diff --git a/Android.mk b/Android.mk
index 273c9cb..42f202b 100644
--- a/Android.mk
+++ b/Android.mk
@@ -16,6 +16,7 @@
 include $(MY_LOCAL_PATH)/visualizer/Android.mk
 include $(MY_LOCAL_PATH)/audiod/Android.mk
 include $(MY_LOCAL_PATH)/post_proc/Android.mk
+include $(MY_LOCAL_PATH)/qahw_api/Android.mk
 endif
 
 endif
diff --git a/configs/msm8937/audio_policy_configuration.xml b/configs/msm8937/audio_policy_configuration.xml
index 238c49e..44abe28 100644
--- a/configs/msm8937/audio_policy_configuration.xml
+++ b/configs/msm8937/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
                 <mixPort name="direct_pcm" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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_24_BIT_PACKED"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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"/>
                 </mixPort>
                 <mixPort name="compressed_offload" role="source"
@@ -245,13 +245,13 @@
             <!-- 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,BT SCO Headset Mic"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Line"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="HDMI"
@@ -268,8 +268,6 @@
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
-                <route type="mix" sink="Telephony Tx"
-                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
             </routes>
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 1202aba..7e5c90f 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -41,7 +41,7 @@
 AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
 BOARD_USES_SRS_TRUEMEDIA := true
 DTS_CODEC_M_ := true
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
 MM_AUDIO_ENABLED_FTM := true
 MM_AUDIO_ENABLED_SAFX := true
 TARGET_USES_QCOM_MM_AUDIO := true
@@ -219,3 +219,7 @@
 #Enable HW AAC Encoder by default
 PRODUCT_PROPERTY_OVERRIDES += \
 qcom.hw.aac.encoder=true
+
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
diff --git a/configs/msm8953/audio_platform_info_extcodec.xml b/configs/msm8953/audio_platform_info_extcodec.xml
index cf68190..ac0eabc 100644
--- a/configs/msm8953/audio_platform_info_extcodec.xml
+++ b/configs/msm8953/audio_platform_info_extcodec.xml
@@ -47,10 +47,13 @@
         <usecase name="USECASE_VOICEMMODE1_CALL" type="out" id="35"/>
         <usecase name="USECASE_VOICEMMODE2_CALL" type="in" id="36"/>
         <usecase name="USECASE_VOICEMMODE2_CALL" type="out" id="36"/>
+        <usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="37"/>
         <usecase name="USECASE_QCHAT_CALL" type="in" id="42"/>
         <usecase name="USECASE_QCHAT_CALL" type="out" id="42"/>
     </pcm_ids>
     <config_params>
+        <param key="spkr_1_tz_name" value="wsatz.11"/>
+        <param key="spkr_2_tz_name" value="wsatz.12"/>
         <param key="native_audio_mode" value="src"/>
         <param key="input_mic_max_count" value="4"/>
     </config_params>
diff --git a/configs/msm8953/audio_policy_configuration.xml b/configs/msm8953/audio_policy_configuration.xml
index 238c49e..44abe28 100644
--- a/configs/msm8953/audio_policy_configuration.xml
+++ b/configs/msm8953/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
                 <mixPort name="direct_pcm" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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_24_BIT_PACKED"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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"/>
                 </mixPort>
                 <mixPort name="compressed_offload" role="source"
@@ -245,13 +245,13 @@
             <!-- 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,BT SCO Headset Mic"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Line"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="HDMI"
@@ -268,8 +268,6 @@
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
-                <route type="mix" sink="Telephony Tx"
-                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
             </routes>
diff --git a/configs/msm8953/mixer_paths.xml b/configs/msm8953/mixer_paths.xml
index 91544a1..a20c6cf 100644
--- a/configs/msm8953/mixer_paths.xml
+++ b/configs/msm8953/mixer_paths.xml
@@ -54,10 +54,10 @@
     <ctl name="RX1 Digital Volume" value="84" />
     <ctl name="RX2 Digital Volume" value="84" />
     <ctl name="RX3 Digital Volume" value="84" />
-    <ctl name="IIR1 INP1 Volume" value="84" />
-    <ctl name="IIR1 INP2 Volume" value="84" />
-    <ctl name="IIR1 INP3 Volume" value="84" />
-    <ctl name="IIR1 INP4 Volume" value="84" />
+    <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="ADC1 Volume" value="4" />
     <ctl name="ADC2 Volume" value="4" />
     <ctl name="ADC3 Volume" value="4" />
diff --git a/configs/msm8953/mixer_paths_mtp.xml b/configs/msm8953/mixer_paths_mtp.xml
index 42a9e68..d618169 100644
--- a/configs/msm8953/mixer_paths_mtp.xml
+++ b/configs/msm8953/mixer_paths_mtp.xml
@@ -57,10 +57,10 @@
     <ctl name="RX1 Digital Volume" value="84" />
     <ctl name="RX2 Digital Volume" value="84" />
     <ctl name="RX3 Digital Volume" value="84" />
-    <ctl name="IIR1 INP1 Volume" value="84" />
-    <ctl name="IIR1 INP2 Volume" value="84" />
-    <ctl name="IIR1 INP3 Volume" value="84" />
-    <ctl name="IIR1 INP4 Volume" value="84" />
+    <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="ADC1 Volume" value="4" />
     <ctl name="ADC2 Volume" value="4" />
     <ctl name="ADC3 Volume" value="4" />
@@ -442,13 +442,22 @@
         <ctl name="QUIN_MI2S_RX Audio Mixer MultiMedia7" value="1" />
     </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">
+        <path name="compress-offload-playback2 afe-proxy" />
+    </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 afe-proxy">
-        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
+    <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-playback transmission-fm">
@@ -1105,6 +1114,11 @@
         <path name="headphones" />
     </path>
 
+    <path name="wsa-speaker-and-headphones">
+        <path name="wsa-speaker" />
+        <path name="headphones" />
+    </path>
+
     <path name="usb-headphones">
     </path>
 
@@ -1119,6 +1133,11 @@
         <path name="usb-headphones" />
     </path>
 
+    <path name="wsa-speaker-and-usb-headphones">
+        <path name="wsa-speaker" />
+        <path name="usb-headphones" />
+    </path>
+
     <path name="voice-rec-mic">
         <path name="handset-mic" />
     </path>
@@ -1270,4 +1289,8 @@
           <path name="speaker-and-headphones" />
     </path>
 
+    <path name="wsa-speaker-and-line">
+          <path name="wsa-speaker-and-headphones" />
+    </path>
+
 </mixer>
diff --git a/configs/msm8953/mixer_paths_qrd_sku3.xml b/configs/msm8953/mixer_paths_qrd_sku3.xml
index 0d68a71..1edb0b4 100644
--- a/configs/msm8953/mixer_paths_qrd_sku3.xml
+++ b/configs/msm8953/mixer_paths_qrd_sku3.xml
@@ -2028,6 +2028,11 @@
         <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
     </path>
 
+    <path name="wsa-speaker-and-headphones">
+        <path name="wsa-speaker" />
+        <path name="headphones" />
+    </path>
+
     <path name="usb-headphones">
     </path>
 
@@ -2045,6 +2050,11 @@
         <path name="usb-headphones" />
     </path>
 
+    <path name="wsa-speaker-and-usb-headphones">
+        <path name="wsa-speaker" />
+        <path name="usb-headphones" />
+    </path>
+
     <path name="speaker-and-hdmi">
         <path name="wsa-speaker" />
         <path name="hdmi" />
@@ -2089,7 +2099,7 @@
         <ctl name="DMIC MUX7" value="DMIC0" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC1" />
+        <ctl name="DMIC MUX8" value="DMIC2" />
         <ctl name="SLIM_0_TX Channels" value="Two" />
     </path>
 
@@ -2101,7 +2111,7 @@
         <ctl name="DMIC MUX7" value="DMIC0" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC1" />
+        <ctl name="DMIC MUX8" value="DMIC2" />
         <ctl name="SLIM_0_TX Channels" value="Two" />
     </path>
 
@@ -2148,7 +2158,7 @@
         <ctl name="DMIC MUX7" value="DMIC0" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC1" />
+        <ctl name="DMIC MUX8" value="DMIC2" />
     </path>
 
     <path name="dmic-broadside">
@@ -2248,4 +2258,9 @@
     <path name="speaker-and-line">
         <path name="speaker-and-headphones" />
     </path>
+
+    <path name="wsa-speaker-and-line">
+        <path name="wsa-speaker" />
+        <path name="headphones" />
+    </path>
 </mixer>
diff --git a/configs/msm8953/mixer_paths_qrd_skuh.xml b/configs/msm8953/mixer_paths_qrd_skuh.xml
index aa70a90..ebf9331 100644
--- a/configs/msm8953/mixer_paths_qrd_skuh.xml
+++ b/configs/msm8953/mixer_paths_qrd_skuh.xml
@@ -57,10 +57,10 @@
     <ctl name="RX1 Digital Volume" value="84" />
     <ctl name="RX2 Digital Volume" value="84" />
     <ctl name="RX3 Digital Volume" value="84" />
-    <ctl name="IIR1 INP1 Volume" value="84" />
-    <ctl name="IIR1 INP2 Volume" value="84" />
-    <ctl name="IIR1 INP3 Volume" value="84" />
-    <ctl name="IIR1 INP4 Volume" value="84" />
+    <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="ADC1 Volume" value="6" />
     <ctl name="ADC2 Volume" value="6" />
     <ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/mixer_paths_qrd_skuhf.xml b/configs/msm8953/mixer_paths_qrd_skuhf.xml
index 84b95bd..1ece99d 100644
--- a/configs/msm8953/mixer_paths_qrd_skuhf.xml
+++ b/configs/msm8953/mixer_paths_qrd_skuhf.xml
@@ -57,10 +57,10 @@
     <ctl name="RX1 Digital Volume" value="84" />
     <ctl name="RX2 Digital Volume" value="84" />
     <ctl name="RX3 Digital Volume" value="84" />
-    <ctl name="IIR1 INP1 Volume" value="84" />
-    <ctl name="IIR1 INP2 Volume" value="84" />
-    <ctl name="IIR1 INP3 Volume" value="84" />
-    <ctl name="IIR1 INP4 Volume" value="84" />
+    <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="ADC1 Volume" value="6" />
     <ctl name="ADC2 Volume" value="6" />
     <ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/mixer_paths_qrd_skui.xml b/configs/msm8953/mixer_paths_qrd_skui.xml
index aa70a90..ebf9331 100644
--- a/configs/msm8953/mixer_paths_qrd_skui.xml
+++ b/configs/msm8953/mixer_paths_qrd_skui.xml
@@ -57,10 +57,10 @@
     <ctl name="RX1 Digital Volume" value="84" />
     <ctl name="RX2 Digital Volume" value="84" />
     <ctl name="RX3 Digital Volume" value="84" />
-    <ctl name="IIR1 INP1 Volume" value="84" />
-    <ctl name="IIR1 INP2 Volume" value="84" />
-    <ctl name="IIR1 INP3 Volume" value="84" />
-    <ctl name="IIR1 INP4 Volume" value="84" />
+    <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="ADC1 Volume" value="6" />
     <ctl name="ADC2 Volume" value="6" />
     <ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/mixer_paths_qrd_skum.xml b/configs/msm8953/mixer_paths_qrd_skum.xml
index 8343847..d504456 100644
--- a/configs/msm8953/mixer_paths_qrd_skum.xml
+++ b/configs/msm8953/mixer_paths_qrd_skum.xml
@@ -57,10 +57,10 @@
     <ctl name="RX1 Digital Volume" value="84" />
     <ctl name="RX2 Digital Volume" value="84" />
     <ctl name="RX3 Digital Volume" value="84" />
-    <ctl name="IIR1 INP1 Volume" value="84" />
-    <ctl name="IIR1 INP2 Volume" value="84" />
-    <ctl name="IIR1 INP3 Volume" value="84" />
-    <ctl name="IIR1 INP4 Volume" value="84" />
+    <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="ADC1 Volume" value="6" />
     <ctl name="ADC2 Volume" value="6" />
     <ctl name="ADC3 Volume" value="6" />
@@ -366,6 +366,28 @@
         <ctl name="INTERNAL_FM_RX Audio Mixer MultiMedia4" value="1" />
     </path>
 
+    <path name="compress-offload-playback2 hdmi">
+        <ctl name="QUIN_MI2S_RX Audio Mixer MultiMedia7" value="1" />
+    </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">
+        <path name="compress-offload-playback2 afe-proxy" />
+    </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-usb-headphones">
+        <path name="compress-offload-playback2 usb-headphones" />
+        <path name="compress-offload-playback2" />
+    </path>
+
     <path name="compress-offload-playback3">
         <ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -795,6 +817,11 @@
         <path name="headphones" />
     </path>
 
+    <path name="wsa-speaker-and-headphones">
+        <path name="wsa-speaker" />
+        <path name="headphones" />
+    </path>
+
     <path name="usb-headphones">
     </path>
 
@@ -809,6 +836,11 @@
         <path name="usb-headphones" />
     </path>
 
+    <path name="wsa-speaker-and-usb-headphones">
+        <path name="wsa-speaker" />
+        <path name="usb-headphones" />
+    </path>
+
     <path name="voice-rec-mic">
         <path name="handset-mic" />
     </path>
@@ -934,4 +966,8 @@
           <path name="speaker-and-headphones" />
     </path>
 
+    <path name="wsa-speaker-and-line">
+          <path name="wsa-speaker-and-headphones" />
+    </path>
+
 </mixer>
diff --git a/configs/msm8953/mixer_paths_skuk.xml b/configs/msm8953/mixer_paths_skuk.xml
index 98a1bab..1538275 100644
--- a/configs/msm8953/mixer_paths_skuk.xml
+++ b/configs/msm8953/mixer_paths_skuk.xml
@@ -57,10 +57,10 @@
     <ctl name="RX1 Digital Volume" value="84" />
     <ctl name="RX2 Digital Volume" value="84" />
     <ctl name="RX3 Digital Volume" value="84" />
-    <ctl name="IIR1 INP1 Volume" value="84" />
-    <ctl name="IIR1 INP2 Volume" value="84" />
-    <ctl name="IIR1 INP3 Volume" value="84" />
-    <ctl name="IIR1 INP4 Volume" value="84" />
+    <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="ADC1 Volume" value="6" />
     <ctl name="ADC2 Volume" value="6" />
     <ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index e646646..2917f9d 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -41,7 +41,7 @@
 BOARD_USES_SRS_TRUEMEDIA := true
 DTS_CODEC_M_ := true
 #AUDIO_FEATURE_ENABLED_MULTIPLE_TUNNEL := true
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
 MM_AUDIO_ENABLED_FTM := true
 MM_AUDIO_ENABLED_SAFX := true
 TARGET_USES_QCOM_MM_AUDIO := true
@@ -219,3 +219,7 @@
 #Enable HW AAC Encoder by default
 PRODUCT_PROPERTY_OVERRIDES += \
 qcom.hw.aac.encoder=true
+
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
diff --git a/configs/msm8996/audio_policy_configuration.xml b/configs/msm8996/audio_policy_configuration.xml
index 56848ad..e8d4cd0 100644
--- a/configs/msm8996/audio_policy_configuration.xml
+++ b/configs/msm8996/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
                 <mixPort name="direct_pcm" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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_24_BIT_PACKED"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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"/>
                 </mixPort>
                 <mixPort name="compressed_offload" role="source"
@@ -139,7 +139,7 @@
                 <mixPort name="voip_rx" role="source"
                          flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                             samplingRates="8000,16000,32000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
                 </mixPort>
 
                 <mixPort name="primary input" role="sink">
@@ -256,13 +256,13 @@
             <!-- 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,BT SCO Headset Mic"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Line"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="HDMI"
@@ -281,8 +281,6 @@
                        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="Telephony Tx"
-                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
             </routes>
diff --git a/configs/msm8996/mixer_paths_tasha.xml b/configs/msm8996/mixer_paths_tasha.xml
index 5741192..9f63413 100644
--- a/configs/msm8996/mixer_paths_tasha.xml
+++ b/configs/msm8996/mixer_paths_tasha.xml
@@ -1975,8 +1975,7 @@
     </path>
 
     <path name="handset-mic-sbc">
-        <path name="adc5" />
-        <ctl name="ADC5 Volume" value="12" />
+        <path name="dmic3" />
     </path>
 
     <path name="three-mic">
diff --git a/configs/msm8996/msm8996.mk b/configs/msm8996/msm8996.mk
index 3b83c24..306fa97 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -41,7 +41,7 @@
 AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
 BOARD_USES_SRS_TRUEMEDIA := true
 DTS_CODEC_M_ := true
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
 MM_AUDIO_ENABLED_FTM := true
 MM_AUDIO_ENABLED_SAFX := true
 TARGET_USES_QCOM_MM_AUDIO := true
@@ -50,6 +50,7 @@
 #DOLBY_DDP := true
 AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
 AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
+AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
 ##AUDIO_FEATURE_FLAGS
 
 #Audio Specific device overlays
@@ -196,3 +197,7 @@
 use.qti.sw.alac.decoder=true
 PRODUCT_PROPERTY_OVERRIDES += \
 use.qti.sw.ape.decoder=true
+
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
diff --git a/configs/msmcobalt/aanc_tuning_mixer.txt b/configs/msmcobalt/aanc_tuning_mixer.txt
index 35743ff..5639bd2 100644
--- a/configs/msmcobalt/aanc_tuning_mixer.txt
+++ b/configs/msmcobalt/aanc_tuning_mixer.txt
@@ -8,10 +8,10 @@
 AIF1_CAP Mixer SLIM TX8:1
 SLIM TX7 MUX:DEC7
 ADC MUX7:DMIC
-DMIC MUX7:DMIC3
+DMIC MUX7:DMIC4
 SLIM TX8 MUX:DEC8
 ADC MUX8:DMIC
-DMIC MUX8:DMIC2
+DMIC MUX8:DMIC0
 SLIM_0_TX Channels:Two
 MultiMedia1 Mixer SLIM_0_TX:1
 
@@ -35,13 +35,13 @@
 SLIM TX7 MUX:DEC7
 ADC MUX7:ANC_FB_TUNE1
 ADC MUX10:DMIC
-DMIC MUX10:DMIC2
+DMIC MUX10:DMIC0
 SLIM TX8 MUX:DEC8
 ADC MUX8:ANC_FB_TUNE2
 ADC MUX12:DMIC
-DMIC MUX12:DMIC2
-ANC0 FB MUX:ANC_IN_EAR
-ANC EAR Enable Switch:1
+DMIC MUX12:DMIC0
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
 SLIM_0_TX Channels:Two
 MultiMedia1 Mixer SLIM_0_TX:1
 
@@ -56,7 +56,7 @@
 DMIC MUX10:ZERO
 DMIC MUX12:ZERO
 ANC0 FB MUX:ZERO
-ANC EAR Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
 
 
 #ANC_TEST_E_PATH_MIC_STEREO Capture
@@ -70,13 +70,13 @@
 SLIM TX7 MUX:DEC7
 ADC MUX7:ANC_FB_TUNE1
 ADC MUX10:DMIC
-DMIC MUX10:DMIC3
+DMIC MUX10:DMIC4
 SLIM TX8 MUX:DEC8
 ADC MUX8:ANC_FB_TUNE2
 ADC MUX12:DMIC
-DMIC MUX12:DMIC3
-ANC0 FB MUX:ANC_IN_EAR
-ANC EAR Enable Switch:1
+DMIC MUX12:DMIC4
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
 SLIM_0_TX Channels:Two
 MultiMedia1 Mixer SLIM_0_TX:1
 
@@ -91,7 +91,7 @@
 DMIC MUX10:ZERO
 DMIC MUX12:ZERO
 ANC0 FB MUX:ZERO
-ANC EAR Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
 
 
 #ANC_TEST_S_PATH_HANDSET_SPKR_ANC_MONO
@@ -103,19 +103,23 @@
 ANC Function:ON
 SLIM RX0 MUX:AIF_MIX1_PB
 SLIM_0_RX Channels:One
-RX INT0_1 MIX1 INP0:RX0
-RX INT0 DEM MUX:CLSH_DSM_OUT
-RX0 Digital Volume:87
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
 ANC Slot:7
-EAR PA Gain:G_6_DB
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
 SLIMBUS_0_RX Audio Mixer MultiMedia1:1
 
 disable
 SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
 ANC Slot:0
 SLIM RX0 MUX:ZERO
-RX INT0_1 MIX1 INP0:ZERO
-RX0 Digital Volume:0
+RX INT7_1 MIX1 INP0:ZERO
 ANC Function:OFF
 
