Merge "hal: Enable Mixer controls for Internal codec"
diff --git a/Android.mk b/Android.mk
index e590854..b970f53 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,4 +1,4 @@
-ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916 msm8994 msm8992 msm8909 msm8996 msm8952 msm8937 thorium msm8953 msmgold msm8998 sdm660 sdm845 sdm670 apq8098_latv,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916 msm8994 msm8992 msm8909 msm8996 msm8952 msm8937 thorium msm8953 msmgold msm8998 sdm660 sdm845 sdm670 apq8098_latv qcs605,$(TARGET_BOARD_PLATFORM)),)
 
 MY_LOCAL_PATH := $(call my-dir)
 
@@ -14,6 +14,7 @@
 include $(MY_LOCAL_PATH)/mm-audio/Android.mk
 include $(MY_LOCAL_PATH)/visualizer/Android.mk
 include $(MY_LOCAL_PATH)/post_proc/Android.mk
+include $(MY_LOCAL_PATH)/qahw/Android.mk
 include $(MY_LOCAL_PATH)/qahw_api/Android.mk
 endif
 
diff --git a/configs/apq8098_latv/apq8098_latv.mk b/configs/apq8098_latv/apq8098_latv.mk
old mode 100644
new mode 100755
index 7a5f6f4..493d5ce
--- a/configs/apq8098_latv/apq8098_latv.mk
+++ b/configs/apq8098_latv/apq8098_latv.mk
@@ -30,7 +30,7 @@
 AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
 AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
 AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
-AUDIO_FEATURE_ENABLED_3D_AUDIO := true
+AUDIO_FEATURE_ENABLED_3D_AUDIO := false
 AUDIO_FEATURE_ENABLED_VOICE_PRINT := false
 USE_LEGACY_AUDIO_DAEMON := false
 USE_LEGACY_AUDIO_MEASUREMENT := false
@@ -149,6 +149,10 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.offload.buffer.size.kb=2
 
+#Minimum duration for offload playback in secc
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.offload.min.duration.secs=0
+
 #Enable offload audio video playback by default
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.offload.video=true
@@ -216,7 +220,7 @@
 
 #Disable reencode
 PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.qaf.reencode=false
+vendor.audio.qaf.reencode=true
 
 #HDMI passthrough out
 PRODUCT_PROPERTY_OVERRIDES += \
diff --git a/configs/apq8098_latv/audio_policy_configuration.xml b/configs/apq8098_latv/audio_policy_configuration.xml
index 484f96a..a66d99f 100755
--- a/configs/apq8098_latv/audio_policy_configuration.xml
+++ b/configs/apq8098_latv/audio_policy_configuration.xml
@@ -49,15 +49,11 @@
         <!-- Primary Audio HAL -->
         <module name="primary" halVersion="2.0">
             <attachedDevices>
-                <item>Earpiece</item>
-                <item>Speaker</item>
                 <item>Telephony Tx</item>
                 <item>Built-In Mic</item>
                 <item>Built-In Back Mic</item>
-                <item>FM Tuner</item>
-                <item>Telephony Rx</item>
             </attachedDevices>
-            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <defaultOutputDevice>HDMI</defaultOutputDevice>
             <mixPorts>
                 <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -72,48 +68,16 @@
                     <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="compress_passthrough" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
                     <profile name="" format="dynamic"
                              samplingRates="dynamic" channelMasks="dynamic"/>
                 </mixPort>
-                <mixPort name="direct_pcm" role="source"
-                        flags="AUDIO_OUTPUT_FLAG_DIRECT">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800,384000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800,384000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                  <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800,384000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                </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_MP2"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_FLAC"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
-                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
-                    <profile name="" format="AUDIO_FORMAT_ALAC"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_APE"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
-                             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"/>
@@ -132,21 +96,6 @@
                     <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
                              samplingRates="32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_DTS"
-                             samplingRates="32000,44100,48000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_DTS_HD"
-                             samplingRates="32000,44100,48000,64000,88200,96000,128000,176400,192000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_WMA"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_WMA_PRO"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
-                             channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
-                    <profile name="" format="AUDIO_FORMAT_VORBIS"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
-                             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"/>
@@ -157,12 +106,6 @@
                              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="dsd_compress_passthrough" role="source"
-                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
-                    <profile name="" format="AUDIO_FORMAT_DSD"
-                             samplingRates="2822400,5644800"
-                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
-                </mixPort>
                 <mixPort name="voice_tx" role="source">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
@@ -194,10 +137,6 @@
                              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"/>
                 </mixPort>
-                <mixPort name="voice_rx" role="sink">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
-                </mixPort>
                 <mixPort name="mmap_no_irq_in" role="sink" flags="AUDIO_INPUT_FLAG_MMAP_NOIRQ">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -207,42 +146,6 @@
 
             <devicePorts>
                 <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
-                <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" address="">
-                    <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="Line" type="AUDIO_DEVICE_OUT_LINE" 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="BT SCO All" type="AUDIO_DEVICE_OUT_ALL_SCO" 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,AUDIO_CHANNEL_OUT_STEREO"/>
@@ -251,14 +154,6 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="dynamic"/>
                 </devicePort>
-                <devicePort tagName="Proxy" type="AUDIO_DEVICE_OUT_PROXY" role="sink">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="dynamic"/>
-                </devicePort>
-                <devicePort tagName="FM" type="AUDIO_DEVICE_OUT_FM" role="sink">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             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">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
@@ -271,11 +166,6 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
-                <devicePort tagName="USB Device Out" type="AUDIO_DEVICE_OUT_USB_DEVICE" role="sink">
-                    <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"/>
-                </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"
@@ -286,119 +176,38 @@
                              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="FM Tuner" type="AUDIO_DEVICE_IN_FM_TUNER" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="48000"
-                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
-                </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="8000,16000,48000" 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>
             </devicePorts>
             <!-- route declaration, i.e. list all available sources for a given sink -->
             <routes>
-                <route type="mix" sink="Earpiece"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
-                <route type="mix" sink="Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
-                <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx,mmap_no_irq_out"/>
-                <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx,mmap_no_irq_out"/>
-                <route type="mix" sink="Line"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="HDMI"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,compress_passthrough"/>
-                <route type="mix" sink="Proxy"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
-                <route type="mix" sink="FM"
-                       sources="primary output"/>
-                <route type="mix" sink="BT SCO All"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
-                <route type="mix" sink="USB Device Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
+                       sources="primary output,raw,compressed_offload,compress_passthrough"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
-                <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,USB Device In,Telephony Rx"/>
+                       sources="BT SCO Headset Mic"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
-                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
+                       sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="mmap_no_irq_in"
-                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,USB Device In"/>
+                       sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="BT A2DP Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,raw,compressed_offload"/>
                 <route type="mix" sink="BT A2DP Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,raw,compressed_offload"/>
                 <route type="mix" sink="BT A2DP Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,raw,compressed_offload"/>
             </routes>
 
         </module>
 
         <!-- A2DP Audio HAL -->
-        <module name="a2dp" halVersion="2.0">
-            <mixPorts>
-                <mixPort name="a2dp input" role="sink">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
-                </mixPort>
-            </mixPorts>
-
-            <devicePorts>
-                <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
-                </devicePort>
-            </devicePorts>
-
-            <routes>
-                <route type="mix" sink="a2dp input"
-                       sources="BT A2DP In"/>
-            </routes>
-        </module>
-
         <!-- Usb Audio HAL -->
-        <module name="usb" halVersion="2.0">
-            <mixPorts>
-                <mixPort name="usb_accessory output" role="source">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
-                </mixPort>
-            </mixPorts>
-            <devicePorts>
-                <devicePort tagName="USB Host Out" type="AUDIO_DEVICE_OUT_USB_ACCESSORY" role="sink">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
-                </devicePort>
-            </devicePorts>
-            <routes>
-                <route type="mix" sink="USB Host Out"
-                       sources="usb_accessory output"/>
-            </routes>
-        </module>
-
         <!-- Remote Submix Audio HAL -->
-        <xi:include href="/vendor/etc/r_submix_audio_policy_configuration.xml"/>
 
     </modules>
     <!-- End of Modules section -->
diff --git a/configs/msm8998/mixer_paths_skuk.xml b/configs/msm8998/mixer_paths_skuk.xml
index f2e9c9a..6eb0e6f 100644
--- a/configs/msm8998/mixer_paths_skuk.xml
+++ b/configs/msm8998/mixer_paths_skuk.xml
@@ -2479,8 +2479,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/msm8998/mixer_paths_tasha.xml b/configs/msm8998/mixer_paths_tasha.xml
index 4ea726b..64d6383 100644
--- a/configs/msm8998/mixer_paths_tasha.xml
+++ b/configs/msm8998/mixer_paths_tasha.xml
@@ -2820,8 +2820,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/msm8998/mixer_paths_tavil.xml b/configs/msm8998/mixer_paths_tavil.xml
index fc5a440..282e7e6 100644
--- a/configs/msm8998/mixer_paths_tavil.xml
+++ b/configs/msm8998/mixer_paths_tavil.xml
@@ -2607,8 +2607,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths.xml b/configs/sdm660/mixer_paths.xml
index 8cca459..d1b662d 100644
--- a/configs/sdm660/mixer_paths.xml
+++ b/configs/sdm660/mixer_paths.xml
@@ -160,6 +160,7 @@
     <ctl name="I2S TX2 INP1" value="ZERO" />
     <ctl name="I2S TX2 INP2" value="ZERO" />
     <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="ADC1_INP1 Switch" value="0" />
     <ctl name="HDMI Mixer MultiMedia4" value="0" />
     <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="ZERO" />
     <ctl name="INT4_MI2S_RX_VI_FB_STEREO_CH_MUX" value="ZERO" />
@@ -1585,6 +1586,7 @@
     <path name="adc1">
         <ctl name="ADC1 Volume" value="6" />
         <ctl name="DEC1 MUX" value="ADC1" />
+        <ctl name="ADC1_INP1 Switch" value="1" />
     </path>
 
     <path name="adc2">
@@ -2142,8 +2144,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths_mtp.xml b/configs/sdm660/mixer_paths_mtp.xml
index 4f61224..fba8372 100644
--- a/configs/sdm660/mixer_paths_mtp.xml
+++ b/configs/sdm660/mixer_paths_mtp.xml
@@ -159,6 +159,7 @@
     <ctl name="I2S TX2 INP1" value="ZERO" />
     <ctl name="I2S TX2 INP2" value="ZERO" />
     <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="ADC1_INP1 Switch" value="0" />
     <ctl name="HDMI Mixer MultiMedia4" value="0" />
     <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="ZERO" />
     <ctl name="INT4_MI2S_RX_VI_FB_STEREO_CH_MUX" value="ZERO" />
@@ -1704,6 +1705,7 @@
     <path name="adc1">
         <ctl name="ADC1 Volume" value="6" />
         <ctl name="DEC1 MUX" value="ADC1" />
+        <ctl name="ADC1_INP1 Switch" value="1" />
     </path>
 
     <path name="adc2">
@@ -2258,8 +2260,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths_skuk.xml b/configs/sdm660/mixer_paths_skuk.xml
index 15b3f13..faafb89 100644
--- a/configs/sdm660/mixer_paths_skuk.xml
+++ b/configs/sdm660/mixer_paths_skuk.xml
@@ -2369,8 +2369,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths_skus.xml b/configs/sdm660/mixer_paths_skus.xml
index bdd0d61..173e268 100644
--- a/configs/sdm660/mixer_paths_skus.xml
+++ b/configs/sdm660/mixer_paths_skus.xml
@@ -2734,8 +2734,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths_skush.xml b/configs/sdm660/mixer_paths_skush.xml
index aa81e7b..1858ecb 100644
--- a/configs/sdm660/mixer_paths_skush.xml
+++ b/configs/sdm660/mixer_paths_skush.xml
@@ -150,6 +150,7 @@
     <ctl name="INT3_MI2S_TX Channels" value="One" />
     <ctl name="I2S TX2 INP1" value="ZERO" />
     <ctl name="I2S TX2 INP2" value="ZERO" />
+    <ctl name="ADC1_INP1 Switch" value="0" />
     <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="HDMI Mixer MultiMedia4" value="0" />
     <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="ZERO" />
@@ -1762,6 +1763,7 @@
     <path name="adc1">
         <ctl name="ADC1 Volume" value="6" />
         <ctl name="DEC1 MUX" value="ADC1" />
+        <ctl name="ADC1_INP1 Switch" value="1" />
     </path>
 
     <path name="adc2">
@@ -2308,8 +2310,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths_wcd9326.xml b/configs/sdm660/mixer_paths_wcd9326.xml
index 9e6623d..b53bd51 100644
--- a/configs/sdm660/mixer_paths_wcd9326.xml
+++ b/configs/sdm660/mixer_paths_wcd9326.xml
@@ -2679,8 +2679,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths_wcd9335.xml b/configs/sdm660/mixer_paths_wcd9335.xml
index d4c098f..a05f3cb 100644
--- a/configs/sdm660/mixer_paths_wcd9335.xml
+++ b/configs/sdm660/mixer_paths_wcd9335.xml
@@ -2701,8 +2701,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm660/mixer_paths_wcd9340.xml b/configs/sdm660/mixer_paths_wcd9340.xml
index 6c6f2cf..8f68011 100644
--- a/configs/sdm660/mixer_paths_wcd9340.xml
+++ b/configs/sdm660/mixer_paths_wcd9340.xml
@@ -2448,8 +2448,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm670/audio_output_policy.conf b/configs/sdm670/audio_output_policy.conf
index 76039c7..52ef791 100644
--- a/configs/sdm670/audio_output_policy.conf
+++ b/configs/sdm670/audio_output_policy.conf
@@ -25,6 +25,13 @@
     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
diff --git a/configs/sdm670/audio_platform_info.xml b/configs/sdm670/audio_platform_info.xml
index 71a8bf6..0597623 100644
--- a/configs/sdm670/audio_platform_info.xml
+++ b/configs/sdm670/audio_platform_info.xml
@@ -27,6 +27,7 @@
 <audio_platform_info>
     <acdb_ids>
         <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="15"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_REVERSE" acdb_id="15"/>
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED" acdb_id="124"/>
         <device name="SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE" acdb_id="131"/>
         <device name="SND_DEVICE_IN_VOICE_REC_TMIC" acdb_id="131"/>
@@ -40,6 +41,11 @@
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1" acdb_id="10"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2" acdb_id="10"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET" acdb_id="45"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_MIC" acdb_id="143"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_STEREO_MIC" acdb_id="144"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_THREE_MIC" acdb_id="145"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_QUAD_MIC" acdb_id="146"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" acdb_id="147"/>
     </acdb_ids>
     <module_ids>
         <aec>
@@ -72,7 +78,6 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD6" type="out" id="30"/>
         <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD7" type="out" id="31"/>
         <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD8" type="out" id="32"/>
-        <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD9" type="out" id="33"/>
         <usecase name="USECASE_VOICEMMODE1_CALL" type="in" id="2"/>
         <usecase name="USECASE_VOICEMMODE1_CALL" type="out" id="2"/>
         <usecase name="USECASE_VOICEMMODE2_CALL" type="in" id="19"/>
@@ -88,8 +93,12 @@
         <usecase name="USECASE_AUDIO_RECORD_LOW_LATENCY" type="in" id="17" />
         <usecase name="USECASE_AUDIO_PLAYBACK_ULL" type="out" id="17" />
         <usecase name="USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE" type="out" id="27" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_MMAP" type="out" id="33" />
+        <usecase name="USECASE_AUDIO_RECORD_MMAP" type="in" id="33" />
         <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="35" />
         <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="35" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="16" />
+        <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="16" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsatz.13"/>
@@ -132,6 +141,8 @@
         <device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO" backend="speaker-and-bt-sco" interface="SLIMBUS_0_RX-and-SLIMBUS_7_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB" backend="speaker-and-bt-sco-wb" interface="SLIMBUS_0_RX-and-SLIMBUS_7_RX"/>
     </backend_names>
 </audio_platform_info>
 
diff --git a/configs/sdm670/audio_platform_info_intcodec.xml b/configs/sdm670/audio_platform_info_intcodec.xml
index d04ced2..94f3853 100644
--- a/configs/sdm670/audio_platform_info_intcodec.xml
+++ b/configs/sdm670/audio_platform_info_intcodec.xml
@@ -42,7 +42,6 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD6" type="out" id="30"/>
         <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD7" type="out" id="31"/>
         <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD8" type="out" id="32"/>
-        <usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD9" type="out" id="33"/>
         <usecase name="USECASE_VOICEMMODE1_CALL" type="in" id="2"/>
         <usecase name="USECASE_VOICEMMODE1_CALL" type="out" id="2"/>
         <usecase name="USECASE_VOICEMMODE2_CALL" type="in" id="19"/>
@@ -58,8 +57,12 @@
         <usecase name="USECASE_AUDIO_RECORD_LOW_LATENCY" type="in" id="17" />
         <usecase name="USECASE_AUDIO_PLAYBACK_ULL" type="out" id="17" />
         <usecase name="USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE" type="out" id="27" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_MMAP" type="out" id="33" />
+        <usecase name="USECASE_AUDIO_RECORD_MMAP" type="in" id="33" />
         <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="12" />
         <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="12" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="16" />
+        <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="16" />
     </pcm_ids>
     <config_params>
         <!-- In the below value string, the value indicates default mono -->
@@ -74,6 +77,7 @@
     </config_params>
     <acdb_ids>
         <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="15"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_REVERSE" acdb_id="15"/>
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED" acdb_id="124"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED" acdb_id="101"/>
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT" acdb_id="124"/>
@@ -88,6 +92,11 @@
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1" acdb_id="10"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2" acdb_id="10"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET" acdb_id="45"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_MIC" acdb_id="143"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_STEREO_MIC" acdb_id="144"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_THREE_MIC" acdb_id="145"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_QUAD_MIC" acdb_id="146"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" acdb_id="147"/>
     </acdb_ids>
     <backend_names>
         <device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="INT0_MI2S_RX"/>
@@ -197,6 +206,8 @@
         <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" interface="INT3_MI2S_TX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_AND_VOICE_HEADPHONES" backend="speaker-and-headphones" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_AND_VOICE_ANC_HEADSET" backend="speaker-and-headphones" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO" backend="speaker-and-bt-sco" interface="INT4_MI2S_RX-and-SLIMBUS_7_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB" backend="speaker-and-bt-sco-wb" interface="INT4_MI2S_RX-and-SLIMBUS_7_RX"/>
         </backend_names>
 </audio_platform_info>
 
diff --git a/configs/sdm670/audio_platform_info_skuw.xml b/configs/sdm670/audio_platform_info_skuw.xml
index 475a441..1107d4a 100644
--- a/configs/sdm670/audio_platform_info_skuw.xml
+++ b/configs/sdm670/audio_platform_info_skuw.xml
@@ -76,6 +76,7 @@
     </config_params>
     <acdb_ids>
         <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="14"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_REVERSE" acdb_id="15"/>
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED" acdb_id="136"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED" acdb_id="136"/>
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT" acdb_id="136"/>
@@ -90,6 +91,11 @@
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1" acdb_id="10"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2" acdb_id="10"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET" acdb_id="45"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_MIC" acdb_id="143"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_STEREO_MIC" acdb_id="144"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_THREE_MIC" acdb_id="145"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_QUAD_MIC" acdb_id="146"/>
+        <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" acdb_id="147"/>
     </acdb_ids>
     <backend_names>
         <device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="INT0_MI2S_RX"/>
@@ -199,6 +205,8 @@
         <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" interface="INT3_MI2S_TX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_AND_VOICE_HEADPHONES" backend="speaker-and-headphones" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_AND_VOICE_ANC_HEADSET" backend="speaker-and-headphones" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO" backend="speaker-and-bt-sco" interface="INT4_MI2S_RX-and-SLIMBUS_7_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB" backend="speaker-and-bt-sco-wb" interface="INT4_MI2S_RX-and-SLIMBUS_7_RX"/>
         </backend_names>
 </audio_platform_info>
 
diff --git a/configs/sdm670/audio_policy_configuration.xml b/configs/sdm670/audio_policy_configuration.xml
index 418cf42..9bb9486 100644
--- a/configs/sdm670/audio_policy_configuration.xml
+++ b/configs/sdm670/audio_policy_configuration.xml
@@ -73,6 +73,10 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
+                <mixPort name="mmap_no_irq_out" role="source" flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_MMAP_NOIRQ">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
                 <mixPort name="hifi_playback" role="source" />
                 <mixPort name="compress_passthrough" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
@@ -192,6 +196,11 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
                 </mixPort>
+                <mixPort name="mmap_no_irq_in" role="sink" flags="AUDIO_INPUT_FLAG_MMAP_NOIRQ">
+                    <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,AUDIO_CHANNEL_INDEX_MASK_3"/>
+                </mixPort>
                 <mixPort name="hifi_input" role="sink" />
             </mixPorts>
 
@@ -304,15 +313,15 @@
             <!-- route declaration, i.e. list all available sources for a given sink -->
             <routes>
                 <route type="mix" sink="Earpiece"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="Line"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="HDMI"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,compress_passthrough"/>
                 <route type="mix" sink="Proxy"
@@ -322,7 +331,7 @@
                 <route type="mix" sink="BT SCO All"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="USB Device Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,hifi_playback"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out,hifi_playback"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
                 <route type="mix" sink="voice_rx"
@@ -333,12 +342,14 @@
                        sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="record_24"
                        sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
+                <route type="mix" sink="mmap_no_irq_in"
+                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,USB Device In"/>
                 <route type="mix" sink="BT A2DP Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,deep_buffer,direct_pcm,compressed_offload"/>
                 <route type="mix" sink="BT A2DP Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,deep_buffer,direct_pcm,compressed_offload"/>
                 <route type="mix" sink="BT A2DP Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,deep_buffer,direct_pcm,compressed_offload"/>
                 <route type="mix" sink="hifi_input" sources="USB Device In" />
             </routes>
 
diff --git a/configs/sdm670/mixer_paths.xml b/configs/sdm670/mixer_paths.xml
index 5f81e65..c526980 100644
--- a/configs/sdm670/mixer_paths.xml
+++ b/configs/sdm670/mixer_paths.xml
@@ -158,6 +158,7 @@
     <ctl name="INT3_MI2S_TX Channels" value="One" />
     <ctl name="I2S TX2 INP1" value="ZERO" />
     <ctl name="I2S TX2 INP2" value="ZERO" />
+    <ctl name="ADC1_INP1 Switch" value="0" />
     <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="HDMI Mixer MultiMedia4" value="0" />
     <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="ZERO" />
@@ -1584,6 +1585,7 @@
     <path name="adc1">
         <ctl name="ADC1 Volume" value="6" />
         <ctl name="DEC1 MUX" value="ADC1" />
+        <ctl name="ADC1_INP1 Switch" value="1" />
     </path>
 
