_path);
-        return ret;
-    }
-
-    /* TERT TDM TX 7 HOSTLESS to SEC TDM RX 7 HOSTLESS */
-    int pcm_dev_rx = 48, pcm_dev_tx = 49;
-    struct pcm_config pcm_config_lb = {
-        .channels = 1,
-        .rate = 48000,
-        .period_size = 240,
-        .period_count = 2,
-        .format = PCM_FORMAT_S16_LE,
-        .start_threshold = 0,
-        .stop_threshold = INT_MAX,
-        .avail_min = 0,
-    };
-
-    auto_hal->hostless.pcm_tx = pcm_open(auto_hal->adev->snd_card,
-                                   pcm_dev_tx,
-                                   PCM_IN, &pcm_config_lb);
-    if (auto_hal->hostless.pcm_tx &&
-        !pcm_is_ready(auto_hal->hostless.pcm_tx)) {
-        ALOGE("%s: %s", __func__,
-            pcm_get_error(auto_hal->hostless.pcm_tx));
-        ret = -EIO;
-        goto error;
-    }
-    auto_hal->hostless.pcm_rx = pcm_open(auto_hal->adev->snd_card,
-                                   pcm_dev_rx,
-                                   PCM_OUT, &pcm_config_lb);
-    if (auto_hal->hostless.pcm_rx &&
-        !pcm_is_ready(auto_hal->hostless.pcm_rx)) {
-        ALOGE("%s: %s", __func__,
-            pcm_get_error(auto_hal->hostless.pcm_rx));
-        ret = -EIO;
-        goto error;
-    }
-
-    if (pcm_start(auto_hal->hostless.pcm_tx) < 0) {
-        ALOGE("%s: pcm start for pcm tx failed", __func__);
-        ret = -EIO;
-        goto error;
-    }
-    if (pcm_start(auto_hal->hostless.pcm_rx) < 0) {
-        ALOGE("%s: pcm start for pcm rx failed", __func__);
-        ret = -EIO;
-        goto error;
-    }
-    return ret;
-
-error:
-    if (auto_hal->hostless.pcm_rx)
-        pcm_close(auto_hal->hostless.pcm_rx);
-    if (auto_hal->hostless.pcm_tx)
-        pcm_close(auto_hal->hostless.pcm_tx);
-    return ret;
-}
-
-void audio_extn_auto_hal_disable_hostless(void)
-{
-    ALOGD("%s: Disable TERT -> SEC Hostless", __func__);
-
-    if (auto_hal == NULL) {
-        ALOGE("%s: Invalid device", __func__);
-        return;
-    }
-
-    if (auto_hal->hostless.pcm_tx) {
-        pcm_close(auto_hal->hostless.pcm_tx);
-        auto_hal->hostless.pcm_tx = NULL;
-    }
-    if (auto_hal->hostless.pcm_rx) {
-        pcm_close(auto_hal->hostless.pcm_rx);
-        auto_hal->hostless.pcm_rx = NULL;
-    }
+    return NULL;
 }
 
 #define MAX_SOURCE_PORTS_PER_PATCH 1
@@ -184,6 +93,11 @@
     char *str = NULL;
     struct str_parms *parms = NULL;
     char *address = NULL;
+    struct audio_usecase *uc_info = NULL;
+    struct audio_patch_record *patch_record = NULL;
+    audio_usecase_t usecase = USECASE_INVALID;
+    audio_io_handle_t input_io_handle = AUDIO_IO_HANDLE_NONE;
+    audio_io_handle_t output_io_handle = AUDIO_IO_HANDLE_NONE;
 
     ALOGV("%s: enter", __func__);
 
@@ -236,6 +150,7 @@
         pthread_mutex_unlock(&adev->lock);
         if(ret)
             return ret;
+        input_io_handle = sinks->ext.mix.handle;
 
         if (strcmp(sources->ext.device.address, "") != 0) {
             address = audio_device_address_to_parameter(
@@ -248,7 +163,7 @@
         if (!parms) {
             ALOGE("%s: failed to allocate mem for parms", __func__);
             ret = -ENOMEM;
-            goto error;
+            goto exit;
         }
         str_parms_add_int(parms, AUDIO_PARAMETER_STREAM_ROUTING,
                         (int)sources->ext.device.type);
@@ -269,6 +184,7 @@
         pthread_mutex_unlock(&adev->lock);
         if(ret)
             return ret;
+        output_io_handle = sources->ext.mix.handle;
 
         if (strcmp(sinks->ext.device.address, "") != 0) {
             address = audio_device_address_to_parameter(
@@ -281,20 +197,93 @@
         if (!parms) {
             ALOGE("%s: failed to allocate mem for parms", __func__);
             ret = -ENOMEM;
-            goto error;
+            goto exit;
         }
         str_parms_add_int(parms, AUDIO_PARAMETER_STREAM_ROUTING,
                         (int)sinks->ext.device.type);
         str = str_parms_to_str(parms);
         out_ctxt->output->stream.common.set_parameters(
                         (struct audio_stream *)out_ctxt->output, str);
+    } else if ((sources->type == AUDIO_PORT_TYPE_DEVICE) &&
+            (sinks->type == AUDIO_PORT_TYPE_DEVICE)) {
+        /* allocate use case and call to plugin driver*/
+        uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+        if (!uc_info) {
+            ALOGE("%s fail to allocate uc_info", __func__);
+            return -ENOMEM;
+        }
+        /* TODO - add sink type check and printout for non speaker sink */
+        switch(sources->ext.device.type) {
+#ifdef FM_TUNER_EXT_ENABLED
+            case AUDIO_DEVICE_IN_FM_TUNER:
+                ALOGV("Creating audio patch for external FM tuner");
+                uc_info->id = USECASE_AUDIO_FM_TUNER_EXT;
+                uc_info->type = PCM_PASSTHROUGH;
+                uc_info->devices = AUDIO_DEVICE_IN_FM_TUNER;
+                uc_info->in_snd_device = SND_DEVICE_IN_CAPTURE_FM;
+                uc_info->out_snd_device = SND_DEVICE_OUT_BUS_MEDIA;
+                break;
+#endif
+            default:
+                ALOGE("%s: Unsupported audio source type %x", __func__,
+                            sources->ext.device.type);
+                goto error;
+        }
+
+        ALOGD("%s: Starting ext hw plugin use case (%d) in_snd_device (%d) out_snd_device (%d)",
+              __func__, uc_info->id, uc_info->in_snd_device, uc_info->out_snd_device);
+
+        ret = audio_extn_ext_hw_plugin_usecase_start(adev->ext_hw_plugin, uc_info);
+        if (ret) {
+            ALOGE("%s: failed to start ext hw plugin use case (%d)",
+                __func__, uc_info->id);
+            goto error;
+        }
+        /* TODO: apply audio port gain to codec if applicable */
+        usecase = uc_info->id;
+        pthread_mutex_lock(&adev->lock);
+        list_add_tail(&adev->usecase_list, &uc_info->list);
+        pthread_mutex_unlock(&adev->lock);
     } else {
-        ALOGW("%s: create device -> device audio patch", __func__);
+        ALOGW("%s: audio patch not supported",__func__);
+        return -EINVAL;
     }
 
+    /* patch created success, add to patch record list */
+    patch_record = (struct audio_patch_record *)calloc(1,
+                    sizeof(struct audio_patch_record));
+    if (!patch_record) {
+        ALOGE("%s fail to allocate patch_record", __func__);
+        ret = -ENOMEM;
+        if (uc_info)
+            list_remove(&uc_info->list);
+        goto error;
+    }
+
+    pthread_mutex_lock(&adev->lock);
+    adev->audio_patch_index++;
+    patch_record->handle = adev->audio_patch_index;
+    patch_record->usecase = usecase;
+    patch_record->input_io_handle = input_io_handle;
+    patch_record->output_io_handle = output_io_handle;
+    memcpy((void *)&patch_record->source, (void *)sources,
+        sizeof(struct audio_port_config));
+    memcpy((void *)&patch_record->sink, (void *)sinks,
+        sizeof(struct audio_port_config));
+    list_add_tail(&adev->audio_patch_record_list, &patch_record->list);
+    pthread_mutex_unlock(&adev->lock);
+
+    *handle = patch_record->handle;
+    goto exit;
+
 error:
+    if(uc_info)
+        free(uc_info);
+exit:
     if (parms)
         str_parms_destroy(parms);
+    if (str)
+        free(str);
     if (address)
         free(address);
     ALOGV("%s: exit: handle 0x%x", __func__, *handle);
@@ -305,6 +294,13 @@
                                 audio_patch_handle_t handle)
 {
     int ret = 0;
+    struct audio_device *adev = (struct audio_device *)dev;
+    struct audio_usecase *uc_info = NULL;
+    struct audio_patch_record *patch_record = NULL;
+    streams_input_ctxt_t *in_ctxt = NULL;
+    streams_output_ctxt_t *out_ctxt = NULL;
+    char *str = NULL;
+    struct str_parms *parms = NULL;
 
     ALOGV("%s: enter: handle 0x%x", __func__, handle);
 
@@ -313,10 +309,92 @@
         return -EINVAL;
     }
 
-    if (handle != AUDIO_PATCH_HANDLE_NONE) {
-        ALOGW("%s: release device -> device audio patch", __func__);
+    if (handle == AUDIO_PATCH_HANDLE_NONE) {
+        ALOGW("%s: null audio patch handle", __func__);
+        return -EINVAL;
     }
 