 #ANC_TEST_E_PATH_HANDSET_SPKR_ANC_MONO
@@ -127,17 +131,21 @@
 ANC Function:ON
 SLIM RX0 MUX:AIF_MIX1_PB
 SLIM_0_RX Channels:One
-RX INT0_1 MIX1 INP0:RX0
-RX INT0 DEM MUX:CLSH_DSM_OUT
-RX0 Digital Volume:87
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
 ANC Slot:8
-EAR PA Gain:G_6_DB
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
 SLIMBUS_0_RX Audio Mixer MultiMedia1:1
 
 disable
 SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
 ANC Slot:0
 SLIM RX0 MUX:ZERO
-RX INT0_1 MIX1 INP0:ZERO
-RX0 Digital Volume:0
+RX INT7_1 MIX1 INP0:ZERO
 ANC Function:OFF
diff --git a/configs/msmcobalt/aanc_tuning_mixer_tavil.txt b/configs/msmcobalt/aanc_tuning_mixer_tavil.txt
new file mode 100644
index 0000000..78156d3
--- /dev/null
+++ b/configs/msmcobalt/aanc_tuning_mixer_tavil.txt
@@ -0,0 +1,151 @@
+#ANC_TEST_P_PATH_MIC_STEREO Capture
+acdb_dev_id:85
+!Capture
+Txdevice:0
+
+enable
+AIF1_CAP Mixer SLIM TX7:1
+AIF1_CAP Mixer SLIM TX8:1
+CDC_IF TX7 MUX:DEC7
+ADC MUX7:DMIC
+DMIC MUX7:DMIC4
+CDC_IF TX8 MUX:DEC8
+ADC MUX8:DMIC
+DMIC MUX8:DMIC0
+SLIM_0_TX Channels:Two
+MultiMedia1 Mixer SLIM_0_TX:1
+
+disable
+MultiMedia1 Mixer SLIM_0_TX:0
+AIF1_CAP Mixer SLIM TX7:0
+AIF1_CAP Mixer SLIM TX8:0
+CDC_IF TX7 MUX:ZERO
+CDC_IF TX8 MUX:ZERO
+DMIC MUX7:ZERO
+DMIC MUX8:ZERO
+
+#ANC_TEST_S_PATH_MIC_STEREO Capture
+acdb_dev_id:88
+!Capture
+Txdevice:0
+
+enable
+AIF1_CAP Mixer SLIM TX7:1
+AIF1_CAP Mixer SLIM TX8:1
+CDC_IF TX7 MUX:DEC7
+ADC MUX7:ANC_FB_TUNE1
+ADC MUX10:DMIC
+DMIC MUX10:DMIC0
+CDC_IF TX8 MUX:DEC8
+ADC MUX8:ANC_FB_TUNE2
+ADC MUX12:DMIC
+DMIC MUX12:DMIC0
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
+SLIM_0_TX Channels:Two
+MultiMedia1 Mixer SLIM_0_TX:1
+
+disable
+MultiMedia1 Mixer SLIM_0_TX:0
+AIF1_CAP Mixer SLIM TX7:0
+AIF1_CAP Mixer SLIM TX8:0
+CDC_IF TX7 MUX:ZERO
+CDC_IF TX8 MUX:ZERO
+ADC MUX7:DMIC
+ADC MUX8:DMIC
+DMIC MUX10:ZERO
+DMIC MUX12:ZERO
+ANC0 FB MUX:ZERO
+ANC OUT EAR SPKR Enable Switch:0
+
+
+#ANC_TEST_E_PATH_MIC_STEREO Capture
+acdb_dev_id:91
+!Capture
+Txdevice:0
+
+enable
+AIF1_CAP Mixer SLIM TX7:1
+AIF1_CAP Mixer SLIM TX8:1
+CDC_IF TX7 MUX:DEC7
+ADC MUX7:ANC_FB_TUNE1
+ADC MUX10:DMIC
+DMIC MUX10:DMIC4
+CDC_IF TX8 MUX:DEC8
+ADC MUX8:ANC_FB_TUNE2
+ADC MUX12:DMIC
+DMIC MUX12:DMIC4
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
+SLIM_0_TX Channels:Two
+MultiMedia1 Mixer SLIM_0_TX:1
+
+disable
+MultiMedia1 Mixer SLIM_0_TX:0
+AIF1_CAP Mixer SLIM TX7:0
+AIF1_CAP Mixer SLIM TX8:0
+CDC_IF TX7 MUX:ZERO
+CDC_IF TX8 MUX:ZERO
+ADC MUX7:DMIC
+ADC MUX8:DMIC
+DMIC MUX10:ZERO
+DMIC MUX12:ZERO
+ANC0 FB MUX:ZERO
+ANC OUT EAR SPKR Enable Switch:0
+
+
+#ANC_TEST_S_PATH_HANDSET_SPKR_ANC_MONO
+acdb_dev_id:86
+!Playback
+Rxdevice:0
+
+enable
+ANC Function:ON
+SLIM RX0 MUX:AIF1_PB
+SLIM_0_RX Channels:One
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
+ANC Slot:7
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
+SLIMBUS_0_RX Audio Mixer MultiMedia1:1
+
+disable
+SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
+ANC Slot:0
+SLIM RX0 MUX:ZERO
+RX INT7_1 MIX1 INP0:ZERO
+ANC Function:OFF
+
+#ANC_TEST_E_PATH_HANDSET_SPKR_ANC_MONO
+acdb_dev_id:89
+!Playback
+Rxdevice:0
+
+enable
+ANC Function:ON
+SLIM RX0 MUX:AIF1_PB
+SLIM_0_RX Channels:One
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
+ANC Slot:8
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
+SLIMBUS_0_RX Audio Mixer MultiMedia1:1
+
+disable
+SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
+ANC Slot:0
+SLIM RX0 MUX:ZERO
+RX INT7_1 MIX1 INP0:ZERO
+ANC Function:OFF
diff --git a/configs/msmcobalt/audio_effects.conf b/configs/msmcobalt/audio_effects.conf
index 5738cf9..d643592 100644
--- a/configs/msmcobalt/audio_effects.conf
+++ b/configs/msmcobalt/audio_effects.conf
@@ -48,6 +48,9 @@
   volume_listener {
     path /system/lib/soundfx/libvolumelistener.so
   }
+  audiosphere {
+    path /system/lib/soundfx/libasphere.so
+  }
 }
 
 # Default pre-processing library. Add to audio_effect.conf "libraries" section if
@@ -252,6 +255,10 @@
     library volume_listener
     uuid 0b776dde-0590-11e5-81ba-0025b32654a0
   }
+  audiosphere {
+    library audiosphere
+    uuid 184e62ab-2d19-4364-9d1b-c0a40733866c
+  }
 }
 
 # additional effect from vendor
diff --git a/configs/msmcobalt/audio_output_policy.conf b/configs/msmcobalt/audio_output_policy.conf
index 67d79bf..e60c664 100644
--- a/configs/msmcobalt/audio_output_policy.conf
+++ b/configs/msmcobalt/audio_output_policy.conf
@@ -42,14 +42,14 @@
   direct_pcm_24 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
     formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
-    sampling_rates 44100|48000|96000|192000
+    sampling_rates 44100|48000|96000|176400|192000|352800
     bit_width 24
     app_type 69940
   }
   compress_passthrough_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING|AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
-    formats AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_E_AC3_JOC|AUDIO_FORMAT_DTS|AUDIO_FORMAT_DTS_HD
-    sampling_rates 32000|44100|48000|88200|96000|176400|192000
+    formats AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_E_AC3_JOC|AUDIO_FORMAT_DTS|AUDIO_FORMAT_DTS_HD|AUDIO_FORMAT_DSD
+    sampling_rates 32000|44100|48000|88200|96000|176400|192000|352800
     bit_width 16
     app_type 69941
   }
diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index 72ed9f3..07839fd 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -53,6 +53,11 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_FM" type="in" id="34"/>
         <usecase name="USECASE_AUDIO_SPKR_CALIB_RX" type="out" id="5"/>
         <usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="35"/>
+        <usecase name="USECASE_AUDIO_PLAYBACK_AFE_PROXY" type="out" id="6"/>
+        <usecase name="USECASE_AUDIO_RECORD_AFE_PROXY" type="in" id="7"/>
+        <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" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsatz.13"/>
@@ -68,9 +73,12 @@
     <backend_names>
         <device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
+        <device name="SND_DEVICE_OUT_ANC_HEADSET" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+        <device name="SND_DEVICE_OUT_VOICE_ANC_HEADSET" backend="headphones" interface="SLIMBUS_6_RX"/>
         <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"/>
diff --git a/configs/msmcobalt/audio_policy.conf b/configs/msmcobalt/audio_policy.conf
index dd827fe..70ab311 100644
--- a/configs/msmcobalt/audio_policy.conf
+++ b/configs/msmcobalt/audio_policy.conf
@@ -26,21 +26,21 @@
         sampling_rates 44100|48000
         channel_masks AUDIO_CHANNEL_OUT_STEREO
         formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE
+        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY
       }
       raw {
         sampling_rates 48000
         channel_masks AUDIO_CHANNEL_OUT_STEREO
         formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE
+        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW
       }
       deep_buffer {
          sampling_rates 44100|48000
          channel_masks AUDIO_CHANNEL_OUT_STEREO
          formats AUDIO_FORMAT_PCM_16_BIT
-         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE
+         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
          flags AUDIO_OUTPUT_FLAG_DEEP_BUFFER
       }
       compress_passthrough {
@@ -58,17 +58,24 @@
         flags AUDIO_OUTPUT_FLAG_DIRECT
       }
       direct_pcm {
-        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
+        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000|352800
         channel_masks 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
         formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
-        devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE
+        devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
       }
       compress_offload {
         sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks 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
         formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
-        devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE
+        devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
+        flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
+      }
+      dsd_compress_passthrough {
+        sampling_rates 2822400|5644800
+        channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO
+        formats AUDIO_FORMAT_DSD
+        devices AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
       }
       incall_music {
@@ -108,14 +115,6 @@
     }
   }
   a2dp {
-    outputs {
-      a2dp {
-        sampling_rates 44100
-        channel_masks AUDIO_CHANNEL_OUT_STEREO
-        formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_ALL_A2DP
-      }
-    }
     inputs {
       a2dp {
         sampling_rates 44100|48000
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index a3876ef..66b7d17 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -86,13 +86,13 @@
                 <mixPort name="direct_pcm" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
                              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="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
                              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"
@@ -118,6 +118,21 @@
                     <profile name="" format="AUDIO_FORMAT_AAC_HE_V2"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
                              channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                    <profile name="" format="AUDIO_FORMAT_AC3"
+                             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_E_AC3"
+                             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_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"/>
@@ -137,6 +152,12 @@
                              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"/>
@@ -144,7 +165,7 @@
                 <mixPort name="voip_rx" role="source"
                          flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                             samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
                 </mixPort>
 
                 <mixPort name="primary input" role="sink">
@@ -228,6 +249,22 @@
                     <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"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink">
+                    <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"
@@ -257,19 +294,25 @@
                     <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,BT SCO Headset Mic"/>
-                <route type="mix" sink="Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
-                <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
-                <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
-                <route type="mix" sink="Line"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                <route type="mix" sink="Speaker"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                <route type="mix" sink="Wired Headset"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
+                <route type="mix" sink="Wired Headphones"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
+                <route type="mix" sink="Line"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
                 <route type="mix" sink="HDMI"
                        sources="primary output,raw,deep_buffer,multichannel,direct_pcm,compressed_offload,compress_passthrough"/>
                 <route type="mix" sink="Proxy"
@@ -278,27 +321,69 @@
                        sources="primary output"/>
                 <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"/>
                 <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,Telephony Rx"/>
+                       sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,Telephony Rx"/>
                 <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"/>
-                <route type="mix" sink="Telephony Tx"
-                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
-                <route type="mix" sink="voice_rx"
-                       sources="Telephony Rx"/>
+                <route type="mix" sink="BT A2DP Out"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                <route type="mix" sink="BT A2DP Headphones"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                <route type="mix" sink="BT A2DP Speaker"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
             </routes>
 
         </module>
 
-        <!-- A2dp Audio HAL -->
-        <xi:include href="a2dp_audio_policy_configuration.xml"/>
+        <!-- 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 -->
-        <xi:include href="usb_audio_policy_configuration.xml"/>
+        <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="r_submix_audio_policy_configuration.xml"/>
diff --git a/configs/msmcobalt/graphite_ipc_platform_info.xml b/configs/msmcobalt/graphite_ipc_platform_info.xml
new file mode 100644
index 0000000..f6775be
--- /dev/null
+++ b/configs/msmcobalt/graphite_ipc_platform_info.xml
@@ -0,0 +1,47 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2016, The Linux Foundation. All rights reserved.       -->
+<!---                                                                           -->
+<!--- Redistribution and use in source and binary forms, with or without        -->
+<!--- modification, are permitted provided that the following conditions are    -->
+<!--- met:                                                                      -->
+<!---     * Redistributions of source code must retain the above copyright      -->
+<!---       notice, this list of conditions and the following disclaimer.       -->
+<!---     * Redistributions in binary form must reproduce the above             -->
+<!---       copyright notice, this list of conditions and the following         -->
+<!---       disclaimer in the documentation and/or other materials provided     -->
+<!---       with the distribution.                                              -->
+<!---     * Neither the name of The Linux Foundation nor the names of its       -->
+<!---       contributors may be used to endorse or promote products derived     -->
+<!---       from this software without specific prior written permission.       -->
+<!---                                                                           -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED              -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF      -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT    -->
+<!--- ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS    -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR    -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF      -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR           -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,     -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE      -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN    -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                             -->
+<graphite_ipc_platform_info>
+    <no_of_glink_channels value="4">
+    </no_of_glink_channels>
+    <!-- channel 1 configuration -->
+    <glink_channel name="g_glink_ctrl" latency_in_us="5000"
+        no_of_intents="1" intents_size="1024">
+    </glink_channel>
+    <!-- channel 2 configuration -->
+    <glink_channel name="g_glink_persistent_data_ild" latency_in_us="30000"
+        no_of_intents="0">
+    </glink_channel>
+    <!-- channel 3 configuration -->
+    <glink_channel name="g_glink_persistent_data_nild" latency_in_us="30000"
+        no_of_intents="0">
+    </glink_channel>
+    <!-- channel 4 configuration -->
+    <glink_channel name="g_glink_audio_data" latency_in_us="10000"
+        no_of_intents="2" intents_size="4096, 4096">
+    </glink_channel>
+</graphite_ipc_platform_info>
diff --git a/configs/msmcobalt/mixer_paths_dtp.xml b/configs/msmcobalt/mixer_paths_dtp.xml
index 9bcf15b..a6c61e4 100644
--- a/configs/msmcobalt/mixer_paths_dtp.xml
+++ b/configs/msmcobalt/mixer_paths_dtp.xml
@@ -138,6 +138,8 @@
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
@@ -617,7 +619,7 @@
     </path>
 
     <path name="audio-ull-playback">
-        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-protected">
@@ -634,11 +636,11 @@
     </path>
 
     <path name="audio-ull-playback hdmi">
-        <ctl name="HDMI Mixer MultiMedia3" value="1" />
+        <ctl name="HDMI Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco">
-        <ctl name="AUX_PCM_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="AUX_PCM_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco-wb">
@@ -652,7 +654,7 @@
     </path>
 
     <path name="audio-ull-playback afe-proxy">
-        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
     </path>
     <path name="multi-channel-playback hdmi">
         <ctl name="HDMI Mixer MultiMedia2" value="1" />
@@ -1103,11 +1105,11 @@
     </path>
 
     <path name="low-latency-record">
-      <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco">
-      <ctl name="MultiMedia5 Mixer AUX_PCM_UL_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer AUX_PCM_UL_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco-wb">
@@ -1116,11 +1118,11 @@
     </path>
 
     <path name="low-latency-record usb-headset-mic">
-        <ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="1" />
+        <ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="1" />
     </path>
 
     <path name="low-latency-record capture-fm">
-      <ctl name="MultiMedia5 Mixer TERT_MI2S_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer TERT_MI2S_TX" value="1" />
     </path>
 