     <path name="adc2">
@@ -2141,8 +2143,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm670/mixer_paths_360cam.xml b/configs/sdm670/mixer_paths_360cam.xml
new file mode 100755
index 0000000..8ef7501
--- /dev/null
+++ b/configs/sdm670/mixer_paths_360cam.xml
@@ -0,0 +1,2336 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!-- Copyright (c) 2015-2017, 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.                          -->
+<mixer>
+    <!-- These are the initial mixer settings -->
+    <ctl name="Voice Rx Device Mute" id="0" value="0" />
+    <ctl name="Voice Rx Device Mute" id="1" value="-1" />
+    <ctl name="Voice Rx Device Mute" id="2" value="20" />
+    <ctl name="Voice Tx Mute" id="0" value="0" />
+    <ctl name="Voice Tx Mute" id="1" value="-1" />
+    <ctl name="Voice Tx Mute" id="2" value="500" />
+    <ctl name="Voice Rx Gain" id="0" value="0" />
+    <ctl name="Voice Rx Gain" id="1" value="-1" />
+    <ctl name="Voice Rx Gain" id="2" value="20" />
+    <ctl name="Voip Tx Mute" id="0" value="0" />
+    <ctl name="Voip Tx Mute" id="1" value="500" />
+    <ctl name="Voip Rx Gain" id="0" value="0" />
+    <ctl name="Voip Rx Gain" id="1" value="20" />
+    <ctl name="Voip Mode Config" value="12" />
+    <ctl name="Voip Rate Config" value="0" />
+    <ctl name="Voip Evrc Min Max Rate Config" id="0" value="1" />
+    <ctl name="Voip Evrc Min Max Rate Config" id="1" value="4" />
+    <ctl name="Voip Dtx Mode" value="0" />
+    <ctl name="TTY Mode" value="OFF" />
+    <ctl name="INT4_MI2S_RX Port Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="INT0_MI2S_RX Port Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia5 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="0" />
+    <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="0" />
+    <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia1 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+    <ctl name="HDMI Mixer MultiMedia1" value="0" />
+    <ctl name="HDMI Mixer MultiMedia2" value="0" />
+    <ctl name="HDMI Mixer MultiMedia3" value="0" />
+    <ctl name="HDMI Mixer MultiMedia4" value="0" />
+    <ctl name="HDMI Mixer MultiMedia5" value="0" />
+    <ctl name="HDMI Mixer MultiMedia7" value="0" />
+    <ctl name="HDMI Mixer MultiMedia9" value="0" />
+    <ctl name="HDMI Mixer MultiMedia10" value="0" />
+    <ctl name="HDMI Mixer MultiMedia11" value="0" />
+    <ctl name="HDMI Mixer MultiMedia12" value="0" />
+    <ctl name="HDMI Mixer MultiMedia13" value="0" />
+    <ctl name="HDMI Mixer MultiMedia14" value="0" />
+    <ctl name="HDMI Mixer MultiMedia15" value="0" />
+    <ctl name="HDMI Mixer MultiMedia16" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia6" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia8" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="0" />
+    <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="0" />
+    <ctl name="Display Port RX Bit Format" value="S16_LE" />
+    <ctl name="Display Port RX SampleRate" value="KHZ_48" />
+    <ctl name="Display Port RX Channels" value="Two" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia2" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia2" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia3" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia3" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia3" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
+    <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
+    <ctl name="USB_AUDIO_RX Channels" value="Two" />
+    <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
+    <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
+    <ctl name="USB_AUDIO_TX Channels" value="One" />
+    <ctl name="MultiMedia6 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="INT4_MI2S_RX Channels" value="One" />
+    <ctl name="INT0_MI2S_RX Channels" value="One" />
+    <ctl name="INT3_MI2S_TX Channels" value="One" />
+    <ctl name="I2S TX2 INP1" value="ZERO" />
+    <ctl name="I2S TX2 INP2" value="ZERO" />
+    <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="ZERO" />
+    <ctl name="INT4_MI2S_RX_VI_FB_STEREO_CH_MUX" value="ZERO" />
+    <ctl name="VI_FEED_TX Channels" value="One" />
+    <ctl name="AIF1_VI_SDW Mixer SPKR_VI_1" value="0" />
+    <ctl name="AIF1_VI_SDW Mixer SPKR_VI_2" value="0" />
+    <ctl name="INT5 MI2S VI MONO" value="Left" />
+    <!-- HFP start -->
+    <ctl name="HFP_SLIM7_UL_HL Switch" value="0" />
+    <ctl name="INT4_MI2S_RX Port Mixer SLIM_7_TX" value="0" />
+    <ctl name="INT0_MI2S_RX Port Mixer SLIM_7_TX" value="0" />
+    <!-- HFP end -->
+    <!-- echo reference -->
+    <ctl name="AUDIO_REF_EC_UL1 MUX" value="None" />
+    <!-- usb headset -->
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="MultiMedia1 Mixer AFE_PCM_TX" value="0" />
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
+    <!-- usb headset end -->
+    <!-- fm -->
+    <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
+    <ctl name="INT4_MI2S_RX Port Mixer SLIM_8_TX" value="0" />
+    <ctl name="INT4_MI2S_RX_DL_HL Switch" value="0" />
+    <ctl name="INT0_MI2S_RX Port Mixer SLIM_8_TX" value="0" />
+    <ctl name="INT0_MI2S_RX_DL_HL Switch" value="0" />
+    <ctl name="MultiMedia1 Mixer SLIM_8_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer SLIM_8_TX" value="0" />
+    <!-- fm end -->
+
+    <!-- Multimode Voice1 -->
+    <ctl name="INT4_MI2S_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="INT0_MI2S_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="VoiceMMode1_Tx Mixer INT3_MI2S_TX_MMode1" value="0" />
+    <!-- Multimode Voice1 HDMI -->
+    <ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="0" />
+    <!-- Multimode Voice1 BTSCO -->
+    <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="0" />
+    <!-- Multimode Voice1 USB headset -->
+    <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="VoiceMMode1_Tx Mixer AFE_PCM_TX_MMode1" value="0" />
+    <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="0" />
+    <!-- Miltimode Voice1 end-->
+
+    <!-- Multimode Voice2 -->
+    <ctl name="INT4_MI2S_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="INT0_MI2S_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="VoiceMMode2_Tx Mixer INT3_MI2S_TX_MMode2" value="0" />
+    <!-- Multimode Voice2 HDMI -->
+    <ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="0" />
+    <!-- Multimode Voice2 BTSCO -->
+    <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="0" />
+    <!-- Multimode Voice2 USB headset -->
+    <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="VoiceMMode2_Tx Mixer AFE_PCM_TX_MMode2" value="0" />
+    <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="0" />
+    <!-- Multimode Voice2 end-->
+
+    <!-- Voice external ec. reference -->
+    <ctl name="VOC_EXT_EC MUX" value="NONE" />
+    <!-- Voice external ec. reference end -->
+
+    <!-- RT Proxy Cal -->
+    <ctl name="RT_PROXY_1_RX SetCalMode" value="CAL_MODE_NONE" />
+    <ctl name="RT_PROXY_1_TX SetCalMode" value="CAL_MODE_NONE" />
+    <!-- RT Proxy Cal end -->
+
+    <!-- Incall Recording -->
+    <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+    <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
+    <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
+    <!-- Incall Recording End -->
+
+    <!-- Incall Music -->
+    <ctl name="Incall_Music Audio Mixer MultiMedia2" value="0" />
+    <!-- Incall Music End -->
+
+    <!-- compress-voip-call start -->
+    <ctl name="INT4_MI2S_RX_Voice Mixer Voip" value="0" />
+    <ctl name="INT0_MI2S_RX_Voice Mixer Voip" value="0" />
+    <ctl name="Voip_Tx Mixer INT3_MI2S_TX_Voip" value="0" />
+    <ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
+    <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
+    <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
+    <ctl name="Voip_Tx Mixer AFE_PCM_TX_Voip" value="0" />
+    <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="0" />
+    <ctl name="Voip_Tx Mixer USB_AUDIO_TX_Voip" value="0" />
+    <!-- compress-voip-call end-->
+
+    <!-- Audio BTSCO -->
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia6" value="0" />
+    <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+    <!-- audio record compress-->
+    <ctl name="MultiMedia8 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="0" />
+    <!-- audio record compress end-->
+
+    <!-- split a2dp -->
+    <ctl name="BT SampleRate" value="KHZ_8" />
+    <ctl name="AFE Input Channels" value="Zero" />
+    <ctl name="SLIM7_RX ADM Channels" value="Zero" />
+    <!-- split a2dp end-->
+
+    <!-- ADSP testfwk -->
+    <ctl name="INT4_MI2S_RX_DL_HL Switch" value="0" />
+    <ctl name="INT0_MI2S_RX_DL_HL Switch" value="0" />
+    <!-- ADSP testfwk end-->
+
+    <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
+
+    <!-- Codec controls -->
+    <!-- WSA controls -->
+    <ctl name="SpkrLeft COMP Switch" value="0" />
+    <ctl name="SpkrRight COMP Switch" value="0" />
+    <ctl name="SpkrLeft BOOST Switch" value="0" />
+    <ctl name="SpkrRight BOOST Switch" value="0" />
+    <ctl name="SpkrLeft VISENSE Switch" value="0" />
+    <ctl name="SpkrRight VISENSE Switch" value="0" />
+    <ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
+    <ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
+    <ctl name="SpkrLeft WSA PA Gain" value="G_0_DB" />
+    <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
+
+    <!-- EAR controls -->
+    <ctl name="EAR_S" value="ZERO" />
+
+    <!-- Volume controls -->
+    <ctl name="HPHL Volume" value="9" />
+    <ctl name="HPHR Volume" value="9" />
+    <ctl name="EAR PA Gain" value="POS_1P5_DB" />
+    <ctl name="EAR PA Boost" value="DISABLE" />
+
+    <ctl name="RX1 Digital Volume" value="84" />
+    <ctl name="RX2 Digital Volume" value="84" />
+    <ctl name="RX3 Digital Volume" value="84" />
+    <ctl name="RX4 Digital Volume" value="84" />
+    <ctl name="RX5 Digital Volume" value="84" />
+    <ctl name="ADC1 Volume" value="4" />
+    <ctl name="ADC2 Volume" value="4" />
+    <ctl name="ADC3 Volume" value="4" />
+    <ctl name="DEC1 Volume" value="84" />
+    <ctl name="DEC2 Volume" value="84" />
+    <ctl name="DEC3 Volume" value="84" />
+    <ctl name="DEC4 Volume" value="84" />
+
+    <!-- Compander controls -->
+    <ctl name="COMP1 Switch" value="1" />
+    <ctl name="COMP2 Switch" value="1" />
+
+    <!-- MUX controls -->
+    <ctl name="DEC1 MUX" value="ZERO" />
+    <ctl name="DEC2 MUX" value="ZERO" />
+    <ctl name="DEC3 MUX" value="ZERO" />
+    <ctl name="DEC4 MUX" value="ZERO" />
+    <ctl name="ADC2 MUX" value="ZERO" />
+    <ctl name="RDAC2 MUX" value="ZERO" />
+    <ctl name="ADC1_INP1 Switch" value="0" />
+
+    <!-- Interpolator chain controls -->
+    <!-- ADD RX Chain Controls -->
+
+    <ctl name="RX1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX2 MIX1 INP1" value="ZERO" />
+    <ctl name="RDAC2 MUX" value="ZERO" />
+    <ctl name="HPHL" value="ZERO" />
+    <ctl name="HPHR" value="ZERO" />
+    <ctl name="RX1 MIX2 INP1" value="ZERO" />
+    <ctl name="RX2 MIX2 INP1" value="ZERO" />
+    <ctl name="RX3 MIX1 INP1" value="ZERO" />
+    <ctl name="RX4 MIX1 INP1" value="ZERO" />
+    <ctl name="RX5 MIX1 INP1" value="ZERO" />
+    <ctl name="RX HPH Mode" value="NORMAL" />
+    <ctl name="COMP0 RX1" value="0" />
+    <ctl name="COMP0 RX2" value="0" />
+
+    <!-- IIR/voice anc -->
+    <ctl name="IIR1 Band1" id ="0" value="268435456" />
+    <ctl name="IIR1 Band1" id ="1" value="0" />
+    <ctl name="IIR1 Band1" id ="2" value="0" />
+    <ctl name="IIR1 Band1" id ="3" value="0" />
+    <ctl name="IIR1 Band1" id ="4" value="0" />
+    <ctl name="IIR1 Band2" id ="0" value="268435456" />
+    <ctl name="IIR1 Band2" id ="1" value="0" />
+    <ctl name="IIR1 Band2" id ="2" value="0" />
+    <ctl name="IIR1 Band2" id ="3" value="0" />
+    <ctl name="IIR1 Band2" id ="4" value="0" />
+    <ctl name="IIR1 Band3" id ="0" value="268435456" />
+    <ctl name="IIR1 Band3" id ="1" value="0" />
+    <ctl name="IIR1 Band3" id ="2" value="0" />
+    <ctl name="IIR1 Band3" id ="3" value="0" />
+    <ctl name="IIR1 Band3" id ="4" value="0" />
+    <ctl name="IIR1 Band4" id ="0" value="268435456" />
+    <ctl name="IIR1 Band4" id ="1" value="0" />
+    <ctl name="IIR1 Band4" id ="2" value="0" />
+    <ctl name="IIR1 Band4" id ="3" value="0" />
+    <ctl name="IIR1 Band4" id ="4" value="0" />
+    <ctl name="IIR1 Band5" id ="0" value="268435456" />
+    <ctl name="IIR1 Band5" id ="1" value="0" />
+    <ctl name="IIR1 Band5" id ="2" value="0" />
+    <ctl name="IIR1 Band5" id ="3" value="0" />
+    <ctl name="IIR1 Band5" id ="4" value="0" />
+    <ctl name="IIR1 Enable Band1" value="0" />
+    <ctl name="IIR1 Enable Band2" value="0" />
+    <ctl name="IIR1 Enable Band3" value="0" />
+    <ctl name="IIR1 Enable Band4" value="0" />
+    <ctl name="IIR1 Enable Band5" value="0" />
+    <ctl name="IIR1 INP1 Volume" value="53" />
+    <ctl name="IIR1 INP2 Volume" value="53" />
+    <ctl name="IIR1 INP3 Volume" value="53" />
+    <ctl name="IIR1 INP4 Volume" value="53" />
+    <ctl name="IIR1 INP1 MUX" value="ZERO" />
+
+    <!-- anc related -->
+
+    <!-- Codec controls end -->
+
+    <!-- These are audio route (FE to BE) specific mixer settings -->
+    <path name="gsm-mode">
+        <ctl name="GSM mode Enable" value="ON" />
+    </path>
+
+    <path name="echo-reference speaker-vbat-mono">
+    </path>
+
+    <path name="echo-reference speaker-vbat-mono-2">
+    </path>
+
+    <path name="echo-reference speaker-vbat">
+    </path>
+
+    <path name="echo-reference">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="INT4_MI2S_RX" />
+    </path>
+
+    <path name="echo-reference headphones">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="INT0_MI2S_RX" />
+    </path>
+
+    <path name="echo-reference headphones-44.1">
+    </path>
+
+    <path name="deep-buffer-playback">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-protected">
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-hdmi">
+        <path name="deep-buffer-playback hdmi" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-display-port">
+        <path name="deep-buffer-playback display-port" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="deep-buffer-playback bt-sco" />
+    </path>
+
+    <path name="deep-buffer-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-usb-headphones">
+        <path name="deep-buffer-playback usb-headphones" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-headphones">
+        <path name="deep-buffer-playback headphones" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="low-latency-playback">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback speaker-protected">
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="low-latency-playback bt-sco" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-hdmi">
+        <path name="low-latency-playback hdmi" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-display-port">
+        <path name="low-latency-playback display-port" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-usb-headphones">
+        <path name="low-latency-playback usb-headphones" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-headphones">
+        <path name="low-latency-playback headphones" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="audio-ull-playback">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback speaker-protected">
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-headphones">
+        <path name="audio-ull-playback" />
+        <path name="audio-ull-playback headphones" />
+    </path>
+
+    <path name="audio-ull-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="audio-ull-playback bt-sco" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-hdmi">
+        <path name="audio-ull-playback hdmi" />
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-display-port">
+        <path name="audio-ull-playback display-port" />
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="multi-channel-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="multi-channel-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="multi-channel-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="compress-offload-playback">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback speaker-protected">
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="silence-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia9" value="1" />
+    </path>
+
+    <path name="compress-offload-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="silence-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="1" />
+    </path>
+
+    <path name="compress-offload-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-hdmi">
+        <path name="compress-offload-playback hdmi" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-display-port">
+        <path name="compress-offload-playback display-port" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-usb-headphones">
+        <path name="compress-offload-playback usb-headphones" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback speaker-and-headphones">
+        <path name="compress-offload-playback headphones" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback2">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 hdmi">
+        <ctl name="HDMI Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback2 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-hdmi">
+        <path name="compress-offload-playback2 hdmi" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-display-port">
+        <path name="compress-offload-playback2 display-port" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-usb-headphones">
+        <path name="compress-offload-playback2 usb-headphones" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-headphones">
+        <path name="compress-offload-playback2 headphones" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback3">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 hdmi">
+        <ctl name="HDMI Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback3 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-hdmi">
+        <path name="compress-offload-playback3 hdmi" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-display-port">
+        <path name="compress-offload-playback3 display-port" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-usb-headphones">
+        <path name="compress-offload-playback3 usb-headphones" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-headphones">
+        <path name="compress-offload-playback3 headphones" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback4">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 hdmi">
+        <ctl name="HDMI Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback4 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-hdmi">
+        <path name="compress-offload-playback4 hdmi" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-display-port">
+        <path name="compress-offload-playback4 display-port" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+
+    <path name="compress-offload-playback4 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-usb-headphones">
+        <path name="compress-offload-playback4 usb-headphones" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback4 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-headphones">
+        <path name="compress-offload-playback4 headphones" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback5">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 hdmi">
+        <ctl name="HDMI Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback5 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-hdmi">
+        <path name="compress-offload-playback5 hdmi" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-display-port">
+        <path name="compress-offload-playback5 display-port" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-usb-headphones">
+        <path name="compress-offload-playback5 usb-headphones" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-headphones">
+        <path name="compress-offload-playback5 headphones" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback6">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 hdmi">
+        <ctl name="HDMI Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback6 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-hdmi">
+        <path name="compress-offload-playback6 hdmi" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-display-port">
+        <path name="compress-offload-playback6 display-port" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-usb-headphones">
+        <path name="compress-offload-playback6 usb-headphones" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-headphones">
+        <path name="compress-offload-playback6 headphones" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback7">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 hdmi">
+        <ctl name="HDMI Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback7 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-hdmi">
+        <path name="compress-offload-playback7 hdmi" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-display-port">
+        <path name="compress-offload-playback7 display-port" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-usb-headphones">
+        <path name="compress-offload-playback7 usb-headphones" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-headphones">
+        <path name="compress-offload-playback7 headphones" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback8">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 hdmi">
+        <ctl name="HDMI Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback8 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-hdmi">
+        <path name="compress-offload-playback8 hdmi" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-display-port">
+        <path name="compress-offload-playback8 display-port" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-usb-headphones">
+        <path name="compress-offload-playback8 usb-headphones" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-headphones">
+        <path name="compress-offload-playback8 headphones" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback9">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 handset">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 hdmi">
+        <ctl name="HDMI Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-offload-playback9 bt-sco" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-hdmi">
+        <path name="compress-offload-playback9 hdmi" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-display-port">
+        <path name="compress-offload-playback9 display-port" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-usb-headphones">
+        <path name="compress-offload-playback9 usb-headphones" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 headphones-44.1">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 headphones-dsd">
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-headphones">
+        <path name="compress-offload-playback9 headphones" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="audio-record">
+        <ctl name="MultiMedia1 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record usb-headset-mic">
+        <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
+    <path name="audio-record bt-sco">
+        <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="audio-record bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="audio-record bt-sco" />
+    </path>
+
+    <path name="audio-record capture-fm">
+        <ctl name="MultiMedia1 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="audio-record-compress">
+        <ctl name="MultiMedia8 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-compress bt-sco">
+        <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="audio-record-compress bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="audio-record-compress bt-sco" />
+    </path>
+
+    <path name="audio-record-compress usb-headset-mic">
+        <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
+    <path name="low-latency-record">
+      <ctl name="MultiMedia8 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="low-latency-record bt-sco">
+      <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="low-latency-record bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="low-latency-record bt-sco" />
+    </path>
+
+    <path name="low-latency-record usb-headset-mic">
+        <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
+    <path name="low-latency-record capture-fm">
+      <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="fm-virtual-record capture-fm">
+        <ctl name="MultiMedia2 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="play-fm">
+        <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+        <ctl name="INT4_MI2S_RX Port Mixer SLIM_8_TX" value="1" />
+        <ctl name="INT4_MI2S_RX_DL_HL Switch" value="1" />
+    </path>
+
+    <path name="play-fm headphones">
+        <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+        <ctl name="INT0_MI2S_RX Port Mixer SLIM_8_TX" value="1" />
+        <ctl name="INT0_MI2S_RX_DL_HL Switch" value="1" />
+    </path>
+
+    <path name="incall-rec-uplink">
+        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+    </path>
+
+    <path name="incall-rec-uplink bt-sco">
+        <path name="incall-rec-uplink" />
+    </path>
+
+    <path name="incall-rec-uplink bt-sco-wb">
+        <path name="incall-rec-uplink" />
+    </path>
+
+    <path name="incall-rec-uplink usb-headset-mic">
+        <path name="incall-rec-uplink" />
+    </path>
+
+    <path name="incall-rec-uplink afe-proxy">
+        <path name="incall-rec-uplink" />
+    </path>
+
+    <path name="incall-rec-uplink-compress">
+        <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="1" />
+    </path>
+
+    <path name="incall-rec-uplink-compress bt-sco">
+        <path name="incall-rec-uplink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-compress bt-sco-wb">
+        <path name="incall-rec-uplink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-compress usb-headset-mic">
+        <path name="incall-rec-uplink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-compress afe-proxy">
+        <path name="incall-rec-uplink-compress" />
+    </path>
+
+    <path name="incall-rec-downlink">
+        <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
+    </path>
+
+    <path name="incall-rec-downlink bt-sco">
+        <path name="incall-rec-downlink" />
+    </path>
+
+    <path name="incall-rec-downlink bt-sco-wb">
+        <path name="incall-rec-downlink" />
+    </path>
+
+    <path name="incall-rec-downlink usb-headset-mic">
+        <path name="incall-rec-downlink" />
+    </path>
+
+    <path name="incall-rec-downlink afe-proxy">
+        <path name="incall-rec-downlink" />
+    </path>
+
+    <path name="incall-rec-downlink-compress">
+        <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="1" />
+    </path>
+
+    <path name="incall-rec-downlink-compress bt-sco">
+        <path name="incall-rec-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-downlink-compress bt-sco-wb">
+        <path name="incall-rec-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-downlink-compress usb-headset-mic">
+        <path name="incall-rec-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-downlink-compress afe-proxy">
+        <path name="incall-rec-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink">
+        <path name="incall-rec-uplink" />
+        <path name="incall-rec-downlink" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink bt-sco">
+        <path name="incall-rec-uplink-and-downlink" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink bt-sco-wb">
+        <path name="incall-rec-uplink-and-downlink" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink usb-headset-mic">
+        <path name="incall-rec-uplink-and-downlink" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink afe-proxy">
+        <path name="incall-rec-uplink-and-downlink" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink-compress">
+        <path name="incall-rec-uplink-compress" />
+        <path name="incall-rec-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink-compress bt-sco">
+        <path name="incall-rec-uplink-and-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink-compress bt-sco-wb">
+        <path name="incall-rec-uplink-and-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink-compress usb-headset-mic">
+        <path name="incall-rec-uplink-and-downlink-compress" />
+    </path>
+
+    <path name="incall-rec-uplink-and-downlink-compress afe-proxy">
+        <path name="incall-rec-uplink-and-downlink-compress" />
+    </path>
+
+    <path name="hfp-sco">
+        <ctl name="HFP_SLIM7_UL_HL Switch" value="1" />
+        <ctl name="INT4_MI2S_RX Port Mixer SLIM_7_TX" value="1" />
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia6" value="1" />
+        <ctl name="MultiMedia6 Mixer INT3_MI2S_TX" value="1" />
+        <ctl name="INT4_MI2S_RX_DL_HL Switch" value="1" />
+   </path>
+
+    <path name="hfp-sco headphones">
+        <ctl name="HFP_SLIM7_UL_HL Switch" value="1" />
+        <ctl name="INT0_MI2S_RX Port Mixer SLIM_7_TX" value="1" />
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia6" value="1" />
+        <ctl name="MultiMedia6 Mixer INT3_MI2S_TX" value="1" />
+        <ctl name="INT0_MI2S_RX_DL_HL Switch" value="1" />
+    </path>
+
+   <path name="hfp-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="hfp-sco" />
+   </path>
+
+    <path name="hfp-sco-wb headphones">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="hfp-sco headphones" />
+    </path>
+
+    <path name="compress-voip-call">
+        <ctl name="INT4_MI2S_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer INT3_MI2S_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call bt-a2dp">
+        <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer INT3_MI2S_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call handset">
+        <ctl name="INT0_MI2S_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer INT3_MI2S_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call headphones">
+        <ctl name="INT0_MI2S_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer INT3_MI2S_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call bt-sco">
+        <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="compress-voip-call bt-sco" />
+    </path>
+
+    <path name="compress-voip-call afe-proxy">
+        <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer AFE_PCM_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call usb-headphones">
+        <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer INT3_MI2S_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call usb-headset">
+        <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer USB_AUDIO_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call voice-speaker-vbat">
+        <path name="echo-reference speaker-vbat-mono" />
+        <path name="compress-voip-call"/>
+    </path>
+
+    <path name="compress-voip-call voice-speaker-2-vbat">
+        <path name="echo-reference speaker-vbat-mono-2" />
+        <path name="compress-voip-call"/>
+    </path>
+
+    <path name="voicemmode1-call">
+        <ctl name="INT4_MI2S_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer INT3_MI2S_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call handset">
+        <ctl name="INT0_MI2S_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer INT3_MI2S_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call headphones">
+        <ctl name="INT0_MI2S_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer INT3_MI2S_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call hdmi">
+        <ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer INT3_MI2S_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call bt-sco">
+        <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="voicemmode1-call bt-sco" />
+    </path>
+
+    <path name="voicemmode1-call afe-proxy">
+        <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer AFE_PCM_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call usb-headphones">
+        <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer INT3_MI2S_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call usb-headset">
+        <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call voice-speaker-vbat">
+        <path name="echo-reference speaker-vbat-mono" />
+        <path name="voicemmode1-call"/>
+    </path>
+
+    <path name="voicemmode1-call voice-speaker-2-vbat">
+        <path name="echo-reference speaker-vbat-mono-2" />
+        <path name="voicemmode1-call"/>
+    </path>
+
+    <path name="voicemmode2-call">
+        <ctl name="INT4_MI2S_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer INT3_MI2S_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call handset">
+        <ctl name="INT0_MI2S_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer INT3_MI2S_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call headphones">
+        <ctl name="INT0_MI2S_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer INT3_MI2S_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call hdmi">
+        <ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer INT3_MI2S_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call bt-sco">
+        <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="voicemmode2-call bt-sco" />
+    </path>
+
+    <path name="voicemmode2-call afe-proxy">
+        <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer AFE_PCM_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call usb-headphones">
+        <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer INT3_MI2S_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call usb-headset">
+        <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call voice-speaker-vbat">
+        <path name="echo-reference speaker-vbat-mono" />
+        <path name="voicemmode2-call"/>
+    </path>
+
+    <path name="voicemmode2-call voice-speaker-2-vbat">
+        <path name="echo-reference speaker-vbat-mono-2" />
+        <path name="voicemmode2-call"/>
+    </path>
+
+   <path name="spkr-rx-calib">
+        <ctl name="INT4_MI2S_RX_DL_HL Switch"  value="1" />
+    </path>
+
+    <path name="spkr-vi-record">
+    </path>
+
+    <!-- These are actual sound device specific mixer settings -->
+    <path name="adc1">
+        <ctl name="ADC1 Volume" value="6" />
+        <ctl name="DEC1 MUX" value="ADC1" />
+        <ctl name="ADC1_INP1 Switch" value="1" />
+    </path>
+
+    <path name="adc2">
+        <ctl name="ADC2 Volume" value="6" />
+        <ctl name="DEC1 MUX" value="ADC2" />
+        <ctl name="ADC2 MUX" value="INP2" />
+    </path>
+
+    <path name="adc3">
+        <ctl name="ADC3 Volume" value="6" />
+        <ctl name="DEC1 MUX" value="ADC2" />
+        <ctl name="ADC2 MUX" value="INP3" />
+    </path>
+
+    <!-- DMIC numbered from 1 to 4 -->
+    <path name="dmic1">
+        <ctl name="DEC1 MUX" value="DMIC1" />
+    </path>
+
+    <path name="dmic2">
+        <ctl name="DEC1 MUX" value="DMIC2" />
+    </path>
+
+    <path name="dmic3">
+        <ctl name="DEC1 MUX" value="DMIC3" />
+    </path>
+
+    <path name="dmic4">
+        <ctl name="DEC1 MUX" value="DMIC4" />
+    </path>
+
+    <path name="speaker">
+        <ctl name="INT4_MI2S_RX Channels" value="One" />
+        <ctl name="RX4 MIX1 INP1" value="RX4" />
+        <ctl name="COMP1 Switch" value="1" />
+        <ctl name="SpkrLeft COMP Switch" value="1" />
+        <ctl name="SpkrLeft BOOST Switch" value="1" />
+        <ctl name="SpkrLeft VISENSE Switch" value="1" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+        <ctl name="SpkrLeft WSA PA Mute" value="0" />
+    </path>
+
+    <path name="speaker-mono">
+        <ctl name="INT4_MI2S_RX Channels" value="One" />
+        <ctl name="RX4 MIX1 INP1" value="RX4" />
+        <ctl name="COMP1 Switch" value="1" />
+        <ctl name="SpkrLeft COMP Switch" value="1" />
+        <ctl name="SpkrLeft BOOST Switch" value="1" />
+        <ctl name="SpkrLeft VISENSE Switch" value="1" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+        <ctl name="SpkrLeft WSA PA Mute" value="0" />
+    </path>
+
+    <path name="speaker-mono-2">
+        <ctl name="INT4_MI2S_RX Channels" value="One" />
+        <ctl name="RX5 MIX1 INP1" value="RX4" />
+        <ctl name="COMP2 Switch" value="1" />
+        <ctl name="SpkrRight COMP Switch" value="1" />
+        <ctl name="SpkrRight BOOST Switch" value="1" />
+        <ctl name="SpkrRight VISENSE Switch" value="1" />
+        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+        <ctl name="SpkrRight WSA PA Mute" value="0" />
+    </path>
+
+    <path name="speaker-fluid">
+        <path name="speaker-mono" />
+    </path>
+
+    <path name="speaker-liquid">
+        <path name="speaker" />
+    </path>
+
+   <path name="speaker-vbat-mono">
+       <path name="speaker-mono" />
+   </path>
+
+   <path name="speaker-vbat-mono-2">
+       <path name="speaker-mono-2" />
+   </path>
+
+   <path name="speaker-vbat">
+       <path name="speaker" />
+   </path>
+
+   <path name="sidetone-iir">
+        <ctl name="IIR1 Enable Band1" value="1" />
+        <ctl name="IIR1 Enable Band2" value="1" />
+        <ctl name="IIR1 Enable Band3" value="1" />
+        <ctl name="IIR1 Enable Band4" value="1" />
+        <ctl name="IIR1 Enable Band5" value="1" />
+    </path>
+
+    <path name="sidetone-headphones">
+        <path name="sidetone-iir" />
+        <ctl name="RX1 MIX2 INP1" value="IIR1" />
+        <ctl name="RX2 MIX2 INP1" value="IIR1" />
+        <ctl name="IIR1 INP1 Volume" value="53" />
+    </path>
+
+    <path name="sidetone-handset">
+        <path name="sidetone-iir" />
+        <ctl name="IIR1 INP1 Volume" value="53" />
+        <ctl name="RX1 MIX2 INP1" value="IIR1" />
+    </path>
+
+    <path name="speaker-mic">
+        <path name="dmic1" />
+    </path>
+
+    <path name="speaker-mic-liquid">
+        <path name="dmic2" />
+    </path>
+
+    <path name="speaker-mic-sbc">
+    </path>
+
+    <path name="speaker-protected">
+        <ctl name="AIF1_VI_SDW Mixer SPKR_VI_1" value="1" />
+        <path name="speaker" />
+        <ctl name="VI_FEED_TX Channels" value="One" />
+        <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="INT5_MI2S_TX" />
+    </path>
+
+    <path name="voice-speaker-protected">
+        <ctl name="AIF1_VI_SDW Mixer SPKR_VI_1" value="1" />
+        <path name="speaker-mono" />
+        <ctl name="VI_FEED_TX Channels" value="One" />
+        <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="INT5_MI2S_TX" />
+    </path>
+
+    <path name="voice-speaker-2-protected">
+        <ctl name="AIF1_VI_SDW Mixer SPKR_VI_2" value="1" />
+        <path name="speaker-mono-2" />
+        <ctl name="VI_FEED_TX Channels" value="One" />
+        <ctl name="INT4_MI2S_RX_VI_FB_MONO_CH_MUX" value="INT5_MI2S_TX" />
+        <ctl name="INT5 MI2S VI MONO" value="Right" />
+    </path>
+
+    <path name="vi-feedback">
+    </path>
+
+    <path name="vi-feedback-mono-1">
+    </path>
+
+    <path name="vi-feedback-mono-2">
+    </path>
+
+    <path name="speaker-protected-vbat">
+        <path name="speaker-protected" />
+    </path>
+
+    <path name="voice-speaker-protected-vbat">
+        <path name="voice-speaker-protected" />
+    </path>
+
+    <path name="voice-speaker-2-protected-vbat">
+        <path name="voice-speaker-2-protected" />
+    </path>
+
+    <path name="handset">
+        <ctl name="INT0_MI2S_RX Channels" value="One" />
+        <ctl name="EAR PA Boost" value="DISABLE" />
+	<ctl name="RX1 MIX1 INP1" value="RX1" />
+	<ctl name="RDAC2 MUX" value="RX1" />
+	<ctl name="EAR_S" value="Switch" />
+    </path>
+
+    <path name="handset-mic">
+        <path name="dmic1" />
+    </path>
+
+    <path name="handset-mic-db">
+    </path>
+
+    <path name="handset-mic-cdp">
+        <path name="adc1" />
+    </path>
+
+    <path name="handset-mic-sbc">
+    </path>
+
+    <path name="three-mic">
+        <ctl name="INT3_MI2S_TX Channels" value="Three" />
+        <ctl name="DEC1 MUX" value="DMIC2" />
+        <ctl name="DEC2 MUX" value="DMIC1" />
+        <ctl name="DEC3 MUX" value="DMIC3" />
+        <ctl name="I2S TX2 INP1" value="DEC3" />
+    </path>
+
+    <path name="anc-handset">
+    </path>
+
+    <path name="headphones">
+        <ctl name="INT0_MI2S_RX Channels" value="Two" />
+        <ctl name="RX1 MIX1 INP1" value="RX1" />
+        <ctl name="RX2 MIX1 INP1" value="RX2" />
+        <ctl name="RDAC2 MUX" value="RX2" />
+        <ctl name="RX HPH Mode" value="HD2" />
+        <ctl name="COMP0 RX1" value="1" />
+        <ctl name="COMP0 RX2" value="1" />
+        <ctl name="HPHL" value="Switch" />
+        <ctl name="HPHR" value="Switch" />
+    </path>
+
+    <path name="headphones-44.1">
+        <path name="headphones" />
+    </path>
+
+    <path name="true-native-mode">
+    </path>
+
+    <path name="line">
+        <path name="headphones" />
+    </path>
+
+    <path name="headset-mic">
+        <path name="dmic2" />
+    </path>
+
+    <path name="headset-mic-liquid">
+        <path name="adc2" />
+    </path>
+
+    <path name="voice-handset">
+        <path name="handset" />
+    </path>
+
+    <path name="voice-handset-tmus">
+        <path name="handset" />
+    </path>
+
+    <path name="voice-speaker">
+        <path name="speaker-mono" />
+    </path>
+
+    <path name="voice-speaker-2">
+        <path name="speaker-mono-2" />
+    </path>
+
+    <path name="voice-speaker-fluid">
+        <path name="speaker-fluid" />
+    </path>
+
+    <path name="voice-speaker-mic">
+        <path name="speaker-mic" />
+    </path>
+
+    <path name="voice-speaker-vbat">
+        <path name="speaker-vbat-mono" />
+    </path>
+
+    <path name="voice-speaker-2-vbat">
+        <path name="speaker-vbat-mono-2" />
+    </path>
+
+    <path name="voice-headphones">
+        <path name="headphones" />
+    </path>
+
+    <path name="voice-line">
+        <path name="voice-headphones" />
+    </path>
+
+    <path name="voice-headset-mic">
+        <path name="headset-mic" />
+    </path>
+
+    <path name="speaker-and-headphones">
+        <path name="headphones" />
+        <path name="speaker" />
+    </path>
+
+    <path name="speaker-and-line">
+        <path name="speaker-and-headphones" />
+    </path>
+
+    <path name="speaker-and-headphones-liquid">
+        <path name="headphones" />
+        <path name="speaker" />
+    </path>
+
+    <path name="speaker-and-line-liquid">
+        <path name="speaker-and-headphones-liquid" />
+    </path>
+
+    <path name="usb-headphones">
+    </path>
+
+    <path name="usb-headset">
+    </path>
+
+    <path name="afe-proxy">
+    </path>
+
+    <path name="anc-headphones">
+        <path name="headphones" />
+    </path>
+
+    <path name="speaker-and-anc-headphones">
+        <path name="anc-headphones" />
+        <path name="speaker" />
+    </path>
+
+    <path name="anc-fb-headphones">
+        <path name="anc-headphones" />
+    </path>
+
+    <path name="speaker-and-anc-fb-headphones">
+        <path name="anc-fb-headphones" />
+        <path name="speaker" />
+    </path>
+
+    <path name="voice-anc-headphones">
+        <path name="voice-headphones" />
+    </path>
+
+    <path name="voice-anc-fb-headphones">
+        <path name="voice-headphones" />
+    </path>
+
+    <path name="speaker-and-anc-headphones-liquid">
+        <path name="anc-headphones" />
+    </path>
+
+    <path name="hdmi">
+    </path>
+
+    <path name="display-port">
+    </path>
+
+    <path name="speaker-and-usb-headphones">
+        <path name="speaker" />
+        <path name="usb-headphones" />
+    </path>
+
+    <path name="speaker-and-hdmi">
+        <path name="speaker" />
+        <path name="hdmi" />
+    </path>
+
+    <path name="speaker-and-display-port">
+        <path name="speaker" />
+        <path name="display-port" />
+    </path>
+
+    <path name="voice-rec-mic">
+        <path name="handset-mic" />
+    </path>
+
+    <path name="camcorder-mic">
+        <path name="handset-mic" />
+    </path>
+
+    <path name="hdmi-tx">
+        <path name="handset-mic" />
+    </path>
+
+    <path name="bt-sco-headset">
+    </path>
+
+    <path name="bt-sco-mic">
+    </path>
+
+    <path name="bt-sco-headset-wb">
+    </path>
+
+    <path name="bt-sco-mic-wb">
+    </path>
+
+    <path name="usb-headset-mic">
+    </path>
+
+    <path name="capture-fm">
+    </path>
+
+    <path name="aanc-path">
+    </path>
+
+    <path name="aanc-handset-mic">
+    </path>
+
+    <!-- Dual MIC devices -->
+    <path name="handset-dmic-endfire">
+        <ctl name="INT3_MI2S_TX Channels" value="Two" />
+        <ctl name="DEC1 MUX" value="DMIC2" />
+        <ctl name="DEC2 MUX" value="DMIC1" />
+    </path>
+
+    <path name="speaker-dmic-endfire">
+        <ctl name="INT3_MI2S_TX Channels" value="Two" />
+        <ctl name="DEC1 MUX" value="DMIC2" />
+        <ctl name="DEC2 MUX" value="DMIC1" />
+    </path>
+
+    <path name="dmic-endfire">
+        <path name="handset-dmic-endfire" />
+        <ctl name="IIR1 INP1 MUX" value="DEC1" />
+    </path>
+
+    <path name="dmic-endfire-liquid">
+        <path name="handset-dmic-endfire" />
+        <ctl name="IIR1 INP1 MUX" value="DEC1" />
+    </path>
+
+    <path name="handset-stereo-dmic-ef">
+        <path name="handset-dmic-endfire" />
+    </path>
+
+    <path name="speaker-stereo-dmic-ef">
+        <path name="speaker-dmic-endfire" />
+    </path>
+
+    <path name="voice-dmic-ef-tmus">
+        <path name="dmic-endfire" />
+    </path>
+
+    <path name="voice-dmic-ef">
+        <path name="dmic-endfire" />
+    </path>
+
+    <path name="voice-speaker-dmic-ef">
+        <path name="speaker-dmic-endfire" />
+    </path>
+
+    <path name="voice-rec-dmic-ef">
+        <path name="dmic-endfire" />
+    </path>
+
+    <path name="voice-rec-dmic-ef-fluence">
+        <path name="dmic-endfire" />
+    </path>
+
+    <path name="handset-stereo-dmic-ef-liquid">
+        <path name="handset-dmic-endfire" />
+    </path>
+
+    <path name="speaker-stereo-dmic-ef-liquid">
+        <path name="speaker-dmic-endfire" />
+    </path>
+
+    <path name="voice-dmic-ef-liquid-liquid">
+        <path name="dmic-endfire-liquid" />
+    </path>
+
+    <path name="voice-speaker-dmic-ef-liquid">
+        <path name="dmic-endfire-liquid" />
+    </path>
+
+    <path name="voice-rec-dmic-ef-liquid">
+        <path name="dmic-endfire-liquid" />
+    </path>
+
+    <path name="voice-rec-dmic-ef-fluence-liquid">
+        <path name="dmic-endfire-liquid" />
+    </path>
+
+    <path name="speaker-dmic-broadside">
+        <ctl name="INT3_MI2S_TX Channels" value="Two" />
+        <ctl name="DEC1 MUX" value="DMIC2" />
+        <ctl name="DEC2 MUX" value="DMIC1" />
+    </path>
+
+    <path name="dmic-broadside">
+        <path name="speaker-dmic-broadside" />
+        <ctl name="IIR1 INP1 MUX" value="DEC1" />
+    </path>
+
+    <path name="voice-speaker-dmic-broadside">
+        <path name="dmic-broadside" />
+    </path>
+
+    <!-- Quad MIC devices -->
+    <path name="speaker-qmic">
+        <ctl name="INT3_MI2S_TX Channels" value="Four" />
+        <ctl name="DEC1 MUX" value="DMIC2" />
+        <ctl name="DEC2 MUX" value="DMIC1" />
+        <ctl name="DEC3 MUX" value="DMIC3" />
+        <ctl name="I2S TX2 INP1" value="DEC3" />
+        <ctl name="DEC4 MUX" value="DMIC4" />
+        <ctl name="I2S TX2 INP2" value="DEC4" />
+    </path>
+
+    <path name="speaker-qmic-liquid">
+    </path>
+
+    <path name="voice-speaker-qmic">
+        <path name="speaker-qmic" />
+    </path>
+
+    <path name="quad-mic">
+        <path name="speaker-qmic" />
+    </path>
+
+    <path name="voice-speaker-qmic-liquid">
+        <path name="speaker-qmic-liquid" />
+    </path>
+
+    <path name="quad-mic-liquid">
+        <path name="speaker-qmic-liquid" />
+    </path>
+
+    <!-- TTY devices -->
+
+    <path name="tty-headphones">
+        <ctl name="RX1 MIX1 INP1" value="RX1" />
+        <ctl name="HPHL" value="Switch" />
+    </path>
+
+    <path name="voice-tty-full-headphones">
+        <ctl name="TTY Mode" value="FULL" />
+        <path name="tty-headphones" />
+    </path>
+
+    <path name="voice-tty-vco-headphones">
+        <ctl name="TTY Mode" value="VCO" />
+        <path name="tty-headphones" />
+    </path>
+
+    <path name="voice-tty-hco-handset">
+        <ctl name="TTY Mode" value="HCO" />
+        <path name="handset" />
+    </path>
+
+    <path name="voice-tty-full-headset-mic">
+        <path name="adc2" />
+    </path>
+
+    <path name="voice-tty-hco-headset-mic">
+        <path name="voice-tty-full-headset-mic" />
+    </path>
+
+    <path name="voice-tty-vco-handset-mic">
+        <path name="dmic4" />
+    </path>
+
+    <path name="unprocessed-handset-mic">
+        <path name="handset-mic" />
+    </path>
+
+    <path name="unprocessed-mic">
+        <path name="unprocessed-handset-mic" />
+    </path>
+    <!-- USB TTY start -->
+
+    <!-- full: both end tty -->
+    <path name="voice-tty-full-usb">
+        <ctl name="TTY Mode" value="FULL" />
+        <path name="usb-headphones" />
+    </path>
+
+    <path name="voice-tty-full-usb-mic">
+        <path name="usb-headset-mic" />
+    </path>
+
+    <!-- vco, in: handset mic use existing, out: tty -->
+    <path name="voice-tty-vco-usb">
+        <ctl name="TTY Mode" value="VCO" />
+        <path name="usb-headphones" />
+    </path>
+
+    <!-- hco, in: tty, out: speaker, use existing handset -->
+    <path name="voice-tty-hco-usb-mic">
+        <path name="voice-tty-full-usb-mic" />
+    </path>
+
+    <!-- USB TTY end   -->
+
+    <!-- Added for ADSP testfwk -->
+    <path name="ADSP testfwk">
+        <ctl name="INT4_MI2S_RX_DL_HL Switch" value="1" />
+    </path>
+
+    <path name="bt-a2dp">
+        <ctl name="BT SampleRate" value="KHZ_48" />
+        <ctl name="AFE Input Channels" value="Two" />
+        <ctl name="SLIM7_RX ADM Channels" value="Two" />
+    </path>
+
+    <path name="speaker-and-bt-a2dp">
+        <path name="speaker" />
+        <path name="bt-a2dp" />
+    </path>
+
+    <path name="deep-buffer-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="low-latency-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="compress-offload-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="audio-ull-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-a2dp">
+        <path name="deep-buffer-playback bt-a2dp" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-a2dp">
+        <path name="compress-offload-playback bt-a2dp" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-a2dp">
+        <path name="low-latency-playback bt-a2dp" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback2 bt-a2dp" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback3 bt-a2dp" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback4 bt-a2dp" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback5 bt-a2dp" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback6 bt-a2dp" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback7 bt-a2dp" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback8 bt-a2dp" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback9 bt-a2dp" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-a2dp">
+        <path name="audio-ull-playback bt-a2dp" />
+        <path name="audio-ull-playback" />
+    </path>
+</mixer>
diff --git a/configs/sdm670/mixer_paths_i2s.xml b/configs/sdm670/mixer_paths_i2s.xml
index bcf433b..fa8377b 100644
--- a/configs/sdm670/mixer_paths_i2s.xml
+++ b/configs/sdm670/mixer_paths_i2s.xml
@@ -73,6 +73,10 @@
     <ctl name="RT_PROXY_1_TX SetCalMode" value="CAL_MODE_NONE" />
     <!-- RT Proxy Cal end -->
 
+    <!-- mixer control to disable lowlatency qos voting -->
+    <ctl name="MultiMedia5_RX QOS Vote" value="Disable" />
+    <!-- mixer control to disable lowlatency qos voting end -->
+
     <!-- These are audio route (FE to BE) specific mixer settings -->
     <path name="deep-buffer-playback">
         <ctl name="AUX_PCM_RX Audio Mixer MultiMedia1" value="1" />
@@ -82,6 +86,10 @@
         <ctl name="AUX_PCM_RX Audio Mixer MultiMedia5" value="1" />
     </path>
 
+    <path name="low-latency-playback resume">
+        <ctl name="MultiMedia5_RX QOS Vote" value="Enable" />
+    </path>
+
     <path name="compress-offload-playback">
         <ctl name="AUX_PCM_RX Audio Mixer MultiMedia4" value="1" />
     </path>
diff --git a/configs/sdm670/mixer_paths_mtp.xml b/configs/sdm670/mixer_paths_mtp.xml
index 9b462a4..c5d2cf2 100644
--- a/configs/sdm670/mixer_paths_mtp.xml
+++ b/configs/sdm670/mixer_paths_mtp.xml
@@ -67,6 +67,9 @@
     <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
     <ctl name="MultiMedia8 Mixer INT3_MI2S_TX" value="0" />
     <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
     <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
     <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
     <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
@@ -135,9 +138,11 @@
     <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
     <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_TX Channels" value="One" />
+    <ctl name="ADC1_INP1 Switch" value="0" />
     <ctl name="MultiMedia6 Mixer INT3_MI2S_TX" value="0" />
     <ctl name="INT4_MI2S_RX Channels" value="One" />
     <ctl name="INT0_MI2S_RX Channels" value="One" />
@@ -373,6 +378,10 @@
 
     <!-- Codec controls end -->
 
+    <!-- mixer control to disable lowlatency qos voting -->
+    <ctl name="MultiMedia5_RX QOS Vote" value="Disable" />
+    <!-- mixer control to disable lowlatency qos voting end -->
+
     <!-- These are audio route (FE to BE) specific mixer settings -->
     <path name="gsm-mode">
         <ctl name="GSM mode Enable" value="ON" />
@@ -446,6 +455,16 @@
         <path name="deep-buffer-playback" />
     </path>
 
+    <path name="deep-buffer-playback speaker-and-bt-sco">
+        <path name="deep-buffer-playback bt-sco" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-sco-wb">
+        <path name="deep-buffer-playback bt-sco-wb" />
+        <path name="deep-buffer-playback" />
+    </path>
+
     <path name="low-latency-playback">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia5" value="1" />
     </path>
@@ -503,6 +522,20 @@
         <path name="low-latency-playback" />
     </path>
 
+    <path name="low-latency-playback resume">
+        <ctl name="MultiMedia5_RX QOS Vote" value="Enable" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco">
+        <path name="low-latency-playback bt-sco" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco-wb">
+        <path name="low-latency-playback bt-sco-wb" />
+        <path name="low-latency-playback" />
+    </path>
+
     <path name="audio-ull-playback">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia8" value="1" />
     </path>
@@ -555,6 +588,16 @@
         <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
+    <path name="audio-ull-playback speaker-and-bt-sco">
+        <path name="audio-ull-playback bt-sco" />
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-sco-wb">
+        <path name="audio-ull-playback bt-sco-wb" />
+        <path name="audio-ull-playback" />
+    </path>
+
     <path name="multi-channel-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
@@ -631,6 +674,16 @@
         <path name="compress-offload-playback" />
     </path>
 
+    <path name="compress-offload-playback speaker-and-bt-sco">
+        <path name="compress-offload-playback bt-sco" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback bt-sco-wb" />
+        <path name="compress-offload-playback" />
+    </path>
+
     <path name="compress-offload-playback2">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia7" value="1" />
     </path>
@@ -691,6 +744,16 @@
         <path name="compress-offload-playback2" />
     </path>
 
+    <path name="compress-offload-playback2 speaker-and-bt-sco">
+        <path name="compress-offload-playback2 bt-sco" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback2 bt-sco-wb" />
+        <path name="compress-offload-playback2" />
+    </path>
+
     <path name="compress-offload-playback3">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -751,6 +814,16 @@
         <path name="compress-offload-playback3" />
     </path>
 
+    <path name="compress-offload-playback3 speaker-and-bt-sco">
+        <path name="compress-offload-playback3 bt-sco" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback3 bt-sco-wb" />
+        <path name="compress-offload-playback3" />
+    </path>
+
     <path name="compress-offload-playback4">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia11" value="1" />
     </path>
@@ -812,6 +885,16 @@
         <path name="compress-offload-playback4" />
     </path>
 
+    <path name="compress-offload-playback4 speaker-and-bt-sco">
+        <path name="compress-offload-playback4 bt-sco" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback4 bt-sco-wb" />
+        <path name="compress-offload-playback4" />
+    </path>
+
     <path name="compress-offload-playback5">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia12" value="1" />
     </path>
@@ -872,6 +955,16 @@
         <path name="compress-offload-playback5" />
     </path>
 
+    <path name="compress-offload-playback5 speaker-and-bt-sco">
+        <path name="compress-offload-playback5 bt-sco" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback5 bt-sco-wb" />
+        <path name="compress-offload-playback5" />
+    </path>
+
     <path name="compress-offload-playback6">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia13" value="1" />
     </path>
@@ -932,6 +1025,16 @@
         <path name="compress-offload-playback6" />
     </path>
 
+    <path name="compress-offload-playback6 speaker-and-bt-sco">
+        <path name="compress-offload-playback6 bt-sco" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback6 bt-sco-wb" />
+        <path name="compress-offload-playback6" />
+    </path>
+
     <path name="compress-offload-playback7">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia14" value="1" />
     </path>
@@ -992,6 +1095,16 @@
         <path name="compress-offload-playback7" />
     </path>
 
+    <path name="compress-offload-playback7 speaker-and-bt-sco">
+        <path name="compress-offload-playback7 bt-sco" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback7 bt-sco-wb" />
+        <path name="compress-offload-playback7" />
+    </path>
+
     <path name="compress-offload-playback8">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia15" value="1" />
     </path>
@@ -1052,6 +1165,16 @@
         <path name="compress-offload-playback8" />
     </path>
 
+    <path name="compress-offload-playback8 speaker-and-bt-sco">
+        <path name="compress-offload-playback8 bt-sco" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback8 bt-sco-wb" />
+        <path name="compress-offload-playback8" />
+    </path>
+
     <path name="compress-offload-playback9">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia16" value="1" />
     </path>
@@ -1112,6 +1235,16 @@
         <path name="compress-offload-playback9" />
     </path>
 
+    <path name="compress-offload-playback9 speaker-and-bt-sco">
+        <path name="compress-offload-playback9 bt-sco" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback9 bt-sco-wb" />
+        <path name="compress-offload-playback9" />
+    </path>
+
     <path name="audio-record">
         <ctl name="MultiMedia1 Mixer INT3_MI2S_TX" value="1" />
     </path>
@@ -1447,6 +1580,63 @@
         <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
     </path>
 
+    <!-- VOIP Rx settings -->
+    <path name="audio-playback-voip">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="audio-playback-voip bt-sco" />
+    </path>
+
+    <path name="audio-playback-voip afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip display-port">
+        <ctl name="DISPLAY_PORT Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip speaker-and-display-port">
+        <path name="audio-playback-voip display-port" />
+        <path name="audio-playback-voip" />
+    </path>
+
+    <!-- VoIP Tx settings -->
+    <path name="audio-record-voip">
+        <ctl name="MultiMedia10 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip usb-headset-mic">
+        <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco">
+        <ctl name="MultiMedia10 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="audio-record-voip bt-sco" />
+   </path>
+
    <path name="spkr-rx-calib">
         <ctl name="INT4_MI2S_RX_DL_HL Switch"  value="1" />
     </path>
@@ -1458,6 +1648,7 @@
     <path name="adc1">
         <ctl name="ADC1 Volume" value="6" />
         <ctl name="DEC1 MUX" value="ADC1" />
+        <ctl name="ADC1_INP1 Switch" value="1" />
     </path>
 
     <path name="adc2">
@@ -1973,6 +2164,23 @@
     <path name="unprocessed-mic">
         <path name="unprocessed-handset-mic" />
     </path>
+
+    <path name="unprocessed-stereo-mic">
+        <path name="voice-rec-dmic-ef" />
+    </path>
+
+    <path name="unprocessed-three-mic">
+        <path name="three-mic" />
+    </path>
+
+    <path name="unprocessed-quad-mic">
+        <path name="quad-mic" />
+    </path>
+
+    <path name="unprocessed-headset-mic">
+        <path name="headset-mic" />
+    </path>
+
     <!-- USB TTY start -->
 
     <!-- full: both end tty -->
@@ -2004,8 +2212,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
@@ -2122,6 +2328,84 @@
         <path name="audio-ull-playback" />
     </path>
 
+    <path name="mmap-playback">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-headphones">
+        <path name="mmap-playback" />
+        <path name="mmap-playback headphones" />
+    </path>
+
+    <path name="mmap-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="16000" />
+        <path name="mmap-playback bt-sco" />
+    </path>
+
+    <path name="mmap-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-buffer-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-hdmi">
+        <path name="mmap-playback hdmi" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-display-port">
+        <path name="mmap-playback display-port" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-usb-headphones">
+        <path name="mmap-playback usb-headphones" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-record">
+      <ctl name="MultiMedia16 Mixer INT3_MI2S_TX " value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco">
+      <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="mmap-record bt-sco" />
+    </path>
+
+    <path name="mmap-record capture-fm">
+      <ctl name="MultiMedia16 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="mmap-record usb-headset-mic">
+       <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
     <path name="hifi-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
diff --git a/configs/sdm670/mixer_paths_skuw.xml b/configs/sdm670/mixer_paths_skuw.xml
index 7ed5142..2b3c8bb 100644
--- a/configs/sdm670/mixer_paths_skuw.xml
+++ b/configs/sdm670/mixer_paths_skuw.xml
@@ -395,6 +395,10 @@
 
     <!-- Codec controls end -->
 
+    <!-- mixer control to disable lowlatency qos voting -->
+    <ctl name="MultiMedia5_RX QOS Vote" value="Disable" />
+    <!-- mixer control to disable lowlatency qos voting end -->
+
     <!-- These are audio route (FE to BE) specific mixer settings -->
     <path name="gsm-mode">
         <ctl name="GSM mode Enable" value="ON" />
@@ -485,6 +489,16 @@
         <path name="deep-buffer-playback" />
     </path>
 
+    <path name="deep-buffer-playback speaker-and-bt-sco">
+        <path name="deep-buffer-playback bt-sco" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-sco-wb">
+        <path name="deep-buffer-playback bt-sco-wb" />
+        <path name="deep-buffer-playback" />
+    </path>
+
     <path name="low-latency-playback">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia5" value="1" />
     </path>
@@ -550,6 +564,20 @@
         <path name="low-latency-playback" />
     </path>
 
+    <path name="low-latency-playback resume">
+        <ctl name="MultiMedia5_RX QOS Vote" value="Enable" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco">
+        <path name="low-latency-playback bt-sco" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco-wb">
+        <path name="low-latency-playback bt-sco-wb" />
+        <path name="low-latency-playback" />
+    </path>
+
     <path name="audio-ull-playback">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia8" value="1" />
     </path>
@@ -610,6 +638,16 @@
         <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
+    <path name="audio-ull-playback speaker-and-bt-sco">
+        <path name="audio-ull-playback bt-sco" />
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-sco-wb">
+        <path name="audio-ull-playback bt-sco-wb" />
+        <path name="audio-ull-playback" />
+    </path>
+
     <path name="multi-channel-playback hdmi">
         <ctl name="HDMI Mixer MultiMedia2" value="1" />
     </path>
@@ -702,6 +740,16 @@
         <path name="compress-offload-playback" />
     </path>
 
+    <path name="compress-offload-playback speaker-and-bt-sco">
+        <path name="compress-offload-playback bt-sco" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback bt-sco-wb" />
+        <path name="compress-offload-playback" />
+    </path>
+
     <path name="compress-offload-playback2">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia7" value="1" />
     </path>
@@ -770,6 +818,16 @@
         <path name="compress-offload-playback2" />
     </path>
 
+    <path name="compress-offload-playback2 speaker-and-bt-sco">
+        <path name="compress-offload-playback2 bt-sco" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback2 bt-sco-wb" />
+        <path name="compress-offload-playback2" />
+    </path>
+
     <path name="compress-offload-playback3">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -838,6 +896,16 @@
         <path name="compress-offload-playback3" />
     </path>
 
+    <path name="compress-offload-playback3 speaker-and-bt-sco">
+        <path name="compress-offload-playback3 bt-sco" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback3 bt-sco-wb" />
+        <path name="compress-offload-playback3" />
+    </path>
+
     <path name="compress-offload-playback4">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia11" value="1" />
     </path>
@@ -907,6 +975,16 @@
         <path name="compress-offload-playback4" />
     </path>
 
+    <path name="compress-offload-playback4 speaker-and-bt-sco">
+        <path name="compress-offload-playback4 bt-sco" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback4 bt-sco-wb" />
+        <path name="compress-offload-playback4" />
+    </path>
+
     <path name="compress-offload-playback5">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia12" value="1" />
     </path>
@@ -975,6 +1053,16 @@
         <path name="compress-offload-playback5" />
     </path>
 
+    <path name="compress-offload-playback5 speaker-and-bt-sco">
+        <path name="compress-offload-playback5 bt-sco" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback5 bt-sco-wb" />
+        <path name="compress-offload-playback5" />
+    </path>
+
     <path name="compress-offload-playback6">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia13" value="1" />
     </path>
@@ -1043,6 +1131,16 @@
         <path name="compress-offload-playback6" />
     </path>
 
+    <path name="compress-offload-playback6 speaker-and-bt-sco">
+        <path name="compress-offload-playback6 bt-sco" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback6 bt-sco-wb" />
+        <path name="compress-offload-playback6" />
+    </path>
+
     <path name="compress-offload-playback7">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia14" value="1" />
     </path>
@@ -1111,6 +1209,16 @@
         <path name="compress-offload-playback7" />
     </path>
 