+    /* get the patch record from handle */
+    pthread_mutex_lock(&adev->lock);
+    patch_record = get_patch_from_list(adev, handle);
+    if(!patch_record) {
+        ALOGE("%s: failed to find the patch record with handle (%d) in the list",
+                __func__, handle);
+        ret = -EINVAL;
+    }
+    pthread_mutex_unlock(&adev->lock);
+    if(ret)
+        goto exit;
+
+    if (patch_record->input_io_handle) {
+        pthread_mutex_lock(&adev->lock);
+        in_ctxt = in_get_stream(adev, patch_record->input_io_handle);
+        if (!in_ctxt) {
+            ALOGE("%s, Could not find input stream", __func__);
+            ret = -EINVAL;
+        }
+        pthread_mutex_unlock(&adev->lock);
+        if(ret)
+            goto exit;
+
+        parms = str_parms_create();
+        str_parms_add_int(parms, AUDIO_PARAMETER_STREAM_ROUTING, 0);
+        str = str_parms_to_str(parms);
+        in_ctxt->input->stream.common.set_parameters(
+                        (struct audio_stream *)in_ctxt->input, str);
+    }
+
+    if (patch_record->output_io_handle) {
+        pthread_mutex_lock(&adev->lock);
+        out_ctxt = out_get_stream(adev, patch_record->output_io_handle);
+        if (!out_ctxt) {
+            ALOGE("%s, Could not find output stream", __func__);
+            ret = -EINVAL;
+        }
+        pthread_mutex_unlock(&adev->lock);
+        if(ret)
+            goto exit;
+
+        parms = str_parms_create();
+        str_parms_add_int(parms, AUDIO_PARAMETER_STREAM_ROUTING, 0);
+        str = str_parms_to_str(parms);
+        out_ctxt->output->stream.common.set_parameters(
+                        (struct audio_stream *)out_ctxt->output, str);
+    }
+
+    if (patch_record->usecase != USECASE_INVALID) {
+        pthread_mutex_lock(&adev->lock);
+        uc_info = get_usecase_from_list(adev, patch_record->usecase);
+        if (!uc_info) {
+            ALOGE("%s: failed to find the usecase (%d)",
+                    __func__, patch_record->usecase);
+            ret = -EINVAL;
+        } else {
+            /* call to plugin to stop the usecase */
+            ret = audio_extn_ext_hw_plugin_usecase_stop(adev->ext_hw_plugin, uc_info);
+            if (ret) {
+                ALOGE("%s: failed to stop ext hw plugin use case (%d)",
+                        __func__, uc_info->id);
+            }
+
+            /* remove usecase from list and free it */
+            list_remove(&uc_info->list);
+            free(uc_info);
+        }
+        pthread_mutex_unlock(&adev->lock);
+    }
+
+    /* remove the patch record from list and free it */
+    pthread_mutex_lock(&adev->lock);
+    list_remove(&patch_record->list);
+    pthread_mutex_unlock(&adev->lock);
+    free(patch_record);
+    if (parms)
+        str_parms_destroy(parms);
+    if (str)
+        free(str);
+
+exit:
     ALOGV("%s: exit", __func__);
     return ret;
 }
@@ -368,11 +446,16 @@
             ret = -EINVAL;
             goto error;
         }
+        if (out->flags == AUDIO_OUTPUT_FLAG_NONE ||
+            out->flags == AUDIO_OUTPUT_FLAG_PRIMARY)
+            out->flags |= AUDIO_OUTPUT_FLAG_MEDIA;
         break;
     case CAR_AUDIO_STREAM_SYS_NOTIFICATION:
         /* sys notification bus stream shares pcm device with low-latency */
         out->usecase = USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION;
         out->config = pcm_config_low_latency;
+        if (out->flags == AUDIO_OUTPUT_FLAG_NONE)
+            out->flags |= AUDIO_OUTPUT_FLAG_SYS_NOTIFICATION;
         break;
     case CAR_AUDIO_STREAM_NAV_GUIDANCE:
         out->usecase = USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE;
@@ -384,10 +467,14 @@
             ret = -EINVAL;
             goto error;
         }
+        if (out->flags == AUDIO_OUTPUT_FLAG_NONE)
+            out->flags |= AUDIO_OUTPUT_FLAG_NAV_GUIDANCE;
         break;
     case CAR_AUDIO_STREAM_PHONE:
         out->usecase = USECASE_AUDIO_PLAYBACK_PHONE;
         out->config = pcm_config_low_latency;
+        if (out->flags == AUDIO_OUTPUT_FLAG_NONE)
+            out->flags |= AUDIO_OUTPUT_FLAG_PHONE;
         break;
     default:
         ALOGE("%s: Car audio stream %x not supported", __func__,
@@ -518,20 +605,21 @@
              *        to be part of port config upon audio patch creation. If not, need
              *        to create a list of audio port configs in adev context.
              */
-#if 0
             list_for_each(node, &adev->audio_patch_record_list) {
                 struct audio_patch_record *patch_record = node_to_item(node,
                                                     struct audio_patch_record,
                                                     list);
-                /* limit audio gain support for bus device only */
-                if (patch_record->sink.type == AUDIO_PORT_TYPE_DEVICE &&
+                /* limit audio gain support for device -> bus device patch */
+                if (patch_record->source.type == AUDIO_PORT_TYPE_DEVICE &&
+                    patch_record->sink.type == AUDIO_PORT_TYPE_DEVICE &&
                     patch_record->sink.role == AUDIO_PORT_ROLE_SINK &&
                     patch_record->sink.ext.device.type == AUDIO_DEVICE_OUT_BUS &&
                     patch_record->sink.ext.device.type == config->ext.device.type &&
                     strcmp(patch_record->sink.ext.device.address,
                         config->ext.device.address) == 0) {
-                    /* cache / update gain per audio patch sink */
-                    patch_record->sink.gain = config->gain;
+                    /* cache audio port configuration for sink */
+                    memcpy((void *)&patch_record->sink, (void *)config,
+                        sizeof(struct audio_port_config));
 
                     struct audio_usecase *uc_info = get_usecase_from_list(adev,
                                                         patch_record->usecase);
@@ -540,18 +628,17 @@
                             __func__, patch_record->usecase);
                         ret = -EINVAL;
                     } else {
-                        volume = config->gain->values[0];
+                        volume = config->gain.values[0];
                         /* linear interpolation from millibel to level */
                         int vol_level = lrint(((volume + (0 - MIN_VOLUME_VALUE_MB)) /
                                                (MAX_VOLUME_VALUE_MB - MIN_VOLUME_VALUE_MB)) * 40);
-                        ALOGV("%s: set volume to patch: %p", __func__,
+                        ALOGV("%s: set volume to patch %x", __func__,
                             patch_record->handle);
                         ret = audio_extn_ext_hw_plugin_set_audio_gain(adev,
                                 uc_info, vol_level);
                     }
                 }
             }
-#endif
             pthread_mutex_unlock(&adev->lock);
         } else if (config->role == AUDIO_PORT_ROLE_SOURCE) {
             // FIXME: handle input devices.
@@ -578,11 +665,9 @@
         ALOGV("%s: snd card status %s", __func__, snd_card_status);
         if (strstr(snd_card_status, "OFFLINE")) {
             auto_hal->card_status = CARD_STATUS_OFFLINE;
-            audio_extn_auto_hal_disable_hostless();
         }
         else if (strstr(snd_card_status, "ONLINE")) {
             auto_hal->card_status = CARD_STATUS_ONLINE;
-            audio_extn_auto_hal_enable_hostless();
         }
     }
 
diff --git a/hal/audio_extn/ext_hw_plugin.c b/hal/audio_extn/ext_hw_plugin.c
index 6b4a718..41faf03 100644
--- a/hal/audio_extn/ext_hw_plugin.c
+++ b/hal/audio_extn/ext_hw_plugin.c
@@ -77,85 +77,6 @@
 /* This can be defined in platform specific file or use compile flag */
 #define LIB_PLUGIN_DRIVER "libaudiohalplugin.so"
 
-/* Note: Due to ADP H/W design, SoC TERT/SEC TDM CLK and FSYNC lines are
- * both connected with CODEC and a single master is needed to provide
- * consistent CLK and FSYNC to slaves, hence configuring SoC TERT TDM as
- * single master and bring up a dummy hostless from TERT to SEC to ensure
- * both slave SoC SEC TDM and CODEC are driven upon system boot. */
-static void ext_hw_plugin_enable_adev_hostless(void *plugin)
-{
-    struct ext_hw_plugin_data *my_plugin =
-        (struct ext_hw_plugin_data *)plugin;
-    char mixer_path[MIXER_PATH_MAX_LENGTH];
-
-    ALOGI("%s: Enable TERT -> SEC Hostless", __func__);
-
-    strlcpy(mixer_path, "dummy-hostless", MIXER_PATH_MAX_LENGTH);
-    ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path);
-    if (audio_route_apply_and_update_path(my_plugin->adev->audio_route,
-            mixer_path)) {
-        ALOGE("%s: %s not supported, continue", __func__, mixer_path);
-        return;
-    }
-
-    /* TERT TDM TX 7 HOSTLESS to SEC TDM RX 7 HOSTLESS */
-    int pcm_dev_rx = 48, pcm_dev_tx = 49;
-    struct pcm_config pcm_config_lb = {
-        .channels = 1,
-        .rate = 48000,
-        .period_size = 240,
-        .period_count = 2,
-        .format = PCM_FORMAT_S16_LE,
-        .start_threshold = 0,
-        .stop_threshold = INT_MAX,
-        .avail_min = 0,
-    };
-
-    my_plugin->adev_hostless.pcm_tx = pcm_open(my_plugin->adev->snd_card,
-                                   pcm_dev_tx,
-                                   PCM_IN, &pcm_config_lb);
-    if (my_plugin->adev_hostless.pcm_tx &&
-        !pcm_is_ready(my_plugin->adev_hostless.pcm_tx)) {
-        ALOGE("%s: %s", __func__,
-            pcm_get_error(my_plugin->adev_hostless.pcm_tx));
-        return;
-    }
-    my_plugin->adev_hostless.pcm_rx = pcm_open(my_plugin->adev->snd_card,
-                                   pcm_dev_rx,
-                                   PCM_OUT, &pcm_config_lb);
-    if (my_plugin->adev_hostless.pcm_rx &&
-        !pcm_is_ready(my_plugin->adev_hostless.pcm_rx)) {
-        ALOGE("%s: %s", __func__,
-            pcm_get_error(my_plugin->adev_hostless.pcm_rx));
-        return;
-    }
-
-    if (pcm_start(my_plugin->adev_hostless.pcm_tx) < 0) {
-        ALOGE("%s: pcm start for pcm tx failed", __func__);
-        return;
-    }
-    if (pcm_start(my_plugin->adev_hostless.pcm_rx) < 0) {
-        ALOGE("%s: pcm start for pcm rx failed", __func__);
-        return;
-    }
-}
-
-static void ext_hw_plugin_disable_adev_hostless(void *plugin)
-{
-    struct ext_hw_plugin_data *my_plugin = (struct ext_hw_plugin_data *)plugin;
-
-    ALOGI("%s: Disable TERT -> SEC Hostless", __func__);
-
-    if (my_plugin->adev_hostless.pcm_tx) {
-        pcm_close(my_plugin->adev_hostless.pcm_tx);
-        my_plugin->adev_hostless.pcm_tx = NULL;
-    }
-    if (my_plugin->adev_hostless.pcm_rx) {
-        pcm_close(my_plugin->adev_hostless.pcm_rx);
-        my_plugin->adev_hostless.pcm_rx = NULL;
-    }
-}
-
 void* ext_hw_plugin_init(struct audio_device *adev, ext_hw_plugin_init_config_t init_config)
 {
     int32_t ret = 0;
@@ -170,7 +91,6 @@
 
     my_plugin->adev = adev;
     fp_audio_route_apply_and_update_path = init_config.fp_audio_route_apply_and_update_path;
-    (void)audio_extn_auto_hal_enable_hostless();
 
     my_plugin->plugin_handle = dlopen(LIB_PLUGIN_DRIVER, RTLD_NOW);
     if (my_plugin->plugin_handle == NULL) {
@@ -209,7 +129,6 @@
             goto plugin_init_fail;
         }
     }
-    ext_hw_plugin_enable_adev_hostless(my_plugin);
     my_plugin->mic_mute = false;
     return my_plugin;
 
@@ -229,7 +148,6 @@
         ALOGE("[%s] NULL plugin pointer",__func__);
         return -EINVAL;
     }
-    ext_hw_plugin_disable_adev_hostless(my_plugin);
     if (my_plugin->audio_hal_plugin_deinit) {
         ret = my_plugin->audio_hal_plugin_deinit();
         if (ret) {
@@ -240,8 +158,6 @@
     if(my_plugin->plugin_handle != NULL)
         dlclose(my_plugin->plugin_handle);
 