     <path name="fm-virtual-record capture-fm">
diff --git a/configs/msmcobalt/mixer_paths_skuk.xml b/configs/msmcobalt/mixer_paths_skuk.xml
new file mode 100644
index 0000000..24499b9
--- /dev/null
+++ b/configs/msmcobalt/mixer_paths_skuk.xml
@@ -0,0 +1,2415 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!-- Copyright (c) 2015-2016, The Linux Foundation. All rights reserved.    -->
+<!--                                                                        -->
+<!-- Redistribution and use in source and binary forms, with or without     -->
+<!-- modification, are permitted provided that the following conditions are -->
+<!-- met:                                                                   -->
+<!--     * Redistributions of source code must retain the above copyright   -->
+<!--       notice, this list of conditions and the following disclaimer.    -->
+<!--     * Redistributions in binary form must reproduce the above          -->
+<!--       copyright notice, this list of conditions and the following      -->
+<!--       disclaimer in the documentation and/or other materials provided  -->
+<!--       with the distribution.                                           -->
+<!--     * Neither the name of The Linux Foundation nor the names of its    -->
+<!--       contributors may be used to endorse or promote products derived  -->
+<!--       from this software without specific prior written permission.    -->
+<!--                                                                        -->
+<!-- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED           -->
+<!-- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF   -->
+<!-- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT -->
+<!-- ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS -->
+<!-- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR -->
+<!-- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF   -->
+<!-- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR        -->
+<!-- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,  -->
+<!-- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE   -->
+<!-- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
+<!-- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                          -->
+<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="SLIMBUS_0_RX Port Mixer SLIM_0_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="SLIMBUS_6_RX Port Mixer SLIM_0_TX" value="0" />
+    <ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia2" value="0" />
+    <ctl name="MultiMedia5 Mixer SLIM_0_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 SLIM_0_TX" value="0" />
+    <ctl name="MultiMedia1 Mixer SLIM_4_TX" value="0" />
+    <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+    <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="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="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia2" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="SLIMBUS_6_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="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_RX Channels" value="One" />
+    <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
+    <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
+    <ctl name="USB_AUDIO_TX Channels" value="One" />
+    <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
+    <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
+    <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
+    <ctl name="SLIM_2_RX Format" value="UNPACKED" />
+    <ctl name="SLIM_2_RX SampleRate" value="KHZ_48" />
+    <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
+    <ctl name="SLIM_0_RX Channels" value="One" />
+    <ctl name="SLIM_5_RX Channels" value="One" />
+    <ctl name="SLIM_6_RX Channels" value="One" />
+    <ctl name="SLIM_2_RX Channels" value="One" />
+    <ctl name="SLIM_0_TX Channels" value="One" />
+    <ctl name="SLIM_1_TX Channels" value="One" />
+    <ctl name="AIF1_CAP Mixer SLIM TX9" value="0"/>
+    <ctl name="AIF1_CAP Mixer SLIM TX8" value="0"/>
+    <ctl name="AIF1_CAP Mixer SLIM TX7" value="0" />
+    <ctl name="AIF1_CAP Mixer SLIM TX6" value="0" />
+    <ctl name="AIF1_CAP Mixer SLIM TX5" value="0"/>
+    <ctl name="AIF1_CAP Mixer SLIM TX4" value="0" />
+    <ctl name="AIF1_CAP Mixer SLIM TX3" value="0"/>
+    <ctl name="AIF1_CAP Mixer SLIM TX2" value="0" />
+    <ctl name="AIF1_CAP Mixer SLIM TX1" value="0"/>
+    <ctl name="AIF1_CAP Mixer SLIM TX0" value="0"/>
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="HDMI Mixer MultiMedia4" value="0" />
+    <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="ZERO" />
+    <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="ZERO" />
+    <ctl name="VI_FEED_TX Channels" value="Two" />
+    <ctl name="AIF4_VI Mixer SPKR_VI_1" value="0" />
+    <ctl name="AIF4_VI Mixer SPKR_VI_2" value="0" />
+    <ctl name="SLIM_4_TX Format" value="UNPACKED" />
+    <ctl name="AANC_SLIM_0_RX MUX" value="ZERO" />
+    <!-- HFP start -->
+    <ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
+    <ctl name="SLIMBUS_0_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="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
+    <ctl name="SLIMBUS_DL_HL Switch" value="0" />
+    <ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
+    <ctl name="SLIMBUS6_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="SLIM_0_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="VoiceMMode1_Tx Mixer SLIM_0_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="SLIM_0_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="VoiceMMode2_Tx Mixer SLIM_0_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" />
+    <ctl name="AIF3_CAP Mixer SLIM TX1" value="0" />
+    <ctl name="AIF3_CAP Mixer SLIM TX2" value="0" />
+    <!-- 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="SLIM_0_RX_Voice Mixer Voip" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer Voip" value="0" />
+    <ctl name="Voip_Tx Mixer SLIM_0_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 SLIM_0_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="SLIMBUS_DL_HL Switch" value="0" />
+    <ctl name="SLIMBUS6_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" />
+
+    <!-- Volume controls -->
+    <ctl name="LINEOUT1 Volume" value="13" />
+    <ctl name="LINEOUT2 Volume" value="13" />
+    <ctl name="HPHL Volume" value="20" />
+    <ctl name="HPHR Volume" value="20" />
+    <ctl name="EAR PA Gain" value="G_6_DB" />
+    <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
+
+    <ctl name="RX0 Digital Volume" value="84" />
+    <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="RX6 Digital Volume" value="84" />
+    <ctl name="RX7 Digital Volume" value="84" />
+    <ctl name="ADC1 Volume" value="12" />
+    <ctl name="ADC2 Volume" value="12" />
+    <ctl name="ADC3 Volume" value="12" />
+    <ctl name="ADC4 Volume" value="12" />
+    <ctl name="DEC0 Volume" value="84" />
+    <ctl name="DEC1 Volume" value="84" />
+    <ctl name="DEC2 Volume" value="84" />
+    <ctl name="DEC3 Volume" value="84" />
+    <ctl name="DEC4 Volume" value="84" />
+    <ctl name="DEC5 Volume" value="84" />
+    <ctl name="DEC6 Volume" value="84" />
+    <ctl name="DEC7 Volume" value="84" />
+    <ctl name="DEC8 Volume" value="84" />
+
+    <!-- Compander controls -->
+    <ctl name="COMP1 Switch" value="1" />
+    <ctl name="COMP2 Switch" value="1" />
+    <ctl name="COMP7 Switch" value="0" />
+    <ctl name="COMP8 Switch" value="0" />
+
+    <!-- ADC, DMIC and AMIC controls -->
+    <ctl name="ADC MUX0" value="AMIC" />
+    <ctl name="ADC MUX1" value="AMIC" />
+    <ctl name="ADC MUX2" value="AMIC" />
+    <ctl name="ADC MUX3" value="AMIC" />
+    <ctl name="ADC MUX4" value="AMIC" />
+    <ctl name="ADC MUX5" value="AMIC" />
+    <ctl name="ADC MUX6" value="AMIC" />
+    <ctl name="ADC MUX7" value="AMIC" />
+    <ctl name="ADC MUX8" value="AMIC" />
+    <ctl name="ADC MUX10" value="AMIC" />
+    <ctl name="ADC MUX11" value="AMIC" />
+    <ctl name="ADC MUX12" value="AMIC" />
+    <ctl name="ADC MUX13" value="AMIC" />
+    <ctl name="DMIC MUX0" value="ZERO" />
+    <ctl name="DMIC MUX1" value="ZERO" />
+    <ctl name="DMIC MUX2" value="ZERO" />
+    <ctl name="DMIC MUX3" value="ZERO" />
+    <ctl name="DMIC MUX4" value="ZERO" />
+    <ctl name="DMIC MUX5" value="ZERO" />
+    <ctl name="DMIC MUX6" value="ZERO" />
+    <ctl name="DMIC MUX7" value="ZERO" />
+    <ctl name="DMIC MUX8" value="ZERO" />
+    <ctl name="DMIC MUX10" value="ZERO" />
+    <ctl name="DMIC MUX11" value="ZERO" />
+    <ctl name="DMIC MUX12" value="ZERO" />
+    <ctl name="DMIC MUX13" value="ZERO" />
+    <ctl name="AMIC MUX0" value="ZERO" />
+    <ctl name="AMIC MUX1" value="ZERO" />
+    <ctl name="AMIC MUX2" value="ZERO" />
+    <ctl name="AMIC MUX3" value="ZERO" />
+    <ctl name="AMIC MUX4" value="ZERO" />
+    <ctl name="AMIC MUX5" value="ZERO" />
+    <ctl name="AMIC MUX6" value="ZERO" />
+    <ctl name="AMIC MUX7" value="ZERO" />
+    <ctl name="AMIC MUX8" value="ZERO" />
+    <ctl name="AMIC MUX10" value="ZERO" />
+    <ctl name="AMIC MUX11" value="ZERO" />
+    <ctl name="AMIC MUX12" value="ZERO" />
+    <ctl name="AMIC MUX13" value="ZERO" />
+
+    <!-- CDC_IF and SLIM controls -->
+    <ctl name="SLIM RX0 MUX" value="ZERO" />
+    <ctl name="SLIM RX1 MUX" value="ZERO" />
+    <ctl name="SLIM RX2 MUX" value="ZERO" />
+    <ctl name="SLIM RX3 MUX" value="ZERO" />
+    <ctl name="SLIM RX4 MUX" value="ZERO" />
+    <ctl name="SLIM RX5 MUX" value="ZERO" />
+    <ctl name="SLIM RX6 MUX" value="ZERO" />
+    <ctl name="SLIM RX7 MUX" value="ZERO" />
+    <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+    <ctl name="CDC_IF RX1 MUX" value="SLIM RX1" />
+    <ctl name="CDC_IF RX2 MUX" value="SLIM RX2" />
+    <ctl name="CDC_IF RX3 MUX" value="SLIM RX3" />
+    <ctl name="CDC_IF RX4 MUX" value="SLIM RX4" />
+    <ctl name="CDC_IF RX5 MUX" value="SLIM RX5" />
+    <ctl name="CDC_IF RX6 MUX" value="SLIM RX6" />
+    <ctl name="CDC_IF RX7 MUX" value="SLIM RX7" />
+    <ctl name="CDC_IF TX0 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX1 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX2 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX3 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX4 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX5 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX6 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX7 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX8 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX9 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX10 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX13 MUX" value="ZERO" />
+
+    <!-- Interpolator chain controls -->
+    <ctl name="RX INT0_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT0_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT0_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT1_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT1_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT1_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT2_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT2_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT2_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT7_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT7_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT8_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT8_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT0_2 MUX" value="ZERO" />
+    <ctl name="RX INT1_2 MUX" value="ZERO" />
+    <ctl name="RX INT2_2 MUX" value="ZERO" />
+    <ctl name="RX INT7_2 MUX" value="ZERO" />
+    <ctl name="RX INT8_2 MUX" value="ZERO" />
+    <ctl name="RX INT1_1 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT2_1 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT1_2 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT2_2 NATIVE MUX" value="OFF" />
+    <ctl name="ASRC0 MUX" value="ZERO" />
+    <ctl name="ASRC1 MUX" value="ZERO" />
+    <ctl name="RX INT1 SEC MIX HPHL Switch" value="0" />
+    <ctl name="RX INT2 SEC MIX HPHR Switch" value="0" />
+    <ctl name="DSD_L IF MUX" value="ZERO" />
+    <ctl name="DSD_R IF MUX" value="ZERO" />
+    <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="0" />
+    <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="0" />
+    <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+    <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+    <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+
+    <!-- Headphone Default mode - uLP -->
+    <ctl name="RX HPH Mode" value="CLS_H_ULP" />
+
+    <!-- IIR/voice anc -->
+    <ctl name="IIR0 Band1" id ="0" value="268435456" />
+    <ctl name="IIR0 Band1" id ="1" value="0" />
+    <ctl name="IIR0 Band1" id ="2" value="0" />
+    <ctl name="IIR0 Band1" id ="3" value="0" />
+    <ctl name="IIR0 Band1" id ="4" value="0" />
+    <ctl name="IIR0 Band2" id ="0" value="268435456" />
+    <ctl name="IIR0 Band2" id ="1" value="0" />
+    <ctl name="IIR0 Band2" id ="2" value="0" />
+    <ctl name="IIR0 Band2" id ="3" value="0" />
+    <ctl name="IIR0 Band2" id ="4" value="0" />
+    <ctl name="IIR0 Band3" id ="0" value="268435456" />
+    <ctl name="IIR0 Band3" id ="1" value="0" />
+    <ctl name="IIR0 Band3" id ="2" value="0" />
+    <ctl name="IIR0 Band3" id ="3" value="0" />
+    <ctl name="IIR0 Band3" id ="4" value="0" />
+    <ctl name="IIR0 Band4" id ="0" value="268435456" />
+    <ctl name="IIR0 Band4" id ="1" value="0" />
+    <ctl name="IIR0 Band4" id ="2" value="0" />
+    <ctl name="IIR0 Band4" id ="3" value="0" />
+    <ctl name="IIR0 Band4" id ="4" value="0" />
+    <ctl name="IIR0 Band5" id ="0" value="268435456" />
+    <ctl name="IIR0 Band5" id ="1" value="0" />
+    <ctl name="IIR0 Band5" id ="2" value="0" />
+    <ctl name="IIR0 Band5" id ="3" value="0" />
+    <ctl name="IIR0 Band5" id ="4" value="0" />
+    <ctl name="IIR0 Enable Band1" value="0" />
+    <ctl name="IIR0 Enable Band2" value="0" />
+    <ctl name="IIR0 Enable Band3" value="0" />
+    <ctl name="IIR0 Enable Band4" value="0" />
+    <ctl name="IIR0 Enable Band5" value="0" />
+    <ctl name="IIR0 INP0 Volume" value="54" />
+    <ctl name="IIR0 INP0 MUX" value="ZERO" />
+    <ctl name="IIR0 INP1 MUX" value="ZERO" />
+    <ctl name="IIR0 INP2 MUX" value="ZERO" />
+    <ctl name="IIR1 INP0 MUX" value="ZERO" />
+    <ctl name="IIR1 INP1 MUX" value="ZERO" />
+    <ctl name="IIR1 INP2 MUX" value="ZERO" />
+
+    <!-- anc related -->
+    <ctl name="ANC Slot" value="0" />
+    <ctl name="ANC Function" value="OFF" />
+    <ctl name="ANC0 FB MUX" value="ZERO" />
+    <ctl name="ANC1 FB MUX" value="ZERO" />
+    <ctl name="ANC OUT EAR Enable Switch" value="0" />
+    <ctl name="ANC OUT EAR SPKR Enable Switch" value="0" />
+    <ctl name="ANC SPKR PA Enable Switch" value="0" />
+
+    <!-- vbat related data -->
+    <!-- vbat related data end -->
+
+    <!-- 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">
+   </path>
+
+    <path name="echo-reference">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
+    </path>
+
+    <path name="echo-reference headphones">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
+    </path>
+
+    <path name="echo-reference headphones-44.1">
+    </path>
+
+    <path name="deep-buffer-playback">
+        <ctl name="SLIMBUS_0_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 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="SLIMBUS_6_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="SLIMBUS_0_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 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="SLIMBUS_6_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="SLIMBUS_0_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="SLIMBUS_6_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 MultiMedia3" 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="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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="1" />
+    </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="SLIMBUS_0_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 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="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 headphones-44.1">
+        <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="1" />
+    </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 SLIM_0_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 SLIM_0_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 SLIM_0_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="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="1" />
+        <ctl name="SLIMBUS_DL_HL Switch" value="1" />
+    </path>
+
+    <path name="play-fm headphones">
+        <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+        <ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="1" />
+        <ctl name="SLIMBUS6_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">
+    </path>
+
+    <path name="hfp-sco headphones">
+    </path>
+
+   <path name="hfp-sco-wb">
+        <path name="hfp-sco" />
+   </path>
+
+    <path name="hfp-sco-wb headphones">
+        <path name="hfp-sco headphones" />
+    </path>
+
+    <path name="compress-voip-call">
+        <ctl name="SLIM_0_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="1" />
+    </path>
+
+    <path name="compress-voip-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer SLIM_0_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 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="voicemmode1-call">
+        <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+    </path>
+
+    <path name="voicemmode1-call hdmi">
+        <ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer SLIM_0_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 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="voicemmode2-call">
+        <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+    </path>
+
+    <path name="voicemmode2-call hdmi">
+        <ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer SLIM_0_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 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="spkr-rx-calib">
+        <ctl name="SLIMBUS_DL_HL Switch"  value="1" />
+    </path>
+
+    <path name="spkr-vi-record">
+    </path>
+
+    <!-- These are actual sound device specific mixer settings -->
+    <path name="amic1">
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1"/>
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="AMIC" />
+        <ctl name="AMIC MUX6" value="ADC1" />
+        <ctl name="IIR0 INP0 MUX" value="DEC6" />
+    </path>
+
+    <path name="amic2">
+        <ctl name="AIF1_CAP Mixer SLIM TX0" value="1"/>
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+        <ctl name="ADC MUX0" value="AMIC" />
+        <ctl name="AMIC MUX0" value="ADC2" />
+        <ctl name="IIR0 INP0 MUX" value="DEC0" />
+    </path>
+
+    <!-- For Tavil, DMIC numbered from 0 to 5 -->
+    <path name="dmic1">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC0" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="dmic2">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="dmic3">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="dmic4">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC3" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="dmic5">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC4" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="dmic6">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC5" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="speaker">
+        <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+        <ctl name="SLIM_0_RX Channels" value="One" />
+        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+        <ctl name="COMP7 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" />
+    </path>
+
+    <path name="speaker-mono">
+        <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+        <ctl name="SLIM_0_RX Channels" value="One" />
+        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+        <ctl name="COMP7 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" />
+    </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">
+       <path name="speaker" />
+   </path>
+
+   <path name="sidetone-iir">
+        <ctl name="IIR0 Enable Band1" value="1" />
+        <ctl name="IIR0 Enable Band2" value="1" />
+        <ctl name="IIR0 Enable Band3" value="1" />
+        <ctl name="IIR0 Enable Band4" value="1" />
+        <ctl name="IIR0 Enable Band5" value="1" />
+    </path>
+
+    <path name="sidetone-headphones">
+        <path name="sidetone-iir" />
+        <ctl name="IIR0 INP0 Volume" value="54" />
+        <ctl name="RX INT1 MIX2 INP" value="SRC0" />
+        <ctl name="RX INT2 MIX2 INP" value="SRC0" />
+    </path>
+
+    <path name="sidetone-handset">
+        <path name="sidetone-iir" />
+        <ctl name="IIR0 INP0 Volume" value="54" />
+        <ctl name="RX INT7 MIX2 INP" value="SRC0" />
+    </path>
+
+    <path name="speaker-mic">
+        <path name="dmic3" />
+    </path>
+
+    <path name="speaker-mic-liquid">
+        <path name="dmic3" />
+    </path>
+
+    <path name="speaker-mic-sbc">
+    </path>
+
+    <path name="speaker-protected">
+        <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+	<ctl name="SLIM_4_TX Format" value="PACKED_16B" />
+        <path name="speaker" />
+        <ctl name="VI_FEED_TX Channels" value="One" />
+        <ctl name="SLIM0_RX_VI_FB_LCH_MUX"  value="SLIM4_TX" />
+    </path>
+
+    <path name="voice-speaker-protected">
+        <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+        <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
+        <path name="speaker-mono" />
+        <ctl name="VI_FEED_TX Channels" value="One" />
+        <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
+    </path>
+
+    <path name="vi-feedback">
+    </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="handset">
+        <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+        <ctl name="SLIM_0_RX Channels" value="One" />
+        <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
+        <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+        <ctl name="EAR PA Gain" value="G_6_DB" />
+    </path>
+
+    <path name="handset-mic">
+        <path name="dmic1" />
+    </path>
+
+    <path name="handset-mic-db">
+    </path>
+
+    <path name="handset-mic-cdp">
+        <path name="amic1" />
+    </path>
+
+    <path name="handset-mic-sbc">
+    </path>
+
+    <path name="three-mic">
+        <ctl name="AIF1_CAP Mixer SLIM TX0" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX1" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX2" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Three" />
+        <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+        <ctl name="ADC MUX0" value="DMIC" />
+        <ctl name="DMIC MUX" value="DMIC0" />
+        <ctl name="CDC_IF TX1 MUX" value="DEC1" />
+        <ctl name="ADC MUX1" value="DMIC" />
+        <ctl name="DMIC MUX1" value="DMIC1" />
+        <ctl name="CDC_IF TX2 MUX" value="DEC2" />
+        <ctl name="ADC MUX2" value="DMIC" />
+        <ctl name="DMIC MUX2" value="DMIC2" />
+    </path>
+
+    <path name="anc-handset">
+        <ctl name="ANC Function" value="ON" />
+        <ctl name="ANC Slot" value="6" />
+        <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+        <ctl name="SLIM_0_RX Channels" value="One" />
+        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+        <ctl name="ANC OUT EAR SPKR Enable Switch" value="1" />
+        <ctl name="ANC SPKR PA Enable Switch" value="1" />
+        <ctl name="SpkrLeft WSA PA Gain" value="G_6_DB" />
+    </path>
+
+    <path name="headphones">
+        <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+        <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+        <ctl name="SLIM_6_RX Channels" value="Two" />
+        <ctl name="RX INT1_2 MUX" value="RX2" />
+        <ctl name="RX INT2_2 MUX" value="RX3" />
+    </path>
+
+    <path name="headphones-44.1">
+        <ctl name="SLIM RX4 MUX" value="AIF3_PB" />
+        <ctl name="SLIM RX5 MUX" value="AIF3_PB" />
+        <ctl name="SLIM_5_RX Channels" value="Two" />
+        <ctl name="CDC_IF RX4 MUX" value="SLIM RX4" />
+        <ctl name="CDC_IF RX5 MUX" value="SLIM RX5" />
+        <ctl name="RX INT1_1 MIX1 INP0" value="RX4" />
+        <ctl name="RX INT2_1 MIX1 INP0" value="RX5" />
+        <ctl name="RX INT1_1 NATIVE MUX" value="ON" />
+        <ctl name="RX INT2_1 NATIVE MUX" value="ON" />
+        <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
+        <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+        <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+    </path>
+
+    <path name="asrc-mode">
+        <ctl name="RX INT1_2 NATIVE MUX" value="ON" />
+        <ctl name="RX INT2_2 NATIVE MUX" value="ON" />
+        <ctl name="ASRC0 MUX" value="ASRC_IN_HPHL" />
+        <ctl name="RX INT1 SEC MIX HPHL Switch" value="1" />
+        <ctl name="ASRC1 MUX" value="ASRC_IN_HPHR" />
+        <ctl name="RX INT2 SEC MIX HPHR Switch" value="1" />
+    </path>
+
+    <path name="headphones-dsd">
+        <ctl name="SLIM RX6 MUX" value="AIF2_PB" />
+        <ctl name="SLIM RX7 MUX" value="AIF2_PB" />
+        <ctl name="SLIM_2_RX Channels" value="Two" />
+        <ctl name="DSD_L IF MUX" value="RX6" />
+        <ctl name="DSD_R IF MUX" value="RX7" />
+        <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="1" />
+        <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="1" />
+        <ctl name="SLIM_2_RX Format" value="DSD_DOP" />
+    </path>
+
+    <path name="hph-highquality-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+    </path>
+
+    <path name="hph-lowpower-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_ULP" />
+    </path>
+
+    <path name="true-native-mode">
+        <ctl name="SLIM RX2 MUX" value="AIF3_PB" />
+        <ctl name="SLIM RX3 MUX" value="AIF3_PB" />
+        <ctl name="CDC_IF RX2 MUX" value="SLIM RX2" />
+        <ctl name="CDC_IF RX3 MUX" value="SLIM RX3" />
+        <ctl name="RX INT1 NATIVE MUX" value="ON" />
+        <ctl name="RX INT2 NATIVE MUX" value="ON" />
+        <ctl name="SLIM_5_RX Channels" value="Two" />
+        <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
+        <ctl name="RX INT2_1 MIX1 INP1" value="RX3" />
+        <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
+        <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+        <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+        <ctl name="COMP1" value="1" />
+        <ctl name="COMP2" value="1" />
+    </path>
+
+    <path name="line">
+        <path name="headphones" />
+    </path>
+
+    <path name="headset-mic">
+        <path name="amic2" />
+    </path>
+
+    <path name="headset-mic-liquid">
+        <path name="amic2" />
+    </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-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-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="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-handset-mic">
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX9" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Three" />
+        <ctl name="AANC_SLIM_0_RX MUX" value="SLIMBUS_0_TX" />
+        <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="DMIC" />
+        <ctl name="DMIC MUX6" value="DMIC2" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
+        <ctl name="CDC_IF TX9 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC0" />
+        <ctl name="IIR0 INP0 MUX" value="DEC6" />
+        <ctl name="ADC MUX10" value="DMIC" />
+        <ctl name="DMIC MUX10" value="DMIC4" />
+        <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
+    </path>
+
+    <!-- Dual MIC devices -->
+    <path name="handset-dmic-endfire">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
+        <ctl name="SLIM_0_TX Channels" value="Two" />
+    </path>
+
+    <path name="speaker-dmic-endfire">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
+        <ctl name="SLIM_0_TX Channels" value="Two" />
+    </path>
+
+    <path name="dmic-endfire">
+        <path name="handset-dmic-endfire" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="dmic-endfire-liquid">
+        <path name="handset-dmic-endfire" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </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="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Two" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC2" />
+    </path>
+
+    <path name="dmic-broadside">
+        <path name="speaker-dmic-broadside" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
+    <path name="voice-speaker-dmic-broadside">
+        <path name="dmic-broadside" />
+    </path>
+
+    <!-- Quad MIC devices -->
+    <path name="speaker-qmic">
+        <ctl name="AIF1_CAP Mixer SLIM TX5" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Four" />
+        <ctl name="CDC_IF TX5 MUX" value="DEC5" />
+        <ctl name="ADC MUX5" value="DMIC" />
+        <ctl name="DMIC MUX5" value="DMIC1" />
+        <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="DMIC" />
+        <ctl name="DMIC MUX6" value="DMIC0" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
+    </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="SLIM RX2 MUX" value="AIF4_PB" />
+        <ctl name="SLIM_6_RX Channels" value="One" />
+        <ctl name="RX INT1_2 MUX" value="RX2" />
+    </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="amic2" />
+    </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="dmic1" />
+    </path>
+
+    <path name="unprocessed-handset-mic">
+        <path name="handset-mic" />
+    </path>
+
+    <path name="unprocessed-mic">
+        <path name="unprocessed-handset-mic" />
+    </path>
+
+    <!-- Added for ADSP testfwk -->
+    <path name="ADSP testfwk">
+        <ctl name="SLIMBUS_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 MultiMedia3" 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/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 860d014..efd275d 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -77,11 +77,10 @@
     <ctl name="DEC8 Volume" value="84" />
     <ctl name="COMP1 Switch" value="1" />
     <ctl name="COMP2 Switch" value="1" />
-    <ctl name="COMP7 Switch" value="1" />
-    <ctl name="COMP8 Switch" value="1" />
+    <ctl name="COMP7 Switch" value="0" />
+    <ctl name="COMP8 Switch" value="0" />
     <ctl name="RX HPH Mode" value="CLS_H_LP" />
     <ctl name="SLIMBUS_3_RX Port Mixer MI2S_TX" value="0" />
-    <ctl name="HDMI_RX Port Mixer MI2S_TX" value="0" />
     <ctl name="SLIMBUS_0_RX Port Mixer SLIM_0_TX" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
@@ -139,6 +138,22 @@
     <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="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
@@ -190,12 +205,6 @@
     <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_RX Channels" value="One" />
-    <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
-    <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
-    <ctl name="USB_AUDIO_TX Channels" value="One" />
-    <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
-    <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
     <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
     <ctl name="IIR0 INP0 MUX" value="ZERO" />
     <ctl name="IIR0 INP1 MUX" value="ZERO" />
@@ -301,8 +310,6 @@
     <ctl name="SPL SRC3 MUX" value="ZERO" />
     <ctl name="RX INT1 SPLINE MIX HPHL Switch" value="0" />
     <ctl name="RX INT3 SPLINE MIX LO1 Switch" value="0" />
-    <ctl name="RX INT1 SPLINE MIX HPHL Native Switch" value="0" />
-    <ctl name="RX INT2 SPLINE MIX HPHR Native Switch" value="0" />
     <ctl name="RX INT2 SPLINE MIX HPHR Switch" value="0" />
     <ctl name="RX INT4 SPLINE MIX LO2 Switch" value="0" />
     <ctl name="RX INT5 SPLINE MIX LO3 Switch" value="0" />
@@ -339,6 +346,7 @@
     <ctl name="SLIM RX4 MUX" value="ZERO" />
     <ctl name="SLIM RX5 MUX" value="ZERO" />
     <ctl name="EAR PA Gain" value="G_6_DB" />
+    <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
     <ctl name="SpkrLeft COMP Switch" value="0" />
     <ctl name="SpkrRight COMP Switch" value="0" />
     <ctl name="SpkrLeft BOOST Switch" value="0" />
@@ -360,6 +368,10 @@
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_6_RX Port Mixer AUX_PCM_UL_TX" value="0" />
     <ctl name="HDMI Mixer MultiMedia4" value="0" />
+    <ctl name= "RX INT1_1 NATIVE MUX" value="OFF" />
+    <ctl name= "RX INT2_1 NATIVE MUX" value="OFF" />
+    <ctl name= "RX INT3_1 NATIVE MUX" value="OFF" />
+    <ctl name= "RX INT4_1 NATIVE MUX" value="OFF" />
     <!-- HFP start -->
     <ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
     <ctl name="SLIMBUS_0_RX Port Mixer SLIM_7_TX" value="0" />
@@ -381,7 +393,7 @@
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
     <!-- usb headset end -->
     <!-- fm -->
-    <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+    <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
     <ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
     <ctl name="SLIMBUS_DL_HL Switch" value="0" />
     <ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
@@ -513,6 +525,9 @@
     <ctl name="ANC0 FB MUX" value="ZERO" />
     <ctl name="ANC1 FB MUX" value="ZERO" />
     <ctl name="ANC EAR Enable Switch" value="0" />
+    <ctl name="ANC OUT EAR SPKR Enable Switch" value="0" />
+    <ctl name="ANC SPKR PA Enable Switch" value="0" />
+    <ctl name="SpkrLeft WSA PA Gain" value="G_0_DB" />
     <!-- anc handset end -->
     <ctl name="ANC Function" value="OFF" />
     <ctl name="ANC HPHL Enable Switch" value="0" />
@@ -548,6 +563,11 @@
     <ctl name="LSM8 MUX" value="None" />
     <ctl name="SLIMBUS_5_TX LSM Function" value="None" />
     <!-- listen 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="SLIMBUS_DL_HL Switch" value="0" />
@@ -604,17 +624,26 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="deep-buffer-playback bt-sco" />
     </path>
 