+    <path name="compress-offload-playback7 speaker-and-bt-sco">
+        <path name="compress-offload-playback7 bt-sco" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback7 bt-sco-wb" />
+        <path name="compress-offload-playback7" />
+    </path>
+
     <path name="compress-offload-playback8">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia15" value="1" />
     </path>
@@ -1179,6 +1287,16 @@
         <path name="compress-offload-playback8" />
     </path>
 
+    <path name="compress-offload-playback8 speaker-and-bt-sco">
+        <path name="compress-offload-playback8 bt-sco" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback8 bt-sco-wb" />
+        <path name="compress-offload-playback8" />
+    </path>
+
     <path name="compress-offload-playback9">
         <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia16" value="1" />
     </path>
@@ -1247,6 +1365,16 @@
         <path name="compress-offload-playback9" />
     </path>
 
+    <path name="compress-offload-playback9 speaker-and-bt-sco">
+        <path name="compress-offload-playback9 bt-sco" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback9 bt-sco-wb" />
+        <path name="compress-offload-playback9" />
+    </path>
+
     <path name="audio-record">
         <ctl name="MultiMedia1 Mixer INT3_MI2S_TX" value="1" />
     </path>
@@ -2253,6 +2381,23 @@
     <path name="unprocessed-mic">
         <path name="unprocessed-handset-mic" />
     </path>
+
+    <path name="unprocessed-stereo-mic">
+        <path name="voice-rec-dmic-ef" />
+    </path>
+
+    <path name="unprocessed-three-mic">
+        <path name="three-mic" />
+    </path>
+
+    <path name="unprocessed-quad-mic">
+        <path name="quad-mic" />
+    </path>
+
+    <path name="unprocessed-headset-mic">
+        <path name="headset-mic" />
+    </path>
+
     <!-- USB TTY start -->
 
     <!-- full: both end tty -->
@@ -2401,4 +2546,82 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="mmap-playback">
+        <ctl name="INT4_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback headphones">
+        <ctl name="INT0_MI2S_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-headphones">
+        <path name="mmap-playback" />
+        <path name="mmap-playback headphones" />
+    </path>
+
+    <path name="mmap-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="16000" />
+        <path name="mmap-playback bt-sco" />
+    </path>
+
+    <path name="mmap-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-buffer-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-hdmi">
+        <path name="mmap-playback hdmi" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-display-port">
+        <path name="mmap-playback display-port" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-usb-headphones">
+        <path name="mmap-playback usb-headphones" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-record">
+      <ctl name="MultiMedia16 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco">
+      <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="mmap-record bt-sco" />
+    </path>
+
+    <path name="mmap-record capture-fm">
+      <ctl name="MultiMedia16 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="mmap-record usb-headset-mic">
+       <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
+    </path>
 </mixer>
diff --git a/configs/sdm670/mixer_paths_tasha.xml b/configs/sdm670/mixer_paths_tasha.xml
index cbb74a5..4db4271 100644
--- a/configs/sdm670/mixer_paths_tasha.xml
+++ b/configs/sdm670/mixer_paths_tasha.xml
@@ -566,6 +566,10 @@
     <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
     <!-- ADSP testfwk end-->
 
+    <!-- mixer control to disable lowlatency qos voting -->
+    <ctl name="MultiMedia5_RX QOS Vote" value="Disable" />
+    <!-- mixer control to disable lowlatency qos voting end -->
+
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
 
     <!-- These are audio route (FE to BE) specific mixer settings -->
@@ -669,6 +673,16 @@
         <path name="deep-buffer-playback" />
     </path>
 
+    <path name="deep-buffer-playback speaker-and-bt-sco">
+        <path name="deep-buffer-playback bt-sco" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-sco-wb">
+        <path name="deep-buffer-playback bt-sco-wb" />
+        <path name="deep-buffer-playback" />
+    </path>
+
     <path name="low-latency-playback">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="1" />
     </path>
@@ -726,6 +740,20 @@
         <path name="low-latency-playback" />
     </path>
 
+    <path name="low-latency-playback resume">
+        <ctl name="MultiMedia5_RX QOS Vote" value="Enable" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco">
+        <path name="low-latency-playback bt-sco" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco-wb">
+        <path name="low-latency-playback bt-sco-wb" />
+        <path name="low-latency-playback" />
+    </path>
+
     <path name="audio-ull-playback">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
@@ -778,6 +806,16 @@
         <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
+    <path name="audio-ull-playback speaker-and-bt-sco">
+        <path name="audio-ull-playback bt-sco" />
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-sco-wb">
+        <path name="audio-ull-playback bt-sco-wb" />
+        <path name="audio-ull-playback" />
+    </path>
+
     <path name="multi-channel-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
@@ -851,6 +889,16 @@
         <path name="compress-offload-playback" />
     </path>
 
+    <path name="compress-offload-playback speaker-and-bt-sco">
+        <path name="compress-offload-playback bt-sco" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback bt-sco-wb" />
+        <path name="compress-offload-playback" />
+    </path>
+
     <path name="compress-offload-playback2">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="1" />
     </path>
@@ -908,6 +956,16 @@
         <path name="compress-offload-playback2" />
     </path>
 
+    <path name="compress-offload-playback2 speaker-and-bt-sco">
+        <path name="compress-offload-playback2 bt-sco" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback2 bt-sco-wb" />
+        <path name="compress-offload-playback2" />
+    </path>
+
     <path name="compress-offload-playback3">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -965,6 +1023,16 @@
         <path name="compress-offload-playback3" />
     </path>
 
+    <path name="compress-offload-playback3 speaker-and-bt-sco">
+        <path name="compress-offload-playback3 bt-sco" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback3 bt-sco-wb" />
+        <path name="compress-offload-playback3" />
+    </path>
+
     <path name="compress-offload-playback4">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="1" />
     </path>
@@ -1023,6 +1091,16 @@
         <path name="compress-offload-playback4" />
     </path>
 
+    <path name="compress-offload-playback4 speaker-and-bt-sco">
+        <path name="compress-offload-playback4 bt-sco" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback4 bt-sco-wb" />
+        <path name="compress-offload-playback4" />
+    </path>
+
     <path name="compress-offload-playback5">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="1" />
     </path>
@@ -1080,6 +1158,16 @@
         <path name="compress-offload-playback5" />
     </path>
 
+    <path name="compress-offload-playback5 speaker-and-bt-sco">
+        <path name="compress-offload-playback5 bt-sco" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback5 bt-sco-wb" />
+        <path name="compress-offload-playback5" />
+    </path>
+
     <path name="compress-offload-playback6">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="1" />
     </path>
@@ -1137,6 +1225,16 @@
         <path name="compress-offload-playback6" />
     </path>
 
+    <path name="compress-offload-playback6 speaker-and-bt-sco">
+        <path name="compress-offload-playback6 bt-sco" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback6 bt-sco-wb" />
+        <path name="compress-offload-playback6" />
+    </path>
+
     <path name="compress-offload-playback7">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="1" />
     </path>
@@ -1194,6 +1292,16 @@
         <path name="compress-offload-playback7" />
     </path>
 
+    <path name="compress-offload-playback7 speaker-and-bt-sco">
+        <path name="compress-offload-playback7 bt-sco" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback7 bt-sco-wb" />
+        <path name="compress-offload-playback7" />
+    </path>
+
     <path name="compress-offload-playback8">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="1" />
     </path>
@@ -1251,6 +1359,16 @@
         <path name="compress-offload-playback8" />
     </path>
 
+    <path name="compress-offload-playback8 speaker-and-bt-sco">
+        <path name="compress-offload-playback8 bt-sco" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback8 bt-sco-wb" />
+        <path name="compress-offload-playback8" />
+    </path>
+
     <path name="compress-offload-playback9">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
     </path>
@@ -1308,6 +1426,16 @@
         <path name="compress-offload-playback9" />
     </path>
 
+    <path name="compress-offload-playback9 speaker-and-bt-sco">
+        <path name="compress-offload-playback9 bt-sco" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback9 bt-sco-wb" />
+        <path name="compress-offload-playback9" />
+    </path>
+
     <path name="audio-record">
         <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="1" />
     </path>
@@ -2604,6 +2732,22 @@
         <path name="unprocessed-handset-mic" />
     </path>
 
+    <path name="unprocessed-stereo-mic">
+        <path name="voice-rec-dmic-ef" />
+    </path>
+
+    <path name="unprocessed-three-mic">
+        <path name="three-mic" />
+    </path>
+
+    <path name="unprocessed-quad-mic">
+        <path name="quad-mic" />
+    </path>
+
+    <path name="unprocessed-headset-mic">
+        <path name="headset-mic" />
+    </path>
+
     <!-- USB TTY start -->
 
     <!-- full: both end tty -->
@@ -2635,8 +2779,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
@@ -2753,6 +2895,84 @@
         <path name="audio-ull-playback" />
     </path>
 
+    <path name="mmap-playback">
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-headphones">
+        <path name="mmap-playback" />
+        <path name="mmap-playback headphones" />
+    </path>
+
+    <path name="mmap-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="16000" />
+        <path name="mmap-playback bt-sco" />
+    </path>
+
+    <path name="mmap-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-buffer-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-hdmi">
+        <path name="mmap-playback hdmi" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-display-port">
+        <path name="mmap-playback display-port" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-usb-headphones">
+        <path name="mmap-playback usb-headphones" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-record">
+      <ctl name="MultiMedia16 Mixer SLIM_0_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco">
+      <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="mmap-record bt-sco" />
+    </path>
+
+    <path name="mmap-record capture-fm">
+      <ctl name="MultiMedia16 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="mmap-record usb-headset-mic">
+       <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
     <path name="hifi-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
diff --git a/configs/sdm670/mixer_paths_tashalite.xml b/configs/sdm670/mixer_paths_tashalite.xml
index e4ea674..97b6911 100644
--- a/configs/sdm670/mixer_paths_tashalite.xml
+++ b/configs/sdm670/mixer_paths_tashalite.xml
@@ -566,6 +566,10 @@
     <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
     <!-- ADSP testfwk end-->
 
+    <!-- mixer control to disable lowlatency qos voting -->
+    <ctl name="MultiMedia5_RX QOS Vote" value="Disable" />
+    <!-- mixer control to disable lowlatency qos voting end -->
+
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
 
     <!-- These are audio route (FE to BE) specific mixer settings -->
@@ -669,6 +673,16 @@
         <path name="deep-buffer-playback" />
     </path>
 
+    <path name="deep-buffer-playback speaker-and-bt-sco">
+        <path name="deep-buffer-playback bt-sco" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-sco-wb">
+        <path name="deep-buffer-playback bt-sco-wb" />
+        <path name="deep-buffer-playback" />
+    </path>
+
     <path name="low-latency-playback">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="1" />
     </path>
@@ -726,6 +740,20 @@
         <path name="low-latency-playback" />
     </path>
 
+    <path name="low-latency-playback resume">
+        <ctl name="MultiMedia5_RX QOS Vote" value="Enable" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco">
+        <path name="low-latency-playback bt-sco" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco-wb">
+        <path name="low-latency-playback bt-sco-wb" />
+        <path name="low-latency-playback" />
+    </path>
+
     <path name="audio-ull-playback">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
@@ -778,6 +806,16 @@
         <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
+    <path name="audio-ull-playback speaker-and-bt-sco">
+        <path name="audio-ull-playback bt-sco" />
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-sco-wb">
+        <path name="audio-ull-playback bt-sco-wb" />
+        <path name="audio-ull-playback" />
+    </path>
+
     <path name="multi-channel-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
@@ -851,6 +889,16 @@
         <path name="compress-offload-playback" />
     </path>
 
+    <path name="compress-offload-playback speaker-and-bt-sco">
+        <path name="compress-offload-playback bt-sco" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback bt-sco-wb" />
+        <path name="compress-offload-playback" />
+    </path>
+
     <path name="compress-offload-playback2">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="1" />
     </path>
@@ -908,6 +956,16 @@
         <path name="compress-offload-playback2" />
     </path>
 
+    <path name="compress-offload-playback2 speaker-and-bt-sco">
+        <path name="compress-offload-playback2 bt-sco" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback2 bt-sco-wb" />
+        <path name="compress-offload-playback2" />
+    </path>
+
     <path name="compress-offload-playback3">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -965,6 +1023,16 @@
         <path name="compress-offload-playback3" />
     </path>
 
+    <path name="compress-offload-playback3 speaker-and-bt-sco">
+        <path name="compress-offload-playback3 bt-sco" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback3 bt-sco-wb" />
+        <path name="compress-offload-playback3" />
+    </path>
+
     <path name="compress-offload-playback4">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="1" />
     </path>
@@ -1023,6 +1091,16 @@
         <path name="compress-offload-playback4" />
     </path>
 
+    <path name="compress-offload-playback4 speaker-and-bt-sco">
+        <path name="compress-offload-playback4 bt-sco" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback4 bt-sco-wb" />
+        <path name="compress-offload-playback4" />
+    </path>
+
     <path name="compress-offload-playback5">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="1" />
     </path>
@@ -1080,6 +1158,16 @@
         <path name="compress-offload-playback5" />
     </path>
 
+    <path name="compress-offload-playback5 speaker-and-bt-sco">
+        <path name="compress-offload-playback5 bt-sco" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback5 bt-sco-wb" />
+        <path name="compress-offload-playback5" />
+    </path>
+
     <path name="compress-offload-playback6">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="1" />
     </path>
@@ -1137,6 +1225,16 @@
         <path name="compress-offload-playback6" />
     </path>
 
+    <path name="compress-offload-playback6 speaker-and-bt-sco">
+        <path name="compress-offload-playback6 bt-sco" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback6 bt-sco-wb" />
+        <path name="compress-offload-playback6" />
+    </path>
+
     <path name="compress-offload-playback7">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="1" />
     </path>
@@ -1194,6 +1292,16 @@
         <path name="compress-offload-playback7" />
     </path>
 
+    <path name="compress-offload-playback7 speaker-and-bt-sco">
+        <path name="compress-offload-playback7 bt-sco" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback7 bt-sco-wb" />
+        <path name="compress-offload-playback7" />
+    </path>
+
     <path name="compress-offload-playback8">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="1" />
     </path>
@@ -1251,6 +1359,16 @@
         <path name="compress-offload-playback8" />
     </path>
 
+    <path name="compress-offload-playback8 speaker-and-bt-sco">
+        <path name="compress-offload-playback8 bt-sco" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback8 bt-sco-wb" />
+        <path name="compress-offload-playback8" />
+    </path>
+
     <path name="compress-offload-playback9">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
     </path>
@@ -1308,6 +1426,16 @@
         <path name="compress-offload-playback9" />
     </path>
 
+    <path name="compress-offload-playback9 speaker-and-bt-sco">
+        <path name="compress-offload-playback9 bt-sco" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback9 bt-sco-wb" />
+        <path name="compress-offload-playback9" />
+    </path>
+
     <path name="audio-record">
         <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="1" />
     </path>
@@ -2582,6 +2710,22 @@
         <path name="unprocessed-handset-mic" />
     </path>
 
+    <path name="unprocessed-stereo-mic">
+        <path name="voice-rec-dmic-ef" />
+    </path>
+
+    <path name="unprocessed-three-mic">
+        <path name="three-mic" />
+    </path>
+
+    <path name="unprocessed-quad-mic">
+        <path name="quad-mic" />
+    </path>
+
+    <path name="unprocessed-headset-mic">
+        <path name="headset-mic" />
+    </path>
+
     <!-- USB TTY start -->
 
     <!-- full: both end tty -->
@@ -2613,8 +2757,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
@@ -2731,6 +2873,84 @@
         <path name="audio-ull-playback" />
     </path>
 
+    <path name="mmap-playback">
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-headphones">
+        <path name="mmap-playback" />
+        <path name="mmap-playback headphones" />
+    </path>
+
+    <path name="mmap-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="16000" />
+        <path name="mmap-playback bt-sco" />
+    </path>
+
+    <path name="mmap-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-buffer-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-hdmi">
+        <path name="mmap-playback hdmi" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-display-port">
+        <path name="mmap-playback display-port" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-usb-headphones">
+        <path name="mmap-playback usb-headphones" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-record">
+      <ctl name="MultiMedia16 Mixer SLIM_0_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco">
+      <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="mmap-record bt-sco" />
+    </path>
+
+    <path name="mmap-record capture-fm">
+      <ctl name="MultiMedia16 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="mmap-record usb-headset-mic">
+       <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
     <path name="hifi-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
diff --git a/configs/sdm670/mixer_paths_tavil.xml b/configs/sdm670/mixer_paths_tavil.xml
index f822423..d77f623 100644
--- a/configs/sdm670/mixer_paths_tavil.xml
+++ b/configs/sdm670/mixer_paths_tavil.xml
@@ -72,6 +72,9 @@
     <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="0" />
     <ctl name="MultiMedia8 Mixer SLIM_4_TX" value="0" />
     <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer SLIM_0_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
     <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
     <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
     <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
@@ -160,6 +163,7 @@
     <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
     <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_TX Channels" value="One" />
@@ -524,6 +528,10 @@
 
     <!-- Codec controls end -->
 
+    <!-- mixer control to disable lowlatency qos voting -->
+    <ctl name="MultiMedia5_RX QOS Vote" value="Disable" />
+    <!-- mixer control to disable lowlatency qos voting end -->
+
     <!-- These are audio route (FE to BE) specific mixer settings -->
     <path name="echo-reference">
         <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
@@ -588,6 +596,16 @@
         <path name="deep-buffer-playback" />
     </path>
 
+    <path name="deep-buffer-playback speaker-and-bt-sco">
+        <path name="deep-buffer-playback bt-sco" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-sco-wb">
+        <path name="deep-buffer-playback bt-sco-wb" />
+        <path name="deep-buffer-playback" />
+    </path>
+
     <path name="low-latency-playback">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="1" />
     </path>
@@ -640,6 +658,20 @@
         <path name="low-latency-playback" />
     </path>
 
+    <path name="low-latency-playback resume">
+        <ctl name="MultiMedia5_RX QOS Vote" value="Enable" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco">
+        <path name="low-latency-playback bt-sco" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-sco-wb">
+        <path name="low-latency-playback bt-sco-wb" />
+        <path name="low-latency-playback" />
+    </path>
+
     <path name="audio-ull-playback">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
@@ -687,6 +719,16 @@
         <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
+    <path name="audio-ull-playback speaker-and-bt-sco">
+        <path name="audio-ull-playback bt-sco" />
+        <path name="audio-ull-playback" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-sco-wb">
+        <path name="audio-ull-playback bt-sco-wb" />
+        <path name="audio-ull-playback" />
+    </path>
+
     <path name="multi-channel-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
@@ -759,6 +801,16 @@
         <path name="compress-offload-playback" />
     </path>
 
+    <path name="compress-offload-playback speaker-and-bt-sco">
+        <path name="compress-offload-playback bt-sco" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback bt-sco-wb" />
+        <path name="compress-offload-playback" />
+    </path>
+
     <path name="compress-offload-playback2">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="1" />
     </path>
@@ -815,6 +867,16 @@
         <path name="compress-offload-playback2" />
     </path>
 
+    <path name="compress-offload-playback2 speaker-and-bt-sco">
+        <path name="compress-offload-playback2 bt-sco" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback2 bt-sco-wb" />
+        <path name="compress-offload-playback2" />
+    </path>
+
     <path name="compress-offload-playback3">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -871,6 +933,16 @@
         <path name="compress-offload-playback3" />
     </path>
 
+    <path name="compress-offload-playback3 speaker-and-bt-sco">
+        <path name="compress-offload-playback3 bt-sco" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback3 bt-sco-wb" />
+        <path name="compress-offload-playback3" />
+    </path>
+
     <path name="compress-offload-playback4">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="1" />
     </path>
@@ -928,6 +1000,16 @@
         <path name="compress-offload-playback4" />
     </path>
 
+    <path name="compress-offload-playback4 speaker-and-bt-sco">
+        <path name="compress-offload-playback4 bt-sco" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback4 bt-sco-wb" />
+        <path name="compress-offload-playback4" />
+    </path>
+
     <path name="compress-offload-playback5">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="1" />
     </path>
@@ -984,6 +1066,16 @@
         <path name="compress-offload-playback5" />
     </path>
 
+    <path name="compress-offload-playback5 speaker-and-bt-sco">
+        <path name="compress-offload-playback5 bt-sco" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback5 bt-sco-wb" />
+        <path name="compress-offload-playback5" />
+    </path>
+
     <path name="compress-offload-playback6">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="1" />
     </path>
@@ -1040,6 +1132,16 @@
         <path name="compress-offload-playback6" />
     </path>
 
+    <path name="compress-offload-playback6 speaker-and-bt-sco">
+        <path name="compress-offload-playback6 bt-sco" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback6 bt-sco-wb" />
+        <path name="compress-offload-playback6" />
+    </path>
+
     <path name="compress-offload-playback7">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="1" />
     </path>
@@ -1096,6 +1198,16 @@
         <path name="compress-offload-playback7" />
     </path>
 
+    <path name="compress-offload-playback7 speaker-and-bt-sco">
+        <path name="compress-offload-playback7 bt-sco" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback7 bt-sco-wb" />
+        <path name="compress-offload-playback7" />
+    </path>
+
     <path name="compress-offload-playback8">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="1" />
     </path>
@@ -1152,6 +1264,16 @@
         <path name="compress-offload-playback8" />
     </path>
 
+    <path name="compress-offload-playback8 speaker-and-bt-sco">
+        <path name="compress-offload-playback8 bt-sco" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback8 bt-sco-wb" />
+        <path name="compress-offload-playback8" />
+    </path>
+
     <path name="compress-offload-playback9">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
     </path>
@@ -1208,6 +1330,16 @@
         <path name="compress-offload-playback9" />
     </path>
 
+    <path name="compress-offload-playback9 speaker-and-bt-sco">
+        <path name="compress-offload-playback9 bt-sco" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-sco-wb">
+        <path name="compress-offload-playback9 bt-sco-wb" />
+        <path name="compress-offload-playback9" />
+    </path>
+
     <path name="audio-record">
         <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="1" />
     </path>
@@ -1541,6 +1673,63 @@
         <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
     </path>
 
+    <!-- VOIP Rx settings -->
+    <path name="audio-playback-voip">
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="audio-playback-voip bt-sco" />
+    </path>
+
+    <path name="audio-playback-voip afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip display-port">
+        <ctl name="DISPLAY_PORT Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="audio-playback-voip speaker-and-display-port">
+        <path name="audio-playback-voip display-port" />
+        <path name="audio-playback-voip" />
+    </path>
+
+    <!-- VoIP Tx settings -->
+    <path name="audio-record-voip">
+        <ctl name="MultiMedia10 Mixer SLIM_0_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip usb-headset-mic">
+        <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco">
+        <ctl name="MultiMedia10 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="audio-record-voip bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="audio-record-voip bt-sco" />
+    </path>
+
    <path name="spkr-rx-calib">
         <ctl name="SLIMBUS_DL_HL Switch"  value="1" />
     </path>
@@ -2240,6 +2429,22 @@
         <path name="unprocessed-handset-mic" />
     </path>
 
+    <path name="unprocessed-stereo-mic">
+        <path name="voice-rec-dmic-ef" />
+    </path>
+
+    <path name="unprocessed-three-mic">
+        <path name="three-mic" />
+    </path>
+
+    <path name="unprocessed-quad-mic">
+        <path name="quad-mic" />
+    </path>
+
+    <path name="unprocessed-headset-mic">
+        <path name="headset-mic" />
+    </path>
+
     <!-- USB TTY start -->
 
     <!-- full: both end tty -->
@@ -2271,8 +2476,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
@@ -2389,6 +2592,84 @@
         <path name="audio-ull-playback" />
     </path>
 
+    <path name="mmap-playback">
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-headphones">
+        <path name="mmap-playback" />
+        <path name="mmap-playback headphones" />
+    </path>
+
+    <path name="mmap-playback bt-sco">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback bt-sco-wb">
+        <ctl name="BT SampleRate" value="16000" />
+        <path name="mmap-playback bt-sco" />
+    </path>
+
+    <path name="mmap-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-buffer-playback hdmi">
+        <ctl name="HDMI Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="mmap-playback speaker-and-hdmi">
+        <path name="mmap-playback hdmi" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-display-port">
+        <path name="mmap-playback display-port" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-playback speaker-and-usb-headphones">
+        <path name="mmap-playback usb-headphones" />
+        <path name="mmap-playback" />
+    </path>
+
+    <path name="mmap-record">
+      <ctl name="MultiMedia16 Mixer SLIM_0_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco">
+      <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="1" />
+    </path>
+
+    <path name="mmap-record bt-sco-wb">
+        <ctl name="BT SampleRate" value="KHZ_16" />
+        <path name="mmap-record bt-sco" />
+    </path>
+
+    <path name="mmap-record capture-fm">
+      <ctl name="MultiMedia16 Mixer SLIM_8_TX" value="1" />
+    </path>
+
+    <path name="mmap-record usb-headset-mic">
+       <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
     <path name="hifi-playback display-port">
         <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
     </path>
diff --git a/configs/sdm670/sdm670.mk b/configs/sdm670/sdm670.mk
index 518ced2..cfe7517 100644
--- a/configs/sdm670/sdm670.mk
+++ b/configs/sdm670/sdm670.mk
@@ -6,7 +6,7 @@
 ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
 USE_CUSTOM_AUDIO_POLICY := 1
 AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
-AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := true
+AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := false
 AUDIO_FEATURE_ENABLED_DYNAMIC_ECNS := false
 AUDIO_FEATURE_ENABLED_EXTN_FORMATS := true
 AUDIO_FEATURE_ENABLED_EXTN_FLAC_DECODER := true
@@ -90,6 +90,7 @@
     hardware/qcom/audio/configs/sdm670/mixer_paths_tashalite.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tashalite.xml \
     hardware/qcom/audio/configs/sdm670/mixer_paths_i2s.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_i2s.xml \
     hardware/qcom/audio/configs/sdm670/mixer_paths_skuw.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_skuw.xml \
+    hardware/qcom/audio/configs/sdm670/mixer_paths_360cam.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_360cam.xml \
     hardware/qcom/audio/configs/sdm670/audio_tuning_mixer.txt:$(TARGET_COPY_OUT_VENDOR)/etc/audio_tuning_mixer.txt \
     hardware/qcom/audio/configs/sdm670/audio_tuning_mixer_tavil.txt:$(TARGET_COPY_OUT_VENDOR)/etc/audio_tuning_mixer_tavil.txt \
     hardware/qcom/audio/configs/sdm670/audio_tuning_mixer_tasha.txt:$(TARGET_COPY_OUT_VENDOR)/etc/audio_tuning_mixer_tasha.txt \
diff --git a/configs/sdm845/audio_policy_configuration.xml b/configs/sdm845/audio_policy_configuration.xml
index a19151c..024a471 100644
--- a/configs/sdm845/audio_policy_configuration.xml
+++ b/configs/sdm845/audio_policy_configuration.xml
@@ -323,7 +323,7 @@
                 <route type="mix" sink="Line"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="HDMI"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,compress_passthrough"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,compress_passthrough,voip_rx"/>
                 <route type="mix" sink="Proxy"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
                 <route type="mix" sink="FM"
@@ -345,11 +345,11 @@
                 <route type="mix" sink="mmap_no_irq_in"
                        sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,USB Device In"/>
                 <route type="mix" sink="BT A2DP Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,deep_buffer,direct_pcm,compressed_offload"/>
                 <route type="mix" sink="BT A2DP Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,deep_buffer,direct_pcm,compressed_offload"/>
                 <route type="mix" sink="BT A2DP Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                       sources="primary output,deep_buffer,direct_pcm,compressed_offload"/>
                 <route type="mix" sink="hifi_input" sources="USB Device In" />
             </routes>
 
diff --git a/configs/sdm845/mixer_paths_qvr.xml b/configs/sdm845/mixer_paths_qvr.xml
index 65ce1b3..d552852 100644
--- a/configs/sdm845/mixer_paths_qvr.xml
+++ b/configs/sdm845/mixer_paths_qvr.xml
@@ -2463,8 +2463,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm845/mixer_paths_skuk.xml b/configs/sdm845/mixer_paths_skuk.xml
index 242fa57..d59b25e 100644
--- a/configs/sdm845/mixer_paths_skuk.xml
+++ b/configs/sdm845/mixer_paths_skuk.xml
@@ -2164,8 +2164,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm845/mixer_paths_tavil.xml b/configs/sdm845/mixer_paths_tavil.xml
index 8a6360b..4a05733 100644
--- a/configs/sdm845/mixer_paths_tavil.xml
+++ b/configs/sdm845/mixer_paths_tavil.xml
@@ -229,6 +229,8 @@
     <ctl name="VoiceMMode1_Tx Mixer AFE_PCM_TX_MMode1" value="0" />
     <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="0" />
     <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="0" />
+    <!-- Multimode Voice1 Display-Port -->
+    <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode1" value="0" />
     <!-- Miltimode Voice1 end-->
 
     <!-- Multimode Voice2 -->
@@ -243,6 +245,8 @@
     <ctl name="VoiceMMode2_Tx Mixer AFE_PCM_TX_MMode2" value="0" />
     <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="0" />
     <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="0" />
+    <!-- Multimode Voice2 Display-Port -->
+    <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode2" value="0" />
     <!-- Multimode Voice2 end-->
 
     <!-- Voice external ec. reference -->
@@ -535,6 +539,10 @@
         <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
     </path>
 
+    <path name="echo-reference display-port">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="DISPLAY_PORT" />
+    </path>
+
     <path name="echo-reference headphones-44.1">
     </path>
 
@@ -1621,6 +1629,16 @@
         <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="1" />
     </path>
 
+    <path name="voicemmode1-call display-port-and-usb-headset-mic">
+        <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call display-port">
+        <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+    </path>
+
     <path name="voicemmode2-call">
         <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode2" value="1" />
         <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
@@ -1656,6 +1674,16 @@
         <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
     </path>
 
+    <path name="voicemmode2-call display-port-and-usb-headset-mic">
+        <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call display-port">
+        <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+    </path>
+
     <!-- VoIP Rx settings -->
     <path name="audio-playback-voip">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
@@ -1841,6 +1869,10 @@
         <path name="speaker-mono-2" />
     </path>
 
+    <path name="speaker-hdk">
+        <path name="speaker-mono" />
+    </path>
+
     <path name="speaker-fluid">
         <path name="speaker-mono" />
     </path>
@@ -2063,6 +2095,14 @@
         <path name="handset-qrd" />
     </path>
 
+    <path name="voice-handset-hdk">
+        <path name="handset-qrd" />
+    </path>
+
+    <path name="voice-handset-tmus-hdk">
+        <path name="handset-qrd" />
+    </path>
+
     <path name="voice-speaker">
         <path name="speaker-mono" />
     </path>
@@ -2071,6 +2111,10 @@
         <path name="speaker-mono-2" />
     </path>
 
+    <path name="voice-speaker-hdk">
+        <path name="speaker-mono" />
+    </path>
+
     <path name="voice-speaker-2">
         <path name="speaker-mono-2" />
     </path>
@@ -2412,6 +2456,11 @@
         <path name="handset-qrd" />
     </path>
 
+    <path name="voice-tty-hco-handset-hdk">
+        <ctl name="TTY Mode" value="HCO" />
+        <path name="handset-qrd" />
+    </path>
+
     <path name="voice-tty-full-headset-mic">
         <path name="amic2" />
         <ctl name="ADC2 Volume" value="0" />
@@ -2480,8 +2529,6 @@
     </path>
 
     <path name="bt-a2dp">
-        <ctl name="BT SampleRate" value="KHZ_48" />
-        <ctl name="AFE Input Channels" value="Two" />
         <ctl name="SLIM7_RX ADM Channels" value="Two" />
     </path>
 
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 83751da..38a84c1 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -231,6 +231,9 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.adm.buffering.ms=2
 
+#enable use of display-port for voice usecases
+vendor.audio.enable.dp.for.voice=false
+
 # for HIDL related packages
 PRODUCT_PACKAGES += \
     android.hardware.audio@2.0-service \
diff --git a/configure.ac b/configure.ac
index 9a17458..805d00c 100644
--- a/configure.ac
+++ b/configure.ac
@@ -75,6 +75,7 @@
 fi
 AC_SUBST([TARGET_CFLAGS])
 
+AM_CONDITIONAL([QTI_AUDIO_SERVER_ENABLED],[test x$BOARD_SUPPORTS_QTI_AUDIO_SERVER = xtrue])
 AM_CONDITIONAL([QAHW_SUPPORT], [test x$BOARD_SUPPORTS_QAHW = xtrue])
 AM_CONDITIONAL([HDMI_EDID],    [test x$AUDIO_FEATURE_ENABLED_HDMI_EDID = xtrue])
 AM_CONDITIONAL([FM_POWER_OPT], [test x$AUDIO_FEATURE_ENABLED_FM_POWER_OPT = xtrue])
diff --git a/hal/Android.mk b/hal/Android.mk
index c8bfb10..9b320f0 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -8,7 +8,7 @@
 
 AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
 
-ifneq ($(filter msm8974 msm8226 msm8610 apq8084 msm8994 msm8992 msm8996 msm8998 apq8098_latv sdm845 sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8610 apq8084 msm8994 msm8992 msm8996 msm8998 apq8098_latv sdm845 sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM = msm8974
   MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -39,6 +39,9 @@
 ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_CFLAGS := -DPLATFORM_SDM670
 endif
+ifneq ($(filter qcs605,$(TARGET_BOARD_PLATFORM)),)
+  LOCAL_CFLAGS := -DPLATFORM_QCS605
+endif
 endif
 
 ifneq ($(filter msm8916 msm8909 msm8952 msm8937 thorium msm8953 msmgold sdm660,$(TARGET_BOARD_PLATFORM)),)
@@ -65,7 +68,7 @@
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
index 8ce3784..c484713 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -35,6 +35,7 @@
 #include "audio_hw.h"
 #include "platform.h"
 #include "platform_api.h"
+#include "audio_extn.h"
 #include <stdlib.h>
 #include <cutils/str_parms.h>
 #include <hardware/audio.h>
@@ -56,6 +57,7 @@
 #define ENC_MEDIA_FMT_APTX_HD                              0x00013200
 #define ENC_MEDIA_FMT_SBC                                  0x00010BF2
 #define ENC_MEDIA_FMT_CELT                                 0x00013221
+#define ENC_MEDIA_FMT_LDAC                                 0x00013224
 #define MEDIA_FMT_AAC_AOT_LC                               2
 #define MEDIA_FMT_AAC_AOT_SBR                              5
 #define MEDIA_FMT_AAC_AOT_PS                               29
@@ -70,6 +72,9 @@
 #define MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR                1
 #define MIXER_ENC_CONFIG_BLOCK     "SLIM_7_RX Encoder Config"
 #define MIXER_ENC_BIT_FORMAT       "AFE Input Bit Format"
+#define MIXER_SCRAMBLER_MODE       "AFE Scrambler Mode"
+#define MIXER_SAMPLE_RATE          "BT SampleRate"
+#define MIXER_AFE_IN_CHANNELS      "AFE Input Channels"
 #define MIXER_ENC_FMT_SBC          "SBC"
 #define MIXER_ENC_FMT_AAC          "AAC"
 #define MIXER_ENC_FMT_APTX         "APTX"
@@ -79,14 +84,16 @@
 #define ENCODER_LATENCY_APTX       40
 #define ENCODER_LATENCY_APTX_HD    20
 #define ENCODER_LATENCY_AAC        70
-//To Do: Fine Tune Encoder CELT latency.
+//To Do: Fine Tune Encoder CELT/LDAC latency.
 #define ENCODER_LATENCY_CELT       40
+#define ENCODER_LATENCY_LDAC       40
 #define DEFAULT_SINK_LATENCY_SBC       140
 #define DEFAULT_SINK_LATENCY_APTX      160
 #define DEFAULT_SINK_LATENCY_APTX_HD   180
 #define DEFAULT_SINK_LATENCY_AAC       180
-//To Do: Fine Tune Default CELT Latency.
+//To Do: Fine Tune Default CELT/LDAC Latency.
 #define DEFAULT_SINK_LATENCY_CELT      180
+#define DEFAULT_SINK_LATENCY_LDAC      180
 
 /*
  * Below enum values are extended from audio_base.h to
@@ -95,12 +102,15 @@
  * between IPC lib and Audio HAL.
  */
 typedef enum {
-    ENC_CODEC_TYPE_INVALID = 4294967295u, // 0xFFFFFFFFUL
-    ENC_CODEC_TYPE_AAC = 67108864u, // 0x04000000UL
-    ENC_CODEC_TYPE_SBC = 520093696u, // 0x1F000000UL
-    ENC_CODEC_TYPE_APTX = 536870912u, // 0x20000000UL
-    ENC_CODEC_TYPE_APTX_HD = 553648128u, // 0x21000000UL
+    ENC_CODEC_TYPE_INVALID = AUDIO_FORMAT_INVALID, // 0xFFFFFFFFUL
+    ENC_CODEC_TYPE_AAC = AUDIO_FORMAT_AAC, // 0x04000000UL
+    ENC_CODEC_TYPE_SBC = AUDIO_FORMAT_SBC, // 0x1F000000UL
+    ENC_CODEC_TYPE_APTX = AUDIO_FORMAT_APTX, // 0x20000000UL
+    ENC_CODEC_TYPE_APTX_HD = AUDIO_FORMAT_APTX_HD, // 0x21000000UL
+#ifndef LINUX_ENABLED
     ENC_CODEC_TYPE_APTX_DUAL_MONO = 570425344u, // 0x22000000UL
+#endif
+    ENC_CODEC_TYPE_LDAC = AUDIO_FORMAT_LDAC, // 0x23000000UL
     ENC_CODEC_TYPE_CELT = 603979776u, // 0x24000000UL
 }enc_codec_t;
 
@@ -115,6 +125,7 @@
                                enc_codec_t *codec_type);
 typedef int (*audio_check_a2dp_ready_t)(void);
 typedef uint16_t (*audio_get_a2dp_sink_latency_t)(void);
+typedef int (*audio_is_scrambling_enabled_t)(void);
 
 enum A2DP_STATE {
     A2DP_STATE_CONNECTED,
@@ -140,9 +151,11 @@
     audio_get_codec_config_t audio_get_codec_config;
     audio_check_a2dp_ready_t audio_check_a2dp_ready;
     audio_get_a2dp_sink_latency_t audio_get_a2dp_sink_latency;
+    audio_is_scrambling_enabled_t audio_is_scrambling_enabled;
     enum A2DP_STATE bt_state;
     enc_codec_t bt_encoder_format;
     uint32_t enc_sampling_rate;
+    uint32_t enc_channels;
     bool a2dp_started;
     bool a2dp_suspended;
     int  a2dp_total_active_session_request;
@@ -171,7 +184,7 @@
     uint16_t      aac_fmt_flag;
     uint16_t      channel_cfg;
     uint32_t      sample_rate;
-} __packed;
+} __attribute__ ((packed));
 
 /* SBC encoder configuration structure. */
 typedef struct sbc_enc_cfg_t sbc_enc_cfg_t;
@@ -191,7 +204,7 @@
     uint32_t      alloc_method;
     uint32_t      bit_rate;
     uint32_t      sample_rate;
-} __packed;
+} __attribute__ ((packed));
 
 
 /* supported num_channels are Mono/Stereo
@@ -207,7 +220,7 @@
     uint16_t      reserved;
     uint8_t       channel_mapping[8];
     uint32_t      custom_size;
-} __packed;
+} __attribute__ ((packed));
 
 struct celt_specific_enc_cfg_t
 {
@@ -216,13 +229,13 @@
     uint16_t      complexity;
     uint16_t      prediction_mode;
     uint16_t      vbr_flag;
-} __packed;
+} __attribute__ ((packed));
 
 struct celt_enc_cfg_t
 {
     struct custom_enc_cfg_t  custom_cfg;
     struct celt_specific_enc_cfg_t celt_cfg;
-} __packed;
+} __attribute__ ((packed));
 
 /* sync_mode introduced with APTX V2 libraries
  * sync mode: 0x0 = stereo sync mode
@@ -232,14 +245,27 @@
 struct aptx_v2_enc_cfg_ext_t
 {
     uint32_t       sync_mode;
-} __packed;
+} __attribute__ ((packed));
 
 /* APTX struct for combining custom enc and V2 fields */
 struct aptx_enc_cfg_t
 {
     struct custom_enc_cfg_t  custom_cfg;
     struct aptx_v2_enc_cfg_ext_t aptx_v2_cfg;
-} __packed;
+} __attribute__ ((packed));
+
+struct ldac_specific_enc_cfg_t
+{
+    uint32_t      bit_rate;
+    uint16_t      channel_mode;
+    uint16_t      mtu;
+} __attribute__ ((packed));
+
+struct ldac_enc_cfg_t
+{
+    struct custom_enc_cfg_t  custom_cfg;
+    struct ldac_specific_enc_cfg_t ldac_cfg;
+} __attribute__ ((packed));
 
 /* In LE BT source code uses system/audio.h for below
  * structure definition. To avoid multiple definition
@@ -313,6 +339,17 @@
     uint32_t bitrate; /*32000 - 1536000, 139500*/
 } audio_celt_encoder_config;
 
+/* Information about BT LDAC encoder configuration
+ * This data is used between audio HAL module and
+ * BT IPC library to configure DSP encoder
+ */
+typedef struct {
+    uint32_t sampling_rate; /*44100,48000,88200,96000*/
+    uint32_t bit_rate; /*303000,606000,909000(in bits per second)*/
+    uint16_t channel_mode; /* 0, 4, 2, 1*/
+    uint16_t mtu; /*679*/
+} audio_ldac_encoder_config;
+
 /*********** END of DSP configurable structures ********************/
 