-    audio_extn_auto_hal_disable_hostless();
-
     free(my_plugin);
     return ret;
 }
@@ -289,6 +205,9 @@
     case USECASE_AUDIO_PLAYBACK_PHONE:
         *plugin_usecase = AUDIO_HAL_PLUGIN_USECASE_PHONE_PLAYBACK;
         break;
+    case USECASE_AUDIO_FM_TUNER_EXT:
+       *plugin_usecase = AUDIO_HAL_PLUGIN_USECASE_FM_TUNER;
+        break;
     default:
         ret = -EINVAL;
     }
@@ -331,7 +250,8 @@
         }
 
         if (((usecase->type == PCM_CAPTURE) || (usecase->type == VOICE_CALL) ||
-                (usecase->type == VOIP_CALL) || (usecase->type == PCM_HFP_CALL)) &&
+              (usecase->type == VOIP_CALL) || (usecase->type == PCM_HFP_CALL) ||
+              (usecase->type == PCM_PASSTHROUGH)) &&
             (usecase->in_snd_device != SND_DEVICE_NONE)) {
             codec_enable.snd_dev = usecase->in_snd_device;
             /* TODO - below should be related with in_snd_dev */
@@ -486,7 +406,8 @@
             my_plugin->out_snd_dev[codec_disable.usecase] = 0;
         }
         if (((usecase->type == PCM_CAPTURE) || (usecase->type == VOICE_CALL) ||
-                (usecase->type == VOIP_CALL) || (usecase->type == PCM_HFP_CALL)) &&
+             (usecase->type == VOIP_CALL) || (usecase->type == PCM_HFP_CALL) ||
+             (usecase->type == PCM_PASSTHROUGH)) &&
             (usecase->in_snd_device != SND_DEVICE_NONE)) {
             codec_disable.snd_dev = usecase->in_snd_device;
 
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index aa13c2b..a175b83 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -84,9 +84,9 @@
     AUDIO_EVENT_CAPTURE_STREAM_INACTIVE,
     AUDIO_EVENT_CAPTURE_STREAM_ACTIVE,
     AUDIO_EVENT_BATTERY_STATUS_CHANGED,
-    AUDIO_EVENT_SCREEN_STATUS_CHANGED,
     AUDIO_EVENT_GET_PARAM,
-    AUDIO_EVENT_UPDATE_ECHO_REF
+    AUDIO_EVENT_UPDATE_ECHO_REF,
+    AUDIO_EVENT_SCREEN_STATUS_CHANGED
 } audio_event_type_t;
 
 typedef enum {
@@ -274,10 +274,42 @@
 
 static void stdev_snd_mon_cb(void * stream __unused, struct str_parms * parms)
 {
+    audio_event_info_t event;
+    char value[32];
+    int ret = 0;
+
     if (!parms)
         return;
 
-    audio_extn_sound_trigger_set_parameters(NULL, parms);
+    ret = str_parms_get_str(parms, "SND_CARD_STATUS", value,
+                            sizeof(value));
+    if (ret > 0) {
+        if (strstr(value, "OFFLINE")) {
+            event.u.status = SND_CARD_STATUS_OFFLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else if (strstr(value, "ONLINE")) {
+            event.u.status = SND_CARD_STATUS_ONLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else
+            ALOGE("%s: unknown snd_card_status", __func__);
+    }
+
+    ret = str_parms_get_str(parms, "CPE_STATUS", value, sizeof(value));
+    if (ret > 0) {
+        if (strstr(value, "OFFLINE")) {
+            event.u.status = CPE_STATUS_OFFLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else if (strstr(value, "ONLINE")) {
+            event.u.status = CPE_STATUS_ONLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else
+            ALOGE("%s: unknown CPE status", __func__);
+    }
+
     return;
 }
 
@@ -630,34 +662,7 @@
         return;
     }
 
-    ret = str_parms_get_str(params, "SND_CARD_STATUS", value,
-                            sizeof(value));
-    if (ret > 0) {
-        if (strstr(value, "OFFLINE")) {
-            event.u.status = SND_CARD_STATUS_OFFLINE;
-            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
-        }
-        else if (strstr(value, "ONLINE")) {
-            event.u.status = SND_CARD_STATUS_ONLINE;
-            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
-        }
-        else
-            ALOGE("%s: unknown snd_card_status", __func__);
-    }
-
-    ret = str_parms_get_str(params, "CPE_STATUS", value, sizeof(value));
-    if (ret > 0) {
-        if (strstr(value, "OFFLINE")) {
-            event.u.status = CPE_STATUS_OFFLINE;
-            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
-        }
-        else if (strstr(value, "ONLINE")) {
-            event.u.status = CPE_STATUS_ONLINE;
-            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
-        }
-        else
-            ALOGE("%s: unknown CPE status", __func__);
-    }
+    stdev_snd_mon_cb(NULL, params);
 
     ret = str_parms_get_int(params, "SVA_NUM_SESSIONS", &val);
     if (ret >= 0) {
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 30bc10d..4f37ac3 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -146,6 +146,12 @@
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_VOIP_RX),
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_BD),
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_INTERACTIVE),
+#ifdef AUDIO_EXTN_AUTO_HAL_ENABLED
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_MEDIA),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_SYS_NOTIFICATION),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NAV_GUIDANCE),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PHONE),
+#endif
     STRING_TO_ENUM(AUDIO_INPUT_FLAG_NONE),
     STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
     STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
@@ -1860,6 +1866,27 @@
     return 0;
 }
 
+int audio_extn_utils_get_codec_variant(int card_num,
+                            char *codec_variant)
+{
+    char procfs_path[50];
+    FILE *fp;
+    snprintf(procfs_path, sizeof(procfs_path),
+             "/proc/asound/card%d/codecs/wcd938x/variant", card_num);
+    if ((fp = fopen(procfs_path, "r")) == NULL) {
+        snprintf(procfs_path, sizeof(procfs_path),
+                 "/proc/asound/card%d/codecs/wcd937x/variant", card_num);
+        if ((fp = fopen(procfs_path, "r")) == NULL) {
+            ALOGE("%s: ERROR. cannot open %s", __func__, procfs_path);
+            return -ENOENT;
+        }
+    }
+    fgets(codec_variant, CODEC_VARIANT_MAX_LENGTH, fp);
+    fclose(fp);
+    ALOGD("%s: codec variant is %s", __func__, codec_variant);
+    return 0;
+}
+
 
 #ifdef AUDIO_EXTERNAL_HDMI_ENABLED
 
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 7d8103b..6377696 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -406,6 +406,7 @@
     [USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION] = "sys-notification-playback",
     [USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE] = "nav-guidance-playback",
     [USECASE_AUDIO_PLAYBACK_PHONE] = "phone-playback",
+    [USECASE_AUDIO_FM_TUNER_EXT] = "fm-tuner-ext",
 };
 
 static const audio_usecase_t offload_usecases[] = {
@@ -505,6 +506,10 @@
 static int out_set_voip_volume(struct audio_stream_out *stream, float left, float right);
 static int out_set_pcm_volume(struct audio_stream_out *stream, float left, float right);
 
+static void adev_snd_mon_cb(void *cookie, struct str_parms *parms);
+static void in_snd_mon_cb(void * stream, struct str_parms * parms);
+static void out_snd_mon_cb(void * stream, struct str_parms * parms);
+
 #ifdef AUDIO_FEATURE_ENABLED_GCOV
 extern void  __gcov_flush();
 static void enable_gcov()
@@ -1528,14 +1533,11 @@
             goto end;
         }
 
-        // NB: case 7 is hypothetical and isn't a practical usecase yet.
-        // But if it does happen, we need to give priority to d2 if
-        // the combo devices active on the existing usecase share a backend.
-        // This is because we cannot have a usecase active on a combo device
-        // and a new usecase requests one device in this combo pair.
         if (platform_check_backends_match(d3[0], d3[1])) {
             return d2; // case 5
         } else {
+            if (popcount(a1) > 1)
+                return d1; //case 7
             // check if d1 is related to any of d3's
             if (d1 == d3[0] || d1 == d3[1])
                 return d1; // case 1
@@ -1611,7 +1613,8 @@
               platform_get_snd_device_name(snd_device),
               platform_get_snd_device_name(usecase->out_snd_device),
               platform_check_backends_match(snd_device, usecase->out_snd_device));
-        if ((usecase->type != PCM_CAPTURE) && (usecase != uc_info)) {
+        if ((usecase->type != PCM_CAPTURE) && (usecase != uc_info) &&
+                (usecase->type != PCM_PASSTHROUGH)) {
             uc_derive_snd_device = derive_playback_snd_device(adev->platform,
                                                usecase, uc_info, snd_device);
             if (((uc_derive_snd_device != usecase->out_snd_device) || force_routing) &&
@@ -2701,16 +2704,6 @@
     }
     enable_audio_route(adev, usecase);
 
-    /* Notify device change info to effect clients registered */
-    if (usecase->type == PCM_PLAYBACK) {
-        audio_extn_gef_notify_device_config(
-                usecase->stream.out->devices,
-                usecase->stream.out->channel_mask,
-                usecase->stream.out->app_type_cfg.sample_rate,
-                platform_get_snd_device_acdb_id(usecase->out_snd_device),
-                usecase->stream.out->app_type_cfg.app_type);
-    }
-
     audio_extn_qdsp_set_device(usecase);
 
     /* If input stream is already running then effect needs to be
@@ -3370,8 +3363,10 @@
         audio_low_latency_hint_end();
     }
 
-    if (out->usecase == USECASE_INCALL_MUSIC_UPLINK)
+    if (out->usecase == USECASE_INCALL_MUSIC_UPLINK ||
+        out->usecase == USECASE_INCALL_MUSIC_UPLINK2) {
         voice_set_device_mute_flag(adev, false);
+    }
 
     /* 1. Get and set stream specific mixer controls */
     disable_audio_route(adev, uc_info);
@@ -3508,6 +3503,16 @@
         goto error_config;
     }
 
+    //Update incall music usecase to reflect correct voice session
+    if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
+        ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
+        if (ret != 0) {
+            ALOGE("%s: Incall music delivery usecase cannot be set error:%d",
+                __func__, ret);
+            goto error_config;
+        }
+    }
+
     if (out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
         if (!audio_extn_a2dp_source_is_ready()) {
             if (out->devices &
@@ -3607,8 +3612,10 @@
          select_devices(adev, out->usecase);
     }
 