@@ -652,12 +681,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="low-latency-playback bt-sco" />
     </path>
 
@@ -666,6 +699,11 @@
         <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>
@@ -689,7 +727,7 @@
     </path>
 
     <path name="audio-ull-playback">
-        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-protected">
@@ -697,7 +735,7 @@
     </path>
 
     <path name="audio-ull-playback headphones">
-        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-and-headphones">
@@ -706,15 +744,19 @@
     </path>
 
     <path name="audio-ull-playback hdmi">
-        <ctl name="HDMI Mixer MultiMedia3" value="1" />
+        <ctl name="HDMI Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco">
-        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-ull-playback bt-sco" />
     </path>
 
@@ -723,18 +765,27 @@
         <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 MultiMedia3" value="1" />
+        <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 MultiMedia3" value="1" />
+        <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>
@@ -755,12 +806,20 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback bt-sco" />
     </path>
 
@@ -769,6 +828,11 @@
         <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>
@@ -803,12 +867,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback2 bt-sco" />
     </path>
 
@@ -817,6 +885,11 @@
         <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>
@@ -851,12 +924,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback3 bt-sco" />
     </path>
 
@@ -865,6 +942,11 @@
         <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>
@@ -899,12 +981,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback4 bt-sco" />
     </path>
 
@@ -913,6 +999,12 @@
         <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>
@@ -947,12 +1039,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback5 bt-sco" />
     </path>
 
@@ -961,6 +1057,11 @@
         <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>
@@ -995,12 +1096,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback6 bt-sco" />
     </path>
 
@@ -1009,6 +1114,11 @@
         <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>
@@ -1043,12 +1153,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback7 bt-sco" />
     </path>
 
@@ -1057,6 +1171,11 @@
         <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>
@@ -1091,12 +1210,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback8 bt-sco" />
     </path>
 
@@ -1105,6 +1228,11 @@
         <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>
@@ -1139,12 +1267,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback9 bt-sco" />
     </path>
 
@@ -1153,6 +1285,11 @@
         <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>
@@ -1192,7 +1329,7 @@
     </path>
 
     <path name="audio-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record bt-sco" />
     </path>
 
@@ -1209,7 +1346,7 @@
     </path>
 
     <path name="audio-record-compress bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record-compress bt-sco" />
     </path>
 
@@ -1218,24 +1355,24 @@
     </path>
 
     <path name="low-latency-record">
-      <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco">
-      <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <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="MultiMedia5 Mixer USB_AUDIO_TX" value="1" />
+        <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
     <path name="low-latency-record capture-fm">
-      <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
     </path>
 
     <path name="fm-virtual-record capture-fm">
@@ -1393,12 +1530,12 @@
     </path>
 
    <path name="hfp-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="hfp-sco" />
    </path>
 
     <path name="hfp-sco-wb headphones">
-        <ctl name="AUX PCM SampleRate" value="16000" />
+        <ctl name="AUX PCM SampleRate" value="KHZ_16" />
         <path name="hfp-sco headphones" />
     </path>
 
@@ -1419,7 +1556,7 @@
     </path>
 
     <path name="compress-voip-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-voip-call bt-sco" />
     </path>
 
@@ -1459,7 +1596,7 @@
     </path>
 
     <path name="vowlan-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="vowlan-call bt-sco" />
     </path>
 
@@ -1499,7 +1636,7 @@
     </path>
 
     <path name="voicemmode1-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode1-call bt-sco" />
     </path>
 
@@ -1539,7 +1676,7 @@
     </path>
 
     <path name="voicemmode2-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode2-call bt-sco" />
     </path>
 
@@ -1636,15 +1773,6 @@
     </path>
 
     <!-- For Tasha, DMIC numbered from 0 to 5 -->
-    <path name="dmic3">
-        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
-        <ctl name="SLIM_0_TX Channels" value="One" />
-        <ctl name="SLIM TX7 MUX" value="DEC7" />
-        <ctl name="ADC MUX7" value="DMIC" />
-        <ctl name="DMIC MUX7" value="DMIC2" />
-        <ctl name="IIR0 INP0 MUX" value="DEC7" />
-    </path>
-
     <path name="dmic1">
         <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
         <ctl name="SLIM_0_TX Channels" value="One" />
@@ -1663,6 +1791,15 @@
         <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
+    <path name="dmic3">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="SLIM TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
+    </path>
+
     <path name="dmic4">
         <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
         <ctl name="SLIM_0_TX Channels" value="One" />
@@ -1691,6 +1828,8 @@
     </path>
 
     <path name="speaker">
+        <ctl name="COMP7 Switch" value="1" />
+        <ctl name="COMP8 Switch" value="1" />
         <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
         <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
         <ctl name="SLIM_0_RX Channels" value="Two" />
@@ -1707,6 +1846,7 @@
     </path>
 
     <path name="speaker-fluid">
+        <ctl name="COMP7 Switch" value="1" />
         <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
         <ctl name="SLIM_0_RX Channels" value="One" />
         <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
@@ -1717,6 +1857,7 @@
     </path>
 
     <path name="speaker-mono">
+        <ctl name="COMP7 Switch" value="1" />
         <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
         <ctl name="SLIM_0_RX Channels" value="One" />
         <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
@@ -1763,11 +1904,11 @@
     </path>
 
     <path name="speaker-mic">
-        <path name="dmic3" />
+        <path name="dmic2" />
     </path>
 
     <path name="speaker-mic-liquid">
-        <path name="dmic3" />
+        <path name="dmic2" />
         <ctl name="DEC7 Volume" value="111" />
     </path>
 
@@ -1809,18 +1950,18 @@
     </path>
 
     <path name="handset">
+        <ctl name="COMP7 Switch" value="1" />
         <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
         <ctl name="SLIM_0_RX Channels" value="One" />
         <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
         <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="RX7 Digital Volume" value="76" />
+        <ctl name="EAR SPKR PA Gain" value="G_6_DB" />
     </path>
 
     <path name="handset-mic">
-        <path name="dmic1" />
+        <path name="dmic3" />
     </path>
 
     <path name="handset-mic-db">
@@ -1847,32 +1988,30 @@
         <ctl name="DMIC MUX5" value="DMIC0" />
         <ctl name="SLIM TX6 MUX" value="DEC6" />
         <ctl name="ADC MUX6" value="DMIC" />
-        <ctl name="DMIC MUX6" value="DMIC4" />
+        <ctl name="DMIC MUX6" value="DMIC2" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC3" />
+        <ctl name="DMIC MUX8" value="DMIC5" />
     </path>
 
     <path name="anc-handset">
         <ctl name="ANC Function" value="ON" />
+        <ctl name="ANC Slot" value="6" />
         <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
         <ctl name="SLIM_0_RX Channels" value="One" />
-        <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
-        <ctl name="RX0 Digital Volume" value="81" />
-        <ctl name="ANC Slot" value="6" />
-        <ctl name="ADC MUX10" value="DMIC" />
-        <ctl name="DMIC MUX10" value="DMIC3" />
-        <ctl name="ANC0 FB MUX" value="ANC_IN_EAR" />
-        <ctl name="ANC EAR Enable Switch" value="1" />
+        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+        <ctl name="ANC OUT EAR SPKR Enable Switch" value="1" />
+        <ctl name="ANC SPKR PA Enable Switch" value="1" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+        <ctl name="SpkrLeft WSA PA Gain" value="G_6_DB" />
     </path>
 
     <path name="headphones">
         <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
         <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
         <ctl name="SLIM_6_RX Channels" value="Two" />
-        <ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
-        <ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
+        <ctl name= "RX INT1_2 MUX" value="RX2" />
+        <ctl name= "RX INT2_2 MUX" value="RX3" />
         <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
     </path>
@@ -1898,8 +2037,16 @@
         <ctl name="RX INT2_2 MUX" value="ZERO" />
         <ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
         <ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
-        <ctl name= "RX INT1 SPLINE MIX HPHL Native Switch" value="1" />
-        <ctl name= "RX INT2 SPLINE MIX HPHR Native Switch" value="1" />
+        <ctl name= "RX INT1_1 NATIVE MUX" value="ON" />
+        <ctl name= "RX INT2_1 NATIVE MUX" value="ON" />
+    </path>
+
+    <path name="hph-highquality-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+    </path>
+
+    <path name="hph-lowpower-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_LP" />
     </path>
 
     <path name="line">
@@ -2098,6 +2245,9 @@
     <path name="hdmi">
     </path>
 
+    <path name="display-port">
+    </path>
+
     <path name="speaker-and-usb-headphones">
         <path name="speaker" />
         <path name="usb-headphones" />
@@ -2108,6 +2258,11 @@
         <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>
@@ -2146,14 +2301,17 @@
         <ctl name="AANC_SLIM_0_RX MUX" value="SLIMBUS_0_TX" />
         <ctl name="SLIM TX6 MUX" value="DEC6" />
         <ctl name="ADC MUX6" value="DMIC" />
-        <ctl name="DMIC MUX6" value="DMIC0" />
+        <ctl name="DMIC MUX6" value="DMIC2" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC3" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
         <ctl name="SLIM TX9 MUX" value="DEC7" />
         <ctl name="ADC MUX7" value="DMIC" />
-        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="DMIC MUX7" value="DMIC0" />
         <ctl name="IIR0 INP0 MUX" value="DEC6" />
+        <ctl name="ADC MUX10" value="DMIC" />
+        <ctl name="DMIC MUX10" value="DMIC4" />
+        <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
     </path>
 
     <!-- Dual MIC devices -->
@@ -2162,10 +2320,10 @@
         <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
         <ctl name="SLIM TX7 MUX" value="DEC7" />
         <ctl name="ADC MUX7" value="DMIC" />
-        <ctl name="DMIC MUX7" value="DMIC0" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC3" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
         <ctl name="SLIM_0_TX Channels" value="Two" />
     </path>
 
@@ -2174,10 +2332,10 @@
         <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
         <ctl name="SLIM TX7 MUX" value="DEC7" />
         <ctl name="ADC MUX7" value="DMIC" />
-        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC3" />
+        <ctl name="DMIC MUX8" value="DMIC5" />
         <ctl name="SLIM_0_TX Channels" value="Two" />
     </path>
 
@@ -2249,7 +2407,7 @@
         <ctl name="SLIM_0_TX Channels" value="Two" />
         <ctl name="SLIM TX7 MUX" value="DEC7" />
         <ctl name="ADC MUX7" value="DMIC" />
-        <ctl name="DMIC MUX7" value="DMIC0" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
         <ctl name="DMIC MUX8" value="DMIC2" />
@@ -2273,16 +2431,16 @@
         <ctl name="SLIM_0_TX Channels" value="Four" />
         <ctl name="SLIM TX5 MUX" value="DEC5" />
         <ctl name="ADC MUX5" value="DMIC" />
-        <ctl name="DMIC MUX5" value="DMIC0" />
+        <ctl name="DMIC MUX5" value="DMIC1" />
         <ctl name="SLIM TX6 MUX" value="DEC6" />
         <ctl name="ADC MUX6" value="DMIC" />
-        <ctl name="DMIC MUX6" value="DMIC2" />
+        <ctl name="DMIC MUX6" value="DMIC0" />
         <ctl name="SLIM TX7 MUX" value="DEC7" />
         <ctl name="ADC MUX7" value="DMIC" />
-        <ctl name="DMIC MUX7" value="DMIC1" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
         <ctl name="SLIM TX8 MUX" value="DEC8" />
         <ctl name="ADC MUX8" value="DMIC" />
-        <ctl name="DMIC MUX8" value="DMIC3" />
+        <ctl name="DMIC MUX8" value="DMIC5" />
     </path>
 
     <path name="speaker-qmic-liquid">
@@ -2360,7 +2518,7 @@
 
     <path name="listen-handset-mic">
         <ctl name="MADONOFF Switch" value="1" />
-        <ctl name="MAD Input" value="DMIC0" />
+        <ctl name="MAD Input" value="DMIC2" />
     </path>
 
     <path name="unprocessed-handset-mic">
@@ -2376,4 +2534,122 @@
         <ctl name="SLIMBUS_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 MultiMedia3" 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/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index ca132c0..29212f9 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -50,6 +50,7 @@
     <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" />
@@ -67,6 +68,9 @@
     <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="0" />
     <ctl name="MultiMedia1 Mixer SLIM_4_TX" value="0" />
     <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+    <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="HDMI Mixer MultiMedia1" value="0" />
     <ctl name="HDMI Mixer MultiMedia2" value="0" />
     <ctl name="HDMI Mixer MultiMedia3" value="0" />
@@ -81,6 +85,22 @@
     <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="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
@@ -88,33 +108,46 @@
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia5" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="0" />
     <ctl name="SLIMBUS_6_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" />
@@ -122,6 +155,7 @@
     <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" />
@@ -132,19 +166,19 @@
     <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_RX Channels" value="One" />
-    <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
-    <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
-    <ctl name="USB_AUDIO_TX Channels" value="One" />
-    <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
-    <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
     <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
+    <ctl name="SLIM_2_RX Format" value="UNPACKED" />
+    <ctl name="SLIM_2_RX SampleRate" value="KHZ_48" />
+    <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
     <ctl name="SLIM_0_RX Channels" value="One" />
     <ctl name="SLIM_5_RX Channels" value="One" />
+    <ctl name="SLIM_6_RX Channels" value="One" />
+    <ctl name="SLIM_2_RX Channels" value="One" />
     <ctl name="SLIM_0_TX Channels" value="One" />
     <ctl name="SLIM_1_TX Channels" value="One" />
-    <ctl name="AIF1_CAP Mixer SLIM TX7" value="0" />
+    <ctl name="AIF1_CAP Mixer SLIM TX9" value="0"/>
     <ctl name="AIF1_CAP Mixer SLIM TX8" value="0"/>
+    <ctl name="AIF1_CAP Mixer SLIM TX7" value="0" />
     <ctl name="AIF1_CAP Mixer SLIM TX6" value="0" />
     <ctl name="AIF1_CAP Mixer SLIM TX5" value="0"/>
     <ctl name="AIF1_CAP Mixer SLIM TX4" value="0" />
@@ -154,6 +188,13 @@
     <ctl name="AIF1_CAP Mixer SLIM TX0" value="0"/>
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="HDMI Mixer MultiMedia4" value="0" />
+    <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="ZERO" />
+    <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="ZERO" />
+    <ctl name="VI_FEED_TX Channels" value="Two" />
+    <ctl name="AIF4_VI Mixer SPKR_VI_1" value="0" />
+    <ctl name="AIF4_VI Mixer SPKR_VI_2" value="0" />
+    <ctl name="SLIM_4_TX Format" value="UNPACKED" />
+    <ctl name="AANC_SLIM_0_RX MUX" value="ZERO" />
     <!-- HFP start -->
     <ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
     <ctl name="SLIMBUS_0_RX Port Mixer SLIM_7_TX" value="0" />
@@ -175,7 +216,7 @@
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
     <!-- usb headset end -->
     <!-- fm -->
-    <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+    <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
     <ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
     <ctl name="SLIMBUS_DL_HL Switch" value="0" />
     <ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
@@ -262,6 +303,12 @@
     <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="SLIMBUS_DL_HL Switch" value="0" />
     <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
@@ -270,14 +317,7 @@
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
 