 /* API to identify DSP encoder captabilities */
@@ -346,6 +383,10 @@
             ALOGD("%s: celt offload supported\n",__func__);
             a2dp.is_a2dp_offload_supported = true;
             break;
+        } else if (strcmp(tok, "ldac") == 0) {
+            ALOGD("%s: ldac offload supported\n",__func__);
+            a2dp.is_a2dp_offload_supported = true;
+            break;
         }
         tok = strtok_r(NULL, "-", &saveptr);
     };
@@ -399,6 +440,8 @@
                         dlsym(a2dp.bt_lib_handle,"audio_check_a2dp_ready");
             a2dp.audio_get_a2dp_sink_latency = (audio_get_a2dp_sink_latency_t)
                         dlsym(a2dp.bt_lib_handle,"audio_get_a2dp_sink_latency");
+            a2dp.audio_is_scrambling_enabled = (audio_is_scrambling_enabled_t)
+                        dlsym(a2dp.bt_lib_handle,"audio_is_scrambling_enabled");
         }
     }
 
@@ -444,11 +487,138 @@
     a2dp.a2dp_suspended = false;
     a2dp.bt_encoder_format = ENC_CODEC_TYPE_INVALID;
     a2dp.enc_sampling_rate = 48000;
+    a2dp.enc_channels = 2;
     a2dp.bt_state = A2DP_STATE_DISCONNECTED;
 
     return 0;
 }
 
+static void a2dp_check_and_set_scrambler()
+{
+    bool scrambler_mode = false;
+    struct mixer_ctl *ctrl_scrambler_mode = NULL;
+    if (a2dp.audio_is_scrambling_enabled && (a2dp.bt_state != A2DP_STATE_DISCONNECTED))
+        scrambler_mode = a2dp.audio_is_scrambling_enabled();
+
+    if (scrambler_mode) {
+        //enable scrambler in dsp
+        ctrl_scrambler_mode = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                            MIXER_SCRAMBLER_MODE);
+        if (!ctrl_scrambler_mode) {
+            ALOGE(" ERROR scrambler mode mixer control not identifed");
+            return;
+        } else {
+            if (mixer_ctl_set_value(ctrl_scrambler_mode, 0, true) != 0) {
+                ALOGE("%s: Could not set scrambler mode", __func__);
+                return;
+            }
+        }
+    }
+}
+
+static void a2dp_set_backend_cfg()
+{
+    char *rate_str = NULL, *in_channels = NULL;
+    uint32_t sampling_rate = a2dp.enc_sampling_rate;
+    struct mixer_ctl *ctl_sample_rate = NULL, *ctrl_in_channels = NULL;
+
+    //For LDAC encoder open slimbus port at 96Khz for 48Khz input
+    //and 88.2Khz for 44.1Khz input.
+    if ((a2dp.bt_encoder_format == ENC_CODEC_TYPE_LDAC) &&
+        (sampling_rate == 48000 || sampling_rate == 44100 )) {
+        sampling_rate = sampling_rate *2;
+    }
+    //Configure backend sampling rate
+    switch (sampling_rate) {
+    case 44100:
+        rate_str = "KHZ_44P1";
+        break;
+    case 48000:
+        rate_str = "KHZ_48";
+        break;
+    case 88200:
+        rate_str = "KHZ_88P2";
+        break;
+    case 96000:
+        rate_str = "KHZ_96";
+        break;
+    default:
+        rate_str = "KHZ_48";
+        break;
+    }
+
+    ALOGD("%s: set backend sample rate =%s", __func__, rate_str);
+    ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                        MIXER_SAMPLE_RATE);
+    if (!ctl_sample_rate) {
+        ALOGE(" ERROR backend sample rate mixer control not identifed");
+        return;
+    } else {
+        if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
+            ALOGE("%s: Failed to set backend sample rate =%s", __func__, rate_str);
+            return;
+        }
+    }
+
+    //Configure AFE input channels
+    switch (a2dp.enc_channels) {
+    case 1:
+        in_channels = "One";
+        break;
+    case 2:
+    default:
+        in_channels = "Two";
+        break;
+    }
+
+    ALOGD("%s: set afe input channels =%d", __func__, a2dp.enc_channels);
+    ctrl_in_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                        MIXER_AFE_IN_CHANNELS);
+    if (!ctrl_in_channels) {
+        ALOGE(" ERROR AFE input channels mixer control not identifed");
+        return;
+    } else {
+        if (mixer_ctl_set_enum_by_string(ctrl_in_channels, in_channels) != 0) {
+            ALOGE("%s: Failed to set AFE in channels =%d", __func__, a2dp.enc_channels);
+            return;
+        }
+    }
+}
+
+static void a2dp_reset_backend_cfg()
+{
+    char *rate_str = "KHZ_8", *in_channels = "Zero";
+    struct mixer_ctl *ctl_sample_rate = NULL, *ctrl_in_channels = NULL;
+
+    //reset backend sampling rate
+    ALOGD("%s: reset backend sample rate =%s", __func__, rate_str);
+    ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                        MIXER_SAMPLE_RATE);
+    if (!ctl_sample_rate) {
+        ALOGE(" ERROR backend sample rate mixer control not identifed");
+        return;
+    } else {
+        if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
+            ALOGE("%s: Failed to reset backend sample rate =%s", __func__, rate_str);
+            return;
+        }
+    }
+
+    //reset AFE input channels
+    ALOGD("%s: reset afe input channels =%s", __func__, in_channels);
+    ctrl_in_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                        MIXER_AFE_IN_CHANNELS);
+    if (!ctrl_in_channels) {
+        ALOGE(" ERROR AFE input channels mixer control not identifed");
+        return;
+    } else {
+        if (mixer_ctl_set_enum_by_string(ctrl_in_channels, in_channels) != 0) {
+            ALOGE("%s: Failed to reset AFE in channels =%d", __func__, a2dp.enc_channels);
+            return;
+        }
+    }
+}
+
 /* API to configure SBC DSP encoder */
 bool configure_sbc_enc_format(audio_sbc_encoder_config *sbc_bt_cfg)
 {
@@ -514,19 +684,119 @@
     is_configured = true;
     a2dp.bt_encoder_format = ENC_CODEC_TYPE_SBC;
     a2dp.enc_sampling_rate = sbc_bt_cfg->sampling_rate;
+
+    if (sbc_dsp_cfg.channel_mode == MEDIA_FMT_SBC_CHANNEL_MODE_MONO)
+        a2dp.enc_channels = 1;
+    else
+        a2dp.enc_channels = 2;
+
     ALOGV("Successfully updated SBC enc format with samplingrate: %d channelmode:%d",
            sbc_dsp_cfg.sample_rate, sbc_dsp_cfg.channel_mode);
 fail:
     return is_configured;
 }
 
+#ifndef LINUX_ENABLED
+static int update_aptx_dsp_config_v2(struct aptx_enc_cfg_t *aptx_dsp_cfg,
+                                     audio_aptx_encoder_config *aptx_bt_cfg)
+{
+    int ret = 0;
+
+    if(aptx_dsp_cfg == NULL || aptx_bt_cfg == NULL) {
+        ALOGE("Invalid param, aptx_dsp_cfg %p aptx_bt_cfg %p",
+              aptx_dsp_cfg, aptx_bt_cfg);
+        return -EINVAL;
+    }
+
+    memset(aptx_dsp_cfg, 0x0, sizeof(struct aptx_enc_cfg_t));
+    aptx_dsp_cfg->custom_cfg.enc_format = ENC_MEDIA_FMT_APTX;
+
+    if (!a2dp.is_aptx_dual_mono_supported) {
+        aptx_dsp_cfg->custom_cfg.sample_rate = aptx_bt_cfg->default_cfg->sampling_rate;
+        aptx_dsp_cfg->custom_cfg.num_channels = aptx_bt_cfg->default_cfg->channels;
+    } else {
+        aptx_dsp_cfg->custom_cfg.sample_rate = aptx_bt_cfg->dual_mono_cfg->sampling_rate;
+        aptx_dsp_cfg->custom_cfg.num_channels = aptx_bt_cfg->dual_mono_cfg->channels;
+        aptx_dsp_cfg->aptx_v2_cfg.sync_mode = aptx_bt_cfg->dual_mono_cfg->sync_mode;
+    }
+
+    switch(aptx_dsp_cfg->custom_cfg.num_channels) {
+        case 1:
+            aptx_dsp_cfg->custom_cfg.channel_mapping[0] = PCM_CHANNEL_C;
+            break;
+        case 2:
+        default:
+            aptx_dsp_cfg->custom_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+            aptx_dsp_cfg->custom_cfg.channel_mapping[1] = PCM_CHANNEL_R;
+            break;
+    }
+    a2dp.enc_channels = aptx_dsp_cfg->custom_cfg.num_channels;
+    if (!a2dp.is_aptx_dual_mono_supported) {
+        a2dp.enc_sampling_rate = aptx_bt_cfg->default_cfg->sampling_rate;
+        ALOGV("Successfully updated APTX enc format with samplingrate: %d \
+               channels:%d", aptx_dsp_cfg->custom_cfg.sample_rate,
+               aptx_dsp_cfg->custom_cfg.num_channels);
+    } else {
+        a2dp.enc_sampling_rate = aptx_bt_cfg->dual_mono_cfg->sampling_rate;
+        ALOGV("Successfully updated APTX dual mono enc format with \
+               samplingrate: %d channels:%d syncmode %d",
+               aptx_dsp_cfg->custom_cfg.sample_rate,
+               aptx_dsp_cfg->custom_cfg.num_channels,
+               aptx_dsp_cfg->aptx_v2_cfg.sync_mode);
+    }
+    return ret;
+}
+#else
+static int update_aptx_dsp_config_v1(struct custom_enc_cfg_t *aptx_dsp_cfg,
+                                     audio_aptx_encoder_config *aptx_bt_cfg)
+{
+    int ret = 0;
+
+    if(aptx_dsp_cfg == NULL || aptx_bt_cfg == NULL) {
+        ALOGE("Invalid param, aptx_dsp_cfg %p aptx_bt_cfg %p",
+              aptx_dsp_cfg, aptx_bt_cfg);
+        return -EINVAL;
+    }
+
+    memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_t));
+    aptx_dsp_cfg->enc_format = ENC_MEDIA_FMT_APTX;
+    aptx_dsp_cfg->sample_rate = aptx_bt_cfg->sampling_rate;
+    aptx_dsp_cfg->num_channels = aptx_bt_cfg->channels;
+    switch(aptx_dsp_cfg->num_channels) {
+        case 1:
+            aptx_dsp_cfg->channel_mapping[0] = PCM_CHANNEL_C;
+            break;
+        case 2:
+        default:
+            aptx_dsp_cfg->channel_mapping[0] = PCM_CHANNEL_L;
+            aptx_dsp_cfg->channel_mapping[1] = PCM_CHANNEL_R;
+            break;
+    }
+
+    ALOGV("Updated APTX enc format with samplingrate: %d channels:%d",
+            aptx_dsp_cfg->sample_rate, aptx_dsp_cfg->num_channels);
+
+    return ret;
+}
+#endif
+
 /* API to configure APTX DSP encoder */
 bool configure_aptx_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
 {
     struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
-    struct aptx_enc_cfg_t aptx_dsp_cfg;
+    int mixer_size;
     bool is_configured = false;
     int ret = 0;
+    int sample_rate_backup;
+#ifndef LINUX_ENABLED
+    struct aptx_enc_cfg_t aptx_dsp_cfg;
+    mixer_size = sizeof(struct aptx_enc_cfg_t);
+    sample_rate_backup = aptx_bt_cfg->default_cfg->sampling_rate;
+#else
+    struct custom_enc_cfg_t aptx_dsp_cfg;
+    mixer_size = sizeof(struct custom_enc_cfg_t);
+    sample_rate_backup = aptx_bt_cfg->sampling_rate;
+#endif
 
     if(aptx_bt_cfg == NULL)
         return false;
@@ -538,30 +808,18 @@
         goto fail;
     }
 
-    memset(&aptx_dsp_cfg, 0x0, sizeof(struct aptx_enc_cfg_t));
-    aptx_dsp_cfg.custom_cfg.enc_format = ENC_MEDIA_FMT_APTX;
+#ifndef LINUX_ENABLED
+    ret = update_aptx_dsp_config_v2(&aptx_dsp_cfg, aptx_bt_cfg);
+#else
+    ret = update_aptx_dsp_config_v1(&aptx_dsp_cfg, aptx_bt_cfg);
+#endif
 
-    if (!a2dp.is_aptx_dual_mono_supported) {
-        aptx_dsp_cfg.custom_cfg.sample_rate = aptx_bt_cfg->default_cfg->sampling_rate;
-        aptx_dsp_cfg.custom_cfg.num_channels = aptx_bt_cfg->default_cfg->channels;
-    } else {
-        aptx_dsp_cfg.custom_cfg.sample_rate = aptx_bt_cfg->dual_mono_cfg->sampling_rate;
-        aptx_dsp_cfg.custom_cfg.num_channels = aptx_bt_cfg->dual_mono_cfg->channels;
-        aptx_dsp_cfg.aptx_v2_cfg.sync_mode = aptx_bt_cfg->dual_mono_cfg->sync_mode;
-    }
-
-    switch(aptx_dsp_cfg.custom_cfg.num_channels) {
-        case 1:
-            aptx_dsp_cfg.custom_cfg.channel_mapping[0] = PCM_CHANNEL_C;
-            break;
-        case 2:
-        default:
-            aptx_dsp_cfg.custom_cfg.channel_mapping[0] = PCM_CHANNEL_L;
-            aptx_dsp_cfg.custom_cfg.channel_mapping[1] = PCM_CHANNEL_R;
-            break;
+    if (ret) {
+        is_configured = false;
+        goto fail;
     }
     ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
-                              sizeof(struct aptx_enc_cfg_t));
+                              mixer_size);
     if (ret != 0) {
         ALOGE("%s: Failed to set APTX encoder config", __func__);
         is_configured = false;
@@ -583,26 +841,20 @@
     }
     is_configured = true;
     a2dp.bt_encoder_format = ENC_CODEC_TYPE_APTX;
-    if (!a2dp.is_aptx_dual_mono_supported) {
-        a2dp.enc_sampling_rate = aptx_bt_cfg->default_cfg->sampling_rate;
-        ALOGV("Successfully updated APTX enc format with samplingrate: %d \
-               channels:%d", aptx_dsp_cfg.custom_cfg.sample_rate,
-               aptx_dsp_cfg.custom_cfg.num_channels);
-    } else {
-        a2dp.enc_sampling_rate = aptx_bt_cfg->dual_mono_cfg->sampling_rate;
-        ALOGV("Successfully updated APTX dual mono enc format with \
-               samplingrate: %d channels:%d syncmode %d",
-               aptx_dsp_cfg.custom_cfg.sample_rate,
-               aptx_dsp_cfg.custom_cfg.num_channels,
-               aptx_dsp_cfg.aptx_v2_cfg.sync_mode);
-    }
 fail:
+    /*restore sample rate */
+    if(!is_configured)
+        a2dp.enc_sampling_rate = sample_rate_backup;
     return is_configured;
 }
 
 /* API to configure APTX HD DSP encoder
  */
+#ifndef LINUX_ENABLED
 bool configure_aptx_hd_enc_format(audio_aptx_default_config *aptx_bt_cfg)
+#else
+bool configure_aptx_hd_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
+#endif
 {
     struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
     struct custom_enc_cfg_t aptx_dsp_cfg;
@@ -655,6 +907,7 @@
     is_configured = true;
     a2dp.bt_encoder_format = ENC_CODEC_TYPE_APTX_HD;
     a2dp.enc_sampling_rate = aptx_bt_cfg->sampling_rate;
+    a2dp.enc_channels = aptx_bt_cfg->channels;
     ALOGV("Successfully updated APTX HD encformat with samplingrate: %d channels:%d",
            aptx_dsp_cfg.sample_rate, aptx_dsp_cfg.num_channels);
 fail:
@@ -719,6 +972,7 @@
     is_configured = true;
     a2dp.bt_encoder_format = ENC_CODEC_TYPE_AAC;
     a2dp.enc_sampling_rate = aac_bt_cfg->sampling_rate;
+    a2dp.enc_channels = aac_bt_cfg->channels;;
     ALOGV("Successfully updated AAC enc format with samplingrate: %d channels:%d",
            aac_dsp_cfg.sample_rate, aac_dsp_cfg.channel_cfg);
 fail:
@@ -786,11 +1040,79 @@
     is_configured = true;
     a2dp.bt_encoder_format = ENC_CODEC_TYPE_CELT;
     a2dp.enc_sampling_rate = celt_bt_cfg->sampling_rate;
+    a2dp.enc_channels = celt_bt_cfg->channels;
     ALOGV("Successfully updated CELT encformat with samplingrate: %d channels:%d",
            celt_dsp_cfg.custom_cfg.sample_rate, celt_dsp_cfg.custom_cfg.num_channels);
 fail:
     return is_configured;
 }
+
+bool configure_ldac_enc_format(audio_ldac_encoder_config *ldac_bt_cfg)
+{
+    struct mixer_ctl *ldac_enc_data = NULL, *ctrl_bit_format = NULL;
+    struct ldac_enc_cfg_t ldac_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+    if(ldac_bt_cfg == NULL)
+        return false;
+
+    ldac_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ldac_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    memset(&ldac_dsp_cfg, 0x0, sizeof(struct ldac_enc_cfg_t));
+
+    ldac_dsp_cfg.custom_cfg.enc_format = ENC_MEDIA_FMT_LDAC;
+    ldac_dsp_cfg.custom_cfg.sample_rate = ldac_bt_cfg->sampling_rate;
+    ldac_dsp_cfg.ldac_cfg.channel_mode = ldac_bt_cfg->channel_mode;
+    switch(ldac_dsp_cfg.ldac_cfg.channel_mode) {
+        case 4:
+            ldac_dsp_cfg.custom_cfg.channel_mapping[0] = PCM_CHANNEL_C;
+            ldac_dsp_cfg.custom_cfg.num_channels = 1;
+            break;
+        case 2:
+        case 1:
+        default:
+            ldac_dsp_cfg.custom_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+            ldac_dsp_cfg.custom_cfg.channel_mapping[1] = PCM_CHANNEL_R;
+            ldac_dsp_cfg.custom_cfg.num_channels = 2;
+            break;
+    }
+
+    ldac_dsp_cfg.custom_cfg.custom_size = sizeof(struct ldac_enc_cfg_t);
+    ldac_dsp_cfg.ldac_cfg.mtu = ldac_bt_cfg->mtu;
+    ldac_dsp_cfg.ldac_cfg.bit_rate = ldac_bt_cfg->bit_rate;
+    ret = mixer_ctl_set_array(ldac_enc_data, (void *)&ldac_dsp_cfg,
+                              sizeof(struct ldac_enc_cfg_t));
+    if (ret != 0) {
+        ALOGE("%s: Failed to set LDAC encoder config", __func__);
+        is_configured = false;
+        goto fail;
+    }
+    ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_BIT_FORMAT);
+    if (!ctrl_bit_format) {
+        ALOGE(" ERROR  bit format CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+    if (ret != 0) {
+        ALOGE("%s: Failed to set bit format to encoder", __func__);
+        is_configured = false;
+        goto fail;
+    }
+    is_configured = true;
+    a2dp.bt_encoder_format = ENC_CODEC_TYPE_LDAC;
+    a2dp.enc_sampling_rate = ldac_bt_cfg->sampling_rate;
+    a2dp.enc_channels = ldac_dsp_cfg.custom_cfg.num_channels;
+    ALOGV("Successfully updated LDAC encformat with samplingrate: %d channels:%d",
+           ldac_dsp_cfg.custom_cfg.sample_rate, ldac_dsp_cfg.custom_cfg.num_channels);
+fail:
+    return is_configured;
+}
+
 bool configure_a2dp_encoder_format()
 {
     void *codec_info = NULL;
@@ -815,16 +1137,24 @@
             break;
         case ENC_CODEC_TYPE_APTX:
             ALOGD(" Received APTX encoder supported BT device");
+#ifndef LINUX_ENABLED
             a2dp.is_aptx_dual_mono_supported = false;
             aptx_encoder_cfg.default_cfg = (audio_aptx_default_config *)codec_info;
+#endif
             is_configured =
               configure_aptx_enc_format(&aptx_encoder_cfg);
             break;
         case ENC_CODEC_TYPE_APTX_HD:
             ALOGD(" Received APTX HD encoder supported BT device");
+#ifndef LINUX_ENABLED
             is_configured =
               configure_aptx_hd_enc_format((audio_aptx_default_config *)codec_info);
+#else
+            is_configured =
+              configure_aptx_hd_enc_format((audio_aptx_encoder_config *)codec_info);
+#endif
             break;
+#ifndef LINUX_ENABLED
         case ENC_CODEC_TYPE_APTX_DUAL_MONO:
             ALOGD(" Received APTX dual mono encoder supported BT device");
             a2dp.is_aptx_dual_mono_supported = true;
@@ -832,6 +1162,7 @@
             is_configured =
               configure_aptx_enc_format(&aptx_encoder_cfg);
             break;
+#endif
         case ENC_CODEC_TYPE_AAC:
             ALOGD(" Received AAC encoder supported BT device");
             is_configured =
@@ -842,6 +1173,11 @@
             is_configured =
               configure_celt_enc_format((audio_celt_encoder_config *)codec_info);
             break;
+        case ENC_CODEC_TYPE_LDAC:
+            ALOGD(" Received LDAC encoder supported BT device");
+            is_configured =
+              configure_ldac_enc_format((audio_ldac_encoder_config *)codec_info);
+            break;
         default:
             ALOGD(" Received Unsupported encoder formar");
             is_configured = false;
@@ -889,8 +1225,11 @@
         }
     }
 
-    if (a2dp.a2dp_started)
+    if (a2dp.a2dp_started) {
         a2dp.a2dp_total_active_session_request++;
+        a2dp_check_and_set_scrambler();
+        a2dp_set_backend_cfg();
+    }
 
     ALOGD("start A2DP playback total active sessions :%d",
           a2dp.a2dp_total_active_session_request);
@@ -947,6 +1286,7 @@
         else
             ALOGV("stop steam to BT IPC lib successful");
         reset_a2dp_enc_config_params();
+        a2dp_reset_backend_cfg();
     }
     if(!a2dp.a2dp_total_active_session_request)
        a2dp.a2dp_started = false;
@@ -1067,13 +1407,8 @@
     return a2dp.is_handoff_in_progress || !a2dp.a2dp_started;
 }
 
-void audio_extn_a2dp_get_apptype_params(uint32_t *sample_rate,
-                                        uint32_t *bit_width)
+void audio_extn_a2dp_get_sample_rate(int *sample_rate)
 {
-    if(a2dp.bt_encoder_format == ENC_CODEC_TYPE_APTX_HD)
-        *bit_width = 24;
-    else
-        *bit_width = 16;
     *sample_rate = a2dp.enc_sampling_rate;
 }
 
@@ -1117,14 +1452,15 @@
 {
     uint32_t latency = 0;
     int avsync_runtime_prop = 0;
-    int sbc_offset = 0, aptx_offset = 0, aptxhd_offset = 0, aac_offset = 0, celt_offset = 0;
+    int sbc_offset = 0, aptx_offset = 0, aptxhd_offset = 0,
+        aac_offset = 0, celt_offset = 0, ldac_offset = 0;
     char value[PROPERTY_VALUE_MAX];
 
     memset(value, '\0', sizeof(char)*PROPERTY_VALUE_MAX);
     avsync_runtime_prop = property_get("vendor.audio.a2dp.codec.latency", value, NULL);
     if (avsync_runtime_prop > 0) {
-        if (sscanf(value, "%d/%d/%d/%d/%d",
-                  &sbc_offset, &aptx_offset, &aptxhd_offset, &aac_offset, &celt_offset) != 5) {
+        if (sscanf(value, "%d/%d/%d/%d/%d%d",
+                  &sbc_offset, &aptx_offset, &aptxhd_offset, &aac_offset, &celt_offset, &ldac_offset) != 6) {
             ALOGI("Failed to parse avsync offset params from '%s'.", value);
             avsync_runtime_prop = 0;
         }
@@ -1156,6 +1492,10 @@
             latency = (avsync_runtime_prop > 0) ? celt_offset : ENCODER_LATENCY_CELT;
             latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_CELT : slatency;
             break;
+        case ENC_CODEC_TYPE_LDAC:
+            latency = (avsync_runtime_prop > 0) ? ldac_offset : ENCODER_LATENCY_LDAC;
+            latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_LDAC : slatency;
+            break;
         default:
             latency = 200;
             break;
diff --git a/hal/audio_extn/adsp_hdlr.h b/hal/audio_extn/adsp_hdlr.h
index b265e42..1c257fc 100644
--- a/hal/audio_extn/adsp_hdlr.h
+++ b/hal/audio_extn/adsp_hdlr.h
@@ -66,7 +66,7 @@
 #define audio_extn_adsp_hdlr_stream_close(handle)                            (0)
 #define audio_extn_adsp_hdlr_stream_set_callback(handle, callback, cookie)   (0)
 #define audio_extn_adsp_hdlr_stream_set_param(handle, cmd, param)            (0)
-#define audio_extn_adsp_hdlr_stream_register_event(stream_data, param, cb)   (0)
+#define audio_extn_adsp_hdlr_stream_register_event(handle, param, cb, cookie) (0)
 #define audio_extn_adsp_hdlr_stream_deregister_event(handle, param)          (0)
 #endif
 
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 81f51e4..2239338 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -99,6 +99,10 @@
 #define AUDIO_FORMAT_AC4  0x22000000UL
 #endif
 
+#ifndef AUDIO_FORMAT_LDAC
+#define AUDIO_FORMAT_LDAC 0x23000000UL
+#endif
+
 #ifndef AUDIO_OUTPUT_FLAG_MAIN
 #define AUDIO_OUTPUT_FLAG_MAIN 0x8000000
 #endif
@@ -216,6 +220,7 @@
 #define audio_extn_usb_get_max_channels(p)                             (0)
 #define audio_extn_usb_get_max_bit_width(p)                            (0)
 #define audio_extn_usb_get_sup_sample_rates(t, s, l)                   (0)
+#define audio_extn_usb_is_tunnel_supported()                           (0)
 #else
 void audio_extn_usb_init(void *adev);
 void audio_extn_usb_deinit();
@@ -232,6 +237,7 @@
 int audio_extn_usb_get_max_channels(bool playback);
 int audio_extn_usb_get_max_bit_width(bool playback);
 int audio_extn_usb_get_sup_sample_rates(int type, uint32_t *sr, uint32_t l);
+bool audio_extn_usb_is_tunnel_supported();
 #endif
 
 #ifndef SPLIT_A2DP_ENABLED
@@ -241,7 +247,7 @@
 #define audio_extn_a2dp_set_parameters(parms)            (0)
 #define audio_extn_a2dp_is_force_device_switch()         (0)
 #define audio_extn_a2dp_set_handoff_mode(is_on)          (0)
-#define audio_extn_a2dp_get_apptype_params(sample_rate,bit_width)    (0)
+#define audio_extn_a2dp_get_sample_rate(sample_rate)     (0)
 #define audio_extn_a2dp_get_encoder_latency()            (0)
 #define audio_extn_a2dp_is_ready()                       (0)
 #define audio_extn_a2dp_is_suspended()                   (0)
@@ -252,8 +258,7 @@
 void audio_extn_a2dp_set_parameters(struct str_parms *parms);
 bool audio_extn_a2dp_is_force_device_switch();
 void audio_extn_a2dp_set_handoff_mode(bool is_on);
-void audio_extn_a2dp_get_apptype_params(uint32_t *sample_rate,
-                                        uint32_t *bit_width);
+void audio_extn_a2dp_get_sample_rate(int *sample_rate);
 uint32_t audio_extn_a2dp_get_encoder_latency();
 bool audio_extn_a2dp_is_ready();
 bool audio_extn_a2dp_is_suspended();
diff --git a/hal/audio_extn/compress_in.c b/hal/audio_extn/compress_in.c
index 49997f5..305ed5b 100644
--- a/hal/audio_extn/compress_in.c
+++ b/hal/audio_extn/compress_in.c
@@ -82,7 +82,9 @@
 static const audio_usecase_t cin_usecases[] = {
     USECASE_AUDIO_RECORD_COMPRESS2,
     USECASE_AUDIO_RECORD_COMPRESS3,
-    USECASE_AUDIO_RECORD_COMPRESS4
+    USECASE_AUDIO_RECORD_COMPRESS4,
+    USECASE_AUDIO_RECORD_COMPRESS5,
+    USECASE_AUDIO_RECORD_COMPRESS6
 };
 
 static pthread_mutex_t cin_lock = PTHREAD_MUTEX_INITIALIZER;
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index db51122..0d9f9ae 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -55,7 +55,7 @@
 #define HFP_RX_VOLUME     "SEC AUXPCM LOOPBACK Volume"
 #elif defined PLATFORM_MSM8996
 #define HFP_RX_VOLUME     "PRI AUXPCM LOOPBACK Volume"
-#elif defined (PLATFORM_MSM8998) || defined (PLATFORM_MSMFALCON) || defined (PLATFORM_SDM845) || defined (PLATFORM_SDM670)
+#elif defined (PLATFORM_MSM8998) || defined (PLATFORM_MSMFALCON) || defined (PLATFORM_SDM845) || defined (PLATFORM_SDM670) || defined (PLATFORM_QCS605)
 #define HFP_RX_VOLUME     "SLIMBUS_7 LOOPBACK Volume"
 #else
 #define HFP_RX_VOLUME     "Internal HFP RX Volume"
diff --git a/hal/audio_extn/ip_hdlr_intf.c b/hal/audio_extn/ip_hdlr_intf.c
old mode 100644
new mode 100755
index a667a22..ece7af3
--- a/hal/audio_extn/ip_hdlr_intf.c
+++ b/hal/audio_extn/ip_hdlr_intf.c
@@ -53,6 +53,9 @@
 #include "audio_hw.h"
 #include "audio_defs.h"
 #include "platform.h"
+#include "audio_extn.h"
+#include "platform_api.h"
+#include "adsp_hdlr.h"
 
 /* These values defined by ADSP */
 #define ADSP_DEC_SERVICE_ID 1
@@ -482,7 +485,7 @@
         ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
                        "Playback ION LIB FD %d", pcm_device_id);
         if (ret < 0) {
-            ALOGE("%s:[%d] snprintf failed",__func__, ip_hdlr->ref_cnt, ret);
+            ALOGE("%s:[%d] snprintf failed %d", __func__, ip_hdlr->ref_cnt, ret);
             goto dlclose;
         }
         ALOGV("%s: fd = %d  pcm_id = %d", __func__, lib_fd, pcm_device_id);
diff --git a/hal/audio_extn/keep_alive.c b/hal/audio_extn/keep_alive.c
index d6a9055..698d8d0 100644
--- a/hal/audio_extn/keep_alive.c
+++ b/hal/audio_extn/keep_alive.c
@@ -30,6 +30,7 @@
 #define LOG_TAG "keep_alive"
 /*#define LOG_NDEBUG 0*/
 
+#include <cutils/properties.h>
 #include <stdlib.h>
 #include <cutils/log.h>
 #include "audio_hw.h"
@@ -375,7 +376,7 @@
     int ret;
 
     if (ka.state == STATE_DISABLED)
-        return;
+        return 0;
 
     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
     if (ret >= 0) {
diff --git a/hal/audio_extn/qaf.c b/hal/audio_extn/qaf.c
old mode 100644
new mode 100755
index 5d41f59..c1c0af7
--- a/hal/audio_extn/qaf.c
+++ b/hal/audio_extn/qaf.c
@@ -131,6 +131,7 @@
 #endif
 
 void set_hdmi_configuration_to_module();
+void set_bt_configuration_to_module();
 
 struct qaf_adsp_hdlr_config_state {
     struct audio_adsp_event event_params;
@@ -228,6 +229,7 @@
 
     pthread_mutex_t lock;
 
+    bool bt_connect;
     bool hdmi_connect;
     int hdmi_sink_channels;
 
@@ -808,7 +810,7 @@
                    / audio_stream_out_frame_size(stream)
                    / out->stream.common.get_sample_rate(&out->stream.common));
         }
-    } else if (ret < bytes) {
+    } else if (ret < (ssize_t)bytes) {
         //partial buffer copied to the module.
         DEBUG_MSG_VV("Not enough space available in mm module, post msg to cb thread");
         (void)qaf_send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
@@ -996,6 +998,36 @@
     return ret;
 }
 
+static int qaf_out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int ret = 0;
+    uint64_t frames=0;
+    struct qaf_module* qaf_mod = NULL;
+    ALOGV("%s, Output Stream %p,dsp frames %d",__func__, stream, (int)dsp_frames);
+
+    qaf_mod = get_qaf_module_for_input_stream(out);
+    if (!qaf_mod) {
+        ret = out->stream.get_render_position(stream, dsp_frames);
+        ALOGV("%s, non qaf_MOD DSP FRAMES %d",__func__, (int)dsp_frames);
+        return ret;
+    }
+
+    if (p_qaf->passthrough_out) {
+        pthread_mutex_lock(&p_qaf->lock);
+        ret = p_qaf->passthrough_out->stream.get_render_position((struct audio_stream_out *)p_qaf->passthrough_out, dsp_frames);
+        pthread_mutex_unlock(&p_qaf->lock);
+        ALOGV("%s, PASS THROUGH DSP FRAMES %p",__func__, dsp_frames);
+        return ret;
+        }
+    frames=*dsp_frames;
+    ret = qaf_get_rendered_frames(out, &frames);
+    *dsp_frames = (uint32_t)frames;
+    ALOGV("%s, DSP FRAMES %d",__func__, (int)dsp_frames);
+    return ret;
+}
+
 static int qaf_out_get_presentation_position(const struct audio_stream_out *stream,
                                              uint64_t *frames,
                                              struct timespec *timestamp)
@@ -1877,7 +1909,7 @@
 {
     ALOGV("%s %d", __func__, __LINE__);
     unsigned char* license_data = NULL;
-    device_license_config_t lic_config = {0};
+    device_license_config_t lic_config = {NULL, 0, 0};
     int ret = -ENOSYS;
 
     struct qaf_module *qaf_mod = NULL;
@@ -1951,8 +1983,10 @@
                                              qaf_mod,
                                              &notify_event_callback,
                                              AUDIO_DATA_EVENT_V2);
-
-    set_hdmi_configuration_to_module();
+    if(p_qaf->bt_connect)
+         set_bt_configuration_to_module();
+    else
+         set_hdmi_configuration_to_module();
 
 #ifdef AUDIO_EXTN_IP_HDLR_ENABLED
     if (mod_type == MS12) {
@@ -1997,7 +2031,7 @@
     if (p_qaf->qaf_mod[mmtype].qaf_audio_session_open == NULL ||
         p_qaf->qaf_mod[mmtype].qaf_audio_stream_open == NULL) {
         ERROR_MSG("Session or Stream is NULL");
-        return status;
+        return -ENOTSUP;
     }
     //Open the module session, if not opened already.
     status = audio_extn_qaf_session_open(mmtype, out);
@@ -2098,7 +2132,7 @@
     if (status != 0) {
         //If no stream is active then close the session.
         qaf_session_close(qaf_mod);
-        return status;
+        return 0;
     }
 
     //If Device is HDMI, QAF passthrough is enabled and there is no previous QAF passthrough input stream.
@@ -2316,6 +2350,7 @@
      */
     out->devices = val;
 
+#ifndef SPLIT_A2DP_ENABLED
     if (val == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) {
         //If device is BT then open the BT stream if not already opened.
         if ( audio_extn_bt_hal_get_output_stream(qaf_mod->bt_hdl) == NULL
@@ -2334,6 +2369,7 @@
             audio_extn_bt_hal_close_output_stream(qaf_mod->bt_hdl);
         }
     }
+#endif
 
     if (p_qaf->passthrough_in == out) { //Device routing is received for QAF passthrough stream.
 
@@ -2512,16 +2548,21 @@
         return ret;
     }
 
+#ifndef LINUX_ENABLED
+//Bypass QAF for dummy PCM session opened by APM during boot time
+    if(flags == 0) {
+        ALOGD("bypassing QAF for flags is equal to none");
+        return ret;
+    }
+#endif
+
     out = (struct stream_out *)*stream_out;
 
     ret = qaf_stream_open(out, config, flags, devices);
-    if (ret == -ENOTSUP) {
+    if (ret < 0) {
+        ERROR_MSG("Error opening QAF stream err[%d]! QAF bypassed.", ret);
         //Stream not supported by QAF, Bypass QAF.
         return 0;
-    } else if (ret < 0) {
-        ERROR_MSG("Error opening QAF stream err[%d]!", ret);
-        adev_close_output_stream(dev, *stream_out);
-        return ret;
     }
 
     /* Override function pointers based on qaf definitions */
@@ -2534,6 +2575,7 @@
     out->stream.common.standby = qaf_out_standby;
     out->stream.common.set_parameters = qaf_out_set_parameters;
     out->stream.get_latency = qaf_out_get_latency;
+    out->stream.get_render_position = qaf_out_get_render_position;
     out->stream.write = qaf_out_write;
     out->stream.get_presentation_position = qaf_out_get_presentation_position;
     out->platform_latency = 0;
@@ -2566,7 +2608,7 @@
     struct qaf_module* qaf_mod = get_qaf_module_for_input_stream(out);
 
     if (!qaf_mod) {
-        DEBUG_MSG("qaf module is NULL, by passing qaf on close output stream");
+        DEBUG_MSG("qaf module is NULL, bypassing qaf on close output stream");
         /*closing non-MS12/default output stream opened with qaf */
         adev_close_output_stream(dev, stream);
         return;
@@ -2609,6 +2651,50 @@
     return (prop_enabled);
 }
 
+void set_bt_configuration_to_module()
+{
+    if (!p_qaf) {
+        return;
+    }
+
+    if (!p_qaf->bt_connect) {
+        DEBUG_MSG("BT is not connected.");
+        return;
+    }
+
+    struct str_parms *qaf_params;
+    char *format_params = NULL;
+
+    qaf_params = str_parms_create();
+    if (qaf_params) {
+        //ms12 wrapper don't support bt, treat this as speaker and routign to bt
+        //will take care as a part of data callback notifier
+        str_parms_add_str(qaf_params,
+            AUDIO_QAF_PARAMETER_KEY_DEVICE,
+            AUDIO_QAF_PARAMETER_VALUE_DEVICE_SPEAKER);
+
+        str_parms_add_str(qaf_params,
+                          AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+                          AUDIO_QAF_PARAMETER_VALUE_PCM);
+        format_params = str_parms_to_str(qaf_params);
+
+        if (p_qaf->qaf_mod[MS12].session_handle && p_qaf->qaf_mod[MS12].qaf_audio_session_set_param) {
+            ALOGE(" Configuring BT/speaker for MS12 wrapper");
+            p_qaf->qaf_mod[MS12].qaf_audio_session_set_param(p_qaf->qaf_mod[MS12].session_handle,
+                                                         format_params);
+        }
+        if (p_qaf->qaf_mod[DTS_M8].session_handle
+                && p_qaf->qaf_mod[DTS_M8].qaf_audio_session_set_param) {
+            ALOGE(" Configuring BT/speaker for MS12 wrapper");
+            p_qaf->qaf_mod[DTS_M8].qaf_audio_session_set_param(p_qaf->qaf_mod[DTS_M8].session_handle,
+                                                           format_params);
+        }
+    }
+    str_parms_destroy(qaf_params);
+
+}
+
+
 /* Query HDMI EDID and sets module output accordingly.*/
 void set_hdmi_configuration_to_module()
 {
@@ -2831,6 +2917,9 @@
             set_hdmi_configuration_to_module();
 
         } else if (val & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) {
+            p_qaf->bt_connect = 1;
+            set_bt_configuration_to_module();
+#ifndef SPLIT_A2DP_ENABLED
             for (k = 0; k < MAX_MM_MODULE_TYPE; k++) {
                 if (!p_qaf->qaf_mod[k].bt_hdl) {
                     DEBUG_MSG("Opening a2dp output...");
@@ -2841,6 +2930,7 @@
                     }
                 }
             }
+#endif
         }
         //TODO else if: Need to consider other devices.
     }
@@ -2876,6 +2966,11 @@
             str_parms_destroy(qaf_params);
             close_qaf_passthrough_stream();
         } else if (val & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) {
+        p_qaf->bt_connect = 0;
+        //reconfig HDMI as end device (if connected)
+        if(p_qaf->hdmi_connect)
+            set_hdmi_configuration_to_module();
+#ifndef SPLIT_A2DP_ENABLED
             DEBUG_MSG("Closing a2dp output...");
             for (k = 0; k < MAX_MM_MODULE_TYPE; k++) {
                 if (p_qaf->qaf_mod[k].bt_hdl) {
@@ -2883,6 +2978,7 @@
                     p_qaf->qaf_mod[k].bt_hdl = NULL;
                 }
             }
+#endif
         }
         //TODO else if: Need to consider other devices.
     }
diff --git a/hal/audio_extn/source_track.c b/hal/audio_extn/source_track.c
index e5e6c06..e078be4 100644
--- a/hal/audio_extn/source_track.c
+++ b/hal/audio_extn/source_track.c
@@ -161,6 +161,7 @@
     case SND_DEVICE_IN_VOICE_SPEAKER_DMIC:
     case SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE:
     case SND_DEVICE_IN_VOICE_SPEAKER_QMIC:
+    case SND_DEVICE_IN_HANDSET_GENERIC_QMIC:
         ret = true;
         break;
     default:
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index 60216b1..8fa47a8 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -1005,6 +1005,11 @@
     return usbmod->is_capture_supported;
 }
 
+bool audio_extn_usb_is_tunnel_supported()
+{
+    return true;
+}
+
 void audio_extn_usb_add_device(audio_devices_t device, int card)
 {
     struct usb_card_config *usb_card_info;
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index cac1369..e03fdc8 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -139,6 +139,7 @@
     STRING_TO_ENUM(AUDIO_INPUT_FLAG_RAW),
     STRING_TO_ENUM(AUDIO_INPUT_FLAG_SYNC),
     STRING_TO_ENUM(AUDIO_INPUT_FLAG_TIMESTAMP),
+    STRING_TO_ENUM(AUDIO_INPUT_FLAG_COMPRESS),
 };
 
 const struct string_to_enum s_format_name_to_enum_table[] = {
@@ -736,12 +737,6 @@
             sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
     }
 