-    if (out->usecase == USECASE_INCALL_MUSIC_UPLINK)
+    if (out->usecase == USECASE_INCALL_MUSIC_UPLINK ||
+        out->usecase == USECASE_INCALL_MUSIC_UPLINK2) {
         voice_set_device_mute_flag(adev, true);
+    }
 
     if (audio_extn_ext_hw_plugin_usecase_start(adev->ext_hw_plugin, uc_info))
         ALOGE("%s: failed to start ext hw plugin", __func__);
@@ -4176,6 +4183,13 @@
             if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
                 do_stop = out->playback_started;
                 out->playback_started = false;
+
+                if (out->mmap_shared_memory_fd >= 0) {
+                    ALOGV("%s: closing mmap_shared_memory_fd = %d",
+                          __func__, out->mmap_shared_memory_fd);
+                    close(out->mmap_shared_memory_fd);
+                    out->mmap_shared_memory_fd = -1;
+                }
             }
         } else {
             ALOGD("copl(%p):standby", out);
@@ -5122,6 +5136,24 @@
             volume[1] = (long)(AmpToDb(right));
             mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
             return 0;
+        } else if ((out->devices & AUDIO_DEVICE_OUT_BUS) &&
+                (audio_extn_auto_hal_get_snd_device_for_car_audio_stream(out) ==
+                    SND_DEVICE_OUT_BUS_MEDIA)) {
+            ALOGD("%s: Overriding offload set volume for media bus stream", __func__);
+            struct listnode *node = NULL;
+            list_for_each(node, &adev->active_outputs_list) {
+                streams_output_ctxt_t *out_ctxt = node_to_item(node,
+                                                    streams_output_ctxt_t,
+                                                    list);
+                if (out_ctxt->output->usecase == USECASE_AUDIO_PLAYBACK_MEDIA) {
+                    out->volume_l = out_ctxt->output->volume_l;
+                    out->volume_r = out_ctxt->output->volume_r;
+                }
+            }
+            if (!out->a2dp_compress_mute) {
+                ret = out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
+            }
+            return ret;
         } else {
             pthread_mutex_lock(&out->compr_mute_lock);
             ALOGV("%s: compress mute %d", __func__, out->a2dp_compress_mute);
@@ -6085,6 +6117,9 @@
         // Fall back to non exclusive mode
         info->shared_memory_fd = pcm_get_poll_fd(out->pcm);
     } else {
+        out->mmap_shared_memory_fd = info->shared_memory_fd; // for closing later
+        ALOGV("%s: opened mmap_shared_memory_fd = %d", __func__, out->mmap_shared_memory_fd);
+
         if (mmap_size < buffer_size) {
             step = "mmap";
             goto exit;
@@ -6231,6 +6266,12 @@
         } else if (in->usecase == USECASE_AUDIO_RECORD_MMAP) {
             do_stop = in->capture_started;
             in->capture_started = false;
+            if (in->mmap_shared_memory_fd >= 0) {
+                ALOGV("%s: closing mmap_shared_memory_fd = %d",
+                      __func__, in->mmap_shared_memory_fd);
+                close(in->mmap_shared_memory_fd);
+                in->mmap_shared_memory_fd = -1;
+            }
         } else {
             if (audio_extn_cin_attached_usecase(in->usecase))
                 audio_extn_cin_close_input_stream(in);
@@ -6959,6 +7000,9 @@
         // Fall back to non exclusive mode
         info->shared_memory_fd = pcm_get_poll_fd(in->pcm);
     } else {
+        in->mmap_shared_memory_fd = info->shared_memory_fd; // for closing later
+        ALOGV("%s: opened mmap_shared_memory_fd = %d", __func__, in->mmap_shared_memory_fd);
+
         if (mmap_size < buffer_size) {
             step = "mmap";
             goto exit;
@@ -7133,6 +7177,15 @@
 
     *stream_out = NULL;
 
+    pthread_mutex_lock(&adev->lock);
+    if (out_get_stream(adev, handle) != NULL) {
+        ALOGW("%s, output stream already opened", __func__);
+        ret = -EEXIST;
+    }
+    pthread_mutex_unlock(&adev->lock);
+    if (ret)
+        return ret;
+
     out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
 
     ALOGD("%s: enter: format(%#x) sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)\
@@ -7174,6 +7227,7 @@
     out->set_dual_mono = false;
     out->prev_card_status_offline = false;
     out->pspd_coeff_sent = false;
+    out->mmap_shared_memory_fd = -1; // not open
 
     if ((flags & AUDIO_OUTPUT_FLAG_BD) &&
         (property_get_bool("vendor.audio.matrix.limiter.enable", false)))
@@ -8052,6 +8106,23 @@
     if (!parms)
         goto error;
 
+    /* notify adev and input/output streams on the snd card status */
+    adev_snd_mon_cb((void *)adev, parms);
+
+    list_for_each(node, &adev->active_outputs_list) {
+        streams_output_ctxt_t *out_ctxt = node_to_item(node,
+                                            streams_output_ctxt_t,
+                                            list);
+        out_snd_mon_cb((void *)out_ctxt->output, parms);
+    }
+
+    list_for_each(node, &adev->active_inputs_list) {
+        streams_input_ctxt_t *in_ctxt = node_to_item(node,
+                                            streams_input_ctxt_t,
+                                            list);
+        in_snd_mon_cb((void *)in_ctxt->input, parms);
+    }
+
     pthread_mutex_lock(&adev->lock);
     ret = str_parms_get_str(parms, "BT_SCO", value, sizeof(value));
     if (ret >= 0) {
@@ -8619,6 +8690,15 @@
             return -EINVAL;
     }
 
+    pthread_mutex_lock(&adev->lock);
+    if (in_get_stream(adev, handle) != NULL) {
+        ALOGW("%s, input stream already opened", __func__);
+        ret = -EEXIST;
+    }
+    pthread_mutex_unlock(&adev->lock);
+    if (ret)
+        return ret;
+
     in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
 
     if (!in) {
@@ -8665,6 +8745,7 @@
     in->zoom = 0;
     list_init(&in->aec_list);
     list_init(&in->ns_list);
+    in->mmap_shared_memory_fd = -1; // not open
 
     ALOGV("%s: source %d, config->channel_mask %#x", __func__, source, config->channel_mask);
     if (source == AUDIO_SOURCE_VOICE_UPLINK ||
@@ -9532,6 +9613,8 @@
     list_init(&adev->usecase_list);
     list_init(&adev->active_inputs_list);
     list_init(&adev->active_outputs_list);
+    list_init(&adev->audio_patch_record_list);
+    adev->audio_patch_index = 0;
     adev->cur_wfd_channels = 2;
     adev->offload_usecases_state = 0;
     adev->pcm_record_uc_state = 0;
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 4810896..ebdaa97 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -228,6 +228,8 @@
     USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE,
     USECASE_AUDIO_PLAYBACK_PHONE,
 
+    /*Audio FM Tuner usecase*/
+    USECASE_AUDIO_FM_TUNER_EXT,
     AUDIO_USECASE_MAX
 };
 
@@ -366,6 +368,7 @@
     bool muted;
     uint64_t written; /* total frames written, not cleared when entering standby */
     int64_t mmap_time_offset_nanos; /* fudge factor to correct inaccuracies in DSP */
+    int     mmap_shared_memory_fd; /* file descriptor associated with MMAP NOIRQ shared memory */
     audio_io_handle_t handle;
     struct stream_app_type_cfg app_type_cfg;
 
@@ -454,6 +457,7 @@
     struct listnode aec_list;
     struct listnode ns_list;
     int64_t mmap_time_offset_nanos; /* fudge factor to correct inaccuracies in DSP */
+    int     mmap_shared_memory_fd; /* file descriptor associated with MMAP NOIRQ shared memory */
     audio_io_handle_t capture_handle;
     audio_input_flags_t flags;
     char profile[MAX_STREAM_PROFILE_STR_LEN];
@@ -494,6 +498,7 @@
     PCM_HFP_CALL,
     TRANSCODE_LOOPBACK_RX,
     TRANSCODE_LOOPBACK_TX,
+    PCM_PASSTHROUGH,
     USECASE_TYPE_MAX
 } usecase_type_t;
 
@@ -673,6 +678,18 @@
     bool use_old_pspd_mix_ctrl;
     int camera_orientation; /* CAMERA_BACK_LANDSCAPE ... CAMERA_FRONT_PORTRAIT */
     bool adm_routing_changed;
+    struct listnode audio_patch_record_list;
+    unsigned int audio_patch_index;
+};
+
+struct audio_patch_record {
+    struct listnode list;
+    audio_patch_handle_t handle;
+    audio_usecase_t usecase;
+    audio_io_handle_t input_io_handle;
+    audio_io_handle_t output_io_handle;
+    struct audio_port_config source;
+    struct audio_port_config sink;
 };
 
 int select_devices(struct audio_device *adev,
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 8b9b53d..64e9383 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -2778,11 +2778,18 @@
 
 struct audio_custom_mtmx_params *
     platform_get_custom_mtmx_params(void *platform,
-                                    struct audio_custom_mtmx_params_info *info)
+                                    struct audio_custom_mtmx_params_info *info,
+                                    uint32_t *idx)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
     struct listnode *node = NULL;
     struct audio_custom_mtmx_params *params = NULL;
+    int i = 0;
+
+    if (!info || !idx) {
+        ALOGE("%s: Invalid params", __func__);
+        return NULL;
+    }
 
     list_for_each(node, &my_data->custom_mtmx_params_list) {
         params = node_to_item(node, struct audio_custom_mtmx_params, list);
@@ -2790,17 +2797,22 @@
             params->info.id == info->id &&
             params->info.ip_channels == info->ip_channels &&
             params->info.op_channels == info->op_channels &&
-            params->info.usecase_id == info->usecase_id &&
             params->info.snd_device == info->snd_device) {
-            ALOGV("%s: found params with ip_ch %d op_ch %d uc_id %d snd_dev %d",
-                  __func__, info->ip_channels, info->op_channels,
-                  info->usecase_id, info->snd_device);
-            return params;
+            while (params->info.usecase_id[i] != 0) {
+                if (params->info.usecase_id[i] == info->usecase_id[0]) {
+                    ALOGV("%s: found params with ip_ch %d op_ch %d uc_id %d snd_dev %d",
+                           __func__, info->ip_channels, info->op_channels,
+                           info->usecase_id[0], info->snd_device);
+                    *idx = i;
+                    return params;
+                }
+                i++;
+            }
         }
     }
     ALOGI("%s: no matching param with id %d ip_ch %d op_ch %d uc_id %d snd_dev %d",
           __func__, info->id, info->ip_channels, info->op_channels,
-          info->usecase_id, info->snd_device);
+          info->usecase_id[0], info->snd_device);
     return NULL;
 }
 
@@ -2810,6 +2822,12 @@
     struct platform_data *my_data = (struct platform_data *)platform;
     struct audio_custom_mtmx_params *params = NULL;
     uint32_t size = sizeof(*params);
+    int i = 0;
+
+    if (!info) {
+        ALOGE("%s: Invalid params", __func__);
+        return NULL;
+    }
 
     if (info->ip_channels > AUDIO_CHANNEL_COUNT_MAX ||
         info->op_channels > AUDIO_CHANNEL_COUNT_MAX) {
@@ -2825,9 +2843,14 @@
         return -ENOMEM;
     }
 