     <!-- Codec controls -->
-    <ctl name="SLIM RX0 MUX" value="ZERO" />
-    <ctl name="SLIM RX1 MUX" value="ZERO" />
-    <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
-    <ctl name="CDC_IF RX1 MUX" value="SLIM RX1" />
-    <ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
-    <ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
-    <ctl name="COMP7 Switch" value="0" />
-    <ctl name="COMP8 Switch" value="0" />
+    <!-- WSA controls -->
     <ctl name="SpkrLeft COMP Switch" value="0" />
     <ctl name="SpkrRight COMP Switch" value="0" />
     <ctl name="SpkrLeft BOOST Switch" value="0" />
@@ -286,22 +326,205 @@
     <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="AIF1_CAP Mixer SLIM TX0" value="0" />
-    <ctl name="CDC_IF TX0 MUX" value="ZERO" />
-    <ctl name="ADC MUX0" value="ZERO" />
+    <!-- Volume controls -->
+    <ctl name="LINEOUT1 Volume" value="13" />
+    <ctl name="LINEOUT2 Volume" value="13" />
+    <ctl name="HPHL Volume" value="20" />
+    <ctl name="HPHR Volume" value="20" />
+    <ctl name="EAR PA Gain" value="G_6_DB" />
+    <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
+
+    <ctl name="RX0 Digital Volume" value="84" />
+    <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="RX6 Digital Volume" value="84" />
+    <ctl name="RX7 Digital Volume" value="84" />
+    <ctl name="ADC1 Volume" value="12" />
+    <ctl name="ADC2 Volume" value="12" />
+    <ctl name="ADC3 Volume" value="12" />
+    <ctl name="ADC4 Volume" value="12" />
+    <ctl name="DEC0 Volume" value="84" />
+    <ctl name="DEC1 Volume" value="84" />
+    <ctl name="DEC2 Volume" value="84" />
+    <ctl name="DEC3 Volume" value="84" />
+    <ctl name="DEC4 Volume" value="84" />
+    <ctl name="DEC5 Volume" value="84" />
+    <ctl name="DEC6 Volume" value="84" />
+    <ctl name="DEC7 Volume" value="84" />
+    <ctl name="DEC8 Volume" value="84" />
+
+    <!-- Compander controls -->
+    <ctl name="COMP1 Switch" value="1" />
+    <ctl name="COMP2 Switch" value="1" />
+    <ctl name="COMP7 Switch" value="0" />
+    <ctl name="COMP8 Switch" value="0" />
+
+    <!-- ADC, DMIC and AMIC controls -->
+    <ctl name="ADC MUX0" value="AMIC" />
+    <ctl name="ADC MUX1" value="AMIC" />
+    <ctl name="ADC MUX2" value="AMIC" />
+    <ctl name="ADC MUX3" value="AMIC" />
+    <ctl name="ADC MUX4" value="AMIC" />
+    <ctl name="ADC MUX5" value="AMIC" />
+    <ctl name="ADC MUX6" value="AMIC" />
+    <ctl name="ADC MUX7" value="AMIC" />
+    <ctl name="ADC MUX8" value="AMIC" />
+    <ctl name="ADC MUX10" value="AMIC" />
+    <ctl name="ADC MUX11" value="AMIC" />
+    <ctl name="ADC MUX12" value="AMIC" />
+    <ctl name="ADC MUX13" value="AMIC" />
     <ctl name="DMIC MUX0" value="ZERO" />
-    <ctl name="DEC0 Volume" value="0" />
+    <ctl name="DMIC MUX1" value="ZERO" />
+    <ctl name="DMIC MUX2" value="ZERO" />
+    <ctl name="DMIC MUX3" value="ZERO" />
+    <ctl name="DMIC MUX4" value="ZERO" />
+    <ctl name="DMIC MUX5" value="ZERO" />
+    <ctl name="DMIC MUX6" value="ZERO" />
+    <ctl name="DMIC MUX7" value="ZERO" />
+    <ctl name="DMIC MUX8" value="ZERO" />
+    <ctl name="DMIC MUX10" value="ZERO" />
+    <ctl name="DMIC MUX11" value="ZERO" />
+    <ctl name="DMIC MUX12" value="ZERO" />
+    <ctl name="DMIC MUX13" value="ZERO" />
+    <ctl name="AMIC MUX0" value="ZERO" />
+    <ctl name="AMIC MUX1" value="ZERO" />
+    <ctl name="AMIC MUX2" value="ZERO" />
+    <ctl name="AMIC MUX3" value="ZERO" />
+    <ctl name="AMIC MUX4" value="ZERO" />
+    <ctl name="AMIC MUX5" value="ZERO" />
+    <ctl name="AMIC MUX6" value="ZERO" />
+    <ctl name="AMIC MUX7" value="ZERO" />
+    <ctl name="AMIC MUX8" value="ZERO" />
+    <ctl name="AMIC MUX10" value="ZERO" />
+    <ctl name="AMIC MUX11" value="ZERO" />
+    <ctl name="AMIC MUX12" value="ZERO" />
+    <ctl name="AMIC MUX13" value="ZERO" />
+
+    <!-- CDC_IF and SLIM controls -->
+    <ctl name="SLIM RX0 MUX" value="ZERO" />
+    <ctl name="SLIM RX1 MUX" value="ZERO" />
+    <ctl name="SLIM RX2 MUX" value="ZERO" />
+    <ctl name="SLIM RX3 MUX" value="ZERO" />
+    <ctl name="SLIM RX4 MUX" value="ZERO" />
+    <ctl name="SLIM RX5 MUX" value="ZERO" />
+    <ctl name="SLIM RX6 MUX" value="ZERO" />
+    <ctl name="SLIM RX7 MUX" value="ZERO" />
+    <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+    <ctl name="CDC_IF RX1 MUX" value="SLIM RX1" />
+    <ctl name="CDC_IF RX2 MUX" value="SLIM RX2" />
+    <ctl name="CDC_IF RX3 MUX" value="SLIM RX3" />
+    <ctl name="CDC_IF RX4 MUX" value="SLIM RX4" />
+    <ctl name="CDC_IF RX5 MUX" value="SLIM RX5" />
+    <ctl name="CDC_IF RX6 MUX" value="SLIM RX6" />
+    <ctl name="CDC_IF RX7 MUX" value="SLIM RX7" />
+    <ctl name="CDC_IF TX0 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX1 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX2 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX3 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX4 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX5 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX6 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX7 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX8 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX9 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX10 MUX" value="ZERO" />
+    <ctl name="CDC_IF TX13 MUX" value="ZERO" />
+
+    <!-- Interpolator chain controls -->
+    <ctl name="RX INT0_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT0_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT0_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT1_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT1_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT1_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT2_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT2_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT2_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT7_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT7_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT8_1 MIX1 INP1" value="ZERO" />
+    <ctl name="RX INT8_1 MIX1 INP2" value="ZERO" />
+    <ctl name="RX INT0_2 MUX" value="ZERO" />
+    <ctl name="RX INT1_2 MUX" value="ZERO" />
+    <ctl name="RX INT2_2 MUX" value="ZERO" />
+    <ctl name="RX INT7_2 MUX" value="ZERO" />
+    <ctl name="RX INT8_2 MUX" value="ZERO" />
+    <ctl name="RX INT1_1 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT2_1 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT1_2 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT2_2 NATIVE MUX" value="OFF" />
+    <ctl name="ASRC0 MUX" value="ZERO" />
+    <ctl name="ASRC1 MUX" value="ZERO" />
+    <ctl name="RX INT1 SEC MIX HPHL Switch" value="0" />
+    <ctl name="RX INT2 SEC MIX HPHR Switch" value="0" />
+    <ctl name="DSD_L IF MUX" value="ZERO" />
+    <ctl name="DSD_R IF MUX" value="ZERO" />
+    <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="0" />
+    <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="0" />
+    <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+    <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+    <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+
+    <!-- Headphone Default mode - uLP -->
+    <ctl name="RX HPH Mode" value="CLS_H_ULP" />
+    <ctl name="ASRC0 Output Mode" value="INT" />
+    <ctl name="ASRC1 Output Mode" value="INT" />
 
     <!-- IIR/voice anc -->
-    <!-- IIR/voice anc end -->
-    <!-- anc handset -->
-    <!-- anc handset end -->
-    <!-- anc headset end -->
-    <!-- aanc handset mic -->
-    <!-- aanc handset mic end -->
-    <!-- ssr qmic -->
-    <!-- ssr qmic end-->
+    <ctl name="IIR0 Band1" id ="0" value="268435456" />
+    <ctl name="IIR0 Band1" id ="1" value="0" />
+    <ctl name="IIR0 Band1" id ="2" value="0" />
+    <ctl name="IIR0 Band1" id ="3" value="0" />
+    <ctl name="IIR0 Band1" id ="4" value="0" />
+    <ctl name="IIR0 Band2" id ="0" value="268435456" />
+    <ctl name="IIR0 Band2" id ="1" value="0" />
+    <ctl name="IIR0 Band2" id ="2" value="0" />
+    <ctl name="IIR0 Band2" id ="3" value="0" />
+    <ctl name="IIR0 Band2" id ="4" value="0" />
+    <ctl name="IIR0 Band3" id ="0" value="268435456" />
+    <ctl name="IIR0 Band3" id ="1" value="0" />
+    <ctl name="IIR0 Band3" id ="2" value="0" />
+    <ctl name="IIR0 Band3" id ="3" value="0" />
+    <ctl name="IIR0 Band3" id ="4" value="0" />
+    <ctl name="IIR0 Band4" id ="0" value="268435456" />
+    <ctl name="IIR0 Band4" id ="1" value="0" />
+    <ctl name="IIR0 Band4" id ="2" value="0" />
+    <ctl name="IIR0 Band4" id ="3" value="0" />
+    <ctl name="IIR0 Band4" id ="4" value="0" />
+    <ctl name="IIR0 Band5" id ="0" value="268435456" />
+    <ctl name="IIR0 Band5" id ="1" value="0" />
+    <ctl name="IIR0 Band5" id ="2" value="0" />
+    <ctl name="IIR0 Band5" id ="3" value="0" />
+    <ctl name="IIR0 Band5" id ="4" value="0" />
+    <ctl name="IIR0 Enable Band1" value="0" />
+    <ctl name="IIR0 Enable Band2" value="0" />
+    <ctl name="IIR0 Enable Band3" value="0" />
+    <ctl name="IIR0 Enable Band4" value="0" />
+    <ctl name="IIR0 Enable Band5" value="0" />
+    <ctl name="IIR0 INP0 Volume" value="54" />
+    <ctl name="IIR0 INP0 MUX" value="ZERO" />
+    <ctl name="IIR0 INP1 MUX" value="ZERO" />
+    <ctl name="IIR0 INP2 MUX" value="ZERO" />
+    <ctl name="IIR1 INP0 MUX" value="ZERO" />
+    <ctl name="IIR1 INP1 MUX" value="ZERO" />
+    <ctl name="IIR1 INP2 MUX" value="ZERO" />
+
+    <!-- anc related -->
+    <ctl name="ANC Slot" value="0" />
+    <ctl name="ANC Function" value="OFF" />
+    <ctl name="ANC0 FB MUX" value="ZERO" />
+    <ctl name="ANC1 FB MUX" value="ZERO" />
+    <ctl name="ANC OUT EAR Enable Switch" value="0" />
+    <ctl name="ANC OUT EAR SPKR Enable Switch" value="0" />
+    <ctl name="ANC SPKR PA Enable Switch" value="0" />
+
     <!-- vbat related data -->
     <!-- vbat related data end -->
 
@@ -319,9 +542,11 @@
    </path>
 
     <path name="echo-reference">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
     </path>
 
     <path name="echo-reference headphones">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
     </path>
 
     <path name="echo-reference headphones-44.1">
@@ -339,17 +564,26 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="deep-buffer-playback bt-sco" />
     </path>
 
@@ -387,12 +621,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="low-latency-playback bt-sco" />
     </path>
 
@@ -401,6 +639,11 @@
         <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>
@@ -424,7 +667,7 @@
     </path>
 
     <path name="audio-ull-playback">
-        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-protected">
@@ -432,7 +675,7 @@
     </path>
 
     <path name="audio-ull-playback headphones">
-        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-and-headphones">
@@ -441,15 +684,19 @@
     </path>
 
     <path name="audio-ull-playback hdmi">
-        <ctl name="HDMI Mixer MultiMedia3" value="1" />
+        <ctl name="HDMI Mixer MultiMedia8" value="1" />
+    </path>
+
+    <path name="audio-ull-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco">
-        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-ull-playback bt-sco" />
     </path>
 
@@ -458,18 +705,27 @@
         <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 MultiMedia3" value="1" />
+        <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 MultiMedia3" value="1" />
+        <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>
@@ -490,12 +746,20 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback bt-sco" />
     </path>
 
@@ -504,6 +768,11 @@
         <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>
@@ -525,6 +794,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="1" />
     </path>
 
+    <path name="compress-offload-playback headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
     <path name="compress-offload-playback speaker-and-headphones">
         <path name="compress-offload-playback headphones" />
         <path name="compress-offload-playback" />
@@ -538,12 +811,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback2 bt-sco" />
     </path>
 
@@ -552,6 +829,11 @@
         <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>
@@ -573,6 +855,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="1" />
     </path>
 
+    <path name="compress-offload-playback2 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
     <path name="compress-offload-playback2 speaker-and-headphones">
         <path name="compress-offload-playback2 headphones" />
         <path name="compress-offload-playback2" />
@@ -586,12 +872,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback3 bt-sco" />
     </path>
 
@@ -600,6 +890,11 @@
         <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>
@@ -621,6 +916,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="compress-offload-playback3 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
     <path name="compress-offload-playback3 speaker-and-headphones">
         <path name="compress-offload-playback3 headphones" />
         <path name="compress-offload-playback3" />
@@ -634,12 +933,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback4 bt-sco" />
     </path>
 
@@ -648,6 +951,12 @@
         <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>
@@ -669,6 +978,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="1" />
     </path>
 
+    <path name="compress-offload-playback4 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
     <path name="compress-offload-playback4 speaker-and-headphones">
         <path name="compress-offload-playback4 headphones" />
         <path name="compress-offload-playback4" />
@@ -682,12 +995,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback5 bt-sco" />
     </path>
 
@@ -696,6 +1013,11 @@
         <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>
@@ -717,6 +1039,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="1" />
     </path>
 
+    <path name="compress-offload-playback5 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
     <path name="compress-offload-playback5 speaker-and-headphones">
         <path name="compress-offload-playback5 headphones" />
         <path name="compress-offload-playback5" />
@@ -730,12 +1056,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback6 bt-sco" />
     </path>
 
@@ -744,6 +1074,11 @@
         <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>
@@ -765,6 +1100,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="1" />
     </path>
 
+    <path name="compress-offload-playback6 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
     <path name="compress-offload-playback6 speaker-and-headphones">
         <path name="compress-offload-playback6 headphones" />
         <path name="compress-offload-playback6" />
@@ -778,12 +1117,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback7 bt-sco" />
     </path>
 
@@ -792,6 +1135,11 @@
         <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>
@@ -813,6 +1161,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="1" />
     </path>
 
+    <path name="compress-offload-playback7 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
     <path name="compress-offload-playback7 speaker-and-headphones">
         <path name="compress-offload-playback7 headphones" />
         <path name="compress-offload-playback7" />
@@ -826,12 +1178,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback8 bt-sco" />
     </path>
 
@@ -840,6 +1196,11 @@
         <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>
@@ -861,6 +1222,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="1" />
     </path>
 
+    <path name="compress-offload-playback8 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
     <path name="compress-offload-playback8 speaker-and-headphones">
         <path name="compress-offload-playback8 headphones" />
         <path name="compress-offload-playback8" />
@@ -874,12 +1239,16 @@
         <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_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback9 bt-sco" />
     </path>
 
@@ -888,6 +1257,11 @@
         <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>
@@ -909,6 +1283,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="compress-offload-playback9 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
     <path name="compress-offload-playback9 speaker-and-headphones">
         <path name="compress-offload-playback9 headphones" />
         <path name="compress-offload-playback9" />
@@ -927,7 +1305,7 @@
     </path>
 
     <path name="audio-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record bt-sco" />
     </path>
 
@@ -944,7 +1322,7 @@
     </path>
 
     <path name="audio-record-compress bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record-compress bt-sco" />
     </path>
 
@@ -953,24 +1331,24 @@
     </path>
 
     <path name="low-latency-record">
-      <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco">
-      <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <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="MultiMedia5 Mixer USB_AUDIO_TX" value="1" />
+        <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
     <path name="low-latency-record capture-fm">
-      <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
     </path>
 
     <path name="fm-virtual-record capture-fm">
@@ -1142,7 +1520,7 @@
     </path>
 
     <path name="compress-voip-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-voip-call bt-sco" />
     </path>
 
@@ -1182,7 +1560,7 @@
     </path>
 
     <path name="voicemmode1-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode1-call bt-sco" />
     </path>
 
@@ -1222,7 +1600,7 @@
     </path>
 
     <path name="voicemmode2-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode2-call bt-sco" />
     </path>
 
@@ -1250,34 +1628,76 @@
 
     <!-- These are actual sound device specific mixer settings -->
     <path name="amic1">
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1"/>
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="AMIC" />
+        <ctl name="AMIC MUX6" value="ADC1" />
+        <ctl name="IIR0 INP0 MUX" value="DEC6" />
     </path>
 
     <path name="amic2">
+        <ctl name="AIF1_CAP Mixer SLIM TX0" value="1"/>
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+        <ctl name="ADC MUX0" value="AMIC" />
+        <ctl name="AMIC MUX0" value="ADC2" />
+        <ctl name="IIR0 INP0 MUX" value="DEC0" />
     </path>
 
     <!-- For Tavil, DMIC numbered from 0 to 5 -->
     <path name="dmic1">
-        <ctl name="AIF1_CAP Mixer SLIM TX0" value="1" />
-        <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
         <ctl name="SLIM_0_TX Channels" value="One" />
-        <ctl name="ADC MUX0" value="DMIC" />
-        <ctl name="DMIC MUX0" value="DMIC0" />
-        <ctl name="DEC0 Volume" value="84" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC0" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="dmic2">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="dmic3">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="dmic4">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC3" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="dmic5">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC4" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="dmic6">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="SLIM_0_TX Channels" value="One" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC5" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="speaker">
@@ -1300,10 +1720,20 @@
         <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
     </path>
 
-    <path name="speaker-fluid">
+    <path name="speaker-mono">
+        <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+        <ctl name="SLIM_0_RX Channels" value="One" />
+        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+        <ctl name="COMP7 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" />
     </path>
 
-    <path name="speaker-mono">
+    <path name="speaker-fluid">
+        <path name="speaker-mono" />
     </path>
 
     <path name="speaker-liquid">
@@ -1319,33 +1749,53 @@
    </path>
 
    <path name="sidetone-iir">
-   </path>
+        <ctl name="IIR0 Enable Band1" value="1" />
+        <ctl name="IIR0 Enable Band2" value="1" />
+        <ctl name="IIR0 Enable Band3" value="1" />
+        <ctl name="IIR0 Enable Band4" value="1" />
+        <ctl name="IIR0 Enable Band5" value="1" />
+    </path>
 
     <path name="sidetone-headphones">
         <path name="sidetone-iir" />
+        <ctl name="IIR0 INP0 Volume" value="54" />
+        <ctl name="RX INT1 MIX2 INP" value="SRC0" />
+        <ctl name="RX INT2 MIX2 INP" value="SRC0" />
     </path>
 
     <path name="sidetone-handset">
         <path name="sidetone-iir" />
+        <ctl name="IIR0 INP0 Volume" value="54" />
+        <ctl name="RX INT7 MIX2 INP" value="SRC0" />
     </path>
 
     <path name="speaker-mic">
-        <path name="dmic3" />
+        <path name="dmic2" />
     </path>
 
     <path name="speaker-mic-liquid">
-        <path name="dmic3" />
+        <path name="dmic2" />
     </path>
 
     <path name="speaker-mic-sbc">
     </path>
 
     <path name="speaker-protected">
+        <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+        <ctl name="AIF4_VI Mixer SPKR_VI_2" value="1" />
+        <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
         <path name="speaker" />
+        <ctl name="VI_FEED_TX Channels" value="Two" />
+        <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
+        <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="SLIM4_TX" />
     </path>
 