-    if(devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
-        //TODO: Handle fractional sampling rate configuration for LL
-        audio_extn_a2dp_get_apptype_params(&sample_rate, &bit_width);
-        ALOGI("%s using %d sampling rate %d bit width for A2DP CoPP",
-              __func__, sample_rate, bit_width);
-    }
 
     ALOGV("%s: flags: %x, format: %x sample_rate %d, profile %s, app_type %d",
            __func__, flags, format, sample_rate, profile, app_type_cfg->app_type);
@@ -988,6 +983,14 @@
             (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
             /* Reset to default if no native stream is active*/
             usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        } else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+                 /*
+                  * For a2dp playback get encoder sampling rate and set copp sampling rate,
+                  * for bit width use the stream param only.
+                  */
+                   audio_extn_a2dp_get_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
+                   ALOGI("%s using %d sample rate rate for A2DP CoPP",
+                        __func__, usecase->stream.out->app_type_cfg.sample_rate);
         }
         sample_rate = usecase->stream.out->app_type_cfg.sample_rate;
 
@@ -2069,8 +2072,8 @@
 #endif
 
 #define MAX_SND_CARD 8
-#define RETRY_US 500000
-#define RETRY_NUMBER 10
+#define RETRY_US 1000000
+#define RETRY_NUMBER 40
 
 int audio_extn_utils_get_snd_card_num()
 {
@@ -2078,24 +2081,31 @@
     void *hw_info = NULL;
     struct mixer *mixer = NULL;
     int retry_num = 0;
-    int snd_card_num = 0;
+    int snd_card_num = 0, min_snd_card_num = 0;
     char* snd_card_name = NULL;
 
-    while (snd_card_num < MAX_SND_CARD) {
-        mixer = mixer_open(snd_card_num);
+    /*
+    * Try with all the sound cards ( 0 to 8 ) and if none of them were detected
+    * sleep for 1 sec and try detections with sound card 0 again.
+    * If sound card gets detected, check if it is relevant, if not check with the
+    * other sound cards. To ensure that the irrelevant sound card is not check again,
+    * we maintain it in min_snd_card_num.
+    */
+    while (retry_num < RETRY_NUMBER) {
 
-        while (!mixer && retry_num < RETRY_NUMBER) {
-            usleep(RETRY_US);
-            mixer = mixer_open(snd_card_num);
-            retry_num++;
+        while (snd_card_num <= MAX_SND_CARD) {
+           mixer = mixer_open(snd_card_num);
+           if (!mixer)
+              snd_card_num++;
+           else
+              break;
         }
 
         if (!mixer) {
-            ALOGE("%s: Unable to open the mixer card: %d", __func__,
-                   snd_card_num);
-            retry_num = 0;
-            snd_card_num++;
-            continue;
+           usleep(RETRY_US);
+           snd_card_num = min_snd_card_num;
+           retry_num++;
+           continue;
         }
 
         snd_card_name = strdup(mixer_get_name(mixer));
@@ -2112,8 +2122,11 @@
             break;
         }
         ALOGE("%s: Failed to init hardware info", __func__);
-        retry_num = 0;
-        snd_card_num++;
+        min_snd_card_num++;
+        snd_card_num = min_snd_card_num;
+
+        if (snd_card_num >= MAX_SND_CARD)
+            break;
 
         free(snd_card_name);
         snd_card_name = NULL;
@@ -2123,12 +2136,14 @@
     }
     if (snd_card_name)
         free(snd_card_name);
+
     if (mixer)
         mixer_close(mixer);
+
     if (hw_info)
         hw_info_deinit(hw_info);
 
-    if (snd_card_num >= MAX_SND_CARD) {
+    if ((snd_card_num >= MAX_SND_CARD) || (retry_num >= RETRY_NUMBER)) {
         ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
         return -1;
     }
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index aa52ef8..4506115 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -116,6 +116,8 @@
     .period_size = VOIP_IO_BUF_SIZE(DEFAULT_VOIP_SAMP_RATE, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE)/2,
     .period_count = 2,
     .format = PCM_FORMAT_S16_LE,
+    .avail_min = VOIP_IO_BUF_SIZE(DEFAULT_VOIP_SAMP_RATE, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE)/2,
+    .stop_threshold = INT_MAX,
 };
 
 #define MIN_CHANNEL_COUNT                1
@@ -300,6 +302,8 @@
     [USECASE_AUDIO_RECORD_COMPRESS2] = "audio-record-compress2",
     [USECASE_AUDIO_RECORD_COMPRESS3] = "audio-record-compress3",
     [USECASE_AUDIO_RECORD_COMPRESS4] = "audio-record-compress4",
+    [USECASE_AUDIO_RECORD_COMPRESS5] = "audio-record-compress5",
+    [USECASE_AUDIO_RECORD_COMPRESS6] = "audio-record-compress6",
     [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
     [USECASE_AUDIO_RECORD_FM_VIRTUAL] = "fm-virtual-record",
     [USECASE_AUDIO_RECORD_MMAP] = "mmap-record",
@@ -934,7 +938,6 @@
 #define check_and_enable_effect(x) ENOSYS
 #endif
 
-
 int pcm_ioctl(struct pcm *pcm, int request, ...)
 {
     va_list ap;
@@ -2733,10 +2736,11 @@
                     ret = -errno;
             }
             else if (ret == -ETIMEDOUT)
-                compress_drain(out->compr);
+                ret = compress_drain(out->compr);
             else
                 ALOGE("%s: Next track returned error %d",__func__, ret);
-            if (ret != -ENETRESET) {
+            if (-ENETRESET != ret && !(-EINTR == ret &&
+                        CARD_STATUS_OFFLINE == out->card_status)) {
                 send_callback = true;
                 pthread_mutex_lock(&out->lock);
                 out->send_new_metadata = 1;
@@ -2749,10 +2753,15 @@
             break;
         case OFFLOAD_CMD_DRAIN:
             ALOGD("copl(%p):calling compress_drain", out);
-            compress_drain(out->compr);
-            ALOGD("copl(%p):calling compress_drain", out);
-            send_callback = true;
-            event = STREAM_CBK_EVENT_DRAIN_READY;
+            ret = compress_drain(out->compr);
+            ALOGD("copl(%p):out of compress_drain", out);
+            // EINTR check avoids drain interruption due to SSR
+            if (-ENETRESET != ret && !(-EINTR == ret &&
+                        CARD_STATUS_OFFLINE == out->card_status)) {
+                send_callback = true;
+                event = STREAM_CBK_EVENT_DRAIN_READY;
+            } else
+                ALOGI("%s: Block drain ready event during SSR", __func__);
             break;
         case OFFLOAD_CMD_ERROR:
             ALOGD("copl(%p): sending error callback to AF", out);
@@ -3349,8 +3358,6 @@
 {
     struct stream_out *out = (struct stream_out *)stream;
     struct audio_device *adev = out->dev;
-    struct audio_usecase *uc_info;
-    struct listnode *node;
     bool do_stop = true;
 
     ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
@@ -3396,13 +3403,6 @@
         if (do_stop) {
             stop_output_stream(out);
         }
-        //restore output device for active usecase when current snd device and output device mismatch
-        list_for_each(node, &adev->usecase_list) {
-            uc_info = node_to_item(node, struct audio_usecase, list);
-            if ((uc_info->type == PCM_PLAYBACK) &&
-                (uc_info->out_snd_device != platform_get_output_snd_device(adev->platform, uc_info->stream.out)))
-                select_devices(adev, uc_info->id);
-        }
         pthread_mutex_unlock(&adev->lock);
     }
     pthread_mutex_unlock(&out->lock);
@@ -3597,7 +3597,8 @@
          * Avoid this by routing audio to speaker until standby.
          */
         if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
-                (val == AUDIO_DEVICE_NONE)) {
+                (val == AUDIO_DEVICE_NONE) &&
+                !audio_extn_a2dp_is_ready()) {
                 val = AUDIO_DEVICE_OUT_SPEAKER;
         }
         /* To avoid a2dp to sco overlapping / BT device improper state
@@ -3646,7 +3647,7 @@
             out->devices = new_dev;
 
             if (output_drives_call(adev, out)) {
-                if(!voice_is_in_call(adev)) {
+                if (!voice_is_call_state_active(adev)) {
                     if (adev->mode == AUDIO_MODE_IN_CALL) {
                         adev->current_call_output = out;
                         ret = voice_start_call(adev);
@@ -3729,7 +3730,9 @@
     }
 
     //suspend, resume handling block
-    if (out->dynamic_pm_qos_enabled) {
+    //remove QOS only if vendor.audio.hal.dynamic.qos.config.supported is set to true
+    // and vendor.audio.hal.output.suspend.supported is set to true
+    if (out->hal_output_suspend_supported && out->dynamic_pm_qos_config_supported) {
         //check suspend parameter only for low latency and if the property
         //is enabled
         if (str_parms_get_str(parms, "suspend_playback", value, sizeof(value)) >= 0) {
@@ -3978,7 +3981,7 @@
     if (str_parms_get_str(query, "supports_hw_suspend", value, sizeof(value)) >= 0) {
         //only low latency track supports suspend_resume
         str_parms_add_int(reply, "supports_hw_suspend",
-                (out->dynamic_pm_qos_enabled));
+                (out->hal_output_suspend_supported));
         if (str)
             free(str);
         str = str_parms_to_str(reply);
@@ -4542,7 +4545,7 @@
                                    uint64_t *frames, struct timespec *timestamp)
 {
     struct stream_out *out = (struct stream_out *)stream;
-    int ret = -1;
+    int ret = -ENODATA;
     unsigned long dsp_frames;
 
     /* below piece of code is not guarded against any lock because audioFliner serializes
@@ -5533,7 +5536,8 @@
     out->convert_buffer = NULL;
     out->started = 0;
     out->a2dp_compress_mute = false;
-    out->dynamic_pm_qos_enabled = 0;
+    out->hal_output_suspend_supported = 0;
+    out->dynamic_pm_qos_config_supported = 0;
 
     if ((flags & AUDIO_OUTPUT_FLAG_BD) &&
         (property_get_bool("audio.matrix.limiter.enable", false)))
@@ -5918,8 +5922,11 @@
             out->stream.get_mmap_position = out_get_mmap_position;
         } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
             out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
-            out->dynamic_pm_qos_enabled = property_get_bool("vendor.audio.dynamic.qos.enable", false);
-            if (!out->dynamic_pm_qos_enabled) {
+            out->hal_output_suspend_supported =
+                property_get_bool("vendor.audio.hal.output.suspend.supported", false);
+            out->dynamic_pm_qos_config_supported =
+                property_get_bool("vendor.audio.hal.dynamic.qos.config.supported", false);
+            if (!out->dynamic_pm_qos_config_supported) {
                 ALOGI("%s: dynamic qos voting not enabled for platform", __func__);
             } else {
                 ALOGI("%s: dynamic qos voting enabled for platform", __func__);
@@ -6253,8 +6260,8 @@
                 goto done;
             }
             platform_cache_edid(adev->platform);
-        } else if ((audio_is_output_device(val) && (val & AUDIO_DEVICE_OUT_USB_DEVICE)) ||
-                   (audio_is_input_device(val) && ((uint32_t)val & AUDIO_DEVICE_IN_USB_DEVICE))) {
+        } else if (((audio_devices_t)val == AUDIO_DEVICE_OUT_USB_DEVICE) ||
+                   ((audio_devices_t)val == AUDIO_DEVICE_IN_USB_DEVICE)) {
             /*
              * Do not allow AFE proxy port usage by WFD source when USB headset is connected.
              * Per AudioPolicyManager, USB device is higher priority than WFD.
@@ -6269,8 +6276,10 @@
                 else
                     audio_extn_usb_add_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value));
             }
-            ALOGV("detected USB connect .. disable proxy");
-            adev->allow_afe_proxy_usage = false;
+            if (!audio_extn_usb_is_tunnel_supported()) {
+                ALOGV("detected USB connect .. disable proxy");
+                adev->allow_afe_proxy_usage = false;
+            }
         }
     }
 
@@ -6283,8 +6292,8 @@
          * invalidated prior to updating sysfs of the disconnect event
          * Invalidate will be handled by audio_extn_ext_disp_set_parameters()
          */
-        if ((audio_is_output_device(val) && (val & AUDIO_DEVICE_OUT_USB_DEVICE)) ||
-            (audio_is_input_device(val) && ((uint32_t)val == AUDIO_DEVICE_IN_USB_DEVICE))) {
+        if (((audio_devices_t)val == AUDIO_DEVICE_OUT_USB_DEVICE) ||
+            ((audio_devices_t)val == AUDIO_DEVICE_IN_USB_DEVICE)) {
             ret = str_parms_get_str(parms, "card", value, sizeof(value));
             if (ret >= 0) {
                 if (audio_is_output_device(val))
@@ -6292,8 +6301,10 @@
                 else
                     audio_extn_usb_remove_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value));
             }
-            ALOGV("detected USB disconnect .. enable proxy");
-            adev->allow_afe_proxy_usage = true;
+            if (!audio_extn_usb_is_tunnel_supported()) {
+                ALOGV("detected USB disconnect .. enable proxy");
+                adev->allow_afe_proxy_usage = true;
+            }
         }
     }
 
@@ -6600,6 +6611,15 @@
     in->standby = 1;
     in->capture_handle = handle;
     in->flags = flags;
+    in->bit_width = 16;
+    in->af_period_multiplier = 1;
+
+    /* Update config params with the requested sample rate and channels */
+    if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) &&
+          (adev->mode != AUDIO_MODE_IN_CALL)) {
+        ret = -EINVAL;
+        goto err_open;
+    }
 
     /* restrict 24 bit capture for unprocessed source only
      * for other sources if 24 bit requested reject 24 and set 16 bit capture only
@@ -6622,7 +6642,9 @@
             goto err_open;
         }
         channel_count = audio_channel_count_from_in_mask(config->channel_mask);
-    } else if (config->format == AUDIO_FORMAT_DEFAULT) {
+    }
+
+    if (config->format == AUDIO_FORMAT_DEFAULT) {
         config->format = AUDIO_FORMAT_PCM_16_BIT;
     } else if ((config->format == AUDIO_FORMAT_PCM_FLOAT) ||
                (config->format == AUDIO_FORMAT_PCM_32_BIT) ||
@@ -6645,11 +6667,8 @@
             if (config->sample_rate > 48000)
                 config->sample_rate = 48000;
             ret_error = true;
-        } else if (config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) {
-            in->config.format = PCM_FORMAT_S24_3LE;
-        } else if (config->format == AUDIO_FORMAT_PCM_8_24_BIT) {
-            in->config.format = PCM_FORMAT_S24_LE;
-        } else {
+        } else if (!(config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+                     config->format == AUDIO_FORMAT_PCM_8_24_BIT)) {
             config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
             ret_error = true;
         }
@@ -6678,32 +6697,18 @@
         in->realtime = 0;
         in->usecase = USECASE_AUDIO_RECORD_MMAP;
         in->config = pcm_config_mmap_capture;
+        in->config.format = pcm_format_from_audio_format(config->format);
         in->stream.start = in_start;
         in->stream.stop = in_stop;
         in->stream.create_mmap_buffer = in_create_mmap_buffer;
         in->stream.get_mmap_position = in_get_mmap_position;
-        in->af_period_multiplier = 1;
         ALOGV("%s: USECASE_AUDIO_RECORD_MMAP", __func__);
     } else if (in->realtime) {
         in->config = pcm_config_audio_capture_rt;
+        in->config.format = pcm_format_from_audio_format(config->format);
         in->sample_rate = in->config.rate;
         in->af_period_multiplier = af_period_multiplier;
-    } else {
-        in->config = pcm_config_audio_capture;
-        in->config.rate = config->sample_rate;
-        in->sample_rate = config->sample_rate;
-        in->af_period_multiplier = 1;
-    }
-    in->bit_width = 16;
-
-    /* Update config params with the requested sample rate and channels */
-    if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) &&
-          (adev->mode != AUDIO_MODE_IN_CALL)) {
-        ret = -EINVAL;
-        goto err_open;
-    }
-
-    if (is_usb_dev && may_use_hifi_record) {
+    } else if (is_usb_dev && may_use_hifi_record) {
         in->usecase = USECASE_AUDIO_RECORD_HIFI;
         in->config = pcm_config_audio_capture;
         frame_size = audio_stream_in_frame_size(&in->stream);
@@ -6713,7 +6718,6 @@
                                             false /*is_low_latency*/);
         in->config.period_size = buffer_size / frame_size;
         in->config.rate = config->sample_rate;
-        in->af_period_multiplier = 1;
         in->config.format = pcm_format_from_audio_format(config->format);
         in->config.channels = channel_count;
     } else if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) ||
@@ -6756,41 +6760,54 @@
         ret = audio_extn_cin_configure_input_stream(in);
         if (ret)
             goto err_open;
-    } else {
-        in->config.channels = channel_count;
-        if (!in->realtime) {
-            in->format = config->format;
-            frame_size = audio_stream_in_frame_size(&in->stream);
-            buffer_size = get_input_buffer_size(config->sample_rate,
-                                                config->format,
-                                                channel_count,
-                                                is_low_latency);
-            in->config.period_size = buffer_size / frame_size;
-        }
-
+    } else if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+        bool valid_rate = (config->sample_rate == 8000 ||
+                           config->sample_rate == 16000 ||
+                           config->sample_rate == 32000 ||
+                           config->sample_rate == 48000);
+        bool valid_ch = audio_channel_count_from_in_mask(in->channel_mask) == 1;
+        //XXX needed for voice_extn_compress_voip_open_input_stream
+        in->config.rate = config->sample_rate;
 #ifndef COMPRESS_VOIP_ENABLED
-        if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
-            (in->config.rate == 8000 || in->config.rate == 16000 ||
-             in->config.rate == 32000 || in->config.rate == 48000) &&
-            (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
-
+        if (valid_rate && valid_ch) {
             in->usecase = USECASE_AUDIO_RECORD_VOIP;
             in->config = default_pcm_config_voip_copp;
-            in->config.period_size = VOIP_IO_BUF_SIZE(in->sample_rate, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;
-            in->config.rate = in->sample_rate;
-#else
-        if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
-               (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||
-                voice_extn_compress_voip_is_active(in->dev)) &&
-               (voice_extn_compress_voip_is_format_supported(in->format)) &&
-               (in->config.rate == 8000 || in->config.rate == 16000 ||
-                in->config.rate == 32000 || in->config.rate == 48000) &&
-               (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
-            voice_extn_compress_voip_open_input_stream(in);
-#endif
+            in->config.period_size = VOIP_IO_BUF_SIZE(in->sample_rate,
+                                                      DEFAULT_VOIP_BUF_DURATION_MS,
+                                                      DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;
         }
+#else
+        if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||
+             voice_extn_compress_voip_is_active(in->dev)) &&
+            (voice_extn_compress_voip_is_format_supported(in->format)) &&
+            valid_rate && valid_ch) {
+            voice_extn_compress_voip_open_input_stream(in);
+        }
+#endif
+        else {
+            ALOGE("%s AUDIO_SOURCE_VOICE_COMMUNICATION invalid args", __func__);
+            ret = -EINVAL;
+            if (!valid_ch) config->channel_mask = 1;
+            if (!valid_rate) config->sample_rate = 48000;
+            goto err_open;
+        }
+        // update back to whatever was overwritten
+        in->config.rate = config->sample_rate;
+        in->sample_rate = config->sample_rate;
+    } else {
+        in->config = pcm_config_audio_capture;
+        in->config.rate = config->sample_rate;
+        in->config.format = pcm_format_from_audio_format(config->format);
+        in->config.channels = channel_count;
+        in->sample_rate = config->sample_rate;
+        in->format = config->format;
+        frame_size = audio_stream_in_frame_size(&in->stream);
+        buffer_size = get_input_buffer_size(config->sample_rate,
+                                            config->format,
+                                            channel_count,
+                                            is_low_latency);
+        in->config.period_size = buffer_size / frame_size;
     }
-
     audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,
                                                 &adev->streams_input_cfg_list,
                                                 devices, flags, in->format, in->sample_rate,
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 3130f93..c5ee848 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -147,6 +147,8 @@
     USECASE_AUDIO_RECORD_COMPRESS2,
     USECASE_AUDIO_RECORD_COMPRESS3,
     USECASE_AUDIO_RECORD_COMPRESS4,
+    USECASE_AUDIO_RECORD_COMPRESS5,
+    USECASE_AUDIO_RECORD_COMPRESS6,
     USECASE_AUDIO_RECORD_LOW_LATENCY,
     USECASE_AUDIO_RECORD_FM_VIRTUAL,
     USECASE_AUDIO_RECORD_HIFI,
@@ -339,7 +341,8 @@
     float volume_r;
 
     char pm_qos_mixer_path[MAX_MIXER_PATH_LEN];
-    int dynamic_pm_qos_enabled;
+    int hal_output_suspend_supported;
+    int dynamic_pm_qos_config_supported;
     bool stream_config_changed;
     mix_matrix_params_t pan_scale_params;
     mix_matrix_params_t downmix_params;
@@ -522,6 +525,7 @@
     bool bt_sco_on;
     struct audio_device_config_param *device_cfg_params;
     unsigned int interactive_usecase_state;
+    bool dp_allowed_for_voice;
 };
 
 int select_devices(struct audio_device *adev,
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index e67d387..56d9aec 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -302,6 +302,7 @@
     bool is_dsd_supported;
     bool is_asrc_supported;
     struct listnode acdb_meta_key_list;
+    bool use_generic_handset;
 };
 
 static bool is_external_codec = false;
@@ -336,6 +337,9 @@
     [USECASE_AUDIO_RECORD_COMPRESS2] = {-1, -1},
     [USECASE_AUDIO_RECORD_COMPRESS3] = {-1, -1},
     [USECASE_AUDIO_RECORD_COMPRESS4] = {-1, -1},
+    [USECASE_AUDIO_RECORD_COMPRESS5] = {-1, -1},
+    [USECASE_AUDIO_RECORD_COMPRESS6] = {-1, -1},
+
     [USECASE_AUDIO_RECORD_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
                                           LOWLATENCY_PCM_DEVICE},
     [USECASE_AUDIO_RECORD_FM_VIRTUAL] = {MULTIMEDIA2_PCM_DEVICE,
@@ -548,6 +552,7 @@
     [SND_DEVICE_IN_HANDSET_8MIC] = "handset-8mic",
     [SND_DEVICE_IN_EC_REF_LOOPBACK_MONO] = "ec-ref-loopback-mono",
     [SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO] = "ec-ref-loopback-stereo",
+    [SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = "quad-mic",
 };
 
 // Platform specific backend bit width table
@@ -710,7 +715,8 @@
     [SND_DEVICE_IN_HANDSET_6MIC] = 4,
     [SND_DEVICE_IN_HANDSET_8MIC] = 4,
     [SND_DEVICE_IN_EC_REF_LOOPBACK_MONO] = 4,
-    [SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO] = 4
+    [SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO] = 4,
+    [SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = 150
 };
 
 struct name_to_index {
@@ -859,6 +865,7 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_8MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_EC_REF_LOOPBACK_MONO)},
     {TO_NAME_INDEX(SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_GENERIC_QMIC)},
 };
 
 static char * backend_tag_table[SND_DEVICE_MAX] = {0};
@@ -884,6 +891,8 @@
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS2)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS3)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS4)},
+    {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS5)},
+    {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS6)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_LOW_LATENCY)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_MMAP)},
     {TO_NAME_INDEX(USECASE_VOICE_CALL)},
@@ -1652,6 +1661,7 @@
     hw_interface_table[SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = strdup("SLIMBUS_0_TX");
     hw_interface_table[SND_DEVICE_IN_UNPROCESSED_QUAD_MIC] = strdup("SLIMBUS_0_TX");
     hw_interface_table[SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = strdup("SLIMBUS_0_TX");
+    hw_interface_table[SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = strdup("SLIMBUS_0_TX");
 
     my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
     /*remove ALAC & APE from DSP decoder list based on software decoder availability*/
@@ -2662,6 +2672,9 @@
         }
     }
 
+    if (property_get_bool("vendor.audio.apptype.multirec.enabled", false))
+        my_data->use_generic_handset = true;
+
     my_data->edid_info = NULL;
     return my_data;
 }
@@ -4386,6 +4399,17 @@
                 snd_device = SND_DEVICE_IN_VOICE_USB_HEADSET_MIC;
             }
         }
+    } else if (my_data->use_generic_handset == true &&  //     system prop is enabled
+               (my_data->source_mic_type & SOURCE_QUAD_MIC) &&  // AND 4mic is available
+               ((in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) ||    // AND device is buit-in mic or back mic
+                (in_device & AUDIO_DEVICE_IN_BACK_MIC)) &&
+               (my_data->fluence_in_audio_rec == true &&       //  AND fluencepro is enabled
+                my_data->fluence_type & FLUENCE_QUAD_MIC) &&
+               (source == AUDIO_SOURCE_CAMCORDER ||           // AND source is cam/mic/unprocessed
+                source == AUDIO_SOURCE_UNPROCESSED ||
+                source == AUDIO_SOURCE_MIC)) {
+                snd_device = SND_DEVICE_IN_HANDSET_GENERIC_QMIC;
+                platform_set_echo_reference(adev, true, out_device);
     } else if (source == AUDIO_SOURCE_CAMCORDER) {
         if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC ||
             in_device & AUDIO_DEVICE_IN_BACK_MIC) {
@@ -4807,6 +4831,7 @@
         goto done_key_audcal;
     }
 
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
     /* parse audio calibration keys */
     ret = parse_audiocal_cfg(parms, &cal);
 
@@ -5100,6 +5125,8 @@
         ret=-EINVAL;
         goto done;
     }
+
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
     /* parse audiocal configuration keys */
     ret = parse_audiocal_cfg(query, &cal);
     if(ret == 0) {
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index d128d0f..aeefcc2 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -228,6 +228,7 @@
     SND_DEVICE_IN_HANDSET_8MIC,
     SND_DEVICE_IN_EC_REF_LOOPBACK_MONO,
     SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO,
+    SND_DEVICE_IN_HANDSET_GENERIC_QMIC,
     SND_DEVICE_IN_END,
 
     SND_DEVICE_MAX = SND_DEVICE_IN_END,
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
old mode 100755
new mode 100644
index 46dca39..bddaf97
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -383,11 +383,25 @@
         hw_info->num_snd_devices = ARRAY_SIZE(tavil_qrd_variant_devices);
         hw_info->is_stereo_spkr = false;
         strlcpy(hw_info->dev_extn, "-qrd", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "sdm845-tavil-hdk-snd-card")) {
+        strlcpy(hw_info->type, " hdk", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "sdm845", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)tavil_qrd_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(tavil_qrd_variant_devices);
+        hw_info->is_stereo_spkr = false;
+        strlcpy(hw_info->dev_extn, "-hdk", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "sdm845-qvr-tavil-snd-card")) {
+        hw_info->is_stereo_spkr = false;
     } else {
         ALOGW("%s: Not a sdm845 device", __func__);
     }
 }
 
+static void  update_hardware_info_sdx(struct hardware_info *hw_info __unused, const char *snd_card_name __unused)
+{
+    ALOGW("%s: Not a sdx device", __func__);
+}
+
 static void  update_hardware_info_8974(struct hardware_info *hw_info, const char *snd_card_name)
 {
     if (!strcmp(snd_card_name, "msm8974-taiko-mtp-snd-card")) {
@@ -544,6 +558,9 @@
     } else if (strstr(snd_card_name, "sdm660") || strstr(snd_card_name, "sdm670")) {
         ALOGV("Bear - variant soundcard");
         update_hardware_info_bear(hw_info, snd_card_name);
+    } else if (strncmp(snd_card_name, "sdx", sizeof("sdx"))) {
+        ALOGV("SDX - variant soundcard");
+        update_hardware_info_sdx(hw_info, snd_card_name);
     } else {
         ALOGE("%s: Unsupported target %s:",__func__, snd_card_name);
         free(hw_info);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index b575063..f2625bc 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -77,7 +77,7 @@
 #endif
 
 #include <linux/msm_audio.h>
-#if defined (PLATFORM_MSM8998) || (PLATFORM_SDM845) || (PLATFORM_SDM670)
+#if defined (PLATFORM_MSM8998) || (PLATFORM_SDM845) || (PLATFORM_SDM670) || defined (PLATFORM_QCS605)
 #include <sound/devdep_params.h>
 #endif
 
@@ -136,6 +136,11 @@
 /* Query external audio device connection status */
 #define AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE "ext_audio_device"
 
+/* Query whether it is ok to select display-port as output
+ * device for voice usecase
+ */
+#define AUDIO_PARAMETER_KEY_DP_FOR_VOICE_USECASE "dp_for_voice"
+
 #define EVENT_EXTERNAL_SPK_1 "qc_ext_spk_1"
 #define EVENT_EXTERNAL_SPK_2 "qc_ext_spk_2"
 #define EVENT_EXTERNAL_MIC   "qc_ext_mic"
@@ -275,6 +280,7 @@
     bool is_dsd_supported;
     bool is_asrc_supported;
     struct listnode acdb_meta_key_list;
+    bool use_generic_handset;
 };
 
 static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -313,6 +319,8 @@
     [USECASE_AUDIO_RECORD_COMPRESS2] = {-1, -1},
     [USECASE_AUDIO_RECORD_COMPRESS3] = {-1, -1},
     [USECASE_AUDIO_RECORD_COMPRESS4] = {-1, -1},
+    [USECASE_AUDIO_RECORD_COMPRESS5] = {-1, -1},
+    [USECASE_AUDIO_RECORD_COMPRESS6] = {-1, -1},
     [USECASE_AUDIO_RECORD_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
                                           LOWLATENCY_PCM_DEVICE},
     [USECASE_AUDIO_RECORD_FM_VIRTUAL] = {MULTIMEDIA2_PCM_DEVICE,
@@ -524,6 +532,7 @@
     [SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = "unprocessed-three-mic",
     [SND_DEVICE_IN_UNPROCESSED_QUAD_MIC] = "unprocessed-quad-mic",
     [SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = "unprocessed-headset-mic",
+    [SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = "quad-mic",
 };
 
 // Platform specific backend bit width table
@@ -682,6 +691,7 @@
     [SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = 145,
     [SND_DEVICE_IN_UNPROCESSED_QUAD_MIC] = 146,
     [SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = 147,
+    [SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = 150,
 };
 
 struct name_to_index {
@@ -824,6 +834,7 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_THREE_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_QUAD_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_GENERIC_QMIC)},
 };
 
 static char * backend_tag_table[SND_DEVICE_MAX] = {0};
@@ -849,6 +860,8 @@
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS2)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS3)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS4)},
+    {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS5)},
+    {TO_NAME_INDEX(USECASE_AUDIO_RECORD_COMPRESS6)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_LOW_LATENCY)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_MMAP)},
     {TO_NAME_INDEX(USECASE_VOICE_CALL)},
@@ -996,6 +1009,8 @@
                   sizeof("sdm670-skuw-snd-card")) ||
          !strncmp(snd_card_name, "sdm660-snd-card-skush",
                   sizeof("sdm660-snd-card-skush")) ||
+         !strncmp(snd_card_name, "sdm670-360cam-snd-card",
+                  sizeof("sdm670-360cam-snd-card")) ||
          !strncmp(snd_card_name, "sdm660-snd-card-mtp",
                   sizeof("sdm660-snd-card-mtp")) ||
          !strncmp(snd_card_name, "sdm670-mtp-snd-card",
@@ -1127,6 +1142,9 @@
         else if (adev->snd_dev_ref_cnt[SND_DEVICE_OUT_SPEAKER_VBAT] > 0)
             strlcpy(my_data->ec_ref_mixer_path, "echo-reference speaker-vbat",
                     sizeof(my_data->ec_ref_mixer_path));
+        else if (adev->snd_dev_ref_cnt[SND_DEVICE_OUT_DISPLAY_PORT] > 0)
+            strlcpy(my_data->ec_ref_mixer_path, "echo-reference display-port",
+                    sizeof(my_data->ec_ref_mixer_path));
         else
             strlcpy(my_data->ec_ref_mixer_path, "echo-reference",
                     sizeof(my_data->ec_ref_mixer_path));
@@ -1299,7 +1317,9 @@
         !strncmp(snd_card_name, "apq8084-taiko-i2s-cdp-snd-card",
                  sizeof("apq8084-taiko-i2s-cdp-snd-card")) ||
         !strncmp(snd_card_name, "apq8096-tasha-i2c-snd-card",
-                 sizeof("apq8096-tasha-i2c-snd-card"))) {
+                 sizeof("apq8096-tasha-i2c-snd-card")) ||
+        !strncmp(snd_card_name, "sdx-tavil-i2s-snd-card",
+                 sizeof("sdx-tavil-i2s-snd-card"))){
         plat_data->is_i2s_ext_modem = true;
     }
     ALOGV("%s, is_i2s_ext_modem:%d soundcard name is %s",__func__,
@@ -1494,6 +1514,7 @@
     hw_interface_table[SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = strdup("SLIMBUS_0_TX");
     hw_interface_table[SND_DEVICE_IN_UNPROCESSED_QUAD_MIC] = strdup("SLIMBUS_0_TX");
     hw_interface_table[SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = strdup("SLIMBUS_0_TX");
+    hw_interface_table[SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = strdup("SLIMBUS_0_TX");
 
     my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
 
@@ -2004,6 +2025,9 @@
         return NULL;
     }
 
+    adev->dp_allowed_for_voice =
+        property_get_bool("vendor.audio.enable.dp.for.voice", false);
+
     my_data->adev = adev;
     my_data->fluence_in_spkr_mode = false;
     my_data->fluence_in_voice_call = false;
@@ -2234,7 +2258,8 @@
     property_get("ro.board.platform", platform, "");
     property_get("ro.baseband", baseband, "");
     if ((!strncmp("apq8084", platform, sizeof("apq8084")) ||
-        !strncmp("msm8996", platform, sizeof("msm8996"))) &&
+        !strncmp("msm8996", platform, sizeof("msm8996")) ||
+        !strncmp("sdx", platform, sizeof("sdx"))) &&
         !strncmp("mdm", baseband, (sizeof("mdm")-1))) {
          my_data->csd = open_csd_client(my_data->is_i2s_ext_modem);
     } else {
@@ -2439,6 +2464,9 @@
     my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].channels_mixer_ctl =
         strdup("USB_AUDIO_RX Channels");
 
+    if (property_get_bool("vendor.audio.apptype.multirec.enabled", false))
+        my_data->use_generic_handset = true;
+
     my_data->edid_info = NULL;
     free(snd_card_name);
     free(snd_card_name_t);
@@ -2566,6 +2594,18 @@
     if (suffix != NULL) {
         strlcat(mixer_path, " ", MIXER_PATH_MAX_LENGTH);
         strlcat(mixer_path, suffix, MIXER_PATH_MAX_LENGTH);
+
+        /* if we can use display-port for voice call and usb mic
+         * is connected, choose dp_rx, usb_tx audio route
+         */
+        if (usecase->type == VOICE_CALL) {
+            struct audio_device *adev = usecase->stream.out->dev;
+            if ((snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
+                adev->dp_allowed_for_voice &&
+                (usecase->in_snd_device == SND_DEVICE_IN_VOICE_USB_HEADSET_MIC)) {
+                strlcat(mixer_path, "-and-usb-headset-mic", MIXER_PATH_MAX_LENGTH);
+            }
+        }
     }
 }
 
@@ -3757,6 +3797,17 @@
         } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
                    devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
             snd_device = SND_DEVICE_OUT_USB_HEADSET;
+        } else if ((devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) &&
+                   adev->dp_allowed_for_voice) {
+            switch(my_data->ext_disp_type) {
+                case EXT_DISPLAY_TYPE_DP:
+                    snd_device = SND_DEVICE_OUT_DISPLAY_PORT;
+                    break;
+                default:
+                    ALOGE("%s: Invalid disp_type %d", __func__,
+                           my_data->ext_disp_type);
+                    goto exit;
+            }
         } else if (devices & AUDIO_DEVICE_OUT_FM_TX) {
             snd_device = SND_DEVICE_OUT_TRANSMISSION_FM;
         } else if (devices & AUDIO_DEVICE_OUT_EARPIECE) {
@@ -4111,6 +4162,15 @@
                 else
                     snd_device = SND_DEVICE_IN_BT_SCO_MIC;
             }
+        } else if ((out_device & AUDIO_DEVICE_OUT_AUX_DIGITAL) &&
+                   adev->dp_allowed_for_voice) {
+            if (audio_extn_usb_is_capture_supported())
+                snd_device = SND_DEVICE_IN_VOICE_USB_HEADSET_MIC;
+            else
+                snd_device = SND_DEVICE_IN_HANDSET_MIC;
+
+            if (voice_is_in_call(adev))
+                platform_set_echo_reference(adev, true, out_device);
         } else if (out_device & AUDIO_DEVICE_OUT_SPEAKER) {
             if (my_data->fluence_type != FLUENCE_NONE &&
                 (my_data->fluence_in_voice_call ||
@@ -4141,6 +4201,17 @@
             snd_device = SND_DEVICE_IN_VOICE_USB_HEADSET_MIC;
           }
         }
+    } else if (my_data->use_generic_handset == true &&  //     system prop is enabled
+               (my_data->source_mic_type & SOURCE_QUAD_MIC) &&  // AND 4mic is available
+               ((in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) ||    // AND device is buit-in mic or back mic
+                (in_device & AUDIO_DEVICE_IN_BACK_MIC)) &&
+               (my_data->fluence_in_audio_rec == true &&       //  AND fluencepro is enabled
+                my_data->fluence_type & FLUENCE_QUAD_MIC) &&
+               (source == AUDIO_SOURCE_CAMCORDER ||           // AND source is cam/mic/unprocessed
+                source == AUDIO_SOURCE_UNPROCESSED ||
+                source == AUDIO_SOURCE_MIC)) {
+                snd_device = SND_DEVICE_IN_HANDSET_GENERIC_QMIC;
+                platform_set_echo_reference(adev, true, out_device);
     } else if (source == AUDIO_SOURCE_CAMCORDER) {
         if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC ||
             in_device & AUDIO_DEVICE_IN_BACK_MIC) {
@@ -4594,6 +4665,7 @@
         goto done_key_audcal;
     }
 
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
     /* parse audio calibration keys */
     ret = parse_audiocal_cfg(parms, &cal);
 
@@ -4970,6 +5042,8 @@
         ret=-EINVAL;
         goto done;
     }
+
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
     /* parse audiocal configuration keys */
     ret = parse_audiocal_cfg(query, &cal);
     if(ret == 0) {
@@ -5053,6 +5127,7 @@
                             struct str_parms *reply)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
     char value[512] = {0};
     int ret;
     char *kv_pairs = NULL;
@@ -5085,6 +5160,19 @@
         str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VOLUME_BOOST, value);
     }
 
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_DP_FOR_VOICE_USECASE,
+                            value, sizeof(value));
+
+    if (ret >= 0) {
+        if (my_data->ext_disp_type == EXT_DISPLAY_TYPE_DP &&
+            adev->dp_allowed_for_voice)
+            strlcpy(value, "true", sizeof(value));
+        else
+            strlcpy(value, "false", sizeof(value));
+
+        str_parms_add_str(reply, AUDIO_PARAMETER_KEY_DP_FOR_VOICE_USECASE, value);
+    }
+
     /* Handle audio calibration keys */
     get_audiocal(platform, query, reply);
     native_audio_get_params(query, reply, value, sizeof(value));
@@ -7641,7 +7729,7 @@
     return sample_rate;
 }
 
-#if defined (PLATFORM_MSM8998) || (PLATFORM_SDM845) || (PLATFORM_SDM670)
+#if defined (PLATFORM_MSM8998) || (PLATFORM_SDM845) || (PLATFORM_SDM670) || defined (PLATFORM_QCS605)
 int platform_get_mmap_data_fd(void *platform, int fe_dev, int dir, int *fd,
                               uint32_t *size)
 {
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 4fa9442..db86cdc 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -228,6 +228,7 @@
     SND_DEVICE_IN_HANDSET_8MIC,
     SND_DEVICE_IN_EC_REF_LOOPBACK_MONO,
     SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO,
+    SND_DEVICE_IN_HANDSET_GENERIC_QMIC,
     SND_DEVICE_IN_END,
 
     SND_DEVICE_MAX = SND_DEVICE_IN_END,
@@ -362,11 +363,11 @@
 #define PLAYBACK_OFFLOAD_DEVICE 9
 
 // Direct_PCM
-#if defined (PLATFORM_MSM8994) || defined (PLATFORM_MSM8996) || defined (PLATFORM_APQ8084) || defined (PLATFORM_MSM8998) || defined (PLATFORM_SDM845) || defined (PLATFORM_SDM670)
+#if defined (PLATFORM_MSM8994) || defined (PLATFORM_MSM8996) || defined (PLATFORM_APQ8084) || defined (PLATFORM_MSM8998) || defined (PLATFORM_SDM845) || defined (PLATFORM_SDM670) ||defined (PLATFORM_QCS605)
 #define PLAYBACK_OFFLOAD_DEVICE2 17
 #endif
 
