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 bcf3e4b..b65a9b6 100644
--- a/configs/atoll/audio_policy_configuration.xml
+++ b/configs/atoll/audio_policy_configuration.xml
@@ -271,20 +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>
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/mixer_paths.xml b/configs/kona/mixer_paths.xml
index 36595c0..201c9fb 100644
--- a/configs/kona/mixer_paths.xml
+++ b/configs/kona/mixer_paths.xml
@@ -2904,4 +2904,80 @@
     <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_qrd.xml b/configs/kona/mixer_paths_qrd.xml
index 65f15db..3cf38a1 100644
--- a/configs/kona/mixer_paths_qrd.xml
+++ b/configs/kona/mixer_paths_qrd.xml
@@ -2313,6 +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 6d14b50..a1e2468 100644
--- a/configs/lito/audio_platform_info.xml
+++ b/configs/lito/audio_platform_info.xml
@@ -211,8 +211,8 @@
         <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"/>
@@ -425,13 +425,13 @@
                         <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 b719ff0..d932652 100644
--- a/configs/lito/audio_policy_configuration.xml
+++ b/configs/lito/audio_policy_configuration.xml
@@ -268,20 +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>
diff --git a/configs/lito/lito.mk b/configs/lito/lito.mk
index 00876db..4022d72 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 4f1aa6f..510fa09 100644
--- a/configs/lito/sound_trigger_platform_info.xml
+++ b/configs/lito/sound_trigger_platform_info.xml
@@ -71,14 +71,14 @@
     <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" />
@@ -112,7 +112,7 @@
             <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" />
@@ -120,6 +120,7 @@
             <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" />
@@ -135,6 +136,7 @@
             <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" !-->
@@ -149,7 +151,6 @@
     <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 -->
@@ -204,7 +205,6 @@
     <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 -->
@@ -257,7 +257,6 @@
     <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 5f861d9..62e75c2 100644
--- a/configs/msm8998/audio_policy_configuration.xml
+++ b/configs/msm8998/audio_policy_configuration.xml
@@ -304,16 +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>
diff --git a/configs/msm8998/msm8998.mk b/configs/msm8998/msm8998.mk
index 7004379..81077c1 100644
--- a/configs/msm8998/msm8998.mk
+++ b/configs/msm8998/msm8998.mk
@@ -252,6 +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 92430bd..1e4e338 100644
--- a/configs/msmnile/audio_policy_configuration.xml
+++ b/configs/msmnile/audio_policy_configuration.xml
@@ -189,7 +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"/>
diff --git a/configs/msmnile/mixer_paths_tavil.xml b/configs/msmnile/mixer_paths_tavil.xml
index cb9d2af..d4ee6fd 100644
--- a/configs/msmnile/mixer_paths_tavil.xml
+++ b/configs/msmnile/mixer_paths_tavil.xml
@@ -3453,4 +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 7d17be7..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
@@ -72,20 +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
-ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
-DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmnile_au/overlay
-endif
+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 \
@@ -104,7 +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/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 e90675c..678e354 100644
--- a/configs/msmsteppe_au/audio_platform_info.xml
+++ b/configs/msmsteppe_au/audio_platform_info.xml
@@ -27,24 +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_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"/>
@@ -141,11 +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 6ab75d8..9fe1345 100644
--- a/configs/msmsteppe_au/audio_policy_configuration.xml
+++ b/configs/msmsteppe_au/audio_policy_configuration.xml
@@ -392,7 +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 75ce9c5..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 -->
diff --git a/configs/msmsteppe_au/msmsteppe_au.mk b/configs/msmsteppe_au/msmsteppe_au.mk
index a0b9cf1..9fac734 100644
--- a/configs/msmsteppe_au/msmsteppe_au.mk
+++ b/configs/msmsteppe_au/msmsteppe_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
@@ -72,20 +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
-ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
-DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmsteppe_au/overlay
-endif
+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 \
@@ -104,7 +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 \
diff --git a/configs/sdm660/audio_policy_configuration.xml b/configs/sdm660/audio_policy_configuration.xml
index e1a0181..5ec3c56 100644
--- a/configs/sdm660/audio_policy_configuration.xml
+++ b/configs/sdm660/audio_policy_configuration.xml
@@ -295,16 +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>
diff --git a/configs/sdm660/sdm660.mk b/configs/sdm660/sdm660.mk
index 03e59e2..b598a2c 100644
--- a/configs/sdm660/sdm660.mk
+++ b/configs/sdm660/sdm660.mk
@@ -263,6 +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 145a811..3f17b95 100644
--- a/configs/sdm710/audio_policy_configuration.xml
+++ b/configs/sdm710/audio_policy_configuration.xml
@@ -304,16 +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>
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index ea71582..5cb6a5a 100644
--- a/configs/sdm710/sdm710.mk
+++ b/configs/sdm710/sdm710.mk
@@ -362,6 +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 fcd0119..65b503c 100644
--- a/configs/sdm845/audio_policy_configuration.xml
+++ b/configs/sdm845/audio_policy_configuration.xml
@@ -298,16 +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>
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 6e56597..77da76c 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -311,6 +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 043df15..a5d7f88 100644
--- a/configs/trinket/audio_policy_configuration.xml
+++ b/configs/trinket/audio_policy_configuration.xml
@@ -268,20 +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>
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.h b/hal/audio_extn/audio_extn.h
index d407f80..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);
@@ -1286,6 +1287,10 @@
 #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);
 int audio_extn_auto_hal_create_audio_patch(struct audio_hw_device *dev,
diff --git a/hal/audio_extn/auto_hal.c b/hal/audio_extn/auto_hal.c
index 7f2163d..304c117 100644
--- a/hal/audio_extn/auto_hal.c
+++ b/hal/audio_extn/auto_hal.c
@@ -65,6 +65,19 @@
     USECASE_AUDIO_PLAYBACK_PHONE,
 };
 