     <path name="voice-speaker-protected">
+        <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+        <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
         <path name="speaker-mono" />
+        <ctl name="VI_FEED_TX Channels" value="One" />
+        <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
     </path>
 
     <path name="vi-feedback">
@@ -1360,10 +1810,19 @@
     </path>
 
     <path name="handset">
+        <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+        <ctl name="SLIM_0_RX Channels" value="One" />
+        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+        <ctl name="COMP7 Switch" value="1" />
+        <ctl name="SpkrLeft COMP Switch" value="1" />
+        <ctl name="SpkrLeft BOOST Switch" value="1" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+        <ctl name="EAR SPKR PA Gain" value="G_6_DB" />
     </path>
 
     <path name="handset-mic">
-        <path name="dmic1" />
+        <path name="dmic3" />
     </path>
 
     <path name="handset-mic-db">
@@ -1377,15 +1836,80 @@
     </path>
 
     <path name="three-mic">
+        <ctl name="AIF1_CAP Mixer SLIM TX5" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Three" />
+        <ctl name="CDC_IF TX5 MUX" value="DEC5" />
+        <ctl name="ADC MUX5" value="DMIC" />
+        <ctl name="DMIC MUX5" value="DMIC0" />
+        <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="DMIC" />
+        <ctl name="DMIC MUX6" value="DMIC2" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC5" />
     </path>
 
     <path name="anc-handset">
+        <ctl name="ANC Function" value="ON" />
+        <ctl name="ANC Slot" value="6" />
+        <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+        <ctl name="SLIM_0_RX Channels" value="One" />
+        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+        <ctl name="ANC OUT EAR SPKR Enable Switch" value="1" />
+        <ctl name="ANC SPKR PA Enable Switch" value="1" />
+        <ctl name="SpkrLeft WSA PA Gain" value="G_6_DB" />
     </path>
 
     <path name="headphones">
+        <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+        <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+        <ctl name="SLIM_6_RX Channels" value="Two" />
+        <ctl name="RX INT1_2 MUX" value="RX2" />
+        <ctl name="RX INT2_2 MUX" value="RX3" />
     </path>
 
     <path name="headphones-44.1">
+        <ctl name="SLIM RX4 MUX" value="AIF3_PB" />
+        <ctl name="SLIM RX5 MUX" value="AIF3_PB" />
+        <ctl name="SLIM_5_RX Channels" value="Two" />
+        <ctl name="RX INT1_1 MIX1 INP0" value="RX4" />
+        <ctl name="RX INT2_1 MIX1 INP0" value="RX5" />
+        <ctl name="RX INT1_1 NATIVE MUX" value="ON" />
+        <ctl name="RX INT2_1 NATIVE MUX" value="ON" />
+    </path>
+
+    <path name="asrc-mode">
+        <ctl name="ASRC0 Output Mode" value="FRAC" />
+        <ctl name="ASRC1 Output Mode" value="FRAC" />
+        <ctl name="RX INT1_2 NATIVE MUX" value="ON" />
+        <ctl name="RX INT2_2 NATIVE MUX" value="ON" />
+        <ctl name="ASRC0 MUX" value="ASRC_IN_HPHL" />
+        <ctl name="RX INT1 SEC MIX HPHL Switch" value="1" />
+        <ctl name="ASRC1 MUX" value="ASRC_IN_HPHR" />
+        <ctl name="RX INT2 SEC MIX HPHR Switch" value="1" />
+    </path>
+
+    <path name="headphones-dsd">
+        <ctl name="SLIM RX6 MUX" value="AIF2_PB" />
+        <ctl name="SLIM RX7 MUX" value="AIF2_PB" />
+        <ctl name="SLIM_2_RX Channels" value="Two" />
+        <ctl name="DSD_L IF MUX" value="RX6" />
+        <ctl name="DSD_R IF MUX" value="RX7" />
+        <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="1" />
+        <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="1" />
+        <ctl name="SLIM_2_RX Format" value="DSD_DOP" />
+    </path>
+
+    <path name="hph-highquality-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+    </path>
+
+    <path name="hph-lowpower-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_ULP" />
     </path>
 
     <path name="true-native-mode">
@@ -1464,6 +1988,7 @@
     </path>
 
     <path name="anc-headphones">
+        <path name="headphones" />
     </path>
 
     <path name="speaker-and-anc-headphones">
@@ -1481,9 +2006,11 @@
     </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">
@@ -1493,6 +2020,9 @@
     <path name="hdmi">
     </path>
 
+    <path name="display-port">
+    </path>
+
     <path name="speaker-and-usb-headphones">
         <path name="speaker" />
         <path name="usb-headphones" />
@@ -1503,6 +2033,11 @@
         <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>
@@ -1534,21 +2069,59 @@
     </path>
 
     <path name="aanc-handset-mic">
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX9" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Three" />
+        <ctl name="AANC_SLIM_0_RX MUX" value="SLIMBUS_0_TX" />
+        <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="DMIC" />
+        <ctl name="DMIC MUX6" value="DMIC2" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
+        <ctl name="CDC_IF TX9 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC0" />
+        <ctl name="IIR0 INP0 MUX" value="DEC6" />
+        <ctl name="ADC MUX10" value="DMIC" />
+        <ctl name="DMIC MUX10" value="DMIC4" />
+        <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
     </path>
 
     <!-- Dual MIC devices -->
     <path name="handset-dmic-endfire">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC4" />
+        <ctl name="SLIM_0_TX Channels" value="Two" />
     </path>
 
     <path name="speaker-dmic-endfire">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC5" />
+        <ctl name="SLIM_0_TX Channels" value="Two" />
     </path>
 
     <path name="dmic-endfire">
         <path name="handset-dmic-endfire" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="dmic-endfire-liquid">
         <path name="handset-dmic-endfire" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="handset-stereo-dmic-ef">
@@ -1604,10 +2177,20 @@
     </path>
 
     <path name="speaker-dmic-broadside">
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Two" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC1" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC2" />
     </path>
 
     <path name="dmic-broadside">
         <path name="speaker-dmic-broadside" />
+        <ctl name="IIR0 INP0 MUX" value="DEC7" />
     </path>
 
     <path name="voice-speaker-dmic-broadside">
@@ -1616,6 +2199,23 @@
 
     <!-- Quad MIC devices -->
     <path name="speaker-qmic">
+        <ctl name="AIF1_CAP Mixer SLIM TX5" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+        <ctl name="SLIM_0_TX Channels" value="Four" />
+        <ctl name="CDC_IF TX5 MUX" value="DEC5" />
+        <ctl name="ADC MUX5" value="DMIC" />
+        <ctl name="DMIC MUX5" value="DMIC1" />
+        <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="DMIC" />
+        <ctl name="DMIC MUX6" value="DMIC0" />
+        <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+        <ctl name="ADC MUX7" value="DMIC" />
+        <ctl name="DMIC MUX7" value="DMIC2" />
+        <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+        <ctl name="ADC MUX8" value="DMIC" />
+        <ctl name="DMIC MUX8" value="DMIC5" />
     </path>
 
     <path name="speaker-qmic-liquid">
@@ -1640,6 +2240,9 @@
     <!-- TTY devices -->
 
     <path name="tty-headphones">
+        <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+        <ctl name="SLIM_6_RX Channels" value="One" />
+        <ctl name="RX INT1_2 MUX" value="RX2" />
     </path>
 
     <path name="voice-tty-full-headphones">
@@ -1682,4 +2285,122 @@
         <ctl name="SLIMBUS_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 MultiMedia3" 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/msmcobalt/msmcobalt.mk b/configs/msmcobalt/msmcobalt.mk
index 4ba276d..9aa0322 100644
--- a/configs/msmcobalt/msmcobalt.mk
+++ b/configs/msmcobalt/msmcobalt.mk
@@ -3,8 +3,8 @@
 #AUDIO_FEATURE_FLAGS
 BOARD_USES_ALSA_AUDIO := true
 USE_CUSTOM_AUDIO_POLICY := 1
-USE_XML_AUDIO_POLICY_CONF := 0
-BOARD_SUPPORTS_SOUND_TRIGGER := true
+USE_XML_AUDIO_POLICY_CONF := 1
+BOARD_SUPPORTS_SOUND_TRIGGER_HAL := true
 AUDIO_USE_LL_AS_PRIMARY_OUTPUT := true
 
 AUDIO_FEATURE_ENABLED_VBAT_MONITOR := true
@@ -19,8 +19,9 @@
 AUDIO_FEATURE_ENABLED_FLUENCE := true
 AUDIO_FEATURE_ENABLED_HDMI_SPK := true
 AUDIO_FEATURE_ENABLED_HDMI_EDID := true
-#AUDIO_FEATURE_ENABLED_HDMI_PASSTHROUGH := true
+AUDIO_FEATURE_ENABLED_HDMI_PASSTHROUGH := true
 #AUDIO_FEATURE_ENABLED_KEEP_ALIVE := true
+AUDIO_FEATURE_ENABLED_DISPLAY_PORT := true
 #AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
 #DOLBY_DDP := true
 AUDIO_FEATURE_ENABLED_HFP := true
@@ -34,7 +35,7 @@
 AUDIO_FEATURE_ENABLED_ALAC_OFFLOAD := true
 AUDIO_FEATURE_ENABLED_APE_OFFLOAD := true
 AUDIO_FEATURE_ENABLED_AAC_ADTS_OFFLOAD := true
-#AUDIO_FEATURE_ENABLED_PROXY_DEVICE := true
+AUDIO_FEATURE_ENABLED_PROXY_DEVICE := true
 AUDIO_FEATURE_ENABLED_KPI_OPTIMIZE := true
 AUDIO_FEATURE_ENABLED_SPKR_PROTECTION := true
 AUDIO_FEATURE_ENABLED_SSR := true
@@ -43,7 +44,7 @@
 AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
 BOARD_USES_SRS_TRUEMEDIA := false
 DTS_CODEC_M_ := false
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
 MM_AUDIO_ENABLED_FTM := true
 MM_AUDIO_ENABLED_SAFX := true
 TARGET_USES_QCOM_MM_AUDIO := true
@@ -51,7 +52,9 @@
 
 AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
 AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
+AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
 AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
+AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
 ##AUDIO_FEATURE_FLAGS
 
 #Audio Specific device overlays
@@ -72,13 +75,18 @@
     hardware/qcom/audio/configs/msmcobalt/mixer_paths.xml:system/etc/mixer_paths.xml \
     hardware/qcom/audio/configs/msmcobalt/mixer_paths_tasha.xml:system/etc/mixer_paths_tasha.xml \
     hardware/qcom/audio/configs/msmcobalt/mixer_paths_tavil.xml:system/etc/mixer_paths_tavil.xml \
+    hardware/qcom/audio/configs/msmcobalt/mixer_paths_skuk.xml:system/etc/mixer_paths_skuk.xml \
+    hardware/qcom/audio/configs/msmcobalt/mixer_paths_skuk.xml:system/etc/mixer_paths_qvr.xml \
     hardware/qcom/audio/configs/msmcobalt/mixer_paths_dtp.xml:system/etc/mixer_paths_dtp.xml \
     hardware/qcom/audio/configs/msmcobalt/mixer_paths_i2s.xml:system/etc/mixer_paths_i2s.xml \
     hardware/qcom/audio/configs/msmcobalt/aanc_tuning_mixer.txt:system/etc/aanc_tuning_mixer.txt \
+    hardware/qcom/audio/configs/msmcobalt/aanc_tuning_mixer_tavil.txt:system/etc/aanc_tuning_mixer_tavil.txt \
     hardware/qcom/audio/configs/msmcobalt/audio_platform_info_i2s.xml:system/etc/audio_platform_info_i2s.xml \
     hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths.xml:system/etc/sound_trigger_mixer_paths.xml \
     hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths_wcd9330.xml:system/etc/sound_trigger_mixer_paths_wcd9330.xml \
+    hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml:system/etc/sound_trigger_mixer_paths_wcd9340.xml \
     hardware/qcom/audio/configs/msmcobalt/sound_trigger_platform_info.xml:system/etc/sound_trigger_platform_info.xml \
+    hardware/qcom/audio/configs/msmcobalt/graphite_ipc_platform_info.xml:system/etc/graphite_ipc_platform_info.xml \
     hardware/qcom/audio/configs/msmcobalt/audio_platform_info.xml:system/etc/audio_platform_info.xml
 
 #XML Audio configuration files
@@ -166,9 +174,9 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.offload.multiple.enabled=false
 
-#Disable Compress passthrough playback
+#Enable Compress passthrough playback
 PRODUCT_PROPERTY_OVERRIDES += \
-audio.offload.passthrough=false
+audio.offload.passthrough=true
 
 #Disable surround sound recording
 PRODUCT_PROPERTY_OVERRIDES += \
@@ -186,3 +194,21 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.parser.ip.buffer.size=262144
 
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
+
+#split a2dp DSP supported encoder list
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.bt.a2dp_offload_cap=sbc-aptx
+
+#enable software decoders for ALAC and APE
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.alac.decoder=true
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.ape.decoder=true
+
+#enable hw aac encoder by default
+PRODUCT_PROPERTY_OVERRIDES += \
+qcom.hw.aac.encoder=true
+
diff --git a/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
new file mode 100755
index 0000000..be77fee
--- /dev/null
+++ b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
@@ -0,0 +1,117 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2014-2016, The Linux Foundation. All rights reserved.       -->
+<!---                                                                           -->
+<!--- Redistribution and use in source and binary forms, with or without        -->
+<!--- modification, are permitted provided that the following conditions are    -->
+<!--- met:                                                                      -->
+<!---     * Redistributions of source code must retain the above copyright      -->
+<!---       notice, this list of conditions and the following disclaimer.       -->
+<!---     * Redistributions in binary form must reproduce the above             -->
+<!---       copyright notice, this list of conditions and the following         -->
+<!---       disclaimer in the documentation and/or other materials provided     -->
+<!---       with the distribution.                                              -->
+<!---     * Neither the name of The Linux Foundation nor the names of its       -->
+<!---       contributors may be used to endorse or promote products derived     -->
+<!---       from this software without specific prior written permission.       -->
+<!---                                                                           -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED              -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF      -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT    -->
+<!--- ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS    -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR    -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF      -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR           -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,     -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE      -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN    -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                             -->
+
+<mixer>
+    <!-- These are the initial mixer settings -->
+    <ctl name="LSM1 MUX" value="None" />
+    <ctl name="LSM2 MUX" value="None" />
+    <ctl name="LSM3 MUX" value="None" />
+    <ctl name="LSM4 MUX" value="None" />
+    <ctl name="LSM5 MUX" value="None" />
+    <ctl name="LSM6 MUX" value="None" />
+    <ctl name="LSM7 MUX" value="None" />
+    <ctl name="LSM8 MUX" value="None" />
+    <ctl name="SLIMBUS_5_TX LSM Function" value="None" />
+    <ctl name="MADONOFF Switch" value="0" />
+    <ctl name="MAD Input" value="DMIC1" />
+    <ctl name="MAD_BROADCAST Switch" value="0" />
+    <ctl name="TX13 INP MUX" value="CDC_DEC_5" />
+    <ctl name="AIF4_MAD Mixer SLIM TX12" value="0" />
+    <ctl name="AIF4_MAD Mixer SLIM TX13" value="0" />
+    <ctl name="CPE AFE MAD Enable" value="0"/>
+    <ctl name="CLK MODE" value="EXTERNAL" />
+    <ctl name="EC BUF MUX INP" value="ZERO" />
+    <ctl name="ADC MUX1" value="DMIC" />
+    <ctl name="DMIC MUX1" value="ZERO" />
+
+    <path name="listen-voice-wakeup-1">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+    </path>
+
+    <path name="listen-voice-wakeup-2">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-3">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-4">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-5">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-6">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-7">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-8">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+    </path>
+
+    <path name="listen-cpe-handset-mic">
+        <ctl name="MAD Input" value="DMIC0" />
+        <ctl name="MAD_SEL MUX" value="SPE" />
+        <ctl name="MAD_INP MUX" value="MAD" />
+        <ctl name="MAD_CPE1 Switch" value="1" />
+    </path>
+
+    <path name="listen-cpe-handset-mic-ecpp">
+        <ctl name="CLK MODE" value="INTERNAL" />
+        <ctl name="EC BUF MUX INP" value="DEC1" />
+        <ctl name="ADC MUX1" value="DMIC" />
+        <ctl name="DMIC MUX1" value="DMIC0" />
+    </path>
+
+    <!-- path name used for low bandwidth FTRT codec interface -->
+    <path name="listen-cpe-handset-mic low-speed-intf">
+        <ctl name="MADONOFF Switch" value="1" />
+        <ctl name="AIF4_MAD Mixer SLIM TX12" value="1" />
+        <ctl name="MAD Input" value="DMIC0" />
+        <ctl name="CPE AFE MAD Enable" value="1"/>
+    </path>
+
+    <path name="listen-ape-handset-mic">
+        <ctl name="MAD Input" value="DMIC0" />
+        <ctl name="MAD_SEL MUX" value="MSM" />
+        <ctl name="MAD_INP MUX" value="MAD" />
+        <ctl name="MAD_BROADCAST Switch" value="1" />
+        <ctl name="CDC_IF TX13 MUX" value="MAD_BRDCST" />
+        <ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
+    </path>
+
+</mixer>
diff --git a/configs/msmcobalt/sound_trigger_platform_info.xml b/configs/msmcobalt/sound_trigger_platform_info.xml
index b92ea48..6c9f55e 100644
--- a/configs/msmcobalt/sound_trigger_platform_info.xml
+++ b/configs/msmcobalt/sound_trigger_platform_info.xml
@@ -27,25 +27,23 @@
 <!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                             -->
 <sound_trigger_platform_info>
     <param version="0x0101" /> <!-- this must be the first param -->
-
     <common_config>
-        <param execution_type="CPE" /> <!-- value: "CPE" "APE" -->
         <param max_cpe_sessions="1" />
+        <param max_wdsp_sessions="2" />
         <param max_ape_sessions="8" />
         <param enable_failure_detection="false" />
     </common_config>
-
     <acdb_ids>
-        <param DEVICE_HANDSET_APE_ACDB_ID="100" />
-        <param DEVICE_HANDSET_CPE_ACDB_ID="128" />
-        <param DEVICE_HANDSET_CPE_ECPP_ACDB_ID="128" />
+        <param DEVICE_HANDSET_MIC_APE="100" />
+        <param DEVICE_HANDSET_MIC_CPE="128" />
+        <param DEVICE_HANDSET_MIC_ECPP_CPE="128" />
     </acdb_ids>
-
-    <!-- Multiple sound_model_config tags can be listed, each with unique    -->
-    <!-- vendor_uuid. The below tag represents QTI SVA engine sound model    -->
-    <!-- configuration. ISV must use their own unique vendor_uuid.           -->
+    <!-- Multiple sound_model_config tags can be listed, each with unique   -->
+    <!-- vendor_uuid. The below tag represents QTI SVA engine sound model   -->
+    <!-- configuration. ISV must use their own unique vendor_uuid.          -->
     <sound_model_config>
         <param vendor_uuid="68ab2d40-e860-11e3-95ef-0002a5d5c51b" />
+        <param execution_type="WDSP" /> <!-- value: "WDSP" "ADSP" "DYNAMIC" -->
         <param app_type="2" /> <!-- app type used in ACDB -->
         <param library="libsmwrapper.so" />
         <param max_cpe_phrases="6" />
@@ -53,8 +51,25 @@
         <param max_ape_phrases="10" />
         <param max_ape_users="10" />
         <param sample_rate="16000" />
-
-        <!-- Module and param ids with which the algorithm is integrated in firmware -->
+        <gcs_usecase>
+            <param uid="0x1" />
+            <!-- module_id, instance_id, param_id -->
+            <param load_sound_model_ids="0x00012C0D, 0x2, 0x00012C14" />
+            <param confidence_levels_ids="0x00012C0D, 0x2, 0x00012C28" />
+            <param detection_event_ids="0x00012C0D, 0x2, 0x00012C29" />
+            <param read_cmd_ids="0x00020013, 0x2, 0x00020015" />
+            <param read_rsp_ids="0x00020013, 0x2, 0x00020016" />
+        </gcs_usecase>
+        <gcs_usecase>
+            <param uid="0x2" />
+            <param load_sound_model_ids="0x00012C0D, 0x3, 0x00012C14" />
+            <param confidence_levels_ids="0x00012C0D, 0x3, 0x00012C28" />
+            <param detection_event_ids="0x00012C0D, 0x3, 0x00012C29" />
+            <param read_cmd_ids="0x00020013, 0x3, 0x00020015" />
+            <param read_rsp_ids="0x00020013, 0x3, 0x00020016" />
+        </gcs_usecase>
+        <!-- Module and param ids with which the algorithm is integrated
+            in non-graphite firmware (note these must come after gcs params) -->
         <param load_sound_model_ids="0x00012C0D, 0x00012C14" />
         <param unload_sound_model_ids="0x00012C0D, 0x00012C15" />
         <param confidence_levels_ids="0x00012C0D, 0x00012C07" />
@@ -62,9 +77,9 @@
 