-#if defined (PLATFORM_APQ8084) || defined (PLATFORM_MSM8996) || defined (PLATFORM_MSM8998) || defined (PLATFORM_SDM845) || defined (PLATFORM_SDM670)
+#if defined (PLATFORM_APQ8084) || defined (PLATFORM_MSM8996) || defined (PLATFORM_MSM8998) || defined (PLATFORM_SDM845) || defined (PLATFORM_SDM670) || defined(PLATFORM_QCS605)
 #define PLAYBACK_OFFLOAD_DEVICE3 18
 #define PLAYBACK_OFFLOAD_DEVICE4 34
 #define PLAYBACK_OFFLOAD_DEVICE5 35
@@ -491,7 +492,7 @@
 #define FM_RX_VOLUME "PRI MI2S LOOPBACK Volume"
 #elif PLATFORM_MSM8996
 #define FM_RX_VOLUME "Tert MI2S LOOPBACK Volume"
-#elif defined (PLATFORM_MSM8998) || defined (PLATFORM_SDM845) || defined (PLATFORM_MSMFALCON) || defined (PLATFORM_SDM670)
+#elif defined (PLATFORM_MSM8998) || defined (PLATFORM_SDM845) || defined (PLATFORM_MSMFALCON) || defined (PLATFORM_SDM670) || defined (PLATFORM_QCS605)
 #define FM_RX_VOLUME "SLIMBUS_8 LOOPBACK Volume"
 #else
 #define FM_RX_VOLUME "Internal FM RX Volume"
diff --git a/hal/voice.c b/hal/voice.c
index b3645be..f9e3562 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -216,8 +216,16 @@
 
     uc_info->id = usecase_id;
     uc_info->type = VOICE_CALL;
-    uc_info->stream.out = adev->current_call_output ;
-    uc_info->devices = adev->current_call_output ->devices;
+    uc_info->stream.out = adev->current_call_output;
+    uc_info->devices = adev->current_call_output->devices;
+
+    if (popcount(uc_info->devices) == 2) {
+        ALOGE("%s: Invalid combo device(%#x) for voice call", __func__,
+              uc_info->devices);
+        ret = -EIO;
+        goto error_start_voice;
+    }
+
     uc_info->in_snd_device = SND_DEVICE_NONE;
     uc_info->out_snd_device = SND_DEVICE_NONE;
 
diff --git a/hal/voice_extn/voice_extn.h b/hal/voice_extn/voice_extn.h
index 7f10d0c..f35344f 100644
--- a/hal/voice_extn/voice_extn.h
+++ b/hal/voice_extn/voice_extn.h
@@ -146,56 +146,56 @@
 static int __unused voice_extn_compress_voip_close_output_stream(
                                      struct audio_stream *stream __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
 static int __unused voice_extn_compress_voip_open_output_stream(
                                          struct stream_out *out __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
 static int __unused voice_extn_compress_voip_close_input_stream(
                                     struct audio_stream *stream __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
 static int __unused voice_extn_compress_voip_open_input_stream(
                                           struct stream_in *in __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
 static int __unused voice_extn_compress_voip_out_get_buffer_size(
                                        struct stream_out *stream __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
 static int __unused voice_extn_compress_voip_in_get_buffer_size(
                                            struct stream_in *in __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
 static int __unused voice_extn_compress_voip_start_input_stream(
                                            struct stream_in *in __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
 static int __unused voice_extn_compress_voip_start_output_stream(
                                           struct stream_out *out __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
@@ -203,7 +203,7 @@
                                 struct audio_device *adev __unused,
                                 bool state __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return 0;
 }
 
@@ -211,7 +211,7 @@
                               struct audio_device *adev __unused,
                               float volume __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return 0;
 }
 
@@ -220,7 +220,7 @@
                                   snd_device_t *out_snd_device __unused,
                                   snd_device_t *in_snd_device __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
@@ -228,7 +228,7 @@
                                   struct audio_device *adev __unused,
                                   struct str_parms *parms __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return -ENOSYS;
 }
 
@@ -236,7 +236,7 @@
                                      struct str_parms *query __unused,
                                      struct str_parms *reply __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
 }
 
 static void __unused voice_extn_compress_voip_out_get_parameters(
@@ -244,7 +244,7 @@
                                           struct str_parms *query __unused,
                                           struct str_parms *reply __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
 }
 
 static void __unused voice_extn_compress_voip_in_get_parameters(
@@ -252,40 +252,40 @@
                                            struct str_parms *query __unused,
                                            struct str_parms *reply __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
 }
 
 static bool __unused voice_extn_compress_voip_pcm_prop_check()
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return false;
 }
 
 static bool __unused  voice_extn_compress_voip_is_active(
                          const struct audio_device *adev __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return false;
 }
 
 static bool __unused voice_extn_compress_voip_is_format_supported(
                                             audio_format_t format __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return true;
 }
 
 static bool __unused voice_extn_compress_voip_is_config_supported(
                                       struct audio_config *config __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return true;
 }
 
 static bool __unused voice_extn_compress_voip_is_started(
                                struct audio_device *adev __unused)
 {
-    ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+    ALOGV("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
     return false;
 }
 
diff --git a/mm-audio/aenc-aac/qdsp6/Android.mk b/mm-audio/aenc-aac/qdsp6/Android.mk
index 2c169b6..b427233 100644
--- a/mm-audio/aenc-aac/qdsp6/Android.mk
+++ b/mm-audio/aenc-aac/qdsp6/Android.mk
@@ -39,7 +39,7 @@
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
diff --git a/mm-audio/aenc-amrnb/qdsp6/Android.mk b/mm-audio/aenc-amrnb/qdsp6/Android.mk
index e8e985b..3fa619e 100644
--- a/mm-audio/aenc-amrnb/qdsp6/Android.mk
+++ b/mm-audio/aenc-amrnb/qdsp6/Android.mk
@@ -39,7 +39,7 @@
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
diff --git a/mm-audio/aenc-evrc/qdsp6/Android.mk b/mm-audio/aenc-evrc/qdsp6/Android.mk
index 49384a4..03965cb 100644
--- a/mm-audio/aenc-evrc/qdsp6/Android.mk
+++ b/mm-audio/aenc-evrc/qdsp6/Android.mk
@@ -39,7 +39,7 @@
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
diff --git a/mm-audio/aenc-g711/qdsp6/Android.mk b/mm-audio/aenc-g711/qdsp6/Android.mk
index 97e105c..6b2b453 100644
--- a/mm-audio/aenc-g711/qdsp6/Android.mk
+++ b/mm-audio/aenc-g711/qdsp6/Android.mk
@@ -40,7 +40,7 @@
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
diff --git a/mm-audio/aenc-qcelp13/qdsp6/Android.mk b/mm-audio/aenc-qcelp13/qdsp6/Android.mk
index 9071eb3..f4b904a 100644
--- a/mm-audio/aenc-qcelp13/qdsp6/Android.mk
+++ b/mm-audio/aenc-qcelp13/qdsp6/Android.mk
@@ -39,7 +39,7 @@
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
 LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 314a73d..86a2ab1 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -52,7 +52,7 @@
 	$(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include \
         $(call include-path-for, audio-effects)
 
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
@@ -89,7 +89,7 @@
 
 ################################################################################
 
-ifneq ($(filter msm8992 msm8994 msm8996 msm8998 sdm660 sdm845 apq8098_latv sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msm8998 sdm660 sdm845 apq8098_latv sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
 
 include $(CLEAR_VARS)
 
@@ -121,7 +121,7 @@
         hardware/qcom/audio/hal/audio_extn \
         external/tinycompress/include
 
-ifneq ($(filter sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
   LOCAL_ADDITIONAL_DEPENDENCIES += $(BOARD_VENDOR_KERNEL_MODULES)
 endif
diff --git a/qahw/Android.mk b/qahw/Android.mk
new file mode 100644
index 0000000..c7df9e3
--- /dev/null
+++ b/qahw/Android.mk
@@ -0,0 +1,38 @@
+ifeq ($(strip $(BOARD_SUPPORTS_QAHW)),true)
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+libqahw-inc := $(LOCAL_PATH)/inc
+
+LOCAL_MODULE := libqahwwrapper
+LOCAL_MODULE_TAGS := optional
+LOCAL_C_INCLUDES   := $(libqahw-inc)
+
+LOCAL_SRC_FILES := \
+    src/qahw.c \
+    src/qahw_effect.c
+
+LOCAL_SHARED_LIBRARIES := \
+    liblog \
+    libcutils \
+    libhardware \
+    libdl
+
+LOCAL_CFLAGS += -Wall -Werror
+
+LOCAL_COPY_HEADERS_TO   := mm-audio/qahw/inc
+LOCAL_COPY_HEADERS      := inc/qahw.h
+LOCAL_COPY_HEADERS      += inc/qahw_effect_api.h
+
+LOCAL_PRELINK_MODULE    := false
+
+include $(BUILD_SHARED_LIBRARY)
+
+include $(CLEAR_VARS)
+LOCAL_COPY_HEADERS_TO   := mm-audio/qahw_api/inc
+LOCAL_COPY_HEADERS      := inc/qahw_defs.h
+
+include $(BUILD_COPY_HEADERS)
+endif
diff --git a/qahw/Makefile.am b/qahw/Makefile.am
new file mode 100644
index 0000000..2f33c9f
--- /dev/null
+++ b/qahw/Makefile.am
@@ -0,0 +1,21 @@
+AM_CFLAGS = -I $(top_srcdir)/inc
+
+h_sources = inc/qahw.h \
+            inc/qahw_effect_api.h
+
+qahw_include_HEADERS = $(h_sources)
+qahw_includedir = $(includedir)/mm-audio/qahw/inc
+
+qahw_api_include_HEADERS = inc/qahw_defs.h
+qahw_api_includedir = $(includedir)/mm-audio/qahw_api/inc
+
+lib_LTLIBRARIES = libqahwwrapper.la
+libqahwwrapper_la_SOURCES = src/qahw.c \
+                     src/qahw_effect.c
+
+libqahwwrapper_la_CFLAGS  = $(AM_CFLAGS)
+libqahwwrapper_la_CFLAGS += -include stddef.h
+libqahwwrapper_la_CFLAGS += -Dstrlcpy=g_strlcpy $(GLIB_CFLAGS) -include glib.h
+libqahwwrapper_la_CFLAGS += -D__unused=__attribute__\(\(__unused__\)\)
+libqahwwrapper_la_CFLAGS += -Werror -Wall
+libqahwwrapper_la_LDFLAGS = -shared -avoid-version -llog -lcutils -lhardware $(GLIB_LIBS)
diff --git a/qahw/configure.ac b/qahw/configure.ac
new file mode 100644
index 0000000..04fde19
--- /dev/null
+++ b/qahw/configure.ac
@@ -0,0 +1,52 @@
+#                                               -*- Autoconf -*-
+# configure.ac -- Autoconf script for halinterface
+#
+
+# Process this file with autoconf to produce a configure script.
+
+# Requires autoconf tool later than 2.61
+AC_PREREQ([2.69])
+# Initialize the audiohalwrapper-interface package version 1.0.0
+AC_INIT(audiohalwrapperinterface,1.0.0)
+# Does not strictly follow GNU Coding standards
+AM_INIT_AUTOMAKE([foreign])
+# Disables auto rebuilding of configure, Makefile.ins
+#AM_MAINTAINER_MODE
+# defines some macros variable to be included by source
+AC_CONFIG_HEADERS([config.h])
+# defines some macros variable to be included by source
+AC_CONFIG_MACRO_DIR([m4])
+
+# Checks for programs.
+AC_PROG_CC
+AM_PROG_CC_C_O
+AC_PROG_CXX
+AC_PROG_LIBTOOL
+AC_PROG_AWK
+AC_PROG_CPP
+AC_PROG_INSTALL
+AC_PROG_LN_S
+AC_PROG_MAKE_SET
+PKG_PROG_PKG_CONFIG
+
+AC_ARG_WITH([glib],
+      AC_HELP_STRING([--with-glib],
+         [enable glib, Build against glib. Use this when building for HLOS systems which use glib]))
+
+if (test "x${with_glib}" = "xyes"); then
+        PKG_CHECK_MODULES(GTHREAD, gthread-2.0 >= 2.16, dummy=yes,
+                                AC_MSG_ERROR(GThread >= 2.16 is required))
+        PKG_CHECK_MODULES(GLIB, glib-2.0 >= 2.16, dummy=yes,
+                                AC_MSG_ERROR(GLib >= 2.16 is required))
+        GLIB_CFLAGS="$GLIB_CFLAGS $GTHREAD_CFLAGS"
+        GLIB_LIBS="$GLIB_LIBS $GTHREAD_LIBS"
+
+        AC_SUBST(GLIB_CFLAGS)
+        AC_SUBST(GLIB_LIBS)
+fi
+
+AC_CONFIG_FILES([ \
+        Makefile
+        ])
+
+AC_OUTPUT
diff --git a/qahw/inc/qahw.h b/qahw/inc/qahw.h
new file mode 100644
index 0000000..b7088ed
--- /dev/null
+++ b/qahw/inc/qahw.h
@@ -0,0 +1,482 @@
+/*
+ * Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2011 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.
+ */
+
+#ifndef QTI_AUDIO_QAHW_H
+#define QTI_AUDIO_QAHW_H
+
+#include <stdint.h>
+#include <strings.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+#include <sys/time.h>
+#include <cutils/bitops.h>
+#include <system/audio.h>
+#include "qahw_defs.h"
+
+__BEGIN_DECLS
+/*
+ * Helper macros for module implementors.
+ *
+ * The derived modules should provide convenience macros for supported
+ * versions so that implementations can explicitly specify module
+ * versions at definition time.
+ */
+
+#define QAHW_MAKE_API_VERSION(maj,min) \
+            ((((maj) & 0xff) << 8) | ((min) & 0xff))
+
+/* First generation of audio devices had version hardcoded to 0. all devices with
+ * versions < 1.0 will be considered of first generation API.
+ */
+#define QAHW_MODULE_API_VERSION_0_0 QAHW_MAKE_API_VERSION(0, 0)
+
+/* Minimal QTI audio HAL version supported by the audio framework */
+#define QAHW_MODULE_API_VERSION_MIN QAHW_MODULE_API_VERSION_0_0
+
+/**
+ * List of known audio HAL modules. This is the base name of the audio HAL
+ * library composed of the "audio." prefix, one of the base names below and
+ * a suffix specific to the device.
+ * e.g: audio.primary.goldfish.so or audio.a2dp.default.so
+ */
+
+#define QAHW_MODULE_ID_PRIMARY     "audio.primary"
+#define QAHW_MODULE_ID_A2DP        "audio.a2dp"
+#define QAHW_MODULE_ID_USB         "audio.usb"
+
+typedef void qahw_module_handle_t;
+typedef void qahw_stream_handle_t;
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+/**************************************/
+/* Output stream specific APIs **/
+
+/*
+ * This method creates and opens the audio hardware output stream.
+ * The "address" parameter qualifies the "devices" audio device type if needed.
+ * The format format depends on the device type:
+ * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
+ * - USB devices use the ALSA card and device numbers in the form  "card=X;device=Y"
+ * - Other devices may use a number or any other string.
+ */
+
+int qahw_open_output_stream_l(qahw_module_handle_t *hw_module,
+                            audio_io_handle_t handle,
+                            audio_devices_t devices,
+                            audio_output_flags_t flags,
+                            struct audio_config *config,
+                            qahw_stream_handle_t **out_handle,
+                            const char *address);
+
+int qahw_close_output_stream_l(qahw_stream_handle_t *out_handle);
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_out_get_sample_rate_l(const qahw_stream_handle_t *stream);
+
+/*
+ *  use set_parameters with key QAHW_PARAMETER_STREAM_SAMPLING_RATE
+ */
+int qahw_out_set_sample_rate_l(qahw_stream_handle_t *stream, uint32_t rate);
+
+/*
+ * Return size of input/output buffer in bytes for this stream - eg. 4800.
+ * It should be a multiple of the frame size.  See also get_input_buffer_size.
+ */
+size_t qahw_out_get_buffer_size_l(const qahw_stream_handle_t *stream);
+
+/*
+ * Return the channel mask -
+ *  e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
+ */
+audio_channel_mask_t qahw_out_get_channels_l(const qahw_stream_handle_t *stream);
+
+/*
+ * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
+ */
+audio_format_t qahw_out_get_format_l(const qahw_stream_handle_t *stream);
+
+/*
+ * Put the audio hardware input/output into standby mode.
+ * Driver should exit from standby mode at the next I/O operation.
+ * Returns 0 on success and <0 on failure.
+ */
+int qahw_out_standby_l(qahw_stream_handle_t *stream);
+
+/*
+ * set/get audio stream parameters. The function accepts a list of
+ * parameter key value pairs in the form: key1=value1;key2=value2;...
+ *
+ * Some keys are reserved for standard parameters (See AudioParameter class)
+ *
+ * If the implementation does not accept a parameter change while
+ * the output is active but the parameter is acceptable otherwise, it must
+ * return -ENOSYS.
+ *
+ * The audio flinger will put the stream in standby and then change the
+ * parameter value.
+ */
+int qahw_out_set_parameters_l(qahw_stream_handle_t *stream, const char*kv_pairs);
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char* qahw_out_get_parameters_l(const qahw_stream_handle_t *stream,
+                               const char *keys);
+
+/* API to set playback stream specific config parameters */
+int qahw_out_set_param_data_l(qahw_stream_handle_t *out_handle,
+                            qahw_param_id param_id,
+                            qahw_param_payload *payload);
+
+/* API to get playback stream specific config parameters */
+int qahw_out_get_param_data_l(qahw_stream_handle_t *out_handle,
+                            qahw_param_id param_id,
+                            qahw_param_payload *payload);
+
+/*
+ * Return the audio hardware driver estimated latency in milliseconds.
+ */
+uint32_t qahw_out_get_latency_l(const qahw_stream_handle_t *stream);
+
+/*
+ * Use this method in situations where audio mixing is done in the
+ * hardware. This method serves as a direct interface with hardware,
+ * allowing you to directly set the volume as apposed to via the framework.
+ * This method might produce multiple PCM outputs or hardware accelerated
+ * codecs, such as MP3 or AAC.
+ */
+int qahw_out_set_volume_l(qahw_stream_handle_t *stream, float left, float right);
+
+/*
+ * Write audio buffer present in meta_data starting from offset
+ * along with timestamp to driver. Returns number of bytes
+ * written or a negative status_t. If at least one frame was written successfully
+ * prior to the error, it is suggested that the driver return that successful
+ * (short) byte count and then return an error in the subsequent call.
+ * timestamp is only sent driver is session has been opened with timestamp flag
+ * otherwise its ignored.
+ *
+ * If set_callback() has previously been called to enable non-blocking mode
+ * the write() is not allowed to block. It must write only the number of
+ * bytes that currently fit in the driver/hardware buffer and then return
+ * this byte count. If this is less than the requested write size the
+ * callback function must be called when more space is available in the
+ * driver/hardware buffer.
+ */
+ssize_t qahw_out_write_l(qahw_stream_handle_t *stream,
+                       qahw_out_buffer_t *out_buf);
+
+/*
+ * return the number of audio frames written by the audio dsp to DAC since
+ * the output has exited standby
+ */
+int qahw_out_get_render_position_l(const qahw_stream_handle_t *stream,
+                                 uint32_t *dsp_frames);
+
+/*
+ * set the callback function for notifying completion of non-blocking
+ * write and drain.
+ * Calling this function implies that all future rite() and drain()
+ * must be non-blocking and use the callback to signal completion.
+ */
+int qahw_out_set_callback_l(qahw_stream_handle_t *stream,
+                          qahw_stream_callback_t callback,
+                          void *cookie);
+
+/*
+ * Notifies to the audio driver to stop playback however the queued buffers are
+ * retained by the hardware. Useful for implementing pause/resume. Empty implementation
+ * if not supported however should be implemented for hardware with non-trivial
+ * latency. In the pause state audio hardware could still be using power. User may
+ * consider calling suspend after a timeout.
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_pause_l(qahw_stream_handle_t *out_handle);
+
+/*
+ * Notifies to the audio driver to resume playback following a pause.
+ * Returns error if called without matching pause.
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_resume_l(qahw_stream_handle_t *out_handle);
+
+/*
+ * Requests notification when data buffered by the driver/hardware has
+ * been played. If set_callback() has previously been called to enable
+ * non-blocking mode, the drain() must not block, instead it should return
+ * quickly and completion of the drain is notified through the callback.
+ * If set_callback() has not been called, the drain() must block until
+ * completion.
+ * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
+ * data has been played.
+ * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
+ * data for the current track has played to allow time for the framework
+ * to perform a gapless track switch.
+ *
+ * Drain must return immediately on stop() and flush() call
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_drain_l(qahw_stream_handle_t *out_handle, qahw_drain_type_t type);
+
+/*
+ * Notifies to the audio driver to flush the queued data. Stream must already
+ * be paused before calling flush().
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_flush_l(qahw_stream_handle_t *out_handle);
+
+/*
+ * Return a recent count of the number of audio frames presented to an external observer.
+ * This excludes frames which have been written but are still in the pipeline.
+ * The count is not reset to zero when output enters standby.
+ * Also returns the value of CLOCK_MONOTONIC as of this presentation count.
+ * The returned count is expected to be 'recent',
+ * but does not need to be the most recent possible value.
+ * However, the associated time should correspond to whatever count is returned.
+ * Example:  assume that N+M frames have been presented, where M is a 'small' number.
+ * Then it is permissible to return N instead of N+M,
+ * and the timestamp should correspond to N rather than N+M.
+ * The terms 'recent' and 'small' are not defined.
+ * They reflect the quality of the implementation.
+ *
+ * 3.0 and higher only.
+ */
+int qahw_out_get_presentation_position_l(const qahw_stream_handle_t *out_handle,
+                                       uint64_t *frames, struct timespec *timestamp);
+
+/* Input stream specific APIs */
+
+/* This method creates and opens the audio hardware input stream */
+int qahw_open_input_stream_l(qahw_module_handle_t *hw_module,
+                           audio_io_handle_t handle,
+                           audio_devices_t devices,
+                           struct audio_config *config,
+                           qahw_stream_handle_t **stream_in,
+                           audio_input_flags_t flags,
+                           const char *address,
+                           audio_source_t source);
+
+int qahw_close_input_stream_l(qahw_stream_handle_t *in_handle);
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_in_get_sample_rate_l(const qahw_stream_handle_t *in_handle);
+
+/*
+ * currently unused - use set_parameters with key
+ *    QAHW_PARAMETER_STREAM_SAMPLING_RATE
+ */
+int qahw_in_set_sample_rate_l(qahw_stream_handle_t *in_handle, uint32_t rate);
+
+/*
+ * Return size of input/output buffer in bytes for this stream - eg. 4800.
+ * It should be a multiple of the frame size.  See also get_input_buffer_size.
+ */
+size_t qahw_in_get_buffer_size_l(const qahw_stream_handle_t *in_handle);
+
+/*
+ * Return the channel mask -
+ *  e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
+ */
+audio_channel_mask_t qahw_in_get_channels_l(const qahw_stream_handle_t *in_handle);
+
+/*
+ * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
+ */
+audio_format_t qahw_in_get_format_l(const qahw_stream_handle_t *in_handle);
+
+/*
+ * currently unused - use set_parameters with key
+ *     QAHW_PARAMETER_STREAM_FORMAT
+ */
+int qahw_in_set_format_l(qahw_stream_handle_t *in_handle, audio_format_t format);
+
+/*
+ * Put the audio hardware input/output into standby mode.
+ * Driver should exit from standby mode at the next I/O operation.
+ * Returns 0 on success and <0 on failure.
+ */
+int qahw_in_standby_l(qahw_stream_handle_t *in_handle);
+
+/*
+ * set/get audio stream parameters. The function accepts a list of
+ * parameter key value pairs in the form: key1=value1;key2=value2;...
+ *
+ * Some keys are reserved for standard parameters (See AudioParameter class)
+ *
+ * If the implementation does not accept a parameter change while
+ * the output is active but the parameter is acceptable otherwise, it must
+ * return -ENOSYS.
+ *
+ * The audio flinger will put the stream in standby and then change the
+ * parameter value.
+ */
+int qahw_in_set_parameters_l(qahw_stream_handle_t *in_handle, const char *kv_pairs);
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char* qahw_in_get_parameters_l(const qahw_stream_handle_t *in_handle,
+                              const char *keys);
+/*
+ * Read audio buffer in from audio driver. Returns number of bytes read, or a
+ * negative status_t. meta_data structure is filled buffer pointer, start
+ * offset and valid catpure timestamp (if session is opened with timetamp flag)
+ * and buffer. if at least one frame was read prior to the error,
+ * read should return that byte count and then return an error in the
+ * subsequent call.
+ */
+ssize_t qahw_in_read_l(qahw_stream_handle_t *in_handle,
+                     qahw_in_buffer_t *in_buf);
+/*
+ * Return the amount of input frames lost in the audio driver since the
+ * last call of this function.
+ * Audio driver is expected to reset the value to 0 and restart counting
+ * upon returning the current value by this function call.
+ * Such loss typically occurs when the user space process is blocked
+ * longer than the capacity of audio driver buffers.
+ *
+ * Unit: the number of input audio frames
+ */
+uint32_t qahw_in_get_input_frames_lost_l(qahw_stream_handle_t *in_handle);
+
+/*
+ * Return a recent count of the number of audio frames received and
+ * the clock time associated with that frame count.
+ *
+ * frames is the total frame count received. This should be as early in
+ *     the capture pipeline as possible. In general,
+ *     frames should be non-negative and should not go "backwards".
+ *
+ * time is the clock MONOTONIC time when frames was measured. In general,
+ *     time should be a positive quantity and should not go "backwards".
+ *
+ * The status returned is 0 on success, -ENOSYS if the device is not
+ * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
+ */
+int qahw_in_get_capture_position_l(const qahw_stream_handle_t *in_handle,
+                                 int64_t *frames, int64_t *time);
+
+/* Module specific APIs */
+
+/* convenience API for opening and closing an audio HAL module */
+qahw_module_handle_t *qahw_load_module_l(const char *hw_module_id);
+
+int qahw_unload_module_l(qahw_module_handle_t *hw_module);
+
+/*
+ * check to see if the audio hardware interface has been initialized.
+ * returns 0 on success, -ENODEV on failure.
+ */
+int qahw_init_check_l(const qahw_module_handle_t *hw_module);
+
+/* set the audio volume of a voice call. Range is between 0.0 and 1.0 */
+int qahw_set_voice_volume_l(qahw_module_handle_t *hw_module, float volume);
+
+/*
+ * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
+ * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
+ * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
+ */
+int qahw_set_mode_l(qahw_module_handle_t *hw_module, audio_mode_t mode);
+
+/* Mute/unmute mic during voice/voip/HFP call */
+int qahw_set_mic_mute_l(qahw_module_handle_t *hw_module, bool state);
+
+/* Get mute/unmute status of mic during voice call */
+int qahw_get_mic_mute_l(qahw_module_handle_t *hw_module, bool *state);
+
+/* set/get global audio parameters */
+int qahw_set_parameters_l(qahw_module_handle_t *hw_module, const char *kv_pairs);
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char* qahw_get_parameters_l(const qahw_module_handle_t *hw_module,
+                           const char *keys);
+
+/* Returns audio input buffer size according to parameters passed or
+ * 0 if one of the parameters is not supported.
+ * See also get_buffer_size which is for a particular stream.
+ */
+size_t qahw_get_input_buffer_size_l(const qahw_module_handle_t *hw_module,
+                                  const struct audio_config *config);
+
+/*returns current QTI HAL version */
+int qahw_get_version_l();
+
+/* Api to implement get parameters based on keyword param_id
+ * and store data in payload.
+ */
+int qahw_get_param_data_l(const qahw_module_handle_t *hw_module,
+                        qahw_param_id param_id,
+                        qahw_param_payload *payload);
+
+/* Api to implement set parameters based on keyword param_id
+ * and data present in payload.
+ */
+int qahw_set_param_data_l(const qahw_module_handle_t *hw_module,
+                        qahw_param_id param_id,
+                        qahw_param_payload *payload);
+
+/* Creates an audio patch between several source and sink ports.
+ * The handle is allocated by the HAL and should be unique for this
+ * audio HAL module.
+ */
+int qahw_create_audio_patch_l(qahw_module_handle_t *hw_module,
+                        unsigned int num_sources,
+                        const struct audio_port_config *sources,
+                        unsigned int num_sinks,
+                        const struct audio_port_config *sinks,
+                        audio_patch_handle_t *handle);
+
+/* Release an audio patch */
+int qahw_release_audio_patch_l(qahw_module_handle_t *hw_module,
+                        audio_patch_handle_t handle);
+/* Fills the list of supported attributes for a given audio port.
+ * As input, "port" contains the information (type, role, address etc...)
+ * needed by the HAL to identify the port.
+ * As output, "port" contains possible attributes (sampling rates, formats,
+ * channel masks, gain controllers...) for this port.
+ */
+int qahw_get_audio_port_l(qahw_module_handle_t *hw_module,
+                      struct audio_port *port);
+
+/* Set audio port configuration */
+int qahw_set_audio_port_config_l(qahw_module_handle_t *hw_module,
+                     const struct audio_port_config *config);
+#ifdef __cplusplus
+}
+#endif
+
+__END_DECLS
+
+#endif  // QTI_AUDIO_QAHW_H
diff --git a/qahw/inc/qahw_defs.h b/qahw/inc/qahw_defs.h
new file mode 100644
index 0000000..c13a1a4
--- /dev/null
+++ b/qahw/inc/qahw_defs.h
@@ -0,0 +1,393 @@
+/*
+ * Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2011 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.
+ */
+
+#include <sys/cdefs.h>
+#include <stdint.h>
+
+#ifndef QTI_AUDIO_HAL_DEFS_H
+#define QTI_AUDIO_HAL_DEFS_H
+
+__BEGIN_DECLS
+
+/**************************************/
+
+/**
+ *  standard audio parameters that the HAL may need to handle
+ */
+
+/**
+ *  audio device parameters
+ */
+
+/* BT SCO Noise Reduction + Echo Cancellation parameters */
+#define QAHW_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
+#define QAHW_PARAMETER_VALUE_ON "on"
+#define QAHW_PARAMETER_VALUE_OFF "off"
+
+/* TTY mode selection */
+#define QAHW_PARAMETER_KEY_TTY_MODE "tty_mode"
+#define QAHW_PARAMETER_VALUE_TTY_OFF "tty_off"
+#define QAHW_PARAMETER_VALUE_TTY_VCO "tty_vco"
+#define QAHW_PARAMETER_VALUE_TTY_HCO "tty_hco"
+#define QAHW_PARAMETER_VALUE_TTY_FULL "tty_full"
+
+/* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off
+   Strings must be in sync with CallFeaturesSetting.java */
+#define QAHW_PARAMETER_KEY_HAC "HACSetting"
+#define QAHW_PARAMETER_VALUE_HAC_ON "ON"
+#define QAHW_PARAMETER_VALUE_HAC_OFF "OFF"
+
+/* A2DP sink address set by framework */
+#define QAHW_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
+
+/* A2DP source address set by framework */
+#define QAHW_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address"
+
+/* Screen state */
+#define QAHW_PARAMETER_KEY_SCREEN_STATE "screen_state"
+
+/* Bluetooth SCO wideband */
+#define QAHW_PARAMETER_KEY_BT_SCO_WB "bt_wbs"
+
+/* Get a new HW synchronization source identifier.
+ * Return a valid source (positive integer) or AUDIO_HW_SYNC_INVALID if an error occurs
+ * or no HW sync is available. */
+#define QAHW_PARAMETER_HW_AV_SYNC "hw_av_sync"
+
+/**
+ *  audio stream parameters
+ */
+
+#define QAHW_PARAMETER_STREAM_ROUTING "routing"             /* audio_devices_t */
+#define QAHW_PARAMETER_STREAM_FORMAT "format"               /* audio_format_t */
+#define QAHW_PARAMETER_STREAM_CHANNELS "channels"           /* audio_channel_mask_t */
+#define QAHW_PARAMETER_STREAM_FRAME_COUNT "frame_count"     /* size_t */
+#define QAHW_PARAMETER_STREAM_INPUT_SOURCE "input_source"   /* audio_source_t */
+#define QAHW_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" /* uint32_t */
+
+#define QAHW_PARAMETER_DEVICE_CONNECT "connect"            /* audio_devices_t */
+#define QAHW_PARAMETER_DEVICE_DISCONNECT "disconnect"      /* audio_devices_t */
+
+/* Query supported formats. The response is a '|' separated list of strings from
+ * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
+#define QAHW_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
+
+/* Query supported channel masks. The response is a '|' separated list of
+ * strings from audio_channel_mask_t enum
+ * e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
+#define QAHW_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
+
+/* Query supported sampling rates. The response is a '|' separated list of
+ * integer values e.g: "sup_sampling_rates=44100|48000" */
+#define QAHW_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
+
+/* Set the HW synchronization source for an output stream. */
+#define QAHW_PARAMETER_STREAM_HW_AV_SYNC "hw_av_sync"
+
+/* Enable mono audio playback if 1, else should be 0. */
+#define QAHW_PARAMETER_MONO_OUTPUT "mono_output"
+
+/**
+ * audio codec parameters
+ */
+
+#define QAHW_OFFLOAD_CODEC_PARAMS           "music_offload_codec_param"
+#define QAHW_OFFLOAD_CODEC_BIT_PER_SAMPLE   "music_offload_bit_per_sample"
+#define QAHW_OFFLOAD_CODEC_BIT_RATE         "music_offload_bit_rate"
+#define QAHW_OFFLOAD_CODEC_AVG_BIT_RATE     "music_offload_avg_bit_rate"
+#define QAHW_OFFLOAD_CODEC_ID               "music_offload_codec_id"
+#define QAHW_OFFLOAD_CODEC_BLOCK_ALIGN      "music_offload_block_align"
+#define QAHW_OFFLOAD_CODEC_SAMPLE_RATE      "music_offload_sample_rate"
+#define QAHW_OFFLOAD_CODEC_ENCODE_OPTION    "music_offload_encode_option"
+#define QAHW_OFFLOAD_CODEC_NUM_CHANNEL      "music_offload_num_channels"
+#define QAHW_OFFLOAD_CODEC_DOWN_SAMPLING    "music_offload_down_sampling"
+#define QAHW_OFFLOAD_CODEC_DELAY_SAMPLES    "delay_samples"
+#define QAHW_OFFLOAD_CODEC_PADDING_SAMPLES  "padding_samples"
+
+/**
+ * extended audio codec parameters
+ */
+
+#define QAHW_OFFLOAD_CODEC_WMA_FORMAT_TAG "music_offload_wma_format_tag"
+#define QAHW_OFFLOAD_CODEC_WMA_BLOCK_ALIGN "music_offload_wma_block_align"
+#define QAHW_OFFLOAD_CODEC_WMA_BIT_PER_SAMPLE "music_offload_wma_bit_per_sample"
+#define QAHW_OFFLOAD_CODEC_WMA_CHANNEL_MASK "music_offload_wma_channel_mask"
+#define QAHW_OFFLOAD_CODEC_WMA_ENCODE_OPTION "music_offload_wma_encode_option"
+#define QAHW_OFFLOAD_CODEC_WMA_ENCODE_OPTION1 "music_offload_wma_encode_option1"
+#define QAHW_OFFLOAD_CODEC_WMA_ENCODE_OPTION2 "music_offload_wma_encode_option2"
+
+#define QAHW_OFFLOAD_CODEC_FLAC_MIN_BLK_SIZE "music_offload_flac_min_blk_size"
+#define QAHW_OFFLOAD_CODEC_FLAC_MAX_BLK_SIZE "music_offload_flac_max_blk_size"
+#define QAHW_OFFLOAD_CODEC_FLAC_MIN_FRAME_SIZE "music_offload_flac_min_frame_size"
+#define QAHW_OFFLOAD_CODEC_FLAC_MAX_FRAME_SIZE "music_offload_flac_max_frame_size"
+
+#define QAHW_OFFLOAD_CODEC_ALAC_FRAME_LENGTH "music_offload_alac_frame_length"
+#define QAHW_OFFLOAD_CODEC_ALAC_COMPATIBLE_VERSION "music_offload_alac_compatible_version"
+#define QAHW_OFFLOAD_CODEC_ALAC_BIT_DEPTH "music_offload_alac_bit_depth"
+#define QAHW_OFFLOAD_CODEC_ALAC_PB "music_offload_alac_pb"
+#define QAHW_OFFLOAD_CODEC_ALAC_MB "music_offload_alac_mb"
+#define QAHW_OFFLOAD_CODEC_ALAC_KB "music_offload_alac_kb"
+#define QAHW_OFFLOAD_CODEC_ALAC_NUM_CHANNELS "music_offload_alac_num_channels"
+#define QAHW_OFFLOAD_CODEC_ALAC_MAX_RUN "music_offload_alac_max_run"
+#define QAHW_OFFLOAD_CODEC_ALAC_MAX_FRAME_BYTES "music_offload_alac_max_frame_bytes"
+#define QAHW_OFFLOAD_CODEC_ALAC_AVG_BIT_RATE "music_offload_alac_avg_bit_rate"
+#define QAHW_OFFLOAD_CODEC_ALAC_SAMPLING_RATE "music_offload_alac_sampling_rate"
+#define QAHW_OFFLOAD_CODEC_ALAC_CHANNEL_LAYOUT_TAG "music_offload_alac_channel_layout_tag"
+
+#define QAHW_OFFLOAD_CODEC_APE_COMPATIBLE_VERSION "music_offload_ape_compatible_version"
+#define QAHW_OFFLOAD_CODEC_APE_COMPRESSION_LEVEL "music_offload_ape_compression_level"
+#define QAHW_OFFLOAD_CODEC_APE_FORMAT_FLAGS "music_offload_ape_format_flags"
+#define QAHW_OFFLOAD_CODEC_APE_BLOCKS_PER_FRAME "music_offload_ape_blocks_per_frame"
+#define QAHW_OFFLOAD_CODEC_APE_FINAL_FRAME_BLOCKS "music_offload_ape_final_frame_blocks"
+#define QAHW_OFFLOAD_CODEC_APE_TOTAL_FRAMES "music_offload_ape_total_frames"
+#define QAHW_OFFLOAD_CODEC_APE_BITS_PER_SAMPLE "music_offload_ape_bits_per_sample"
+#define QAHW_OFFLOAD_CODEC_APE_NUM_CHANNELS "music_offload_ape_num_channels"
+#define QAHW_OFFLOAD_CODEC_APE_SAMPLE_RATE "music_offload_ape_sample_rate"
+#define QAHW_OFFLOAD_CODEC_APE_SEEK_TABLE_PRESENT "music_offload_seek_table_present"
+
+#define QAHW_OFFLOAD_CODEC_VORBIS_BITSTREAM_FMT "music_offload_vorbis_bitstream_fmt"
+
+/* Set or Query stream profile type */
+#define QAHW_PARAMETER_STREAM_PROFILE "audio_stream_profile"
+
+/* audio input flags for compress and timestamp mode.
+ * check other input flags defined in audio.h for conflicts
+ */
+#define QAHW_INPUT_FLAG_TIMESTAMP 0x80000000
+#define QAHW_INPUT_FLAG_COMPRESS  0x40000000
+
+/* Query fm volume */
+#define QAHW_PARAMETER_KEY_FM_VOLUME "fm_volume"
+
+/* Query if a2dp  is supported */
+#define QAHW_PARAMETER_KEY_HANDLE_A2DP_DEVICE "isA2dpDeviceSupported"
+
+#define MAX_OUT_CHANNELS 8
+#define MAX_INP_CHANNELS 8
+
+#define QAHW_PCM_CHANNEL_FL    1  /* Front left channel.                           */
+#define QAHW_PCM_CHANNEL_FR    2  /* Front right channel.                          */
+#define QAHW_PCM_CHANNEL_FC    3  /* Front center channel.                         */
+#define QAHW_PCM_CHANNEL_LS    4  /* Left surround channel.                        */
+#define QAHW_PCM_CHANNEL_RS    5  /* Right surround channel.                       */
+#define QAHW_PCM_CHANNEL_LFE   6  /* Low frequency effect channel.                 */
+#define QAHW_PCM_CHANNEL_CS    7  /* Center surround channel; Rear center channel. */
+#define QAHW_PCM_CHANNEL_LB    8  /* Left back channel; Rear left channel.         */
+#define QAHW_PCM_CHANNEL_RB    9  /* Right back channel; Rear right channel.       */
+#define QAHW_PCM_CHANNEL_TS   10  /* Top surround channel.                         */
+#define QAHW_PCM_CHANNEL_CVH  11  /* Center vertical height channel.               */
+#define QAHW_PCM_CHANNEL_MS   12  /* Mono surround channel.                        */
+#define QAHW_PCM_CHANNEL_FLC  13  /* Front left of center.                         */
+#define QAHW_PCM_CHANNEL_FRC  14  /* Front right of center.                        */
+#define QAHW_PCM_CHANNEL_RLC  15  /* Rear left of center.                          */
+#define QAHW_PCM_CHANNEL_RRC  16  /* Rear right of center.                         */
+
+/* type of asynchronous write callback events. Mutually exclusive */
+typedef enum {
+    QAHW_STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
+    QAHW_STREAM_CBK_EVENT_DRAIN_READY,  /* drain completed */
+    QAHW_STREAM_CBK_EVENT_ERROR,  /* stream hit some error */
+
+    QAHW_STREAM_CBK_EVENT_ADSP = 0x100    /* callback event from ADSP PP,
+                                           * corresponding payload will be
+                                           * sent as is to the client
+                                           */
+} qahw_stream_callback_event_t;
+
+typedef int qahw_stream_callback_t(qahw_stream_callback_event_t event,
+                                   void *param,
+                                   void *cookie);
+
+/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
+typedef enum {
+    QAHW_DRAIN_ALL,            /* drain() returns when all data has been played */
+    QAHW_DRAIN_EARLY_NOTIFY    /* drain() returns a short time before all data
+                                  from the current track has been played to
+                                  give time for gapless track switch */
+} qahw_drain_type_t;
+
+/* meta data flags */
+/*TBD: Extend this based on stb requirement*/
+typedef enum {
+ QAHW_META_DATA_FLAGS_NONE = 0,
+} qahw_meta_data_flags_t;
+
+typedef struct {
+    const void *buffer;    /* write buffer pointer */
+    size_t bytes;          /* size of buffer */
+    size_t offset;         /* offset in buffer from where valid byte starts */
+    int64_t *timestamp;    /* timestmap */
+    qahw_meta_data_flags_t flags; /* meta data flags */
+    uint32_t reserved[64]; /*reserved for future */
+} qahw_out_buffer_t;
+
+typedef struct {
+    void *buffer;          /* read buffer pointer */
+    size_t bytes;          /* size of buffer */
+    size_t offset;         /* offset in buffer from where valid byte starts */
+    int64_t *timestamp;    /* timestmap */
+    uint32_t reserved[64]; /*reserved for future */
+} qahw_in_buffer_t;
+
+#define MAX_SECTORS 8
+
+struct qahw_source_tracking_param {
+    uint8_t   vad[MAX_SECTORS];
+    uint16_t  doa_speech;
+    uint16_t  doa_noise[3];
+    uint8_t   polar_activity[360];
+};
+
+struct qahw_sound_focus_param {
+    uint16_t  start_angle[MAX_SECTORS];
+    uint8_t   enable[MAX_SECTORS];
+    uint16_t  gain_step;
+};
+
+struct aptx_dec_bt_addr {
+    uint32_t nap;
+    uint32_t uap;
+    uint32_t lap;
+};
+
+struct qahw_aptx_dec_param {
+   struct aptx_dec_bt_addr bt_addr;
+};
+
+struct qahw_avt_device_drift_param {
+   /* Flag to indicate if resync is required on the client side for
+    * drift correction. Flag is set to TRUE for the first get_param response
+    * after device interface starts. This flag value can be used by client
+    * to identify if device interface restart has happened and if any
+    * re-sync is required at their end for drift correction.
+    */
+    uint32_t        resync_flag;
+    /* Accumulated drift value in microseconds.
+     * Positive drift value indicates AV timer is running faster than device.
+     * Negative drift value indicates AV timer is running slower than device.
+     */
+    int32_t         avt_device_drift_value;
+    /* 64-bit absolute timestamp of reference */
+    uint64_t        ref_timer_abs_ts;
+};
+
+/*use these for setting infine window.i.e free run mode */
+#define QAHW_MAX_RENDER_START_WINDOW 0x8000000000000000
+#define QAHW_MAX_RENDER_END_WINDOW   0x7FFFFFFFFFFFFFFF
+
+struct qahw_out_render_window_param {
+   uint64_t        render_ws; /* render window start value microseconds*/
+   uint64_t        render_we; /* render window end value microseconds*/
+};
+
+struct qahw_out_start_delay_param {
+   uint64_t       start_delay; /* session start delay in microseconds*/
+};
+
+struct qahw_out_enable_drift_correction {
+   bool        enable; /* enable drift correction*/
+};
+
+struct qahw_out_correct_drift {
+    /*
+     * adjust time in microseconds, a positive value
+     * to advance the clock or a negative value to
+     * delay the clock.
+     */
+    int64_t        adjust_time;
+};
+
+#define QAHW_MAX_ADSP_STREAM_CMD_PAYLOAD_LEN 512
+
+typedef enum {
+    QAHW_STREAM_PP_EVENT = 0,
+    QAHW_STREAM_ENCDEC_EVENT = 1,
+} qahw_event_id;
+
+/* payload format for HAL parameter
+ * QAHW_PARAM_ADSP_STREAM_CMD
+ */
+struct qahw_adsp_event {
+    qahw_event_id event_type;      /* type of the event */
+    uint32_t payload_length;       /* length in bytes of the payload */
+    void *payload;                 /* the actual payload */
+};
+
+struct qahw_out_channel_map_param {
+   uint8_t       channels;                               /* Input Channels */
+   uint8_t       channel_map[AUDIO_CHANNEL_COUNT_MAX];   /* Input Channel Map */
+};
+
+struct qahw_device_cfg_param {
+   uint32_t   sample_rate;
+   uint32_t   channels;
+   uint32_t   bit_width;
+   audio_format_t format;
+   audio_devices_t device;
+   uint8_t    channel_map[AUDIO_CHANNEL_COUNT_MAX];
+   uint16_t   channel_allocation;
+};
+
+typedef struct qahw_mix_matrix_params {
+    uint16_t num_output_channels;
+    uint16_t num_input_channels;
+    uint8_t has_output_channel_map;
+    uint32_t output_channel_map[AUDIO_CHANNEL_COUNT_MAX];
+    uint8_t has_input_channel_map;
+    uint32_t input_channel_map[AUDIO_CHANNEL_COUNT_MAX];
+    uint8_t has_mixer_coeffs;
+    float mixer_coeffs[AUDIO_CHANNEL_COUNT_MAX][AUDIO_CHANNEL_COUNT_MAX];
+} qahw_mix_matrix_params_t;
+
+typedef union {
+    struct qahw_source_tracking_param st_params;
+    struct qahw_sound_focus_param sf_params;
+    struct qahw_aptx_dec_param aptx_params;
+    struct qahw_avt_device_drift_param drift_params;
+    struct qahw_out_render_window_param render_window_params;
+    struct qahw_out_start_delay_param start_delay;
+    struct qahw_out_enable_drift_correction drift_enable_param;
+    struct qahw_out_correct_drift drift_correction_param;
+    struct qahw_adsp_event adsp_event_params;
+    struct qahw_out_channel_map_param channel_map_params;
+    struct qahw_device_cfg_param device_cfg_params;
+    struct qahw_mix_matrix_params mix_matrix_params;
+} qahw_param_payload;
+
+typedef enum {
+    QAHW_PARAM_SOURCE_TRACK,
+    QAHW_PARAM_SOUND_FOCUS,
+    QAHW_PARAM_APTX_DEC,
+    QAHW_PARAM_AVT_DEVICE_DRIFT,  /* PARAM to query AV timer vs device drift */
+    QAHW_PARAM_OUT_RENDER_WINDOW, /* PARAM to set render window */
+    QAHW_PARAM_OUT_START_DELAY, /* PARAM to set session start delay*/
+    /* enable adsp drift correction this must be called before out_write */
+    QAHW_PARAM_OUT_ENABLE_DRIFT_CORRECTION,
+    /* param to set drift value to be adjusted by dsp */
+    QAHW_PARAM_OUT_CORRECT_DRIFT,
+    QAHW_PARAM_ADSP_STREAM_CMD,
+    QAHW_PARAM_OUT_CHANNEL_MAP,    /* PARAM to set i/p channel map */
+    QAHW_PARAM_DEVICE_CONFIG,      /* PARAM to set device config */
+    QAHW_PARAM_OUT_MIX_MATRIX_PARAMS,
+    QAHW_PARAM_CH_MIX_MATRIX_PARAMS,
+} qahw_param_id;
+
+__END_DECLS
+
+#endif  // QTI_AUDIO_HAL_DEFS_H
+
diff --git a/qahw_api/inc/qahw_effect_api.h b/qahw/inc/qahw_effect_api.h
similarity index 99%
rename from qahw_api/inc/qahw_effect_api.h
rename to qahw/inc/qahw_effect_api.h
index 5b380b1..de53cd3 100644
--- a/qahw_api/inc/qahw_effect_api.h
+++ b/qahw/inc/qahw_effect_api.h
@@ -31,7 +31,7 @@
 