+static struct audio_patch_record *get_patch_from_list(struct audio_device *adev,
+                                                    audio_patch_handle_t patch_id)
+{
+    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;
+    }
+    return NULL;
+}
+
 #define MAX_SOURCE_PORTS_PER_PATCH 1
 #define MAX_SINK_PORTS_PER_PATCH 1
 
@@ -80,6 +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__);
 
@@ -132,6 +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(
@@ -144,7 +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);
@@ -165,6 +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(
@@ -177,20 +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);
@@ -201,6 +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);
 
@@ -209,10 +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;
 }
@@ -264,11 +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;
@@ -280,10 +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__,
@@ -414,20 +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);
@@ -436,18 +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.
diff --git a/hal/audio_extn/ext_hw_plugin.c b/hal/audio_extn/ext_hw_plugin.c
index 619ecfc..41faf03 100644
--- a/hal/audio_extn/ext_hw_plugin.c
+++ b/hal/audio_extn/ext_hw_plugin.c
@@ -205,6 +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;
     }
@@ -247,7 +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 */
@@ -402,7 +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/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 b759f4d..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[] = {
@@ -1532,14 +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
@@ -1615,7 +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) &&
@@ -2705,16 +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
@@ -3374,8 +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);
@@ -3512,6 +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 &
@@ -3611,8 +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__);
@@ -4180,6 +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);
@@ -6107,6 +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;
@@ -6253,6 +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);
@@ -6981,6 +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;
@@ -7155,6 +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)\
@@ -7196,6 +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)))
@@ -8658,6 +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) {
@@ -8704,6 +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 ||
@@ -9571,6 +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 5fffd87..130c017 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -3719,6 +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 &&
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index e3d1b7b..90105cd 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -559,6 +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 0321714..6e7bafd 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 */
@@ -3663,6 +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);
@@ -4981,6 +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 &&
@@ -5866,7 +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)
@@ -6418,7 +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) &&
@@ -8043,7 +8059,9 @@
     }
 
     /* 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);
 
@@ -8626,6 +8644,21 @@
                         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 7b4647a..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
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 473edc2..d278742 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -684,7 +684,16 @@
                                                   struct stream_out *out)
 {
     if(voice_extn_incall_music_enabled) {
-        out->usecase = USECASE_INCALL_MUSIC_UPLINK;
+        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;
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)