         <!-- format: "ADPCM_packet" or "PCM_packet" !-->
         <!-- transfer_mode: "FTRT" or "RT" -->
-        <!--  kw_duration is in milli seconds. It is valid only for FTRT transfer mode -->
+        <!--  kw_duration is in milli seconds. It is valid only for FTRT
+            transfer mode -->
         <param capture_keyword="PCM_packet, RT, 2000" />
         <param client_capture_read_delay="2000" />
     </sound_model_config>
-
 </sound_trigger_platform_info>
diff --git a/configs/msmfalcon/audio_policy_configuration.xml b/configs/msmfalcon/audio_policy_configuration.xml
index 56848ad..b1ea1b9 100644
--- a/configs/msmfalcon/audio_policy_configuration.xml
+++ b/configs/msmfalcon/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
                 <mixPort name="direct_pcm" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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_24_BIT_PACKED"
-                             samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+                             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"/>
                 </mixPort>
                 <mixPort name="compressed_offload" role="source"
@@ -256,13 +256,13 @@
             <!-- 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,BT SCO Headset Mic"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Speaker"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Line"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="HDMI"
@@ -281,8 +281,6 @@
                        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="Telephony Tx"
-                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
             </routes>
diff --git a/configs/msmfalcon/msmfalcon.mk b/configs/msmfalcon/msmfalcon.mk
index 2e29d7a..554f32b 100644
--- a/configs/msmfalcon/msmfalcon.mk
+++ b/configs/msmfalcon/msmfalcon.mk
@@ -39,7 +39,7 @@
 AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
 BOARD_USES_SRS_TRUEMEDIA := false
 DTS_CODEC_M_ := false
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
 MM_AUDIO_ENABLED_FTM := true
 MM_AUDIO_ENABLED_SAFX := true
 TARGET_USES_QCOM_MM_AUDIO := true
diff --git a/hal/Android.mk b/hal/Android.mk
index 83787e3..daf7397 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -243,6 +243,17 @@
     LOCAL_SRC_FILES += audio_extn/source_track.c
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SPLIT_A2DP)),true)
+    LOCAL_CFLAGS += -DSPLIT_A2DP_ENABLED
+    LOCAL_SRC_FILES += audio_extn/a2dp.c
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_QAF)),true)
+    LOCAL_CFLAGS += -DQAF_EXTN_ENABLED
+    LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/qaf/
+    LOCAL_SRC_FILES += audio_extn/qaf.c
+endif
+
 LOCAL_SHARED_LIBRARIES := \
 	liblog \
 	libcutils \
@@ -251,6 +262,7 @@
 	libaudioroute \
 	libdl \
 	libaudioutils \
+	libhardware \
 	libexpat
 
 LOCAL_C_INCLUDES += \
@@ -279,6 +291,14 @@
 endif
 
 ifeq ($(strip $(BOARD_SUPPORTS_SOUND_TRIGGER)),true)
+    ST_FEATURE_ENABLE := true
+endif
+
+ifeq ($(strip $(BOARD_SUPPORTS_SOUND_TRIGGER_HAL)),true)
+    ST_FEATURE_ENABLE := true
+endif
+
+ifeq ($(ST_FEATURE_ENABLE), true)
     LOCAL_CFLAGS += -DSOUND_TRIGGER_ENABLED
     LOCAL_CFLAGS += -DSOUND_TRIGGER_PLATFORM_NAME=$(TARGET_BOARD_PLATFORM)
     LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/sound_trigger
@@ -296,11 +316,25 @@
     LOCAL_SHARED_LIBRARIES += libperipheral_client
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_DISPLAY_PORT)),true)
+    LOCAL_CFLAGS += -DDISPLAY_PORT_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_GEF_SUPPORT)),true)
+    LOCAL_CFLAGS += -DAUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED
+    LOCAL_SRC_FILES += audio_extn/gef.c
+endif
+
 LOCAL_CFLAGS += -Wall -Werror
 
 LOCAL_COPY_HEADERS_TO   := mm-audio
 LOCAL_COPY_HEADERS      := audio_extn/audio_defs.h
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SND_MONITOR)), true)
+    LOCAL_CFLAGS += -DSND_MONITOR_ENABLED
+    LOCAL_SRC_FILES += audio_extn/sndmonitor.c
+endif
+
 LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
 
 LOCAL_MODULE_RELATIVE_PATH := hw
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
new file mode 100644
index 0000000..e72cb76
--- /dev/null
+++ b/hal/audio_extn/a2dp.c
@@ -0,0 +1,787 @@
+/*
+* Copyright (c) 2015-16, 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 "split_a2dp"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+#include <errno.h>
+#include <cutils/log.h>
+#include <dlfcn.h>
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <stdlib.h>
+#include <cutils/str_parms.h>
+#include <hardware/audio.h>
+#include <hardware/hardware.h>
+#include <cutils/properties.h>
+
+#ifdef SPLIT_A2DP_ENABLED
+#define AUDIO_PARAMETER_A2DP_STARTED "A2dpStarted"
+#define BT_IPC_LIB_NAME  "libbthost_if.so"
+#define ENC_MEDIA_FMT_NONE                                     0
+#define ENC_MEDIA_FMT_AAC                                  0x00010DA6
+#define ENC_MEDIA_FMT_APTX                                 0x000131ff
+#define ENC_MEDIA_FMT_APTX_HD                              0x00013200
+#define ENC_MEDIA_FMT_SBC                                  0x00010BF2
+#define MEDIA_FMT_AAC_AOT_LC                               2
+#define MEDIA_FMT_AAC_AOT_SBR                              5
+#define MEDIA_FMT_AAC_AOT_PS                               29
+#define PCM_CHANNEL_L                                      1
+#define PCM_CHANNEL_R                                      2
+#define PCM_CHANNEL_C                                      3
+#define MEDIA_FMT_SBC_CHANNEL_MODE_MONO                    1
+#define MEDIA_FMT_SBC_CHANNEL_MODE_STEREO                  2
+#define MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO               8
+#define MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO            9
+#define MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS           0
+#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_ENC_FMT_SBC          "SBC"
+#define MIXER_ENC_FMT_AAC          "AAC"
+#define MIXER_ENC_FMT_APTX         "APTX"
+#define MIXER_ENC_FMT_APTXHD       "APTXHD"
+#define MIXER_ENC_FMT_NONE         "NONE"
+
+
+typedef int (*audio_stream_open_t)(void);
+typedef int (*audio_stream_close_t)(void);
+typedef int (*audio_start_stream_t)(void);
+typedef int (*audio_stop_stream_t)(void);
+typedef int (*audio_suspend_stream_t)(void);
+typedef void (*audio_handoff_triggered_t)(void);
+typedef void (*clear_a2dpsuspend_flag_t)(void);
+typedef void * (*audio_get_codec_config_t)(uint8_t *multicast_status,uint8_t *num_dev,
+                               audio_format_t *codec_type);
+
+enum A2DP_STATE {
+    A2DP_STATE_CONNECTED,
+    A2DP_STATE_STARTED,
+    A2DP_STATE_STOPPED,
+    A2DP_STATE_DISCONNECTED,
+};
+
+/* structure used to  update a2dp state machine
+ * to communicate IPC library
+ * to store DSP encoder configuration information
+ */
+struct a2dp_data {
+    struct audio_device *adev;
+    void *bt_lib_handle;
+    audio_stream_open_t audio_stream_open;
+    audio_stream_close_t audio_stream_close;
+    audio_start_stream_t audio_start_stream;
+    audio_stop_stream_t audio_stop_stream;
+    audio_suspend_stream_t audio_suspend_stream;
+    audio_handoff_triggered_t audio_handoff_triggered;
+    clear_a2dpsuspend_flag_t clear_a2dpsuspend_flag;
+    audio_get_codec_config_t audio_get_codec_config;
+    enum A2DP_STATE bt_state;
+    audio_format_t bt_encoder_format;
+    uint32_t enc_sampling_rate;
+    bool a2dp_started;
+    bool a2dp_suspended;
+    int  a2dp_total_active_session_request;
+    bool is_a2dp_offload_supported;
+    bool is_handoff_in_progress;
+};
+
+struct a2dp_data a2dp;
+
+/* START of DSP configurable structures
+ * These values should match with DSP interface defintion
+ */
+
+/* AAC encoder configuration structure. */
+typedef struct aac_enc_cfg_t aac_enc_cfg_t;
+
+/* supported enc_mode are AAC_LC, AAC_SBR, AAC_PS
+ * supported aac_fmt_flag are ADTS/RAW
+ * supported channel_cfg are Native mode, Mono , Stereo
+ */
+struct aac_enc_cfg_t {
+    uint32_t      enc_format;
+    uint32_t      bit_rate;
+    uint32_t      enc_mode;
+    uint16_t      aac_fmt_flag;
+    uint16_t      channel_cfg;
+    uint32_t      sample_rate;
+} ;
+
+/* SBC encoder configuration structure. */
+typedef struct sbc_enc_cfg_t sbc_enc_cfg_t;
+
+/* supported num_subbands are 4/8
+ * supported blk_len are 4, 8, 12, 16
+ * supported channel_mode are MONO, STEREO, DUAL_MONO, JOINT_STEREO
+ * supported alloc_method are LOUNDNESS/SNR
+ * supported bit_rate for mono channel is max 320kbps
+ * supported bit rate for stereo channel is max 512 kbps
+ */
+struct sbc_enc_cfg_t{
+    uint32_t      enc_format;
+    uint32_t      num_subbands;
+    uint32_t      blk_len;
+    uint32_t      channel_mode;
+    uint32_t      alloc_method;
+    uint32_t      bit_rate;
+    uint32_t      sample_rate;
+};
+
+
+/* supported num_channels are Mono/Stereo
+ * supported channel_mapping for mono is CHANNEL_C
+ * supported channel mapping for stereo is CHANNEL_L and CHANNEL_R
+ * custom size and reserved are not used(for future enhancement)
+  */
+struct custom_enc_cfg_aptx_t
+{
+    uint32_t      enc_format;
+    uint32_t      sample_rate;
+    uint16_t      num_channels;
+    uint16_t      reserved;
+    uint8_t       channel_mapping[8];
+    uint32_t      custom_size;
+};
+
+/*********** END of DSP configurable structures ********************/
+
+/* API to identify DSP encoder captabilities */
+static void a2dp_offload_codec_cap_parser(char *value)
+{
+    char *tok = NULL,*saveptr;
+
+    tok = strtok_r(value, "-", &saveptr);
+    while (tok != NULL) {
+        if (strcmp(tok, "sbc") == 0) {
+            ALOGD("%s: SBC offload supported\n",__func__);
+            a2dp.is_a2dp_offload_supported = true;
+            break;
+        } else if (strcmp(tok, "aptx") == 0) {
+            ALOGD("%s: aptx offload supported\n",__func__);
+            a2dp.is_a2dp_offload_supported = true;
+            break;
+        }
+        tok = strtok_r(NULL, "-", &saveptr);
+    };
+}
+
+static void update_offload_codec_capabilities()
+{
+    char value[PROPERTY_VALUE_MAX] = {'\0'};
+
+    property_get("persist.bt.a2dp_offload_cap", value, "false");
+    ALOGD("get_offload_codec_capabilities = %s",value);
+    a2dp.is_a2dp_offload_supported =
+            property_get_bool("persist.bt.a2dp_offload_cap", false);
+    if (strcmp(value, "false") != 0)
+        a2dp_offload_codec_cap_parser(value);
+    ALOGD("%s: codec cap = %s",__func__,value);
+}
+
+/* API to open BT IPC library to start IPC communication */
+static void open_a2dp_output()
+{
+    int ret = 0;
+
+    ALOGD(" Open A2DP output start ");
+    if (a2dp.bt_lib_handle == NULL){
+        ALOGD(" Requesting for BT lib handle");
+        a2dp.bt_lib_handle = dlopen(BT_IPC_LIB_NAME, RTLD_NOW);
+
+        if (a2dp.bt_lib_handle == NULL) {
+            ALOGE("%s: DLOPEN failed for %s", __func__, BT_IPC_LIB_NAME);
+            ret = -ENOSYS;
+            goto init_fail;
+        } else {
+            a2dp.audio_stream_open = (audio_stream_open_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_stream_open");
+            a2dp.audio_start_stream = (audio_start_stream_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_start_stream");
+            a2dp.audio_get_codec_config = (audio_get_codec_config_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_get_codec_config");
+            a2dp.audio_suspend_stream = (audio_suspend_stream_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_suspend_stream");
+            a2dp.audio_handoff_triggered = (audio_handoff_triggered_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_handoff_triggered");
+            a2dp.clear_a2dpsuspend_flag = (clear_a2dpsuspend_flag_t)
+                          dlsym(a2dp.bt_lib_handle, "clear_a2dpsuspend_flag");
+            a2dp.audio_stop_stream = (audio_stop_stream_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_stop_stream");
+            a2dp.audio_stream_close = (audio_stream_close_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_stream_close");
+        }
+    }
+
+    if (a2dp.bt_lib_handle && a2dp.audio_stream_open) {
+        if (a2dp.bt_state == A2DP_STATE_DISCONNECTED) {
+            ALOGD("calling BT stream open");
+            ret = a2dp.audio_stream_open();
+            if(ret != 0) {
+                ALOGE("Failed to open output stream for a2dp: status %d", ret);
+                goto init_fail;
+            }
+            a2dp.bt_state = A2DP_STATE_CONNECTED;
+        } else {
+            ALOGD("Called a2dp open with improper state, Ignoring request state %d", a2dp.bt_state);
+        }
+    } else {
+        ALOGE("a2dp handle is not identified, Ignoring open request");
+        a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+        goto init_fail;
+    }
+
+init_fail:
+    if(ret != 0 && (a2dp.bt_lib_handle != NULL)) {
+        dlclose(a2dp.bt_lib_handle);
+        a2dp.bt_lib_handle = NULL;
+    }
+}
+
+static int close_a2dp_output()
+{
+    ALOGV("%s\n",__func__);
+    if (!(a2dp.bt_lib_handle && a2dp.audio_stream_close)) {
+        ALOGE("a2dp handle is not identified, Ignoring close request");
+        return -ENOSYS;
+    }
+    if ((a2dp.bt_state == A2DP_STATE_CONNECTED) &&
+        (a2dp.bt_state == A2DP_STATE_STARTED) &&
+        (a2dp.bt_state == A2DP_STATE_STOPPED)) {
+        ALOGD("calling BT stream close");
+        if(a2dp.audio_stream_close() == false)
+            ALOGE("failed close a2dp control path from BT library");
+        a2dp.a2dp_started = false;
+        a2dp.a2dp_total_active_session_request = 0;
+        a2dp.a2dp_suspended = false;
+        a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+        a2dp.enc_sampling_rate = 48000;
+        a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+    } else {
+        ALOGD("close a2dp called in improper state");
+        a2dp.a2dp_started = false;
+        a2dp.a2dp_total_active_session_request = 0;
+        a2dp.a2dp_suspended = false;
+        a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+        a2dp.enc_sampling_rate = 48000;
+        a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+    }
+
+    return 0;
+}
+
+/* API to configure SBC DSP encoder */
+bool configure_sbc_enc_format(audio_sbc_encoder_config *sbc_bt_cfg)
+{
+    struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
+    struct sbc_enc_cfg_t sbc_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(sbc_bt_cfg == NULL)
+        return false;
+
+   ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    a2dp.bt_encoder_format = AUDIO_FORMAT_SBC;
+    memset(&sbc_dsp_cfg, 0x0, sizeof(struct sbc_enc_cfg_t));
+    sbc_dsp_cfg.enc_format = ENC_MEDIA_FMT_SBC;
+    sbc_dsp_cfg.num_subbands = sbc_bt_cfg->subband;
+    sbc_dsp_cfg.blk_len = sbc_bt_cfg->blk_len;
+    switch(sbc_bt_cfg->channels) {
+        case 0:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_MONO;
+            break;
+        case 1:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO;
+            break;
+        case 3:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO;
+            break;
+        case 2:
+        default:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_STEREO;
+            break;
+    }
+    if (sbc_bt_cfg->alloc)
+        sbc_dsp_cfg.alloc_method = MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS;
+    else
+        sbc_dsp_cfg.alloc_method = MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR;
+    sbc_dsp_cfg.bit_rate = sbc_bt_cfg->bitrate;
+    sbc_dsp_cfg.sample_rate = sbc_bt_cfg->sampling_rate;
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&sbc_dsp_cfg,
+                                    sizeof(struct sbc_enc_cfg_t));
+    if (ret != 0) {
+        ALOGE("%s: failed to set SBC 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.enc_sampling_rate = sbc_bt_cfg->sampling_rate;
+    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;
+}
+
+/* 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 custom_enc_cfg_aptx_t aptx_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(aptx_bt_cfg == NULL)
+        return false;
+
+    ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    a2dp.bt_encoder_format = AUDIO_FORMAT_APTX;
+    memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_aptx_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;
+    }
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+                              sizeof(struct custom_enc_cfg_aptx_t));
+    if (ret != 0) {
+        ALOGE("%s: Failed to set APTX 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;
+    } else {
+        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.enc_sampling_rate = aptx_bt_cfg->sampling_rate;
+    ALOGV("Successfully updated APTX enc format with samplingrate: %d channels:%d",
+           aptx_dsp_cfg.sample_rate, aptx_dsp_cfg.num_channels);
+fail:
+    return is_configured;
+}
+
+/* API to configure APTX HD DSP encoder
+ */
+bool configure_aptx_hd_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
+{
+    struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
+    struct custom_enc_cfg_aptx_t aptx_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(aptx_bt_cfg == NULL)
+        return false;
+
+    ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+
+    a2dp.bt_encoder_format = AUDIO_FORMAT_APTX_HD;
+    memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_aptx_t));
+    aptx_dsp_cfg.enc_format = ENC_MEDIA_FMT_APTX_HD;
+    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;
+    }
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+                              sizeof(struct custom_enc_cfg_aptx_t));
+    if (ret != 0) {
+        ALOGE("%s: Failed to set APTX HD 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, "S24_LE");
+    if (ret != 0) {
+        ALOGE("%s: Failed to set APTX HD encoder config", __func__);
+        is_configured = false;
+        goto fail;
+    }
+    is_configured = true;
+    a2dp.enc_sampling_rate = aptx_bt_cfg->sampling_rate;
+    ALOGV("Successfully updated APTX HD encformat with samplingrate: %d channels:%d",
+           aptx_dsp_cfg.sample_rate, aptx_dsp_cfg.num_channels);
+fail:
+    return is_configured;
+}
+
+/* API to configure AAC DSP encoder */
+bool configure_aac_enc_format(audio_aac_encoder_config *aac_bt_cfg)
+{
+    struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
+    struct aac_enc_cfg_t aac_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(aac_bt_cfg == NULL)
+        return false;
+
+    ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    a2dp.bt_encoder_format = AUDIO_FORMAT_AAC;
+    memset(&aac_dsp_cfg, 0x0, sizeof(struct aac_enc_cfg_t));
+    aac_dsp_cfg.enc_format = ENC_MEDIA_FMT_AAC;
+    aac_dsp_cfg.bit_rate = aac_bt_cfg->bitrate;
+    aac_dsp_cfg.sample_rate = aac_bt_cfg->sampling_rate;
+    switch(aac_bt_cfg->enc_mode) {
+        case 0:
+            aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_LC;
+            break;
+        case 2:
+            aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_PS;
+            break;
+        case 1:
+        default:
+            aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_SBR;
+            break;
+    }
+    aac_dsp_cfg.aac_fmt_flag = aac_bt_cfg->format_flag;
+    aac_dsp_cfg.channel_cfg = aac_bt_cfg->channels;
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aac_dsp_cfg,
+                              sizeof(struct aac_enc_cfg_t));
+    if (ret != 0) {
+        ALOGE("%s: failed to set SBC 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) {
+        is_configured = false;
+        ALOGE(" ERROR  bit format CONFIG data mixer control not identifed");
+        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.enc_sampling_rate = aac_bt_cfg->sampling_rate;
+    ALOGV("Successfully updated AAC enc format with samplingrate: %d channels:%d",
+           aac_dsp_cfg.sample_rate, aac_dsp_cfg.channel_cfg);
+fail:
+    return is_configured;
+}
+
+bool configure_a2dp_encoder_format()
+{
+    void *codec_info = NULL;
+    uint8_t multi_cast = 0, num_dev = 1;
+    audio_format_t codec_type = AUDIO_FORMAT_INVALID;
+    bool is_configured = false;
+
+    if (!a2dp.audio_get_codec_config) {
+        ALOGE(" a2dp handle is not identified, ignoring a2dp encoder config");
+        return false;
+    }
+    ALOGD("configure_a2dp_encoder_format start");
+    codec_info = a2dp.audio_get_codec_config(&multi_cast, &num_dev,
+                               &codec_type);
+
+    switch(codec_type) {
+        case AUDIO_FORMAT_SBC:
+            ALOGD(" Received SBC encoder supported BT device");
+            is_configured =
+               configure_sbc_enc_format((audio_sbc_encoder_config *)codec_info);
+            break;
+        case AUDIO_FORMAT_APTX:
+            ALOGD(" Received APTX encoder supported BT device");
+            is_configured =
+              configure_aptx_enc_format((audio_aptx_encoder_config *)codec_info);
+            break;
+        case AUDIO_FORMAT_APTX_HD:
+            ALOGD(" Received APTX HD encoder supported BT device");
+            is_configured =
+             configure_aptx_hd_enc_format((audio_aptx_encoder_config *)codec_info);
+            break;
+        case AUDIO_FORMAT_AAC:
+            ALOGD(" Received AAC encoder supported BT device");
+            is_configured =
+              configure_aac_enc_format((audio_aac_encoder_config *)codec_info);
+            break;
+        default:
+            ALOGD(" Received Unsupported encoder formar");
+            is_configured = false;
+            break;
+    }
+    return is_configured;
+}
+
+int audio_extn_a2dp_start_playback()
+{
+    int ret = 0;
+
+    ALOGD("audio_extn_a2dp_start_playback start");
+
+    if(!(a2dp.bt_lib_handle && a2dp.audio_start_stream
+       && a2dp.audio_get_codec_config)) {
+        ALOGE("a2dp handle is not identified, Ignoring start request");
+        return -ENOSYS;
+    }
+
+    if(a2dp.a2dp_suspended == true) {
+        //session will be restarted after suspend completion
+        ALOGD("a2dp start requested during suspend state");
+        return -ENOSYS;
+    }
+
+    if (!a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+        ALOGD("calling BT module stream start");
+        /* This call indicates BT IPC lib to start playback */
+        ret =  a2dp.audio_start_stream();
+        ALOGE("BT controller start return = %d",ret);
+        if (ret != 0 ) {
+           ALOGE("BT controller start failed");
+           a2dp.a2dp_started = false;
+           ret = -ETIMEDOUT;
+        } else {
+           if(configure_a2dp_encoder_format() == true) {
+                a2dp.a2dp_started = true;
+                ret = 0;
+                ALOGD("Start playback successful to BT library");
+           } else {
+                ALOGD(" unable to configure DSP encoder");
+                a2dp.a2dp_started = false;
+                ret = -ETIMEDOUT;
+           }
+        }
+    }
+
+    if (a2dp.a2dp_started)
+        a2dp.a2dp_total_active_session_request++;
+
+    ALOGD("start A2DP playback total active sessions :%d",
+          a2dp.a2dp_total_active_session_request);
+    return ret;
+}
+
+int audio_extn_a2dp_stop_playback()
+{
+    int ret =0;
+
+    ALOGV("audio_extn_a2dp_stop_playback start");
+    if(!(a2dp.bt_lib_handle && a2dp.audio_stop_stream)) {
+        ALOGE("a2dp handle is not identified, Ignoring start request");
+        return -ENOSYS;
+    }
+
+    if (a2dp.a2dp_started && (a2dp.a2dp_total_active_session_request > 0))
+        a2dp.a2dp_total_active_session_request--;
+
+    if ( a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+        struct mixer_ctl *ctl_enc_config, *ctrl_bit_format;
+        struct sbc_enc_cfg_t dummy_reset_config;
+
+        ALOGV("calling BT module stream stop");
+        ret = a2dp.audio_stop_stream();
+        if (ret < 0)
+            ALOGE("stop stream to BT IPC lib failed");
+        else
+            ALOGV("stop steam to BT IPC lib successful");
+         memset(&dummy_reset_config, 0x0, sizeof(struct sbc_enc_cfg_t));
+        ctl_enc_config = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                               MIXER_ENC_CONFIG_BLOCK);
+        if (!ctl_enc_config) {
+            ALOGE(" ERROR  a2dp encoder format mixer control not identifed");
+        } else {
+            ret = mixer_ctl_set_array(ctl_enc_config, (void *)&dummy_reset_config,
+                                            sizeof(struct sbc_enc_cfg_t));
+             a2dp.bt_encoder_format = ENC_MEDIA_FMT_NONE;
+        }
+        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");
+        } else {
+            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__);
+            }
+        }
+    }
+    if(!a2dp.a2dp_total_active_session_request)
+       a2dp.a2dp_started = false;
+    ALOGD("Stop A2DP playback total active sessions :%d",
+          a2dp.a2dp_total_active_session_request);
+    return 0;
+}
+
+void audio_extn_a2dp_set_parameters(struct str_parms *parms)
+{
+     int ret, val;
+     char value[32]={0};
+
+     if(a2dp.is_a2dp_offload_supported == false) {
+        ALOGV("no supported encoders identified,ignoring a2dp setparam");
+        return;
+     }
+
+     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value,
+                            sizeof(value));
+     if( ret >= 0) {
+         val = atoi(value);
+         if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
+             ALOGV("Received device connect request for A2DP");
+             open_a2dp_output();
+         }
+         goto param_handled;
+     }
+
+     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value,
+                         sizeof(value));
+
+     if( ret >= 0) {
+         val = atoi(value);
+         if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
+             ALOGV("Received device dis- connect request");
+             close_a2dp_output();
+         }
+         goto param_handled;
+     }
+
+     ret = str_parms_get_str(parms, "A2dpSuspended", value, sizeof(value));
+     if (ret >= 0) {
+         if (a2dp.bt_lib_handle && (a2dp.bt_state != A2DP_STATE_DISCONNECTED) ) {
+             if ((!strncmp(value,"true",sizeof(value)))) {
+                ALOGD("Setting a2dp to suspend state");
+                a2dp.a2dp_suspended = true;
+                if(a2dp.audio_suspend_stream)
+                   a2dp.audio_suspend_stream();
+            } else if (a2dp.a2dp_suspended == true) {
+                ALOGD("Resetting a2dp suspend state");
+                if(a2dp.clear_a2dpsuspend_flag)
+                    a2dp.clear_a2dpsuspend_flag();
+                a2dp.a2dp_suspended = false;
+            }
+        }
+        goto param_handled;
+     }
+param_handled:
+     ALOGV("end of a2dp setparam");
+}
+
+void audio_extn_a2dp_set_handoff_mode(bool is_on)
+{
+    a2dp.is_handoff_in_progress = is_on;
+}
+
+bool audio_extn_a2dp_is_force_device_switch()
+{
+    //During encoder reconfiguration mode, force a2dp device switch
+    return a2dp.is_handoff_in_progress;
+}
+
+void audio_extn_a2dp_get_apptype_params(uint32_t *sample_rate,
+                                        uint32_t *bit_width)
+{
+    if(a2dp.bt_encoder_format == AUDIO_FORMAT_APTX_HD)
+        *bit_width = 24;
+    else
+        *bit_width = 16;
+    *sample_rate = a2dp.enc_sampling_rate;
+}
+void audio_extn_a2dp_init (void *adev)
+{
+  a2dp.adev = (struct audio_device*)adev;
+  a2dp.bt_lib_handle = NULL;
+  a2dp.a2dp_started = false;
+  a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+  a2dp.a2dp_total_active_session_request = 0;
+  a2dp.a2dp_suspended = false;
+  a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+  a2dp.enc_sampling_rate = 48000;
+  a2dp.is_a2dp_offload_supported = false;
+  a2dp.is_handoff_in_progress = false;
+  update_offload_codec_capabilities();
+}
+#endif // SPLIT_A2DP_ENABLED
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 49e649c..083b925 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -96,7 +96,7 @@
 * this is done when device switch happens by setting audioparamter
 */
 