 #include <system/audio.h>
 
-#include "qahw_api.h"
+#include "qahw.h"
 
 __BEGIN_DECLS
 
diff --git a/qahw_api/src/qahw.c b/qahw/src/qahw.c
similarity index 87%
rename from qahw_api/src/qahw.c
rename to qahw/src/qahw.c
index b5a85d7..91fc5bf 100644
--- a/qahw_api/src/qahw.c
+++ b/qahw/src/qahw.c
@@ -37,7 +37,7 @@
 #include <cutils/list.h>
 
 #include <hardware/audio.h>
-#include "qahw_api.h"
+#include "qahw.h"
 
 #define NO_ERROR 0
 #define MAX_MODULE_NAME_LENGTH  100
@@ -119,7 +119,7 @@
 /******************************************************************************/
 
 /* call this function without anylock held */
-static bool is_valid_qahw_stream(void *qahw_stream,
+static bool is_valid_qahw_stream_l(void *qahw_stream,
                                  qahw_stream_direction_t dir)
 {
 
@@ -174,7 +174,7 @@
 }
 
 /* call this fucntion with ahw_module_init_lock held*/
-static qahw_module_t* get_qahw_module_by_ptr(qahw_module_t *qahw_module)
+static qahw_module_t* get_qahw_module_by_ptr_l(qahw_module_t *qahw_module)
 {
     struct listnode *node = NULL;
     qahw_module_t *module = NULL, *module_temp = NULL;
@@ -194,7 +194,7 @@
 }
 
 /* call this function with qahw_module_init_lock held*/
-static qahw_module_t* get_qahw_module_by_name(const char *qahw_name)
+static qahw_module_t* get_qahw_module_by_name_l(const char *qahw_name)
 {
     struct listnode *node = NULL;
     qahw_module_t *module = NULL, *module_temp = NULL;
@@ -217,13 +217,13 @@
 /*
  * Return the sampling rate in Hz - eg. 44100.
  */
-uint32_t qahw_out_get_sample_rate(const qahw_stream_handle_t *out_handle)
+uint32_t qahw_out_get_sample_rate_l(const qahw_stream_handle_t *out_handle)
 {
     uint32_t rate = 0;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGV("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -244,13 +244,13 @@
  * currently unused - use set_parameters with key
  *    AUDIO_PARAMETER_STREAM_SAMPLING_RATE
  */
-int qahw_out_set_sample_rate(qahw_stream_handle_t *out_handle, uint32_t rate)
+int qahw_out_set_sample_rate_l(qahw_stream_handle_t *out_handle, uint32_t rate)
 {
     int32_t rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -267,13 +267,13 @@
     return rc;
 }
 
-size_t qahw_out_get_buffer_size(const qahw_stream_handle_t *out_handle)
+size_t qahw_out_get_buffer_size_l(const qahw_stream_handle_t *out_handle)
 {
     size_t buf_size = 0;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -290,13 +290,13 @@
     return buf_size;
 }
 
-audio_channel_mask_t qahw_out_get_channels(const qahw_stream_handle_t *out_handle)
+audio_channel_mask_t qahw_out_get_channels_l(const qahw_stream_handle_t *out_handle)
 {
     audio_channel_mask_t ch_mask = 0;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -313,13 +313,13 @@
     return ch_mask;
 }
 
-audio_format_t qahw_out_get_format(const qahw_stream_handle_t *out_handle)
+audio_format_t qahw_out_get_format_l(const qahw_stream_handle_t *out_handle)
 {
     audio_format_t format = AUDIO_FORMAT_INVALID;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -336,13 +336,13 @@
     return format;
 }
 
-int qahw_out_standby(qahw_stream_handle_t *out_handle)
+int qahw_out_standby_l(qahw_stream_handle_t *out_handle)
 {
     int32_t rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -361,13 +361,13 @@
     return rc;
 }
 
-int qahw_out_set_parameters(qahw_stream_handle_t *out_handle, const char *kv_pairs)
+int qahw_out_set_parameters_l(qahw_stream_handle_t *out_handle, const char *kv_pairs)
 {
     int rc = NO_ERROR;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         rc = -EINVAL;
         goto exit;
@@ -387,14 +387,14 @@
     return rc;
 }
 
-char *qahw_out_get_parameters(const qahw_stream_handle_t *out_handle,
+char *qahw_out_get_parameters_l(const qahw_stream_handle_t *out_handle,
                                const char *keys)
 {
     char *str_param = NULL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -413,7 +413,7 @@
 }
 
 /* API to get playback stream specific config parameters */
-int qahw_out_set_param_data(qahw_stream_handle_t *out_handle,
+int qahw_out_set_param_data_l(qahw_stream_handle_t *out_handle,
                             qahw_param_id param_id,
                             qahw_param_payload *payload)
 {
@@ -426,7 +426,7 @@
         goto exit;
     }
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -446,7 +446,7 @@
 }
 
 /* API to get playback stream specific config parameters */
-int qahw_out_get_param_data(qahw_stream_handle_t *out_handle,
+int qahw_out_get_param_data_l(qahw_stream_handle_t *out_handle,
                             qahw_param_id param_id,
                             qahw_param_payload *payload)
 {
@@ -454,7 +454,7 @@
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -473,13 +473,13 @@
     return rc;
 }
 
-uint32_t qahw_out_get_latency(const qahw_stream_handle_t *out_handle)
+uint32_t qahw_out_get_latency_l(const qahw_stream_handle_t *out_handle)
 {
     uint32_t latency = 0;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -497,13 +497,13 @@
     return latency;
 }
 
-int qahw_out_set_volume(qahw_stream_handle_t *out_handle, float left, float right)
+int qahw_out_set_volume_l(qahw_stream_handle_t *out_handle, float left, float right)
 {
     int rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -522,7 +522,7 @@
    return rc;
 }
 
-ssize_t qahw_out_write(qahw_stream_handle_t *out_handle,
+ssize_t qahw_out_write_l(qahw_stream_handle_t *out_handle,
         qahw_out_buffer_t *out_buf)
 {
     int rc = -EINVAL;
@@ -534,7 +534,7 @@
         goto exit;
     }
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -557,14 +557,14 @@
     return rc;
 }
 
-int qahw_out_get_render_position(const qahw_stream_handle_t *out_handle,
+int qahw_out_get_render_position_l(const qahw_stream_handle_t *out_handle,
                                  uint32_t *dsp_frames)
 {
     int rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -582,7 +582,7 @@
     return rc;
 }
 
-int qahw_out_set_callback(qahw_stream_handle_t *out_handle,
+int qahw_out_set_callback_l(qahw_stream_handle_t *out_handle,
                           qahw_stream_callback_t callback,
                           void *cookie)
 {
@@ -591,7 +591,7 @@
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -610,14 +610,14 @@
     return rc;
 }
 
-int qahw_out_pause(qahw_stream_handle_t *out_handle)
+int qahw_out_pause_l(qahw_stream_handle_t *out_handle)
 {
     /*TBD:load hal func pointer and call */
     int rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -636,14 +636,14 @@
     return rc;
 }
 
-int qahw_out_resume(qahw_stream_handle_t *out_handle)
+int qahw_out_resume_l(qahw_stream_handle_t *out_handle)
 {
     /*TBD:load hal func pointer and call */
     int rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -662,14 +662,14 @@
     return rc;
 }
 
-int qahw_out_drain(qahw_stream_handle_t *out_handle, qahw_drain_type_t type )
+int qahw_out_drain_l(qahw_stream_handle_t *out_handle, qahw_drain_type_t type )
 {
     /*TBD:load hal func pointer and call */
     int rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -688,13 +688,13 @@
     return rc;
 }
 
-int qahw_out_flush(qahw_stream_handle_t *out_handle)
+int qahw_out_flush_l(qahw_stream_handle_t *out_handle)
 {
     int rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -713,14 +713,14 @@
     return rc;
 }
 
-int qahw_out_get_presentation_position(const qahw_stream_handle_t *out_handle,
+int qahw_out_get_presentation_position_l(const qahw_stream_handle_t *out_handle,
                            uint64_t *frames, struct timespec *timestamp)
 {
     int rc = -EINVAL;
     qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
     audio_stream_out_t *out = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         goto exit;
     }
@@ -740,13 +740,13 @@
 }
 
 /* Input stream specific APIs */
-uint32_t qahw_in_get_sample_rate(const qahw_stream_handle_t *in_handle)
+uint32_t qahw_in_get_sample_rate_l(const qahw_stream_handle_t *in_handle)
 {
     uint32_t rate = 0;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -768,13 +768,13 @@
  * currently unused - use set_parameters with key
  *    AUDIO_PARAMETER_STREAM_SAMPLING_RATE
  */
-int qahw_in_set_sample_rate(qahw_stream_handle_t *in_handle, uint32_t rate)
+int qahw_in_set_sample_rate_l(qahw_stream_handle_t *in_handle, uint32_t rate)
 {
     int rc = -EINVAL;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -793,13 +793,13 @@
     return rc;
 }
 
-size_t qahw_in_get_buffer_size(const qahw_stream_handle_t *in_handle)
+size_t qahw_in_get_buffer_size_l(const qahw_stream_handle_t *in_handle)
 {
     size_t buf_size = 0;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -818,13 +818,13 @@
 }
 
 
-audio_channel_mask_t qahw_in_get_channels(const qahw_stream_handle_t *in_handle)
+audio_channel_mask_t qahw_in_get_channels_l(const qahw_stream_handle_t *in_handle)
 {
     audio_channel_mask_t ch_mask = 0;;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -842,13 +842,13 @@
     return ch_mask;
 }
 
-audio_format_t qahw_in_get_format(const qahw_stream_handle_t *in_handle)
+audio_format_t qahw_in_get_format_l(const qahw_stream_handle_t *in_handle)
 {
     audio_format_t format = AUDIO_FORMAT_INVALID;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -870,13 +870,13 @@
  * currently unused - use set_parameters with key
  *     AUDIO_PARAMETER_STREAM_FORMAT
  */
-int qahw_in_set_format(qahw_stream_handle_t *in_handle, audio_format_t format)
+int qahw_in_set_format_l(qahw_stream_handle_t *in_handle, audio_format_t format)
 {
     int rc = -EINVAL;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -895,13 +895,13 @@
     return rc;
 }
 
-int qahw_in_standby(qahw_stream_handle_t *in_handle)
+int qahw_in_standby_l(qahw_stream_handle_t *in_handle)
 {
     int rc = -EINVAL;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -933,13 +933,13 @@
  * The audio flinger will put the stream in standby and then change the
  * parameter value.
  */
-int qahw_in_set_parameters(qahw_stream_handle_t *in_handle, const char *kv_pairs)
+int qahw_in_set_parameters_l(qahw_stream_handle_t *in_handle, const char *kv_pairs)
 {
     int rc = -EINVAL;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -961,14 +961,14 @@
  * Returns a pointer to a heap allocated string. The caller is responsible
  * for freeing the memory for it using free().
  */
-char * qahw_in_get_parameters(const qahw_stream_handle_t *in_handle,
+char * qahw_in_get_parameters_l(const qahw_stream_handle_t *in_handle,
                               const char *keys)
 {
     char *str_param = NULL;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -991,7 +991,7 @@
  *  negative status_t. If at least one frame was read prior to the error,
  *  read should return that byte count and then return an error in the subsequent call.
  */
-ssize_t qahw_in_read(qahw_stream_handle_t *in_handle,
+ssize_t qahw_in_read_l(qahw_stream_handle_t *in_handle,
                      qahw_in_buffer_t *in_buf)
 {
     int rc = -EINVAL;
@@ -1003,7 +1003,7 @@
         goto exit;
     }
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -1037,13 +1037,13 @@
  *
  * Unit: the number of input audio frames
  */
-uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
+uint32_t qahw_in_get_input_frames_lost_l(qahw_stream_handle_t *in_handle)
 {
     uint32_t rc = 0;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     audio_stream_in_t *in = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         goto exit;
     }
@@ -1076,7 +1076,7 @@
  * The status returned is 0 on success, -ENOSYS if the device is not
  * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
  */
-int qahw_in_get_capture_position(const qahw_stream_handle_t *in_handle __unused,
+int qahw_in_get_capture_position_l(const qahw_stream_handle_t *in_handle __unused,
                                  int64_t *frames __unused, int64_t *time __unused)
 {
     /*TBD:: do we need this*/
@@ -1087,14 +1087,14 @@
  * check to see if the audio hardware interface has been initialized.
  * returns 0 on success, -ENODEV on failure.
  */
-int qahw_init_check(const qahw_module_handle_t *hw_module)
+int qahw_init_check_l(const qahw_module_handle_t *hw_module)
 {
     int rc = -EINVAL;
     qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1114,14 +1114,14 @@
     return rc;
 }
 /* set the audio volume of a voice call. Range is between 0.0 and 1.0 */
-int qahw_set_voice_volume(qahw_module_handle_t *hw_module, float volume)
+int qahw_set_voice_volume_l(qahw_module_handle_t *hw_module, float volume)
 {
     int rc = -EINVAL;
     qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1147,14 +1147,14 @@
  * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
  * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
  */
-int qahw_set_mode(qahw_module_handle_t *hw_module, audio_mode_t mode)
+int qahw_set_mode_l(qahw_module_handle_t *hw_module, audio_mode_t mode)
 {
     int rc = -EINVAL;
     qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1175,14 +1175,14 @@
     return rc;
 }
 
-int qahw_set_mic_mute(qahw_module_handle_t *hw_module, bool state)
+int qahw_set_mic_mute_l(qahw_module_handle_t *hw_module, bool state)
 {
     int rc = -EINVAL;
     qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1203,7 +1203,7 @@
     return rc;
 }
 
-int qahw_get_mic_mute(qahw_module_handle_t *hw_module, bool *state)
+int qahw_get_mic_mute_l(qahw_module_handle_t *hw_module, bool *state)
 {
     size_t rc = 0;
     qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
@@ -1211,7 +1211,7 @@
     audio_hw_device_t *audio_device;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1233,7 +1233,7 @@
 }
 
 /* set/get global audio parameters */
-int qahw_set_parameters(qahw_module_handle_t *hw_module, const char *kv_pairs)
+int qahw_set_parameters_l(qahw_module_handle_t *hw_module, const char *kv_pairs)
 {
     int rc = -EINVAL;
     qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
@@ -1241,7 +1241,7 @@
     audio_hw_device_t *audio_device;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1266,7 +1266,7 @@
  * Returns a pointer to a heap allocated string. The caller is responsible
  * for freeing the memory for it using free().
  */
-char * qahw_get_parameters(const qahw_module_handle_t *hw_module,
+char * qahw_get_parameters_l(const qahw_module_handle_t *hw_module,
                            const char *keys)
 {
     char *str_param = NULL;
@@ -1275,7 +1275,7 @@
     audio_hw_device_t *audio_device;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1298,7 +1298,7 @@
 /* Api to implement get parameters  based on keyword param_id
  * and store data in payload.
  */
-int qahw_get_param_data(const qahw_module_handle_t *hw_module,
+int qahw_get_param_data_l(const qahw_module_handle_t *hw_module,
                         qahw_param_id param_id,
                         qahw_param_payload *payload)
 {
@@ -1307,7 +1307,7 @@
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1332,7 +1332,7 @@
 /* Api to implement set parameters  based on keyword param_id
  * and data present in payload.
  */
-int qahw_set_param_data(const qahw_module_handle_t *hw_module,
+int qahw_set_param_data_l(const qahw_module_handle_t *hw_module,
                         qahw_param_id param_id,
                         qahw_param_payload *payload)
 {
@@ -1341,7 +1341,7 @@
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1367,7 +1367,7 @@
  * The handle is allocated by the HAL and should be unique for this
  * audio HAL module.
  */
-int qahw_create_audio_patch(qahw_module_handle_t *hw_module,
+int qahw_create_audio_patch_l(qahw_module_handle_t *hw_module,
                         unsigned int num_sources,
                         const struct audio_port_config *sources,
                         unsigned int num_sinks,
@@ -1379,7 +1379,7 @@
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1406,7 +1406,7 @@
 }
 
 /* Release an audio patch */
-int qahw_release_audio_patch(qahw_module_handle_t *hw_module,
+int qahw_release_audio_patch_l(qahw_module_handle_t *hw_module,
                         audio_patch_handle_t handle)
 {
     int ret = 0;
@@ -1414,7 +1414,7 @@
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1442,7 +1442,7 @@
  * As output, "port" contains possible attributes (sampling rates, formats,
  * channel masks, gain controllers...) for this port.
  */
-int qahw_get_audio_port(qahw_module_handle_t *hw_module,
+int qahw_get_audio_port_l(qahw_module_handle_t *hw_module,
                       struct audio_port *port)
 {
     int ret = 0;
@@ -1450,7 +1450,7 @@
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1473,7 +1473,7 @@
 }
 
 /* Set audio port configuration */
-int qahw_set_audio_port_config(qahw_module_handle_t *hw_module,
+int qahw_set_audio_port_config_l(qahw_module_handle_t *hw_module,
                      const struct audio_port_config *config)
 {
     int ret = 0;
@@ -1481,7 +1481,7 @@
     qahw_module_t *qahw_module_temp;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1507,7 +1507,7 @@
  * 0 if one of the parameters is not supported.
  * See also get_buffer_size which is for a particular stream.
  */
-size_t qahw_get_input_buffer_size(const qahw_module_handle_t *hw_module,
+size_t qahw_get_input_buffer_size_l(const qahw_module_handle_t *hw_module,
                                   const struct audio_config *config)
 {
     size_t rc = 0;
@@ -1516,7 +1516,7 @@
     audio_hw_device_t *audio_device;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1546,7 +1546,7 @@
  * - USB devices use the ALSA card and device numbers in the form  "card=X;device=Y"
  * - Other devices may use a number or any other string.
  */
-int qahw_open_output_stream(qahw_module_handle_t *hw_module,
+int qahw_open_output_stream_l(qahw_module_handle_t *hw_module,
                             audio_io_handle_t handle,
                             audio_devices_t devices,
                             audio_output_flags_t flags,
@@ -1561,7 +1561,7 @@
     qahw_stream_out_t *qahw_stream_out = NULL;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1634,7 +1634,7 @@
     return rc;
 }
 
-int qahw_close_output_stream(qahw_stream_handle_t *out_handle)
+int qahw_close_output_stream_l(qahw_stream_handle_t *out_handle)
 {
 
     int rc = 0;
@@ -1642,7 +1642,7 @@
     qahw_module_t *qahw_module = NULL;
     audio_hw_device_t *audio_device = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_out, STREAM_DIR_OUT)) {
         ALOGE("%s::Invalid out handle %p", __func__, out_handle);
         rc = -EINVAL;
         goto exit;
@@ -1669,7 +1669,7 @@
 }
 
 /* This method creates and opens the audio hardware input stream */
-int qahw_open_input_stream(qahw_module_handle_t *hw_module,
+int qahw_open_input_stream_l(qahw_module_handle_t *hw_module,
                            audio_io_handle_t handle,
                            audio_devices_t devices,
                            struct audio_config *config,
@@ -1685,7 +1685,7 @@
     qahw_stream_in_t *qahw_stream_in = NULL;
 
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     pthread_mutex_unlock(&qahw_module_init_lock);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
@@ -1738,14 +1738,14 @@
     return rc;
 }
 
-int qahw_close_input_stream(qahw_stream_handle_t *in_handle)
+int qahw_close_input_stream_l(qahw_stream_handle_t *in_handle)
 {
     int rc = 0;
     qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
     qahw_module_t *qahw_module = NULL;
     audio_hw_device_t *audio_device = NULL;
 
-    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+    if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
         ALOGV("%s::Invalid in handle %p", __func__, in_handle);
         rc = -EINVAL;
         goto exit;
@@ -1772,13 +1772,13 @@
 }
 
 /*returns current QTI HAL verison */
-int qahw_get_version() {
+int qahw_get_version_l() {
     return QAHW_MODULE_API_VERSION_CURRENT;
 }
 
 /* convenience API for opening and closing an audio HAL module */
 
-qahw_module_handle_t *qahw_load_module(const char *hw_module_id)
+qahw_module_handle_t *qahw_load_module_l(const char *hw_module_id)
 {
     int rc = -EINVAL;
     qahw_module_handle_t *qahw_mod_handle = NULL;
@@ -1806,7 +1806,7 @@
     /* return exiting module ptr if already loaded */
     pthread_mutex_lock(&qahw_module_init_lock);
     if (qahw_list_count > 0) {
-        qahw_module = get_qahw_module_by_name(hw_module_id);
+        qahw_module = get_qahw_module_by_name_l(hw_module_id);
         if(qahw_module != NULL) {
             qahw_mod_handle = (void *)qahw_module;
             pthread_mutex_lock(&qahw_module->lock);
@@ -1876,7 +1876,7 @@
     return qahw_mod_handle;
 }
 
-int qahw_unload_module(qahw_module_handle_t *hw_module)
+int qahw_unload_module_l(qahw_module_handle_t *hw_module)
 {
     int rc = -EINVAL;
     bool is_empty = false;
@@ -1887,7 +1887,7 @@
      * it is closed
     */
     pthread_mutex_lock(&qahw_module_init_lock);
-    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
     if (qahw_module_temp == NULL) {
         ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
         goto error_exit;
diff --git a/qahw_api/src/qahw_effect.c b/qahw/src/qahw_effect.c
similarity index 100%
rename from qahw_api/src/qahw_effect.c
rename to qahw/src/qahw_effect.c
diff --git a/qahw_api/Android.mk b/qahw_api/Android.mk
index ee06a30..ba402ba 100644
--- a/qahw_api/Android.mk
+++ b/qahw_api/Android.mk
@@ -4,28 +4,27 @@
 
 include $(CLEAR_VARS)
 
-libqahw-inc := $(LOCAL_PATH)/inc
+libqahwapi-inc := $(LOCAL_PATH)/inc
 
 LOCAL_MODULE := libqahw
 LOCAL_MODULE_TAGS := optional
-LOCAL_C_INCLUDES   := $(libqahw-inc)
+LOCAL_C_INCLUDES   := $(libqahwapi-inc)
+LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/qahw/inc
 
 LOCAL_SRC_FILES := \
-    src/qahw.c \
-    src/qahw_effect.c
+    src/qahw_api.cpp
 
 LOCAL_SHARED_LIBRARIES := \
     liblog \
     libcutils \
     libhardware \
-    libdl
+    libdl \
+    libqahwwrapper
 
 LOCAL_CFLAGS += -Wall -Werror
 
 LOCAL_COPY_HEADERS_TO   := mm-audio/qahw_api/inc
 LOCAL_COPY_HEADERS      := inc/qahw_api.h
-LOCAL_COPY_HEADERS      += inc/qahw_defs.h
-LOCAL_COPY_HEADERS      += inc/qahw_effect_api.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_audiosphere.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_bassboost.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_environmentalreverb.h
@@ -40,4 +39,5 @@
 
 #test app compilation
 include $(LOCAL_PATH)/test/Android.mk
+
 endif
diff --git a/qahw_api/Makefile.am b/qahw_api/Makefile.am
index 7c74d25..13fe417 100644
--- a/qahw_api/Makefile.am
+++ b/qahw_api/Makefile.am
@@ -1,8 +1,4 @@
-AM_CFLAGS = -I $(top_srcdir)/qahw_api/inc
-
 h_sources = inc/qahw_api.h \
-            inc/qahw_defs.h \
-            inc/qahw_effect_api.h \
             inc/qahw_effect_audiosphere.h \
             inc/qahw_effect_bassboost.h \
             inc/qahw_effect_environmentalreverb.h \
@@ -11,15 +7,22 @@
             inc/qahw_effect_virtualizer.h \
             inc/qahw_effect_visualizer.h
 
+AM_CPPFLAGS := -I $(top_srcdir)/qahw_api/inc/
+AM_CPPFLAGS += -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/mm-audio/audio-hal-server
+AM_CPPFLAGS += -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/mm-audio/qahw/inc
+AM_CPPFLAGS += -D__unused=__attribute__\(\(__unused__\)\)
+
 library_include_HEADERS = $(h_sources)
 library_includedir = $(includedir)/mm-audio/qahw_api/inc
 
-lib_LTLIBRARIES = libqahw.la
-libqahw_la_SOURCES = src/qahw.c \
-                     src/qahw_effect.c
-libqahw_la_CFLAGS = -Dstrlcpy=g_strlcpy $(GLIB_CFLAGS) -include glib.h $(AM_CFLAGS) -include stddef.h
-libqahw_la_CFLAGS += -D__unused=__attribute__\(\(__unused__\)\)
-libqahw_la_CFLAGS += -Wno-sign-compare -Werror
-libqahw_la_CFLAGS += -Dstrlcat=g_strlcat -Dstrlcpy=g_strlcpy -include glib.h $(GLIB_CFLAGS)
-libqahw_la_LIBADD = $(GLIB_LIBS) -llog -lcutils -lhardware -ldl
-libqahw_la_LDFLAGS = -module -shared -avoid-version
+lib_LTLIBRARIES     = libqahw.la
+libqahw_la_SOURCES   = src/qahw_api.cpp
+libqahw_la_LIBADD    = -lqahwwrapper
+libqahw_la_CPPFLAGS := $(AM_CPPFLAGS)
+libqahw_la_CPPFLAGS += -std=c++11 -DHAVE_PTHREADS -DHAVE_ANDROID_OS
+libqahw_la_CPPFLAGS += -DDEBUG_REFS_CALLSTACK_ENABLED=0
+libqahw_la_LDFLAGS   = -ltinyalsa -lhardware -lexpat -lcutils -llog -ldl -lbinder -shared -avoid-version -llog -lcutils -lpthread -lutils
+if QTI_AUDIO_SERVER_ENABLED
+AM_CPPFLAGS += -DQTI_AUDIO_SERVER_ENABLED
+libqahw_la_LDFLAGS +=  -lqtiaudioserver
+endif
diff --git a/qahw_api/inc/qahw_api.h b/qahw_api/inc/qahw_api.h
index 5e0e661..0aa3c79 100644
--- a/qahw_api/inc/qahw_api.h
+++ b/qahw_api/inc/qahw_api.h
@@ -16,8 +16,8 @@
  * limitations under the License.
  */
 
-#ifndef QTI_AUDIO_HAL_API_H
-#define QTI_AUDIO_HAL_API_H
+#ifndef QTI_AUDIO_QAHW_API_H
+#define QTI_AUDIO_QAHW_API_H
 
 #include <stdint.h>
 #include <strings.h>
@@ -61,6 +61,7 @@
 
 typedef void qahw_module_handle_t;
 typedef void qahw_stream_handle_t;
+typedef void (*audio_error_callback)(void* context);
 
 /**************************************/
 /* Output stream specific APIs **/
@@ -471,6 +472,8 @@
 int qahw_set_audio_port_config(qahw_module_handle_t *hw_module,
                      const struct audio_port_config *config);
 
+void qahw_register_qas_death_notify_cb(audio_error_callback cb, void* context);
+
 __END_DECLS
 