-    ALOGI("%s: adding mtmx params with id %d ip_ch %d op_ch %d uc_id %d snd_dev %d",
+    ALOGI("%s: adding mtmx params with id %d ip_ch %d op_ch %d snd_dev %d",
           __func__, info->id, info->ip_channels, info->op_channels,
-          info->usecase_id, info->snd_device);
+          info->snd_device);
+    while (info->usecase_id[i] != 0) {
+        ALOGI("%s: supported usecase ids for added mtmx params %d",
+              __func__, info->usecase_id[i]);
+        i++;
+    }
 
     params->info = *info;
     list_add_tail(&my_data->custom_mtmx_params_list, &params->list);
@@ -3696,6 +3719,17 @@
                   __func__, new_snd_device[i]);
             return -EINVAL;
         }
+
+        /* Notify device change info to effect clients registered */
+        if (usecase->type == PCM_PLAYBACK) {
+            audio_extn_gef_notify_device_config(
+                    usecase->stream.out->devices,
+                    usecase->stream.out->channel_mask,
+                    sample_rate,
+                    acdb_dev_id,
+                    usecase->stream.out->app_type_cfg.app_type);
+        }
+
         ALOGV("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
               __func__, new_snd_device[i], acdb_dev_id);
         if (new_snd_device[i] >= SND_DEVICE_OUT_BEGIN &&
@@ -5876,7 +5910,6 @@
     case USECASE_VOICEMMODE1_CALL:
     case USECASE_VOICEMMODE2_CALL:
     case USECASE_COMPRESS_VOIP_CALL:
-    case USECASE_AUDIO_RECORD_FM_VIRTUAL:
     case USECASE_INCALL_REC_UPLINK:
     case USECASE_INCALL_REC_DOWNLINK:
     case USECASE_INCALL_REC_UPLINK_AND_DOWNLINK:
@@ -5885,8 +5918,6 @@
     case USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS:
     case USECASE_INCALL_MUSIC_UPLINK:
     case USECASE_INCALL_MUSIC_UPLINK2:
-    case USECASE_AUDIO_SPKR_CALIB_RX:
-    case USECASE_AUDIO_SPKR_CALIB_TX:
     case USECASE_AUDIO_RECORD_VOIP:
         needs_event = true;
         break;
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 137e700..90105cd 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -357,7 +357,8 @@
     platform_get_custom_mtmx_params
     (
         void *platform __unused,
-        struct audio_custom_mtmx_params_info *info __unused
+        struct audio_custom_mtmx_params_info *info __unused,
+        uint32_t *idx __unused
     )
 {
     ALOGW("%s: not implemented!", __func__);
@@ -558,6 +559,17 @@
               __func__, snd_device);
         return -EINVAL;
     }
+
+    /* Notify device change info to effect clients registered */
+    if (usecase->type == PCM_PLAYBACK) {
+        audio_extn_gef_notify_device_config(
+                usecase->stream.out->devices,
+                usecase->stream.out->channel_mask,
+                usecase->stream.out->app_type_cfg.sample_rate,
+                acdb_dev_id,
+                usecase->stream.out->app_type_cfg.app_type);
+    }
+
     if (my_data->acdb_send_audio_cal) {
         ("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
               __func__, snd_device, acdb_dev_id);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index c7d2090..42c9256 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -335,6 +335,7 @@
     char ec_ref_mixer_path[MIXER_PATH_MAX_LENGTH];
     codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
     char codec_version[CODEC_VERSION_MAX_LENGTH];
+    char codec_variant[CODEC_VARIANT_MAX_LENGTH];
     int hw_dep_fd;
     char cvd_version[MAX_CVD_VERSION_STRING_SIZE];
     char snd_card_name[MAX_SND_CARD_STRING_SIZE];
@@ -480,7 +481,7 @@
                                              NAV_GUIDANCE_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_PHONE] = {PHONE_PCM_DEVICE,
                                       PHONE_PCM_DEVICE},
-
+    [USECASE_AUDIO_FM_TUNER_EXT] = {-1, -1},
 };
 
 /* Array to store sound devices */
@@ -2019,7 +2020,9 @@
         !strncmp(snd_card_name, "sdx-tavil-i2s-snd-card",
                  sizeof("sdx-tavil-i2s-snd-card")) ||
         !strncmp(snd_card_name, "sda845-tavil-i2s-snd-card",
-                 sizeof("sda845-tavil-i2s-snd-card"))) {
+                 sizeof("sda845-tavil-i2s-snd-card")) ||
+        !strncmp(snd_card_name, "sa6155-adp-star-snd-card",
+                 sizeof("sa6155-adp-star-snd-card"))) {
         plat_data->is_i2s_ext_modem = true;
     }
     ALOGV("%s, is_i2s_ext_modem:%d soundcard name is %s",__func__,
@@ -2028,6 +2031,20 @@
     return plat_data->is_i2s_ext_modem;
 }
 
+static bool is_auto_snd_card(const char *snd_card_name)
+{
+    bool is_auto_snd_card = false;
+
+    if (!strncmp(snd_card_name, "sa6155-adp-star-snd-card",
+                 sizeof("sa6155-adp-star-snd-card"))) {
+        is_auto_snd_card = true;
+        ALOGV("%s : Auto snd card detected: soundcard name is %s",__func__,
+               snd_card_name);
+    }
+
+    return is_auto_snd_card;
+}
+
 static void set_platform_defaults(struct platform_data * my_data)
 {
     int32_t dev;
@@ -2925,7 +2942,8 @@
         return NULL;
     }
 
-    if (platform_is_i2s_ext_modem(snd_card_name, my_data)) {
+    if (platform_is_i2s_ext_modem(snd_card_name, my_data) &&
+        !is_auto_snd_card(snd_card_name)) {
         ALOGD("%s: Call MIXER_XML_PATH_I2S", __func__);
 
         adev->audio_route = audio_route_init(adev->snd_card,
@@ -3163,7 +3181,7 @@
 
 
     /* Initialize ACDB ID's */
-    if (my_data->is_i2s_ext_modem)
+    if (my_data->is_i2s_ext_modem && !is_auto_snd_card(snd_card_name))
         platform_info_init(PLATFORM_INFO_XML_PATH_I2S, my_data, PLATFORM);
     else if (!strncmp(snd_card_name, "sdm660-snd-card-skush",
                sizeof("sdm660-snd-card-skush")))
@@ -3362,6 +3380,7 @@
     property_get("ro.baseband", baseband, "");
     if ((!strncmp("apq8084", platform, sizeof("apq8084")) ||
         !strncmp("msm8996", platform, sizeof("msm8996")) ||
+        !strncmp("sm6150", platform, sizeof("sm6150")) ||
         !strncmp("sdx", platform, sizeof("sdx")) ||
         !strncmp("sdm845", platform, sizeof("sdm845"))) &&
         ( !strncmp("mdm", baseband, (sizeof("mdm")-1)) ||
@@ -3645,6 +3664,8 @@
         }
     }
 
+    ret = audio_extn_utils_get_codec_variant(my_data->adev->snd_card,
+                                             my_data->codec_variant);
     ret = audio_extn_utils_get_codec_version(snd_card_name,
                                              my_data->adev->snd_card,
                                              my_data->codec_version);
@@ -3682,11 +3703,18 @@
 
 struct audio_custom_mtmx_params *
     platform_get_custom_mtmx_params(void *platform,
-                                    struct audio_custom_mtmx_params_info *info)
+                                    struct audio_custom_mtmx_params_info *info,
+                                    uint32_t *idx)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
     struct listnode *node = NULL;
     struct audio_custom_mtmx_params *params = NULL;
+    int i = 0;
+
+    if (!info || !idx) {
+        ALOGE("%s: Invalid params", __func__);
+        return NULL;
+    }
 
     list_for_each(node, &my_data->custom_mtmx_params_list) {
         params = node_to_item(node, struct audio_custom_mtmx_params, list);
@@ -3694,17 +3722,22 @@
             params->info.id == info->id &&
             params->info.ip_channels == info->ip_channels &&
             params->info.op_channels == info->op_channels &&
-            params->info.usecase_id == info->usecase_id &&
             params->info.snd_device == info->snd_device) {
-            ALOGV("%s: found params with ip_ch %d op_ch %d uc_id %d snd_dev %d",
-                  __func__, info->ip_channels, info->op_channels,
-                  info->usecase_id, info->snd_device);
-            return params;
+            while (params->info.usecase_id[i] != 0) {
+                if (params->info.usecase_id[i] == info->usecase_id[0]) {
+                    ALOGV("%s: found params with ip_ch %d op_ch %d uc_id %d snd_dev %d",
+                           __func__, info->ip_channels, info->op_channels,
+                           info->usecase_id[0], info->snd_device);
+                    *idx = i;
+                    return params;
+                }
+                i++;
+            }
         }
     }
     ALOGI("%s: no matching param with id %d ip_ch %d op_ch %d uc_id %d snd_dev %d",
           __func__, info->id, info->ip_channels, info->op_channels,
-          info->usecase_id, info->snd_device);
+          info->usecase_id[0], info->snd_device);
     return NULL;
 }
 
@@ -3714,6 +3747,12 @@
     struct platform_data *my_data = (struct platform_data *)platform;
     struct audio_custom_mtmx_params *params = NULL;
     uint32_t size = sizeof(*params);
+    int i = 0;
+
+    if (!info) {
+        ALOGE("%s: Invalid params", __func__);
+        return -EINVAL;
+    }
 
     if (info->ip_channels > AUDIO_CHANNEL_COUNT_MAX ||
         info->op_channels > AUDIO_CHANNEL_COUNT_MAX) {
@@ -3729,9 +3768,14 @@
         return -ENOMEM;
     }
 
-    ALOGI("%s: adding mtmx params with id %d ip_ch %d op_ch %d uc_id %d snd_dev %d",
+    ALOGI("%s: adding mtmx params with id %d ip_ch %d op_ch %d snd_dev %d",
           __func__, info->id, info->ip_channels, info->op_channels,
-          info->usecase_id, info->snd_device);
+          info->snd_device);
+    while (info->usecase_id[i] != 0) {
+        ALOGI("%s: supported usecase ids for added mtmx params %d",
+              __func__, info->usecase_id[i]);
+        i++;
+    }
 
     params->info = *info;
     list_add_tail(&my_data->custom_mtmx_params_list, &params->list);
@@ -3755,20 +3799,30 @@
     struct platform_data *my_data = (struct platform_data *)platform;
     struct listnode *node = NULL;
     struct audio_custom_mtmx_in_params *params = NULL;
+    int i = 0;
+
+    if (!info) {
+        ALOGE("%s: Invalid params", __func__);
+        return NULL;
+    }
 
     list_for_each(node, &my_data->custom_mtmx_in_params_list) {
         params = node_to_item(node, struct audio_custom_mtmx_in_params, list);
         if (params &&
-            params->in_info.op_channels == info->op_channels &&
-            params->in_info.usecase_id == info->usecase_id) {
-            ALOGV("%s: found params with op_ch %d uc_id %d",
-                  __func__, info->op_channels, info->usecase_id);
-            return params;
+            params->in_info.op_channels == info->op_channels) {
+            while (params->in_info.usecase_id[i] != 0) {
+                if (params->in_info.usecase_id[i] == info->usecase_id[0]) {
+                    ALOGV("%s: found params with op_ch %d uc_id %d",
+                          __func__, info->op_channels, info->usecase_id[0]);
+                    return params;
+                }
+                i++;
+            }
         }
     }
 
     ALOGI("%s: no matching param with op_ch %d uc_id %d",
-           __func__, info->op_channels, info->usecase_id);
+           __func__, info->op_channels, info->usecase_id[0]);
     return NULL;
 }
 
@@ -3778,6 +3832,12 @@
     struct platform_data *my_data = (struct platform_data *)platform;
     struct audio_custom_mtmx_in_params *params = NULL;
     uint32_t size = sizeof(*params);
+    int i = 0;
+
+    if (!info) {
+        ALOGE("%s: Invalid params", __func__);
+        return -EINVAL;
+    }
 
     if (info->op_channels > AUDIO_CHANNEL_COUNT_MAX) {
         ALOGE("%s: unusupported channels in %d", __func__, info->op_channels);
@@ -3790,8 +3850,14 @@
         return -ENOMEM;
     }
 