-#endif  // QTI_AUDIO_HAL_API_H
+#endif  // QTI_AUDIO_QAHW_API_H
diff --git a/qahw_api/src/qahw_api.cpp b/qahw_api/src/qahw_api.cpp
new file mode 100644
index 0000000..8967bd6
--- /dev/null
+++ b/qahw_api/src/qahw_api.cpp
@@ -0,0 +1,1626 @@
+/*
+* Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions are
+* met:
+*     * Redistributions of source code must retain the above copyright
+*       notice, this list of conditions and the following disclaimer.
+*     * Redistributions in binary form must reproduce the above
+*       copyright notice, this list of conditions and the following
+*       disclaimer in the documentation and/or other materials provided
+*       with the distribution.
+*     * Neither the name of The Linux Foundation nor the names of its
+*       contributors may be used to endorse or promote products derived
+*       from this software without specific prior written permission.
+*
+* THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+* ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+* BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+* OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#define LOG_TAG "qahw_api"
+#define LOG_NDEBUG 0
+#define LOG_NDDEBUG 0
+
+#include <utils/Errors.h>
+#include <dlfcn.h>
+#include <utils/Log.h>
+#include <stdlib.h>
+#include <cutils/list.h>
+#include <assert.h>
+
+#include <hardware/audio.h>
+#include <cutils/properties.h>
+#include "qahw_api.h"
+#include "qahw.h"
+
+#if QTI_AUDIO_SERVER_ENABLED
+#include <mm-audio/qti-audio-server/qti_audio_server.h>
+#include <mm-audio/qti-audio-server/qti_audio_server_client.h>
+
+using namespace audiohal;
+extern struct listnode stream_list;
+extern pthread_mutex_t list_lock;
+
+/* Flag to indicate if QAS is enabled or not */
+bool g_binder_enabled = false;
+/* QTI audio server handle */
+sp<Iqti_audio_server> g_qas = NULL;
+/* Handle for client context*/
+void* g_ctxt = NULL;
+/* Death notification handle */
+sp<death_notifier> g_death_notifier = NULL;
+/* Client callback handle */
+audio_error_callback g_audio_err_cb = NULL;
+/* Flag to indicate qas status */
+bool g_qas_died = false;
+/* Count how many times hal is loaded */
+static unsigned int g_qas_load_count;
+/* Store HAL handle */
+qahw_module_handle_t *g_qas_handle = NULL;
+
+void death_notifier::binderDied(const wp<IBinder>& who)
+{
+    struct listnode *node;
+    p_stream_handle *handle;
+
+    if (g_audio_err_cb) {
+        ALOGD("%s %d", __func__, __LINE__);
+        g_audio_err_cb(g_ctxt);
+    }
+    g_qas_died = true;
+
+    pthread_mutex_lock(&list_lock);
+    list_for_each(node, &stream_list) {
+        handle = node_to_item(node, p_stream_handle, list);
+         if (handle != NULL) {
+            sh_mem_data *shmem_data = handle->shmem_data;
+            ALOGD("%s: %d: signal to unblock any wait conditions", __func__, __LINE__);
+            pthread_cond_signal(&shmem_data->c_cond);
+            shmem_data->status = 0;
+        }
+    }
+    pthread_mutex_unlock(&list_lock);
+
+}
+
+void qahw_register_qas_death_notify_cb(audio_error_callback cb, void* context)
+{
+    ALOGD("%s %d", __func__, __LINE__);
+    g_audio_err_cb = cb;
+    g_ctxt = context;
+}
+
+death_notifier::death_notifier()
+{
+    ALOGV("%s %d", __func__, __LINE__);
+    sp<ProcessState> proc(ProcessState::self());
+    proc->startThreadPool();
+}
+
+sp<Iqti_audio_server> get_qti_audio_server() {
+    sp<IServiceManager> sm;
+    sp<IBinder> binder;
+    int retry_cnt = 5;
+
+    if (g_qas == 0) {
+        sm = defaultServiceManager();
+        if (sm != NULL) {
+            do {
+                binder = sm->getService(String16(QTI_AUDIO_SERVER));
+                if (binder != 0)
+                    break;
+                else
+                    ALOGE("%d:%s: get qas service failed",__LINE__, __func__);
+
+                 ALOGW("qti_audio_server not published, waiting...");
+                usleep(500000);
+            } while (--retry_cnt);
+        } else {
+            ALOGE("%d:%s: defaultServiceManager failed",__LINE__, __func__);
+        }
+        if (binder == NULL)
+            return NULL;
+
+        if (g_death_notifier == NULL) {
+            g_death_notifier = new death_notifier();
+            if (g_death_notifier == NULL) {
+                ALOGE("%d: %s() unable to allocate death notifier", __LINE__, __func__);
+                return NULL;
+            }
+        }
+        binder->linkToDeath(g_death_notifier);
+        g_qas = interface_cast<Iqti_audio_server>(binder);
+        assert(g_qas != 0);
+    }
+    return g_qas;
+}
+
+uint32_t qahw_out_get_sample_rate(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_get_sample_rate(out_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_get_sample_rate_l(out_handle);
+    }
+}
+
+int qahw_out_set_sample_rate(qahw_stream_handle_t *out_handle, uint32_t rate)
+{
+    ALOGV("%d:%s %d",__LINE__, __func__, rate);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_set_sample_rate(out_handle, rate);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_set_sample_rate_l(out_handle, rate);
+    }
+}
+
+size_t qahw_out_get_buffer_size(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_get_buffer_size(out_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_get_buffer_size_l(out_handle);
+    }
+}
+
+audio_channel_mask_t qahw_out_get_channels(const qahw_stream_handle_t
+                                              *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return (audio_channel_mask_t)(-ENODEV);
+            }
+            return qas->qahw_out_get_channels(out_handle);
+        } else {
+            return (audio_channel_mask_t)(-ENODEV);
+        }
+    } else {
+        return qahw_out_get_channels_l(out_handle);
+    }
+}
+
+audio_format_t qahw_out_get_format(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return AUDIO_FORMAT_INVALID;
+            }
+            return qas->qahw_out_get_format(out_handle);
+        } else {
+            return AUDIO_FORMAT_INVALID;;
+        }
+    } else {
+        return qahw_out_get_format_l(out_handle);
+    }
+}
+
+int qahw_out_standby(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_standby(out_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_standby_l(out_handle);
+    }
+}
+
+int qahw_out_set_parameters(qahw_stream_handle_t *out_handle,
+                                const char *kv_pairs)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_set_parameters(out_handle, kv_pairs);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_set_parameters_l(out_handle, kv_pairs);
+    }
+}
+
+char *qahw_out_get_parameters(const qahw_stream_handle_t *out_handle,
+                                 const char *keys)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return NULL;
+            }
+            return qas->qahw_out_get_parameters(out_handle, keys);
+        } else {
+            return NULL;
+        }
+    } else {
+        return qahw_out_get_parameters_l(out_handle, keys);
+    }
+}
+
+int qahw_out_set_param_data(qahw_stream_handle_t *out_handle,
+                            qahw_param_id param_id,
+                            qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_set_param_data(out_handle, param_id, payload);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_set_param_data_l(out_handle, param_id, payload);
+    }
+}
+
+int qahw_out_get_param_data(qahw_stream_handle_t *out_handle,
+                            qahw_param_id param_id,
+                            qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_get_param_data(out_handle, param_id, payload);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_get_param_data_l(out_handle, param_id, payload);
+    }
+}
+
+uint32_t qahw_out_get_latency(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_get_latency(out_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_get_latency_l(out_handle);
+    }
+}
+
+int qahw_out_set_volume(qahw_stream_handle_t *out_handle, float left, float right)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_set_volume(out_handle, left, right);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_set_volume_l(out_handle, left, right);
+    }
+}
+
+ssize_t qahw_out_write(qahw_stream_handle_t *out_handle,
+                        qahw_out_buffer_t *out_buf)
+{
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_write(out_handle, out_buf);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_write_l(out_handle, out_buf);
+    }
+}
+
+int qahw_out_get_render_position(const qahw_stream_handle_t *out_handle,
+                                 uint32_t *dsp_frames)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_get_render_position(out_handle, dsp_frames);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_get_render_position_l(out_handle, dsp_frames);
+    }
+}
+
+int qahw_out_set_callback(qahw_stream_handle_t *out_handle,
+                          qahw_stream_callback_t callback,
+                          void *cookie)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_set_callback(out_handle, callback, cookie);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_set_callback_l(out_handle, callback, cookie);
+    }
+}
+
+int qahw_out_pause(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_pause(out_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_pause_l(out_handle);
+    }
+}
+
+int qahw_out_resume(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_resume(out_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_resume_l(out_handle);
+    }
+}
+
+int qahw_out_drain(qahw_stream_handle_t *out_handle, qahw_drain_type_t type )
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+                ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+                return -ENODEV;
+            }
+            return qas->qahw_out_drain(out_handle, type);
+        } else {
+            return -EINVAL;
+        }
+    } else {
+        return qahw_out_drain_l(out_handle, type);
+    }
+}
+
+int qahw_out_flush(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_flush(out_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_flush_l(out_handle);
+    }
+}
+
+int qahw_out_get_presentation_position(const qahw_stream_handle_t *out_handle,
+                           uint64_t *frames, struct timespec *timestamp)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_out_get_presentation_position(out_handle,
+                                                 frames, timestamp);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_out_get_presentation_position_l(out_handle,
+                                         frames, timestamp);
+    }
+}
+
+uint32_t qahw_in_get_sample_rate(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_get_sample_rate(in_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_get_sample_rate_l(in_handle);
+    }
+}
+
+int qahw_in_set_sample_rate(qahw_stream_handle_t *in_handle, uint32_t rate)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_set_sample_rate(in_handle, rate);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_set_sample_rate_l(in_handle, rate);
+    }
+}
+
+size_t qahw_in_get_buffer_size(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_get_buffer_size(in_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_get_buffer_size_l(in_handle);
+    }
+}
+
+audio_channel_mask_t qahw_in_get_channels(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_get_channels(in_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_get_channels_l(in_handle);
+    }
+}
+
+audio_format_t qahw_in_get_format(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return (audio_format_t)-ENODEV;
+            }
+            return qas->qahw_in_get_format(in_handle);
+        } else {
+            return (audio_format_t)-ENODEV;
+        }
+    } else {
+        return qahw_in_get_format_l(in_handle);
+    }
+}
+
+int qahw_in_set_format(qahw_stream_handle_t *in_handle, audio_format_t format)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return (audio_format_t)-ENODEV;
+            }
+            return qas->qahw_in_set_format(in_handle, format);
+        } else {
+            return (audio_format_t)-ENODEV;
+        }
+    } else {
+        return qahw_in_set_format_l(in_handle, format);
+    }
+}
+
+int qahw_in_standby(qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_standby(in_handle);
+        } else {
+            return -EINVAL;
+        }
+    } else {
+        return qahw_in_standby_l(in_handle);
+    }
+}
+
+int qahw_in_set_parameters(qahw_stream_handle_t *in_handle, const char *kv_pairs)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_set_parameters(in_handle, kv_pairs);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_set_parameters_l(in_handle, kv_pairs);
+    }
+}
+
+char* qahw_in_get_parameters(const qahw_stream_handle_t *in_handle,
+                              const char *keys)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return NULL;
+            }
+            return qas->qahw_in_get_parameters(in_handle, keys);
+        } else {
+            return NULL;
+        }
+    } else {
+        return qahw_in_get_parameters_l(in_handle, keys);
+    }
+}
+
+ssize_t qahw_in_read(qahw_stream_handle_t *in_handle,
+                     qahw_in_buffer_t *in_buf)
+{
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_read(in_handle, in_buf);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_read_l(in_handle, in_buf);
+    }
+}
+
+uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_get_input_frames_lost(in_handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_get_input_frames_lost_l(in_handle);
+    }
+}
+
+int qahw_in_get_capture_position(const qahw_stream_handle_t *in_handle,
+                                 int64_t *frames, int64_t *time)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_in_get_capture_position(in_handle, frames, time);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_in_get_capture_position_l(in_handle, frames, time);
+    }
+}
+
+int qahw_init_check(const qahw_module_handle_t *hw_module)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_init_check(hw_module);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_init_check_l(hw_module);
+    }
+}
+
+int qahw_set_voice_volume(qahw_module_handle_t *hw_module, float volume)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_set_voice_volume(hw_module, volume);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_set_voice_volume_l(hw_module, volume);
+    }
+}
+
+int qahw_set_mode(qahw_module_handle_t *hw_module, audio_mode_t mode)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_set_mode(hw_module, mode);;
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_set_mode_l(hw_module, mode);
+    }
+}
+
+int qahw_set_mic_mute(qahw_module_handle_t *hw_module, bool state)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_set_mic_mute(hw_module, state);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_set_mic_mute_l(hw_module, state);
+    }
+}
+
+int qahw_get_mic_mute(qahw_module_handle_t *hw_module, bool *state)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_get_mic_mute(hw_module, state);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_get_mic_mute_l(hw_module, state);
+    }
+}
+
+int qahw_set_parameters(qahw_module_handle_t *hw_module, const char *kv_pairs)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_set_parameters(hw_module, kv_pairs);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_set_parameters_l(hw_module, kv_pairs);
+    }
+}
+
+char* qahw_get_parameters(const qahw_module_handle_t *hw_module,
+                           const char *keys)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return NULL;
+            }
+            return qas->qahw_get_parameters(hw_module, keys);;
+        } else {
+            return NULL;
+        }
+    } else {
+        return qahw_get_parameters_l(hw_module, keys);
+    }
+}
+
+int qahw_get_param_data(const qahw_module_handle_t *hw_module,
+                        qahw_param_id param_id,
+                        qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_get_param_data(hw_module, param_id, payload);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_get_param_data_l(hw_module, param_id, payload);
+    }
+}
+
+int qahw_set_param_data(const qahw_module_handle_t *hw_module,
+                        qahw_param_id param_id,
+                        qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_set_param_data(hw_module, param_id, payload);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_set_param_data_l(hw_module, param_id, payload);
+    }
+}
+
+int qahw_create_audio_patch(qahw_module_handle_t *hw_module,
+                        unsigned int num_sources,
+                        const struct audio_port_config *sources,
+                        unsigned int num_sinks,
+                        const struct audio_port_config *sinks,
+                        audio_patch_handle_t *handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_create_audio_patch(hw_module, num_sources,
+                                         sources, num_sinks, sinks,
+                                         handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_create_audio_patch_l(hw_module, num_sources,
+                                         sources, num_sinks, sinks,
+                                         handle);
+    }
+}
+
+int qahw_release_audio_patch(qahw_module_handle_t *hw_module,
+                        audio_patch_handle_t handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_release_audio_patch(hw_module, handle);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_release_audio_patch_l(hw_module, handle);
+    }
+}
+
+int qahw_get_audio_port(qahw_module_handle_t *hw_module,
+                      struct audio_port *port)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_get_audio_port(hw_module, port);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_get_audio_port_l(hw_module, port);
+    }
+}
+
+int qahw_set_audio_port_config(qahw_module_handle_t *hw_module,
+                     const struct audio_port_config *config)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_set_audio_port_config(hw_module, config);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_set_audio_port_config_l(hw_module, config);
+    }
+}
+
+size_t qahw_get_input_buffer_size(const qahw_module_handle_t *hw_module,
+                                  const struct audio_config *config)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_get_input_buffer_size(hw_module, config);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_get_input_buffer_size_l(hw_module, config);
+    }
+}
+
+int qahw_open_output_stream(qahw_module_handle_t *hw_module,
+                            audio_io_handle_t handle,
+                            audio_devices_t devices,
+                            audio_output_flags_t flags,
+                            struct audio_config *config,
+                            qahw_stream_handle_t **out_handle,
+                            const char *address)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_open_output_stream(hw_module, handle, devices,
+                                                 flags, config, out_handle,
+                                                 address);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_open_output_stream_l(hw_module, handle, devices,
+                                           flags, config, out_handle,
+                                           address);
+    }
+}
+
+int qahw_close_output_stream(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    int status;
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_close_output_stream(out_handle);
+        } else {
+            p_stream_handle *handle;
+            struct listnode *node, *tempnode;
+            pthread_mutex_lock(&list_lock);
+            list_for_each_safe(node, tempnode, &stream_list) {
+                handle = node_to_item(node, p_stream_handle, list);
+                p_stream_handle *p_stream = (p_stream_handle *)out_handle;
+                if (handle != NULL && handle == p_stream) {
+                    sh_mem_data *shmem_data = handle->shmem_data;
+                    ALOGD("%s %d: clear memory of handle %p &handle %p", __func__, __LINE__, handle, &handle);
+                    handle->sh_mem_dealer.clear();
+                    handle->sh_mem_handle.clear();
+                }
+                list_remove(node);
+                free(node_to_item(node, p_stream_handle, list));
+                ALOGD("%s %d: Freed node", __func__, __LINE__);
+            }
+            pthread_mutex_unlock(&list_lock);
+            return -ENODEV;
+        }
+    } else {
+        return qahw_close_output_stream_l(out_handle);
+    }
+}
+
+int qahw_open_input_stream(qahw_module_handle_t *hw_module,
+                           audio_io_handle_t handle,
+                           audio_devices_t devices,
+                           struct audio_config *config,
+                           qahw_stream_handle_t **in_handle,
+                           audio_input_flags_t flags,
+                           const char *address,
+                           audio_source_t source)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_open_input_stream(hw_module, handle, devices,
+                                           config, in_handle, flags,
+                                           address, source);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_open_input_stream_l(hw_module, handle, devices,
+                                       config, in_handle, flags,
+                                       address, source);
+    }
+}
+
+int qahw_close_input_stream(qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_close_input_stream(in_handle);
+        } else {
+            p_stream_handle *handle;
+            struct listnode *node, *tempnode;
+            pthread_mutex_lock(&list_lock);
+            list_for_each_safe(node, tempnode, &stream_list) {
+                ALOGD("%s %d", __func__, __LINE__);
+                handle = node_to_item(node, p_stream_handle, list);
+                p_stream_handle *p_stream = (p_stream_handle *)in_handle;
+                if (handle != NULL && handle == p_stream) {
+                    sh_mem_data *shmem_data = handle->shmem_data;
+                    ALOGD("%s %d: clear memory of handle %p &handle %p", __func__, __LINE__, handle, &handle);
+                    handle->sh_mem_dealer.clear();
+                    handle->sh_mem_handle.clear();
+                }
+                list_remove(node);
+                free(node_to_item(node, p_stream_handle, list));
+                ALOGD("%s %d: Freed node", __func__, __LINE__);
+            }
+            pthread_mutex_unlock(&list_lock);
+            return -EINVAL;
+        }
+    } else {
+        return qahw_close_input_stream_l(in_handle);
+    }
+}
+
+int qahw_get_version()
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_get_version();
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_get_version_l();
+    }
+}
+
+int qahw_unload_module(qahw_module_handle_t *hw_module)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    if (g_binder_enabled) {
+        if (!g_qas_died && (--g_qas_load_count == 0)) {
+            sp<Iqti_audio_server> qas = get_qti_audio_server();
+            if (qas == 0) {
+               ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+               return -ENODEV;
+            }
+            return qas->qahw_unload_module(hw_module);
+        } else {
+            return -ENODEV;
+        }
+    } else {
+        return qahw_unload_module_l(hw_module);
+    }
+}
+
+qahw_module_handle_t *qahw_load_module(const char *hw_module_id)
+{
+    char value[PROPERTY_VALUE_MAX];
+
+    ALOGV("%d:%s",__LINE__, __func__);
+    g_binder_enabled = property_get_bool("persist.vendor.audio.qas.enabled", false);
+    ALOGV("%d:%s: g_binder_enabled %d",__LINE__, __func__, g_binder_enabled);
+    if (g_binder_enabled) {
+        sp<Iqti_audio_server> qas = get_qti_audio_server();
+        if (qas == 0) {
+           ALOGE("%d:%s: invalid HAL handle",__LINE__, __func__);
+           return (void*)(-ENODEV);
+        }
+        g_qas_handle = qas->qahw_load_module(hw_module_id);
+        if (g_qas_handle == NULL) {
+            ALOGE("%s: HAL loading failed", __func__);
+        } else if (g_qas_load_count == 0) {
+            g_qas_load_count++;
+            g_qas_died = false;
+            pthread_mutex_init(&list_lock, (const pthread_mutexattr_t *) NULL);
+            list_init(&stream_list);
+            ALOGV("%s %d: stream_list %p", __func__, __LINE__, stream_list);
+        } else {
+            g_qas_load_count++;
+            ALOGD("%s: returning existing instance of hal", __func__);
+        }
+    } else {
+        g_qas_handle = qahw_load_module_l(hw_module_id);
+    }
+    return g_qas_handle;
+}
+#else
+void qahw_register_qas_death_notify_cb(audio_error_callback cb __unused, void* context __unused)
+{
+}
+
+uint32_t qahw_out_get_sample_rate(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_sample_rate_l(out_handle);
+}
+
+int qahw_out_set_sample_rate(qahw_stream_handle_t *out_handle, uint32_t rate)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_set_sample_rate_l(out_handle, rate);
+}
+
+size_t qahw_out_get_buffer_size(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_buffer_size_l(out_handle);
+}
+
+audio_channel_mask_t qahw_out_get_channels(const qahw_stream_handle_t
+                                              *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_channels_l(out_handle);
+}
+
+audio_format_t qahw_out_get_format(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_format_l(out_handle);
+}
+
+int qahw_out_standby(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_standby_l(out_handle);
+}
+
+int qahw_out_set_parameters(qahw_stream_handle_t *out_handle,
+                                const char *kv_pairs)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_set_parameters_l(out_handle, kv_pairs);
+}
+
+char *qahw_out_get_parameters(const qahw_stream_handle_t *out_handle,
+                                 const char *keys)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_parameters_l(out_handle, keys);
+}
+
+int qahw_out_set_param_data(qahw_stream_handle_t *out_handle,
+                            qahw_param_id param_id,
+                            qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_set_param_data_l(out_handle, param_id, payload);
+}
+
+int qahw_out_get_param_data(qahw_stream_handle_t *out_handle,
+                            qahw_param_id param_id,
+                            qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_param_data_l(out_handle, param_id, payload);
+}
+
+uint32_t qahw_out_get_latency(const qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_latency_l(out_handle);
+}
+
+int qahw_out_set_volume(qahw_stream_handle_t *out_handle, float left, float right)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_set_volume_l(out_handle, left, right);
+}
+
+ssize_t qahw_out_write(qahw_stream_handle_t *out_handle,
+                        qahw_out_buffer_t *out_buf)
+{
+    return qahw_out_write_l(out_handle, out_buf);
+}
+
+int qahw_out_get_render_position(const qahw_stream_handle_t *out_handle,
+                                 uint32_t *dsp_frames)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_render_position_l(out_handle, dsp_frames);
+}
+
+int qahw_out_set_callback(qahw_stream_handle_t *out_handle,
+                          qahw_stream_callback_t callback,
+                          void *cookie)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_set_callback_l(out_handle, callback, cookie);
+}
+
+int qahw_out_pause(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_pause_l(out_handle);
+}
+
+int qahw_out_resume(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_resume_l(out_handle);
+}
+
+int qahw_out_drain(qahw_stream_handle_t *out_handle, qahw_drain_type_t type )
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_drain_l(out_handle, type);
+}
+
+int qahw_out_flush(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_flush_l(out_handle);
+}
+
+int qahw_out_get_presentation_position(const qahw_stream_handle_t *out_handle,
+                           uint64_t *frames, struct timespec *timestamp)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_out_get_presentation_position_l(out_handle,
+                                     frames, timestamp);
+}
+
+uint32_t qahw_in_get_sample_rate(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_get_sample_rate_l(in_handle);
+}
+
+int qahw_in_set_sample_rate(qahw_stream_handle_t *in_handle, uint32_t rate)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_set_sample_rate_l(in_handle, rate);
+}
+
+size_t qahw_in_get_buffer_size(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_get_buffer_size_l(in_handle);
+}
+
+audio_channel_mask_t qahw_in_get_channels(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_get_channels_l(in_handle);
+}
+
+audio_format_t qahw_in_get_format(const qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_get_format_l(in_handle);
+}
+
+int qahw_in_set_format(qahw_stream_handle_t *in_handle, audio_format_t format)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_set_format_l(in_handle, format);
+}
+
+int qahw_in_standby(qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_standby_l(in_handle);
+}
+
+int qahw_in_set_parameters(qahw_stream_handle_t *in_handle, const char *kv_pairs)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_set_parameters_l(in_handle, kv_pairs);
+}
+
+char* qahw_in_get_parameters(const qahw_stream_handle_t *in_handle,
+                              const char *keys)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_get_parameters_l(in_handle, keys);
+}
+
+ssize_t qahw_in_read(qahw_stream_handle_t *in_handle,
+                     qahw_in_buffer_t *in_buf)
+{
+    return qahw_in_read_l(in_handle, in_buf);
+}
+
+uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_get_input_frames_lost_l(in_handle);
+}
+
+int qahw_in_get_capture_position(const qahw_stream_handle_t *in_handle,
+                                 int64_t *frames, int64_t *time)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_in_get_capture_position_l(in_handle, frames, time);
+}
+
+int qahw_init_check(const qahw_module_handle_t *hw_module)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_init_check_l(hw_module);
+}
+
+int qahw_set_voice_volume(qahw_module_handle_t *hw_module, float volume)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_set_voice_volume_l(hw_module, volume);
+}
+
+int qahw_set_mode(qahw_module_handle_t *hw_module, audio_mode_t mode)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_set_mode_l(hw_module, mode);
+}
+
+int qahw_set_mic_mute(qahw_module_handle_t *hw_module, bool state)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_set_mic_mute_l(hw_module, state);
+}
+
+int qahw_get_mic_mute(qahw_module_handle_t *hw_module, bool *state)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_get_mic_mute_l(hw_module, state);
+}
+
+int qahw_set_parameters(qahw_module_handle_t *hw_module, const char *kv_pairs)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_set_parameters_l(hw_module, kv_pairs);
+}
+
+char* qahw_get_parameters(const qahw_module_handle_t *hw_module,
+                           const char *keys)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_get_parameters_l(hw_module, keys);
+}
+
+int qahw_get_param_data(const qahw_module_handle_t *hw_module,
+                        qahw_param_id param_id,
+                        qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_get_param_data_l(hw_module, param_id, payload);
+}
+
+int qahw_set_param_data(const qahw_module_handle_t *hw_module,
+                        qahw_param_id param_id,
+                        qahw_param_payload *payload)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_set_param_data_l(hw_module, param_id, payload);
+}
+
+int qahw_create_audio_patch(qahw_module_handle_t *hw_module,
+                        unsigned int num_sources,
+                        const struct audio_port_config *sources,
+                        unsigned int num_sinks,
+                        const struct audio_port_config *sinks,
+                        audio_patch_handle_t *handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_create_audio_patch_l(hw_module, num_sources,
+                                     sources, num_sinks, sinks,
+                                     handle);
+}
+
+int qahw_release_audio_patch(qahw_module_handle_t *hw_module,
+                        audio_patch_handle_t handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_release_audio_patch_l(hw_module, handle);
+}
+
+int qahw_get_audio_port(qahw_module_handle_t *hw_module,
+                      struct audio_port *port)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_get_audio_port_l(hw_module, port);
+}
+
+int qahw_set_audio_port_config(qahw_module_handle_t *hw_module,
+                     const struct audio_port_config *config)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_set_audio_port_config_l(hw_module, config);
+}
+
+size_t qahw_get_input_buffer_size(const qahw_module_handle_t *hw_module,
+                                  const struct audio_config *config)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_get_input_buffer_size_l(hw_module, config);
+}
+
+int qahw_open_output_stream(qahw_module_handle_t *hw_module,
+                            audio_io_handle_t handle,
+                            audio_devices_t devices,
+                            audio_output_flags_t flags,
+                            struct audio_config *config,
+                            qahw_stream_handle_t **out_handle,
+                            const char *address)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_open_output_stream_l(hw_module, handle, devices,
+                                       flags, config, out_handle,
+                                       address);
+}
+
+int qahw_close_output_stream(qahw_stream_handle_t *out_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_close_output_stream_l(out_handle);
+}
+
+int qahw_open_input_stream(qahw_module_handle_t *hw_module,
+                           audio_io_handle_t handle,
+                           audio_devices_t devices,
+                           struct audio_config *config,
+                           qahw_stream_handle_t **in_handle,
+                           audio_input_flags_t flags,
+                           const char *address,
+                           audio_source_t source)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_open_input_stream_l(hw_module, handle, devices,
+                                   config, in_handle, flags,
+                                   address, source);
+}
+
+int qahw_close_input_stream(qahw_stream_handle_t *in_handle)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_close_input_stream_l(in_handle);
+}
+
+int qahw_get_version()
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_get_version_l();
+}
+
+int qahw_unload_module(qahw_module_handle_t *hw_module)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_unload_module_l(hw_module);
+}
+
+qahw_module_handle_t *qahw_load_module(const char *hw_module_id)
+{
+    ALOGV("%d:%s",__LINE__, __func__);
+    return qahw_load_module_l(hw_module_id);
+}
+#endif
diff --git a/qahw_api/test/Android.mk b/qahw_api/test/Android.mk
index 887f416..ec4e698 100644
--- a/qahw_api/test/Android.mk
+++ b/qahw_api/test/Android.mk
@@ -8,6 +8,7 @@
 LOCAL_MODULE := hal_play_test
 
 hal-play-inc     = $(TARGET_OUT_HEADERS)/mm-audio/qahw_api/inc
+hal-play-inc    += $(TARGET_OUT_HEADERS)/mm-audio/qahw/inc
 hal-play-inc    += external/tinyalsa/include
 
 LOCAL_CFLAGS += -Wall -Werror -Wno-sign-compare
@@ -15,6 +16,7 @@
 LOCAL_SHARED_LIBRARIES := \
     libaudioutils\
     libqahw \
+    libqahwwrapper \
     libutils \
     libcutils
 
diff --git a/qahw_api/test/Makefile.am b/qahw_api/test/Makefile.am
index b4bbea7..40cc24a 100644
--- a/qahw_api/test/Makefile.am
+++ b/qahw_api/test/Makefile.am
@@ -1,34 +1,35 @@
 bin_PROGRAMS = hal_play_test
 
-PLAY_CPPFLAGS = -Wno-sign-compare -Werror
-PLAY_CPPFLAGS += -D__unused=__attribute__\(\(__unused__\)\)
+PLAY_CFLAGS = -Wno-sign-compare -Werror
+PLAY_CFLAGS += -D__unused=__attribute__\(\(__unused__\)\)
 PLAY_INCLUDES = -I $(top_srcdir)/qahw_api/inc
+PLAY_INCLUDES += -I $(top_srcdir)/qahw/inc
 
 hal_play_test_SOURCES = qahw_playback_test.c \
                         qahw_effect_test.c
 
 hal_play_test_LDADD = -lutils ../libqahw.la
+AM_CFLAGS = -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/mm-audio/qahw/inc
 
 if QAP
-AM_CFLAGS = -DQAP
+AM_CFLAGS += -DQAP
 AM_CFLAGS += -I ${WORKSPACE}/audio/mm-audio/qap_wrapper/inc/
 hal_play_test_SOURCES += qap_wrapper_extn.c
 hal_play_test_LDADD += -lqap_wrapper
 endif
 
-hal_play_test_CPPFLAGS = $(PLAY_CPPFLAGS) $(PLAY_INCLUDES)
-hal_play_test_CFLAGS = $(AM_CFLAGS)
-hal_play_test_CFLAGS = -DLINUX_ENABLED
+hal_play_test_CFLAGS = $(PLAY_CFLAGS) $(PLAY_INCLUDES)
+hal_play_test_CFLAGS += $(AM_CFLAGS) -DLINUX_ENABLED
 
 bin_PROGRAMS += hal_rec_test
 
-REC_CPPFLAGS = $(PLAY_CPPFLAGS) -Dnullptr=NULL
+REC_CFLAGS = $(PLAY_CFLAGS) -Dnullptr=NULL
 REC_INCLUDES = $(PLAY_INCLUDES)
 
 hal_rec_test_SOURCES = qahw_multi_record_test.c
-hal_rec_test_CPPFLAGS = -Dstrlcat=g_strlcat $(GLIB_CFLAGS) -include glib.h
-hal_rec_test_CPPFLAGS += $(REC_CPPFLAGS) $(REC_INCLUDES)
-hal_play_test_CFLAGS += $(AM_CFLAGS)
+hal_rec_test_CFLAGS = -Dstrlcat=g_strlcat $(GLIB_CFLAGS) -include glib.h
+hal_rec_test_CFLAGS += $(REC_CFLAGS) $(REC_INCLUDES)
+hal_rec_test_CFLAGS += $(AM_CFLAGS)
 hal_rec_test_LDADD = -lutils ../libqahw.la $(GLIB_LIBS)
 
 bin_PROGRAMS += trans_loopback_test
@@ -37,6 +38,6 @@
 
 trans_loopback_test_SOURCES = trans_loopback_test.c
 
-trans_loopback_test_CFLAGS  = $(CFLAGS) -Wno-sign-compare -Werror
+trans_loopback_test_CFLAGS  = $(AM_CFLAGS) -Wno-sign-compare -Werror
 trans_loopback_test_CFLAGS  += $(trans_loopback_test_INCLUDES)
-trans_loopback_test_LDADD = -llog  -lutils ../libqahw.la -lcutils
+trans_loopback_test_LDADD = -llog  -lutils ../libqahw.la -lcutils -lm
diff --git a/qahw_api/test/qahw_multi_record_test.c b/qahw_api/test/qahw_multi_record_test.c
index 5b07b0f..d618101 100644
--- a/qahw_api/test/qahw_multi_record_test.c
+++ b/qahw_api/test/qahw_multi_record_test.c
@@ -90,6 +90,7 @@
 int sourcetrack_done = 0;
 static pthread_mutex_t sourcetrack_lock;
 struct qahw_sound_focus_param sound_focus_data;
+void *context = NULL;
 
 static bool request_wake_lock(bool wakelock_acquired, bool enable)
 {
@@ -130,7 +131,6 @@
    stop_record = true;
 }
 
-
 void read_soundfocus_param(void)
 {
     uint16_t start_angle[4] = {0};
@@ -249,7 +249,9 @@
 {
   int rc = 0, ret = 0, count = 0;
   FILE *fdLatencyNode = nullptr;
-  struct timespec tsColdI, tsColdF, tsCont;
+  struct timespec tsColdI = { 0, 0 };
+  struct timespec tsColdF = { 0, 0 };
+  struct timespec tsCont = { 0, 0 };
   uint64_t tCold, tCont, tsec, tusec;
   char latencyBuf[200] = {0};
   time_t start_time = time(0);
@@ -352,6 +354,7 @@
       fprintf(log_file, "File open failed \n");
       if (log_file != stdout)
           fprintf(stdout, "File open failed \n");
+      free(buffer);
       test_end();
       pthread_exit(0);
   }
@@ -442,6 +445,7 @@
   hdr.riff_sz = data_sz + 44 - 8;
   fseek(fd, 0, SEEK_SET);
   fwrite(&hdr, 1, sizeof(hdr), fd);
+  free(buffer);
   fclose(fd);
 
   /* capture latency kpis if required */
@@ -450,7 +454,8 @@
               tsColdF.tv_nsec/1000000 - tsColdI.tv_nsec/1000000;
 
       fread((void *) latencyBuf, 100, 1, fdLatencyNode);
-      fclose(fdLatencyNode);
+      if (fdLatencyNode)
+          fclose(fdLatencyNode);
       sscanf(latencyBuf, " %llu,%llu", &tsec, &tusec);
       tCont = ((uint64_t)tsCont.tv_sec)*1000 - tsec*1000 + ((uint64_t)tsCont.tv_nsec)/1000000 - tusec/1000;
       if (log_file != stdout) {
@@ -583,6 +588,12 @@
     printf("                                               For mono channel 16kHz rate for 30seconds\n\n");
 }
 
+static void qti_audio_server_death_notify_cb(void *ctxt) {
+    fprintf(log_file, "qas died\n");
+    fprintf(stderr, "qas died\n");
+    stop_record = true;
+}
+
 int main(int argc, char* argv[]) {
     int max_recordings_requested = 0, status = 0;
     int thread_active[MAX_RECORD_SESSIONS] = {0};
@@ -675,6 +686,8 @@
                 break;
          }
     }
+    fprintf(log_file, "registering qas callback");
+    qahw_register_qas_death_notify_cb((audio_error_callback)qti_audio_server_death_notify_cb, context);
 
     wakelock_acquired = request_wake_lock(wakelock_acquired, true);
     qahw_mod_handle = qahw_load_module(mod_name);
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
index fc55f83..fceff8b 100644
--- a/qahw_api/test/qahw_playback_test.c
+++ b/qahw_api/test/qahw_playback_test.c
@@ -54,6 +54,10 @@
 #define DTSHD_CHUNK_STREAM_KEYWORD "STRMDATA"
 #define DTSHD_META_KEYWORD_SIZE 8 /*in bytes */
 
+#ifndef AUDIO_OUTPUT_FLAG_MAIN
+#define AUDIO_OUTPUT_FLAG_MAIN 0x8000000
+#endif
+
 static ssize_t get_bytes_to_read(FILE* file, int filetype);
 static void init_streams(void);
 int pthread_cancel(pthread_t thread);
@@ -351,7 +355,7 @@
     struct proxy_data* params = (struct proxy_data*) data;
     qahw_module_handle_t *qahw_mod_handle = params->acp.qahw_mod_handle;
     qahw_in_buffer_t in_buf;
-    char *buffer;
+    char *buffer = NULL;
     int rc = 0;
     int bytes_to_read, bytes_written = 0, bytes_wrote = 0;
     FILE *fp = NULL;
@@ -378,6 +382,7 @@
         if ((fp = fopen(params->acp.file_name,"w"))== NULL) {
             fprintf(log_file, "Cannot open file to dump proxy data\n");
             fprintf(stderr, "Cannot open file to dump proxy data\n");
+            free(buffer);
             pthread_exit(0);
         }
         else {
@@ -420,6 +425,7 @@
             fprintf(stderr, "could not close input stream %d \n", rc);
         }
         fprintf(log_file, "pcm data saved to file %s", params->acp.file_name);
+        free(buffer);
     }
     return 0;
 }
@@ -463,7 +469,7 @@
     return NULL;
 }
 
-static int is_eof(stream_config *stream) {
+static int __unused is_eof (stream_config *stream) {
     if (stream->filename) {
         if (feof(stream->file_stream)) {
             fprintf(log_file, "stream %d: error in fread, error %d\n", stream->stream_index, ferror(stream->file_stream));
@@ -505,7 +511,7 @@
     ret = qahw_out_write(out_handle, &out_buf);
     if (ret < 0) {
         fprintf(log_file, "stream %d: writing data to hal failed (ret = %zd)\n", stream_params->stream_index, ret);
-    } else if (ret != bytes) {
+    } else if ((ret != bytes) && (!stop_playback)) {
         fprintf(log_file, "stream %d: provided bytes %zd, written bytes %d\n",stream_params->stream_index, bytes, ret);
         fprintf(log_file, "stream %d: waiting for event write ready\n", stream_params->stream_index);
         pthread_cond_wait(&stream_params->write_cond, &stream_params->write_lock);
@@ -516,7 +522,7 @@
     return ret;
 }
 
-static bool is_assoc_active()
+static bool __unused is_assoc_active()
 {
     int i = 0;
     bool is_assoc_active = false;
@@ -530,7 +536,7 @@
     return is_assoc_active;
 }
 
-static int get_assoc_index()
+static int __unused get_assoc_index()
 {
     int i = 0;
 
@@ -1132,6 +1138,7 @@
     if (data == NULL) {
         fprintf(log_file, "calloc failed!!\n");
         fprintf(stderr, "calloc failed!!\n");
+        fclose(fd_latency_node);
         return -ENOMEM;
     }
 
@@ -1144,7 +1151,8 @@
             if (ret) {
                 fprintf(log_file, "error(%d) fetching start time for cold latency", ret);
                 fprintf(stderr, "error(%d) fetching start time for cold latency", ret);
-                return -1;
+                rc = -1;
+                goto exit;
             }
         } else if (count == 16) {
             int *d = (int *)data;
@@ -1153,7 +1161,8 @@
             if (ret) {
                 fprintf(log_file, "error(%d) fetching start time for continuous latency", ret);
                 fprintf(stderr, "error(%d) fetching start time for continuous latency", ret);
-                return -1;
+                rc = -1;
+                goto exit;
             }
         }
 
@@ -1179,6 +1188,9 @@
     fprintf(log_file, "\n values from debug node %s\n", latency_buf);
     fprintf(log_file, " cold latency %llums, continuous latency %llums,\n", tcold, tcont);
     fprintf(log_file, " **Note: please add DSP Pipe/PP latency numbers to this, for final latency values\n");
+exit:
+    fclose(fd_latency_node);
+    free(data);
     return rc;
 }
 
@@ -1221,7 +1233,7 @@
 
 void parse_aptx_dec_bt_addr(char *value, struct qahw_aptx_dec_param *aptx_cfg)
 {
-    int ba[6];
+    int ba[6] = {0, 0, 0, 0, 0, 0};
     char *str, *tok;
     uint32_t addr[3];
     int i = 0;
@@ -1693,7 +1705,7 @@
                 ret = fread(&read_chunk_size, 1, DTSHD_META_KEYWORD_SIZE, file);
                 chunk_size = convert_BE_to_LE(read_chunk_size);
                 if (ret != DTSHD_META_KEYWORD_SIZE) {
-                    fprintf(stderr,"%s %d file read error ret %\n",
+                    fprintf(stderr,"%s %d file read error ret %d \n",
                             __func__, __LINE__, ret);
                     file_read_size = -EINVAL;
                     break;
@@ -1836,6 +1848,7 @@
     char *init_ptr = NULL;
     char *token = NULL;
     char *saveptr = NULL;
+    int rc = 0;
 
     if (NULL == channel_map)
         return -EINVAL;
@@ -1849,20 +1862,24 @@
         init_ptr = token_string;
         token = strtok_r(token_string, ",", &saveptr);
         int index = 0;
-        if (NULL == token)
-            return -EINVAL;
+        if (NULL == token) {
+            rc = -EINVAL;
+            goto exit;
+        }
         else
             channel_map[index++] = get_channel_mask_for_name(token);
 
         while(NULL !=(token = strtok_r(NULL,",",&saveptr)))
             channel_map[index++] = get_channel_mask_for_name(token);
 
-        free(init_ptr);
-        init_ptr = NULL;
-        token_string = NULL;
+        goto exit;
     } else
         return -EINVAL;
-    return 0;
+exit:
+    free(init_ptr);
+    init_ptr = NULL;
+    token_string = NULL;
+    return rc;
 }
 
 int extract_mixer_coeffs(qahw_mix_matrix_params_t * mm_params, const char * arg_string){
@@ -1872,7 +1889,7 @@
     char *token = NULL;
     char *saveptr = NULL;
     int i = 0;
-    int j = 0;
+    int j = 0, rc = 0;
 
     if (NULL == mm_params)
         return -EINVAL;
@@ -1885,9 +1902,10 @@
     if(token_string != NULL) {
         init_ptr = token_string;
         token = strtok_r(token_string, ",", &saveptr);
-        int index = 0;
-        if (NULL == token)
-            return -EINVAL;
+        if (NULL == token) {
+            rc = -EINVAL;
+            goto exit;
+        }
         else {
             mm_params->mixer_coeffs[i][j] = atof(token);
             j++;
@@ -1902,11 +1920,14 @@
                 break;
             mm_params->mixer_coeffs[i][j++] = atof(token);
         }
-        free(init_ptr);
-        init_ptr = NULL;
-        token_string = NULL;
+        goto exit;
     } else
         return -EINVAL;
+exit:
+    free(init_ptr);
+    init_ptr = NULL;
+    token_string = NULL;
+    return rc;
 }
 
 #ifdef QAP
@@ -1956,6 +1977,16 @@
 }
 #endif
 
+static void qti_audio_server_death_notify_cb(void *ctxt) {
+    fprintf(log_file, "qas died\n");
+    fprintf(stderr, "qas died\n");
+
+    stream_config *s_params = (stream_config*) ctxt;
+    pthread_cond_signal(&s_params->write_cond);
+    pthread_cond_signal(&s_params->drain_cond);
+    stop_playback = true;
+}
+
 int main(int argc, char* argv[]) {
     char *ba = NULL;
     qahw_param_payload payload;
@@ -2115,14 +2146,14 @@
                 get_kvpairs_string(optarg, "O", output_ch_map);
                 get_kvpairs_string(optarg, "M", mixer_coeff);
 
-                extract_channel_mapping(stream_param[i].mm_params_downmix.input_channel_map, input_ch_map);
+                extract_channel_mapping((uint16_t *)(stream_param[i].mm_params_downmix.input_channel_map), input_ch_map);
                 stream_param[i].mm_params_downmix.has_input_channel_map = 1;
                 fprintf(log_file, "\ndownmix Input channel mapping: ");
                 for (iter_i= 0; iter_i < stream_param[i].mm_params_downmix.num_input_channels; iter_i++) {
                     fprintf(log_file, "0x%x, ", stream_param[i].mm_params_downmix.input_channel_map[iter_i]);
                 }
 
-                extract_channel_mapping(stream_param[i].mm_params_downmix.output_channel_map, output_ch_map);
+                extract_channel_mapping((uint16_t *)(stream_param[i].mm_params_downmix.output_channel_map), output_ch_map);
                 stream_param[i].mm_params_downmix.has_output_channel_map = 1;
                 fprintf(log_file, "\ndownmix Output channel mapping: ");
                 for (iter_i = 0; iter_i < stream_param[i].mm_params_downmix.num_output_channels; iter_i++)
@@ -2150,14 +2181,14 @@
                 get_kvpairs_string(optarg, "O", output_ch_map);
                 get_kvpairs_string(optarg, "M", mixer_coeff);
 
-                extract_channel_mapping(stream_param[i].mm_params_pan_scale.input_channel_map, input_ch_map);
+                extract_channel_mapping((uint16_t *)(stream_param[i].mm_params_pan_scale.input_channel_map), input_ch_map);
                 stream_param[i].mm_params_pan_scale.has_input_channel_map = 1;
                 fprintf(log_file, "\n pan_sclae Input channel mapping: ");
                 for (iter_i= 0; iter_i < stream_param[i].mm_params_pan_scale.num_input_channels; iter_i++) {
                     fprintf(log_file, "0x%x, ", stream_param[i].mm_params_pan_scale.input_channel_map[iter_i]);
                 }
 
-                extract_channel_mapping(stream_param[i].mm_params_pan_scale.output_channel_map, output_ch_map);
+                extract_channel_mapping((uint16_t *)(stream_param[i].mm_params_pan_scale.output_channel_map), output_ch_map);
                 stream_param[i].mm_params_pan_scale.has_output_channel_map = 1;
                 fprintf(log_file, "\n pan_scale Output channel mapping: ");
                 for (iter_i = 0; iter_i < stream_param[i].mm_params_pan_scale.num_output_channels; iter_i++)
@@ -2313,6 +2344,8 @@
 
         }
     }
+    fprintf(log_file, "registering qas callback");
+    qahw_register_qas_death_notify_cb((audio_error_callback)qti_audio_server_death_notify_cb, &stream_param);
 
     wakelock_acquired = request_wake_lock(wakelock_acquired, true);
     num_of_streams = i+1;
diff --git a/qahw_api/test/qap_wrapper_extn.c b/qahw_api/test/qap_wrapper_extn.c
index 0f24412..de79ccf 100644
--- a/qahw_api/test/qap_wrapper_extn.c
+++ b/qahw_api/test/qap_wrapper_extn.c
@@ -41,7 +41,11 @@
 /*#define LOG_NDEBUG 0*/
 
 #if LINUX_ENABLED
-#define QAC_LIB_MS12 "libdolby_ms12_wrapper.so"
+#if defined(__LP64__)
+#define QAC_LIB_MS12 "/usr/lib64/libdolby_ms12_wrapper.so"
+#else
+#define QAC_LIB_MS12 "/usr/lib/libdolby_ms12_wrapper.so"
+#endif
 #define QAC_LIB_M8   "libdts_m8_wrapper.so"
 #else
 #define QAC_LIB_MS12 "/system/lib/libdolby_ms12_wrapper.so"
@@ -105,6 +109,8 @@
 pthread_mutex_t main_eos_lock;
 pthread_cond_t sec_eos_cond;
 pthread_mutex_t sec_eos_lock;
+bool main_eos_received = false;
+bool sec_eos_received = false;
 
 dlb_ms12_session_param_t dlb_param;
 dlb_ms12_session_param_t dlb_param_hp;
@@ -802,6 +808,7 @@
                 stream_cnt--;
             pthread_mutex_lock(&main_eos_lock);
             pthread_cond_signal(&main_eos_cond);
+            main_eos_received = true;
             pthread_mutex_unlock(&main_eos_lock);
 
             ALOGE("%s %d Received Main Input EOS ", __func__, __LINE__);
@@ -820,6 +827,7 @@
                 ALOGV("%s %d Received Secondary Input EOS", __func__, __LINE__);
                 pthread_mutex_lock(&sec_eos_lock);
                 pthread_cond_signal(&sec_eos_cond);
+                sec_eos_received = true;
                 pthread_mutex_unlock(&sec_eos_lock);
             }
             if (!stream_cnt)
@@ -1580,16 +1588,22 @@
 
 wait_for_eos:
     if (stream_info->sec_input) {
-        pthread_mutex_lock(&sec_eos_lock);
-        pthread_cond_wait(&sec_eos_cond, &sec_eos_lock);
-        pthread_mutex_unlock(&sec_eos_lock);
+        if (!sec_eos_received) {
+            pthread_mutex_lock(&sec_eos_lock);
+            pthread_cond_wait(&sec_eos_cond, &sec_eos_lock);
+            pthread_mutex_unlock(&sec_eos_lock);
+        }
+        sec_eos_received = false;
         fprintf(stdout, "Received EOS event for secondary input\n");
         ALOGV("Received EOS event for secondary input\n");
     }
     if (!(stream_info->system_input || stream_info->sec_input)){
-        pthread_mutex_lock(&main_eos_lock);
-        pthread_cond_wait(&main_eos_cond, &main_eos_lock);
-        pthread_mutex_unlock(&main_eos_lock);
+        if (!main_eos_received) {
+            pthread_mutex_lock(&main_eos_lock);
+            pthread_cond_wait(&main_eos_cond, &main_eos_lock);
+            pthread_mutex_unlock(&main_eos_lock);
+        }
+        main_eos_received = false;
         fprintf(stdout, "Received EOS event for main input\n");
         ALOGV("Received EOS event for main input\n");
     }
diff --git a/visualizer/Android.mk b/visualizer/Android.mk
index af4bea4..d932ce0 100644
--- a/visualizer/Android.mk
+++ b/visualizer/Android.mk
@@ -21,7 +21,7 @@
 
 LOCAL_CFLAGS+= -O2 -fvisibility=hidden
 
-ifneq ($(filter sdm660 sdm845 msm8998 apq8098_latv sdm670,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm660 sdm845 msm8998 apq8098_latv sdm670 qcs605,$(TARGET_BOARD_PLATFORM)),)
     LOCAL_CFLAGS += -DCAPTURE_DEVICE=7
 endif