-    ALOGI("%s: adding mtmx in params with op_ch %d uc_id %d",
-          __func__, info->op_channels, info->usecase_id);
+    ALOGI("%s: adding mtmx in params with op_ch %d",
+          __func__, info->op_channels);
+
+    while (info->usecase_id[i] != 0) {
+        ALOGI("%s: supported usecase ids for added mtmx in params %d",
+              __func__, info->usecase_id[i]);
+        i++;
+    }
 
     params->in_info = *info;
     list_add_tail(&my_data->custom_mtmx_in_params_list, &params->list);
@@ -4918,6 +4984,17 @@
                   __func__, new_snd_device[i]);
             return -EINVAL;
         }
+
+        /* Notify device change info to effect clients registered */
+        if (usecase->type == PCM_PLAYBACK) {
+            audio_extn_gef_notify_device_config(
+                    usecase->stream.out->devices,
+                    usecase->stream.out->channel_mask,
+                    sample_rate,
+                    acdb_dev_id,
+                    usecase->stream.out->app_type_cfg.app_type);
+        }
+
         ALOGV("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
               __func__, new_snd_device[i], acdb_dev_id);
         if (new_snd_device[i] >= SND_DEVICE_OUT_BEGIN &&
@@ -5803,7 +5880,8 @@
                 snd_device = SND_DEVICE_OUT_BT_SCO_WB;
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
-        } else if (devices & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
+        } else if ((devices & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE)) ||
+                   (devices & AUDIO_DEVICE_OUT_BUS)) {
             if (my_data->is_vbat_speaker || my_data->is_bcl_speaker) {
                 if (hw_info_is_stereo_spkr(my_data->hw_info)) {
                     if (my_data->mono_speaker == SPKR_1)
@@ -6213,6 +6291,7 @@
     int str_bitwidth = (in == NULL) ? CODEC_BACKEND_DEFAULT_BIT_WIDTH : in->bit_width;
     int sample_rate = (in == NULL) ? 8000 : in->sample_rate;
     struct audio_usecase *usecase = NULL;
+    audio_usecase_t uc_id = (in == NULL) ? USECASE_AUDIO_RECORD : in->usecase;
 
     ALOGV("%s: enter: out_device(%#x) in_device(%#x) channel_count (%d) channel_mask (0x%x)",
           __func__, out_device, in_device, channel_count, channel_mask);
@@ -6354,7 +6433,8 @@
         } else if (out_device & AUDIO_DEVICE_OUT_SPEAKER ||
                    out_device & AUDIO_DEVICE_OUT_SPEAKER_SAFE ||
                    out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
-                   out_device & AUDIO_DEVICE_OUT_LINE) {
+                   out_device & AUDIO_DEVICE_OUT_LINE ||
+                   out_device & AUDIO_DEVICE_OUT_BUS) {
             if (my_data->fluence_type != FLUENCE_NONE &&
                 (my_data->fluence_in_voice_call ||
                  my_data->fluence_in_hfp_call) &&
@@ -6622,7 +6702,7 @@
             }
         } else if (in_device & AUDIO_DEVICE_IN_LOOPBACK) {
             if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-                usecase = get_usecase_from_list(adev, USECASE_AUDIO_RECORD);
+                usecase = get_usecase_from_list(adev, uc_id);
                 if (usecase == NULL) {
                     ALOGE("%s: Could not find the record usecase", __func__);
                     snd_device = SND_DEVICE_NONE;
@@ -7943,9 +8023,6 @@
     case USECASE_INCALL_MUSIC_UPLINK:
     case USECASE_INCALL_MUSIC_UPLINK2:
     case USECASE_AUDIO_RECORD_VOIP:
-    case USECASE_AUDIO_RECORD_FM_VIRTUAL:
-    case USECASE_AUDIO_SPKR_CALIB_RX:
-    case USECASE_AUDIO_SPKR_CALIB_TX:
         needs_event = true;
         break;
     default:
@@ -7979,7 +8056,9 @@
     }
 
     /* Use client specified buffer size if mentioned */
-    if ((info != NULL) && (info->duration_us > 0)) {
+    if ((info != NULL) &&
+        (info->duration_us >= MIN_OFFLOAD_BUFFER_DURATION_MS) &&
+        (info->duration_us <= MAX_OFFLOAD_BUFFER_DURATION_MS)) {
         duration_ms = info->duration_us / 1000;
         channel_count = audio_channel_count_from_in_mask(info->channel_mask);
 
@@ -8562,6 +8641,21 @@
                         sample_rate = curr_out->sample_rate;
                     }
                 }
+
+                /* WCD9380 support SR upto 192Khz only, hence reset
+                 * SR > 192Khz to 192Khz.
+                 */
+                if (strstr(my_data->codec_variant, "WCD9380")) {
+                    switch (sample_rate) {
+                        case 352800:
+                        case 384000:
+                            sample_rate = 192000;
+                            ALOGD("%s:Reset Sampling rate to %d",  __func__, sample_rate);
+                            break;
+                        default:
+                            break;
+                    }
+                }
             }
         } else if (na_mode != NATIVE_AUDIO_MODE_MULTIPLE_MIX_IN_CODEC) {
             /*
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 13f85e6..3816f77 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -372,6 +372,7 @@
 #define DEFAULT_VOLUME_RAMP_DURATION_MS 20
 #define MIXER_PATH_MAX_LENGTH 100
 #define CODEC_VERSION_MAX_LENGTH 100
+#define CODEC_VARIANT_MAX_LENGTH 100
 
 #define MAX_VOL_INDEX 5
 #define MIN_VOL_INDEX 0
@@ -444,7 +445,12 @@
 #define MULTIMEDIA9_PCM_DEVICE 32
 #define FM_PLAYBACK_PCM_DEVICE 5
 #define FM_CAPTURE_PCM_DEVICE  6
+
+#ifdef PLATFORM_AUTO
+#define HFP_PCM_RX 36
+#else
 #define HFP_PCM_RX 5
+#endif
 
 #define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
 
@@ -456,6 +462,10 @@
 #define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
 #elif PLATFORM_APQ8084
 #define INCALL_MUSIC_UPLINK2_PCM_DEVICE 34
+#elif PLATFORM_MSMNILE
+#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 27
+#elif PLATFORM_KONA
+#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 23
 #else
 #define INCALL_MUSIC_UPLINK2_PCM_DEVICE 35
 #endif
@@ -586,6 +596,14 @@
 #define VOLTE_CALL_PCM_DEVICE 15
 #define QCHAT_CALL_PCM_DEVICE 37
 #define VOWLAN_CALL_PCM_DEVICE 16
+#elif PLATFORM_AUTO
+#define HOST_LESS_RX_ID 41
+#define HOST_LESS_TX_ID 42
+#define VOICE_CALL_PCM_DEVICE 8
+#define VOICE2_CALL_PCM_DEVICE -1
+#define VOLTE_CALL_PCM_DEVICE -1
+#define QCHAT_CALL_PCM_DEVICE -1
+#define VOWLAN_CALL_PCM_DEVICE -1
 #else
 #define VOICE_CALL_PCM_DEVICE 2
 #define VOICE2_CALL_PCM_DEVICE 22
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 394310a..30a10c5 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -35,6 +35,7 @@
 #define PRODUCT_FFV      "ffv"
 #define PRODUCT_ALLPLAY  "allplay"
 #define MAX_IN_CHANNELS 32
+#define CUSTOM_MTRX_PARAMS_MAX_USECASE 8
 
 typedef enum {
     PLATFORM,
@@ -103,9 +104,9 @@
     uint32_t id;
     uint32_t ip_channels;
     uint32_t op_channels;
-    uint32_t usecase_id;
+    uint32_t usecase_id[CUSTOM_MTRX_PARAMS_MAX_USECASE];
     uint32_t snd_device;
-    char fe_name[128];
+    uint32_t fe_id[CUSTOM_MTRX_PARAMS_MAX_USECASE];
 };
 
 struct audio_custom_mtmx_params {
@@ -116,7 +117,7 @@
 
 struct audio_custom_mtmx_in_params_info {
     uint32_t op_channels;
-    uint32_t usecase_id;
+    uint32_t usecase_id[CUSTOM_MTRX_PARAMS_MAX_USECASE];
 };
 
 struct audio_custom_mtmx_params_in_ch_info {
@@ -377,7 +378,8 @@
 int platform_get_delay(void *platform, int pcm_device_id);
 struct audio_custom_mtmx_params *
     platform_get_custom_mtmx_params(void *platform,
-                                    struct audio_custom_mtmx_params_info *info);
+                                    struct audio_custom_mtmx_params_info *info,
+                                    uint32_t *idx);
 int platform_add_custom_mtmx_params(void *platform,
                                     struct audio_custom_mtmx_params_info *info);
 /* callback functions from platform to common audio HAL */
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 8ee8b07..d73792c 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -653,6 +653,8 @@
     }
 
     str_parms_add_str(my_data.kvpairs, (char*)attr[1], (char*)attr[3]);
+    if (my_data.caller == PLATFORM)
+        platform_set_parameters(my_data.platform, my_data.kvpairs);
 done:
     return;
 }
@@ -1068,13 +1070,19 @@
 
 static void process_custom_mtmx_in_params(const XML_Char **attr)
 {
-    int attr_idx = 0;
+    int attr_idx = 0, i = 0;
+    char *context = NULL, *value = NULL;
 
     if (strcmp(attr[attr_idx++], "usecase") != 0) {
         ALOGE("%s: 'usecase' not found", __func__);
         return;
     }
-    mtmx_in_params_info.usecase_id = platform_get_usecase_index((char *)attr[attr_idx++]);
+    /* Check if multi usecases are supported for this custom mtrx params */
+    value = strtok_r((char *)attr[attr_idx++], ",", &context);
+    while (value && (i < CUSTOM_MTRX_PARAMS_MAX_USECASE)) {
+        mtmx_in_params_info.usecase_id[i++] = platform_get_usecase_index(value);
+        value = strtok_r(NULL, ",", &context);
+    }
 
     if (strcmp(attr[attr_idx++], "out_channel_count") != 0) {
         ALOGE("%s: 'out_channel_count' not found", __func__);
@@ -1089,7 +1097,7 @@
 static void process_custom_mtmx_param_coeffs(const XML_Char **attr)
 {
     uint32_t attr_idx = 0, out_ch_idx = -1, ch_coeff_count = 0;
-    uint32_t ip_channels = 0, op_channels = 0;
+    uint32_t ip_channels = 0, op_channels = 0, idx = 0;
     char *context = NULL, *ch_coeff_value = NULL;
     struct audio_custom_mtmx_params *mtmx_params = NULL;
 
@@ -1109,7 +1117,7 @@
         return;
     }
     mtmx_params = platform_get_custom_mtmx_params((void *)my_data.platform,
-                                                  &mtmx_params_info);
+                                                  &mtmx_params_info, &idx);
     if (mtmx_params == NULL) {
         ALOGE("%s: mtmx params with given param info, not found", __func__);
         return;
@@ -1129,7 +1137,10 @@
 
 static void process_custom_mtmx_params(const XML_Char **attr)
 {
-    int attr_idx = 0;
+    int attr_idx = 0, i = 0;
+    char *context = NULL, *value = NULL;
+
+    memset(&mtmx_params_info, 0, sizeof(mtmx_params_info));
 
     if (strcmp(attr[attr_idx++], "param_id") != 0) {
         ALOGE("%s: 'param_id' not found", __func__);
@@ -1153,7 +1164,13 @@
         ALOGE("%s: 'usecase' not found", __func__);
         return;
     }
-    mtmx_params_info.usecase_id = platform_get_usecase_index((char *)attr[attr_idx++]);
+
+    /* check if multi usecases are supported for this custom mtrx params */
+    value = strtok_r((char *)attr[attr_idx++], ",", &context);
+    while (value && (i < CUSTOM_MTRX_PARAMS_MAX_USECASE)) {
+        mtmx_params_info.usecase_id[i++] = platform_get_usecase_index(value);
+        value = strtok_r(NULL, ",", &context);
+    }
 
     if (strcmp(attr[attr_idx++], "snd_device") != 0) {
         ALOGE("%s: 'snd_device' not found", __func__);
@@ -1161,12 +1178,15 @@
     }
     mtmx_params_info.snd_device = platform_get_snd_device_index((char *)attr[attr_idx++]);
 
-    if ((attr[attr_idx] != NULL) && (strcmp(attr[attr_idx++], "fe_name") == 0)) {
-        strlcpy(mtmx_params_info.fe_name, (char *)attr[attr_idx++],
-                sizeof(mtmx_params_info.fe_name));
-    } else {
-        ALOGD("%s: 'fe_name' not found", __func__);
-        mtmx_params_info.fe_name[0] = '\0';
+    if ((attr[attr_idx] != NULL) && (strcmp(attr[attr_idx++], "fe_id") == 0)) {
+        i = 0;
+        value = strtok_r((char *)attr[attr_idx++], ",", &context);
+        while (value && (i < CUSTOM_MTRX_PARAMS_MAX_USECASE)) {
+            mtmx_params_info.fe_id[i++] = atoi(value);
+            value = strtok_r(NULL, ",", &context);
+        }
+
+        attr_idx++;
     }
 
     platform_add_custom_mtmx_params((void *)my_data.platform, &mtmx_params_info);
@@ -1387,9 +1407,6 @@
         section = ROOT;
     } else if (strcmp(tag_name, "config_params") == 0) {
         section = ROOT;
-        if (my_data.caller == PLATFORM) {
-            platform_set_parameters(my_data.platform, my_data.kvpairs);
-        }
     } else if (strcmp(tag_name, "operator_specific") == 0) {
         section = ROOT;
     } else if (strcmp(tag_name, "interface_names") == 0) {
diff --git a/hal/voice.c b/hal/voice.c
index 006dd08..0000d72 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -48,6 +48,10 @@
     .format = PCM_FORMAT_S16_LE,
 };
 
+#ifdef PLATFORM_AUTO
+struct pcm *voice_loopback_tx = NULL;
+struct pcm *voice_loopback_rx = NULL;
+#endif
 static struct voice_session *voice_get_session_from_use_case(struct audio_device *adev,
                               audio_usecase_t usecase_id)
 {
@@ -182,6 +186,16 @@
         session->pcm_tx = NULL;
     }
 
+#ifdef PLATFORM_AUTO
+    if(voice_loopback_rx) {
+        pcm_close(voice_loopback_rx);
+        voice_loopback_rx = NULL;
+    }
+    if(voice_loopback_tx) {
+        pcm_close(voice_loopback_tx);
+        voice_loopback_tx = NULL;
+    }
+#endif
     /* 2. Get and set stream specific mixer controls */
     disable_audio_route(adev, uc_info);
 
@@ -201,6 +215,9 @@
     int ret = 0;
     struct audio_usecase *uc_info;
     int pcm_dev_rx_id, pcm_dev_tx_id;
+#ifdef PLATFORM_AUTO
+    int pcm_dev_loopback_rx_id, pcm_dev_loopback_tx_id;
+#endif
     uint32_t sample_rate = 8000;
     struct voice_session *session = NULL;
     struct pcm_config voice_config = pcm_config_voice_call;
@@ -246,6 +263,10 @@
 
     select_devices(adev, usecase_id);
 
+#ifdef PLATFORM_AUTO
+    pcm_dev_loopback_rx_id = HOST_LESS_RX_ID;
+    pcm_dev_loopback_tx_id = HOST_LESS_TX_ID;
+#endif
     pcm_dev_rx_id = platform_get_pcm_device_id(uc_info->id, PCM_PLAYBACK);
     pcm_dev_tx_id = platform_get_pcm_device_id(uc_info->id, PCM_CAPTURE);
 
@@ -287,7 +308,29 @@
         goto error_start_voice;
     }
 
-    if(adev->mic_break_enabled)
+#ifdef PLATFORM_AUTO
+    voice_loopback_rx = pcm_open(adev->snd_card,
+                                 pcm_dev_loopback_rx_id,
+                                 PCM_OUT, &voice_config);
+    if (voice_loopback_rx < 0 || !pcm_is_ready(voice_loopback_rx)) {
+        ALOGE("%s: Either could not open pcm_dev_loopback_rx_id %d or %s",
+              __func__, pcm_dev_loopback_rx_id, pcm_get_error(voice_loopback_rx));
+        ret = -EIO;
+        goto error_start_voice;
+    }
+
+    voice_loopback_tx = pcm_open(adev->snd_card,
+                                 pcm_dev_loopback_tx_id,
+                                 PCM_IN, &voice_config);
+    if (voice_loopback_tx < 0 || !pcm_is_ready(voice_loopback_tx)) {
+         ALOGE("%s: Either could not open pcm_dev_loopback_tx_id %d or %s",
+               __func__, pcm_dev_loopback_tx_id, pcm_get_error(voice_loopback_tx));
+         ret = -EIO;
+         goto error_start_voice;
+    }
+#endif
+
+    if (adev->mic_break_enabled)
         platform_set_mic_break_det(adev->platform, true);
 
     ret = pcm_start(session->pcm_tx);
@@ -302,6 +345,20 @@
         goto error_start_voice;
     }
 
+#ifdef PLATFORM_AUTO
+    ret = pcm_start(voice_loopback_tx);
+    if (ret != 0) {
+        ALOGE("%s: %s", __func__, pcm_get_error(voice_loopback_tx));
+        goto error_start_voice;
+    }
+
+    ret = pcm_start(voice_loopback_rx);
+    if (ret != 0) {
+        ALOGE("%s: %s", __func__, pcm_get_error(voice_loopback_rx));
+        goto error_start_voice;
+    }
+#endif
+
     /* Enable aanc only when no calls are active */
     if (!voice_is_call_state_active(adev))
         voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, true);
@@ -729,6 +786,10 @@
 void voice_init(struct audio_device *adev)
 {
     int i = 0;
+    int max_voice_sessions = MAX_VOICE_SESSIONS;
+
+    if (!voice_extn_is_multi_session_supported())
+        max_voice_sessions = 1;
 
     memset(&adev->voice, 0, sizeof(adev->voice));
     adev->voice.tty_mode = TTY_MODE_OFF;
@@ -736,7 +797,7 @@
     adev->voice.volume = 1.0f;
     adev->voice.mic_mute = false;
     adev->voice.in_call = false;
-    for (i = 0; i < MAX_VOICE_SESSIONS; i++) {
+    for (i = 0; i < max_voice_sessions; i++) {
         adev->voice.session[i].pcm_rx = NULL;
         adev->voice.session[i].pcm_tx = NULL;
         adev->voice.session[i].state.current = CALL_INACTIVE;
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 473edc2..e6a4ed6 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -87,6 +87,7 @@
 static bool voice_extn_compress_voip_enabled = false;
 static bool voice_extn_dynamic_ecns_feature_enabled = false;
 static bool voice_extn_incall_music_enabled = false;
+static bool voice_extn_multi_session_enabled = false;
 
 int voice_extn_is_call_state_active(struct audio_device *adev, bool *is_call_active);
 
@@ -184,10 +185,14 @@
     struct voice_session *session = NULL;
     int i = 0;
     uint32_t session_id = 0;
+    int max_voice_sessions = MAX_VOICE_SESSIONS;
 
-    for (i = 0; i < MAX_VOICE_SESSIONS; i++) {
+    if (!voice_extn_is_multi_session_supported())
+        max_voice_sessions = 1;
+
+    for (i = 0; i < max_voice_sessions; i++) {
         session = &adev->voice.session[i];
-        if(session->state.current == call_state){
+        if (session->state.current == call_state){
             session_id = session->vsid;
             break;
         }
@@ -203,10 +208,13 @@
     enum voice_lch_mode lch_mode;
     struct voice_session *session = NULL;
     int ret = 0;
+    int max_voice_sessions = MAX_VOICE_SESSIONS;
 
     ALOGD("%s: enter:", __func__);
+    if (!voice_extn_is_multi_session_supported())
+        max_voice_sessions = 1;
 
-    for (i = 0; i < MAX_VOICE_SESSIONS; i++) {
+    for (i = 0; i < max_voice_sessions; i++) {
         usecase_id = voice_extn_get_usecase_for_session_idx(i);
         session = &adev->voice.session[i];
         ALOGD("%s: cur_state=%d new_state=%d vsid=%x",
@@ -220,7 +228,7 @@
             case CALL_INACTIVE:
                 ALOGD("%s: INACTIVE -> ACTIVE vsid:%x", __func__, session->vsid);
                 ret = voice_start_usecase(adev, usecase_id);
-                if(ret < 0) {
+                if (ret < 0) {
                     ALOGE("%s: voice_start_usecase() failed for usecase: %d\n",
                           __func__, usecase_id);
                 } else {
@@ -258,7 +266,7 @@
             case CALL_LOCAL_HOLD:
                 ALOGD("%s: ACTIVE/HOLD/LOCAL_HOLD -> INACTIVE vsid:%x", __func__, session->vsid);
                 ret = voice_stop_usecase(adev, usecase_id);
-                if(ret < 0) {
+                if (ret < 0) {
                     ALOGE("%s: voice_stop_usecase() failed for usecase: %d\n",
                           __func__, usecase_id);
                 } else {
@@ -334,8 +342,12 @@
     struct voice_session *session = NULL;
     int i = 0;
     bool is_call_active;
+    int max_voice_sessions = MAX_VOICE_SESSIONS;
 
-    for (i = 0; i < MAX_VOICE_SESSIONS; i++) {
+    if (!voice_extn_is_multi_session_supported())
+        max_voice_sessions = 1;
+
+    for (i = 0; i < max_voice_sessions; i++) {
         if (vsid == adev->voice.session[i].vsid) {
             session = &adev->voice.session[i];
             break;
@@ -370,6 +382,9 @@
 int voice_extn_get_active_session_id(struct audio_device *adev,
                                      uint32_t *session_id)
 {
+    if (!voice_extn_is_multi_session_supported())
+        return -ENOSYS;
+
     *session_id = get_session_id_with_state(adev, CALL_ACTIVE);
     return 0;
 }
@@ -378,11 +393,14 @@
 {
     struct voice_session *session = NULL;
     int i = 0;
-    *is_call_active = false;
 
+    if (!voice_extn_is_multi_session_supported())
+        return -ENOSYS;
+
+    *is_call_active = false;
     for (i = 0; i < MAX_VOICE_SESSIONS; i++) {
         session = &adev->voice.session[i];
-        if(session->state.current != CALL_INACTIVE){
+        if (session->state.current != CALL_INACTIVE){
             *is_call_active = true;
             break;
         }
@@ -403,14 +421,17 @@
     return voice_extn_dynamic_ecns_feature_enabled;
 }
 
-// START: INCALL_MUSIC ===================================================================
 void incall_music_feature_init(bool is_feature_enabled)
 {
     voice_extn_incall_music_enabled = is_feature_enabled;
     ALOGV("%s: ---- Feature INCALL_MUSIC is %s----", __func__,
                                 is_feature_enabled ? "ENABLED" : "NOT ENABLED");
 }
-// END: INCALL_MUSIC ===================================================================
+
+bool voice_extn_is_incall_music_enabled()
+{
+    return voice_extn_incall_music_enabled;
+}
 
 void compr_voip_feature_init(bool is_feature_enabled)
 {
@@ -424,6 +445,18 @@
     return voice_extn_compress_voip_enabled;
 }
 
+void multi_voice_session_feature_init(bool is_feature_enabled)
+{
+    voice_extn_multi_session_enabled = is_feature_enabled;
+    ALOGV("%s:: ---- Feature MULTI VOICE SESSION is %s ----", __func__,
+                                is_feature_enabled ? "ENABLED" : "NOT ENABLED");
+}
+
+bool voice_extn_is_multi_session_supported()
+{
+    return voice_extn_multi_session_enabled;
+}
+
 void voice_extn_feature_init()
 {
     // Register feature function here
@@ -437,10 +470,17 @@
     incall_music_feature_init(
        property_get_bool("vendor.audio.feature.incall_music.enable",
                           true));
+    multi_voice_session_feature_init(
+       property_get_bool("vendor.audio.feature.multi_voice_session.enable",
+                          true));
+
 }
 
 void voice_extn_init(struct audio_device *adev)
 {
+    if (!voice_extn_is_multi_session_supported())
+        return;
+
     adev->voice.session[VOICE_SESS_IDX].vsid =  VOICE_VSID;
     adev->voice.session[VOICE2_SESS_IDX].vsid = VOICE2_VSID;
     adev->voice.session[VOLTE_SESS_IDX].vsid =  VOLTE_VSID;
@@ -454,6 +494,8 @@
                                          const audio_usecase_t usecase_id,
                                          struct voice_session **session)
 {
+    if (!voice_extn_is_multi_session_supported())
+        return -ENOSYS;
 
     switch(usecase_id)
     {
@@ -500,6 +542,9 @@
      * udpated.
      */
     ALOGV("%s: enter:", __func__);
+    if (!voice_extn_is_multi_session_supported())
+        return -ENOSYS;
+
     return update_calls(adev);
 }
 
@@ -510,6 +555,9 @@
 
     ALOGV("%s: enter:", __func__);
 
+    if (!voice_extn_is_multi_session_supported())
+        return -ENOSYS;
+
     /* If BT device is enabled and voice calls are ended, telephony will call
      * set_mode(AUDIO_MODE_NORMAL) which will trigger audio policy manager to
      * set routing with device BT A2DP profile. Hence end all voice calls when
@@ -616,8 +664,12 @@
     int ret = 0;
     char *cur_ptr = value;
     int i, len=0;
+    int max_voice_sessions = MAX_VOICE_SESSIONS;
 
-    for (i = 0; i < MAX_VOICE_SESSIONS; i++) {
+    if (!voice_extn_is_multi_session_supported())
+        max_voice_sessions = 1;
+
+    for (i = 0; i < max_voice_sessions; i++) {
         snprintf(cur_ptr, VOICE_EXTN_PARAMETER_VALUE_MAX_LEN - len,
                  "%d:%d,",adev->voice.session[i].vsid,
                  adev->voice.session[i].state.current);
@@ -655,7 +707,7 @@
         }
         str_parms_add_str(reply, AUDIO_PARAMETER_KEY_ALL_CALL_STATES, value);
     }
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         voice_extn_compress_voip_get_parameters(query, reply);
 
     str = str_parms_to_str(reply);
@@ -667,7 +719,7 @@
                                    struct str_parms *query,
                                    struct str_parms *reply)
 {
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         voice_extn_compress_voip_out_get_parameters(out, query, reply);
 }
 
@@ -675,7 +727,7 @@
                                   struct str_parms *query,
                                   struct str_parms *reply)
 {
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         voice_extn_compress_voip_in_get_parameters(in, query, reply);
 }
 
@@ -683,8 +735,17 @@
 int voice_extn_check_and_set_incall_music_usecase(struct audio_device *adev,
                                                   struct stream_out *out)
 {
-    if(voice_extn_incall_music_enabled) {
-        out->usecase = USECASE_INCALL_MUSIC_UPLINK;
+    if (voice_extn_incall_music_enabled) {
+        uint32_t session_id = get_session_id_with_state(adev, CALL_ACTIVE);
+        if (session_id == VOICEMMODE1_VSID) {
+            out->usecase = USECASE_INCALL_MUSIC_UPLINK;
+        } else if (session_id == VOICEMMODE2_VSID) {
+            out->usecase = USECASE_INCALL_MUSIC_UPLINK2;
+        } else {
+            ALOGE("%s: Invalid session id %x", __func__, session_id);
+            out->usecase = USECASE_INCALL_MUSIC_UPLINK;
+        }
+
         out->config = pcm_config_incall_music;
         //FIXME: add support for MONO stream configuration when audioflinger mixer supports it
         out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
@@ -698,10 +759,10 @@
 
 
 int voice_extn_compress_voip_set_parameters(struct audio_device *adev,
-                                             struct str_parms *parms) 
+                                            struct str_parms *parms)
 {
     int ret = -ENOSYS;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_set_parameters(adev, parms);
     return ret;
 }
@@ -709,7 +770,7 @@
 void voice_extn_compress_voip_get_parameters(struct str_parms *query,
                                              struct str_parms *reply)
 {
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         compress_voip_get_parameters(query, reply);
 }
 
@@ -718,7 +779,7 @@
                                                  struct str_parms *query,
                                                  struct str_parms *reply)
 {
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         compress_voip_out_get_parameters(out, query, reply);
 }
 
@@ -726,14 +787,14 @@
                                                 struct str_parms *query,
                                                 struct str_parms *reply)
 {
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         compress_voip_in_get_parameters(in, query, reply);
 }
 
 int voice_extn_compress_voip_out_get_buffer_size(struct stream_out *out)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_out_get_buffer_size(out);
     return ret;
 }
@@ -741,7 +802,7 @@
 int voice_extn_compress_voip_in_get_buffer_size(struct stream_in *in)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_in_get_buffer_size(in);
     return ret;
 }
@@ -750,7 +811,7 @@
 int voice_extn_compress_voip_start_output_stream(struct stream_out *out)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_start_output_stream(out);
     return ret;
 }
@@ -759,7 +820,7 @@
 int voice_extn_compress_voip_start_input_stream(struct stream_in *in)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_start_input_stream(in);
     return ret;
 }
@@ -767,7 +828,7 @@
 int voice_extn_compress_voip_close_output_stream(struct audio_stream *stream)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_close_output_stream(stream);
     return ret;
 }
@@ -776,7 +837,7 @@
 int voice_extn_compress_voip_close_input_stream(struct audio_stream *stream)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_close_input_stream(stream);
     return ret;
 }
@@ -784,7 +845,7 @@
 int voice_extn_compress_voip_open_output_stream(struct stream_out *out)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_open_output_stream(out);
     return ret;
 }
@@ -793,7 +854,7 @@
 int voice_extn_compress_voip_open_input_stream(struct stream_in *in)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_open_input_stream(in);
     return ret;
 }
@@ -801,7 +862,7 @@
 int voice_extn_compress_voip_set_volume(struct audio_device *adev, float volume)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_set_volume(adev, volume);
     return ret;
 }
@@ -809,7 +870,7 @@
 int voice_extn_compress_voip_set_mic_mute(struct audio_device *adev, bool state)
 {
     int ret = -1;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_set_mic_mute(adev, state);
     return ret;
 }
@@ -817,7 +878,7 @@
 bool voice_extn_compress_voip_pcm_prop_check()
 {
     bool ret = false;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_pcm_prop_check();
     return ret;
 }
@@ -825,7 +886,7 @@
 bool voice_extn_compress_voip_is_active(const struct audio_device *adev)
 {
     bool ret = false;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_is_active(adev);
     return ret;
 }
@@ -833,7 +894,7 @@
 bool voice_extn_compress_voip_is_format_supported(audio_format_t format)
 {
     bool ret = false;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_is_format_supported(format);
     return ret;
 }
@@ -841,7 +902,7 @@
 bool voice_extn_compress_voip_is_config_supported(struct audio_config *config)
 {
     bool ret = false;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_is_config_supported(config);
     return ret;
 }
@@ -849,7 +910,7 @@
 bool voice_extn_compress_voip_is_started(struct audio_device *adev)
 {
     bool ret = false;
-    if(voice_extn_compress_voip_enabled)
+    if (voice_extn_compress_voip_enabled)
         ret = compress_voip_is_started(adev);
     return ret;
 }
diff --git a/hal/voice_extn/voice_extn.h b/hal/voice_extn/voice_extn.h
index c4f0d49..b2aab7e 100644
--- a/hal/voice_extn/voice_extn.h
+++ b/hal/voice_extn/voice_extn.h
@@ -78,10 +78,14 @@
 bool voice_extn_compress_voip_is_config_supported(struct audio_config *config);
 bool voice_extn_compress_voip_is_started(struct audio_device *adev);
 void voice_extn_feature_init();
-void compr_voip_feature_init(bool is_feature_enabled);
-bool voice_extn_is_compress_voip_supported();
 void dynamic_ecns_feature_init(bool is_feature_enabled);
 bool voice_extn_is_dynamic_ecns_enabled();
+void incall_music_feature_init(bool is_feature_enabled);
+bool voice_extn_is_incall_music_enabled();
+void compr_voip_feature_init(bool is_feature_enabled);
+bool voice_extn_is_compress_voip_supported();
+void multi_voice_session_feature_init(bool is_feature_enabled);
+bool voice_extn_is_multi_session_supported();
 
 
 #endif //VOICE_EXTN_H
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index fb42514..76a42b1 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -142,7 +142,7 @@
 
 ################################################################################
 
-ifneq ($(filter msm8992 msm8994 msm8996 msm8998 sdm660 sdm845 apq8098_latv sdm710 msm8953 msm8937 qcs605 msmnile kona atoll $(MSMSTEPPE) $(TRINKET) lito,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msm8998 sdm660 sdm845 apq8098_latv sdm710 msm8953 msm8937 qcs605 sdmshrike msmnile kona atoll $(MSMSTEPPE) $(TRINKET) lito,$(TARGET_BOARD_PLATFORM)),)
 
 include $(CLEAR_VARS)