-#define HDMI_PLUG_STATUS_NOTIFY_ENABLE 0x30
+#define EXT_DISPLAY_PLUG_STATUS_NOTIFY_ENABLE 0x30
 
 static ssize_t update_sysfs_node(const char *path, const char *data, size_t len)
 {
@@ -121,56 +121,62 @@
     return err;
 }
 
-static int get_hdmi_sysfs_node_index()
+static int get_ext_disp_sysfs_node_index(int ext_disp_type)
 {
-    static int node_index = -1;
+    int node_index = -1;
     char fbvalue[80] = {0};
     char fbpath[80] = {0};
     int i = 0;
-    FILE *hdmi_fp = NULL;
+    FILE *ext_disp_fd = NULL;
 
-    if(node_index >= 0) {
-        //hdmi sysfs node will not change so we just need to get the index once.
-        ALOGV("HDMI sysfs node is at fb%d", node_index);
-        return node_index;
-    }
-
-    for(i = 0; i < 3; i++) {
+    while (1) {
         snprintf(fbpath, sizeof(fbpath),
                   "/sys/class/graphics/fb%d/msm_fb_type", i);
-        hdmi_fp = fopen(fbpath, "r");
-        if(hdmi_fp) {
-            fread(fbvalue, sizeof(char), 80, hdmi_fp);
-            if(strncmp(fbvalue, "dtv panel", strlen("dtv panel")) == 0) {
-                node_index = i;
-                ALOGV("HDMI is at fb%d",i);
-                fclose(hdmi_fp);
-                return node_index;
+        ext_disp_fd = fopen(fbpath, "r");
+        if (ext_disp_fd) {
+            if (fread(fbvalue, sizeof(char), 80, ext_disp_fd)) {
+                if(((strncmp(fbvalue, "dtv panel", strlen("dtv panel")) == 0) &&
+                    (ext_disp_type == EXT_DISPLAY_TYPE_HDMI)) ||
+                   ((strncmp(fbvalue, "dp panel", strlen("dp panel")) == 0) &&
+                    (ext_disp_type == EXT_DISPLAY_TYPE_DP))) {
+                    node_index = i;
+                    ALOGD("%s: Ext Disp:%d is at fb%d", __func__, ext_disp_type, i);
+                    fclose(ext_disp_fd);
+                    return node_index;
+                }
             }
-            fclose(hdmi_fp);
+            fclose(ext_disp_fd);
+            i++;
         } else {
-            ALOGE("Failed to open fb node %d",i);
+            ALOGE("%s: Scanned till end of fbs or Failed to open fb node %d", __func__, i);
+            break;
         }
     }
 
     return -1;
 }
 
-static int update_hdmi_sysfs_node(int node_value)
+static int update_ext_disp_sysfs_node(const struct audio_device *adev, int node_value)
 {
-    char hdmi_ack_path[80] = {0};
-    char hdmi_ack_value[3] = {0};
+    char ext_disp_ack_path[80] = {0};
+    char ext_disp_ack_value[3] = {0};
     int index, ret = -1;
+    int ext_disp_type = platform_get_ext_disp_type(adev->platform);
 
-    index = get_hdmi_sysfs_node_index();
+    if (ext_disp_type < 0) {
+        ALOGE("%s, Unable to get the external display type, err:%d",
+              __func__, ext_disp_type);
+        return -EINVAL;
+    }
 
+    index = get_ext_disp_sysfs_node_index(ext_disp_type);
     if (index >= 0) {
-        snprintf(hdmi_ack_value, sizeof(hdmi_ack_value), "%d", node_value);
-        snprintf(hdmi_ack_path, sizeof(hdmi_ack_path),
+        snprintf(ext_disp_ack_value, sizeof(ext_disp_ack_value), "%d", node_value);
+        snprintf(ext_disp_ack_path, sizeof(ext_disp_ack_path),
                   "/sys/class/graphics/fb%d/hdmi_audio_cb", index);
 
-        ret = update_sysfs_node(hdmi_ack_path, hdmi_ack_value,
-                sizeof(hdmi_ack_value));
+        ret = update_sysfs_node(ext_disp_ack_path, ext_disp_ack_value,
+                sizeof(ext_disp_ack_value));
 
         ALOGI("update hdmi_audio_cb at fb[%d] to:[%d] %s",
             index, node_value, (ret >= 0) ? "success":"fail");
@@ -179,25 +185,27 @@
     return ret;
 }
 
-static void check_and_set_hdmi_connection_status(struct str_parms *parms)
+static void check_and_set_ext_disp_connection_status(const struct audio_device *adev,
+                                                     struct str_parms *parms)
 {
     char value[32] = {0};
     static bool is_hdmi_sysfs_node_init = false;
 
     if (str_parms_get_str(parms, "connect", value, sizeof(value)) >= 0
-            && (atoi(value) & AUDIO_DEVICE_OUT_HDMI)) {
-        //params = "connect=1024" for HDMI connection.
+            && (atoi(value) & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+        //params = "connect=1024" for external display connection.
         if (is_hdmi_sysfs_node_init == false) {
+            //check if this is different for dp and hdmi
             is_hdmi_sysfs_node_init = true;
-            update_hdmi_sysfs_node(HDMI_PLUG_STATUS_NOTIFY_ENABLE);
+            update_ext_disp_sysfs_node(adev, EXT_DISPLAY_PLUG_STATUS_NOTIFY_ENABLE);
         }
-        update_hdmi_sysfs_node(1);
+        update_ext_disp_sysfs_node(adev, 1);
     } else if(str_parms_get_str(parms, "disconnect", value, sizeof(value)) >= 0
-            && (atoi(value) & AUDIO_DEVICE_OUT_HDMI)){
-        //params = "disconnect=1024" for HDMI disconnection.
-        update_hdmi_sysfs_node(0);
+            && (atoi(value) & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
+        //params = "disconnect=1024" for external display disconnection.
+        update_ext_disp_sysfs_node(adev, 0);
     } else {
-        // handle hdmi devices only
+        // handle ext disp devices only
         return;
     }
 }
@@ -755,6 +763,7 @@
    audio_extn_ssr_set_parameters(adev, parms);
    audio_extn_hfp_set_parameters(adev, parms);
    audio_extn_dts_eagle_set_parameters(adev, parms);
+   audio_extn_a2dp_set_parameters(parms);
    audio_extn_ddp_set_parameters(adev, parms);
    audio_extn_ds2_set_parameters(adev, parms);
    audio_extn_customstereo_set_parameters(adev, parms);
@@ -762,7 +771,8 @@
    audio_extn_pm_set_parameters(parms);
    audio_extn_source_track_set_parameters(adev, parms);
    audio_extn_fbsp_set_parameters(parms);
-   check_and_set_hdmi_connection_status(parms);
+   audio_extn_keep_alive_set_parameters(adev, parms);
+   check_and_set_ext_disp_connection_status(adev, parms);
    if (adev->offload_effects_set_parameters != NULL)
        adev->offload_effects_set_parameters(parms);
 }
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index fe3fe95..cd9763e 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -155,7 +155,7 @@
 #define audio_extn_usb_deinit()                                        (0)
 #define audio_extn_usb_add_device(device, card)                        (0)
 #define audio_extn_usb_remove_device(device, card)                     (0)
-#define audio_extn_usb_is_config_supported(bit_width, sample_rate, ch) (0)
+#define audio_extn_usb_is_config_supported(bit_width, sample_rate, ch, pb) (0)
 #define audio_extn_usb_enable_sidetone(device, enable)                 (0)
 #define audio_extn_usb_set_sidetone_gain(parms, value, len)            (0)
 #else
@@ -165,14 +165,36 @@
 void audio_extn_usb_remove_device(audio_devices_t device, int card);
 bool audio_extn_usb_is_config_supported(unsigned int *bit_width,
                                         unsigned int *sample_rate,
-                                        unsigned int ch);
+                                        unsigned int *ch,
+                                        bool is_playback);
 int audio_extn_usb_enable_sidetone(int device, bool enable);
 int audio_extn_usb_set_sidetone_gain(struct str_parms *parms,
                                      char *value, int len);
 #endif
 
+#ifndef SPLIT_A2DP_ENABLED
+#define audio_extn_a2dp_init(adev)                       (0)
+#define audio_extn_a2dp_start_playback()                 (0)
+#define audio_extn_a2dp_stop_playback()                  (0)
+#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)
+
+#else
+void audio_extn_a2dp_init(void *adev);
+int audio_extn_a2dp_start_playback();
+void audio_extn_a2dp_stop_playback();
+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);
+
+#endif
+
 #ifndef SSR_ENABLED
-#define audio_extn_ssr_check_and_set_usecase(in)      (0)
+#define audio_extn_ssr_check_and_set_usecase(in)      (-1)
 #define audio_extn_ssr_init(in, num_out_chan)         (0)
 #define audio_extn_ssr_deinit()                       (0)
 #define audio_extn_ssr_update_enabled()               (0)
@@ -387,6 +409,16 @@
 
 #endif
 
+#ifndef AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
+#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH  0x10000
+#endif
+
+enum {
+    EXT_DISPLAY_TYPE_NONE,
+    EXT_DISPLAY_TYPE_HDMI,
+    EXT_DISPLAY_TYPE_DP
+};
+
 #ifndef HDMI_PASSTHROUGH_ENABLED
 #define audio_extn_passthru_update_stream_configuration(adev, out)            (0)
 #define audio_extn_passthru_is_convert_supported(adev, out)                   (0)
@@ -405,8 +437,6 @@
 #define audio_extn_passthru_set_parameters(a, p) (-ENOSYS)
 #define audio_extn_passthru_init(a) do {} while(0)
 #define audio_extn_passthru_should_standby(o) (1)
-
-#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH  0x1000
 #else
 bool audio_extn_passthru_is_convert_supported(struct audio_device *adev,
                                                  struct stream_out *out);
@@ -434,9 +464,11 @@
 #ifndef HFP_ENABLED
 #define audio_extn_hfp_is_active(adev)                  (0)
 #define audio_extn_hfp_get_usecase()                    (-1)
+#define hfp_set_mic_mute(dev, state)                    (0)
 #else
 bool audio_extn_hfp_is_active(struct audio_device *adev);
 audio_usecase_t audio_extn_hfp_get_usecase();
+int hfp_set_mic_mute(struct audio_device *dev, bool state);
 #endif
 
 #ifndef DEV_ARBI_ENABLED
@@ -556,6 +588,29 @@
 void audio_utils_set_hdmi_channel_status(struct stream_out *out, char * buffer, size_t bytes);
 #endif
 
+#ifdef QAF_EXTN_ENABLED
+bool audio_extn_qaf_is_enabled();
+void audio_extn_qaf_deinit();
+void audio_extn_qaf_close_output_stream(struct audio_hw_device *dev __unused,
+                                        struct audio_stream_out *stream);
+int audio_extn_qaf_open_output_stream(struct audio_hw_device *dev,
+                                   audio_io_handle_t handle,
+                                   audio_devices_t devices,
+                                   audio_output_flags_t flags,
+                                   struct audio_config *config,
+                                   struct audio_stream_out **stream_out,
+                                   const char *address __unused);
+int audio_extn_qaf_init(struct audio_device *adev);
+int audio_extn_qaf_set_parameters(struct audio_device *adev, struct str_parms *parms);
+#else
+#define audio_extn_qaf_is_enabled()                                     (0)
+#define audio_extn_qaf_deinit()                                         (0)
+#define audio_extn_qaf_close_output_stream         adev_close_output_stream
+#define audio_extn_qaf_open_output_stream           adev_open_output_stream
+#define audio_extn_qaf_init(adev)                                       (0)
+#define audio_extn_qaf_set_parameters(adev, parms)                      (0)
+#endif
+
 #ifndef KEEP_ALIVE_ENABLED
 #define audio_extn_keep_alive_init(a) do {} while(0)
 #define audio_extn_keep_alive_start() do {} while(0)
@@ -571,5 +626,53 @@
                                          struct str_parms *parms);
 #endif
 
+#ifndef AUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED
+
+#define audio_extn_gef_init(adev) (0)
+#define audio_extn_gef_deinit() (0)
+#define audio_extn_gef_notify_device_config(devices, cmask, acdb_id) (0)
+#define audio_extn_gef_send_audio_cal(dev, acdb_dev_id, acdb_device_type,\
+    app_type, topology_id, sample_rate, module_id, param_id, data, length, persist) (0)
+#define audio_extn_gef_get_audio_cal(adev, acdb_dev_id, acdb_device_type,\
+    app_type, topology_id, sample_rate, module_id, param_id, data, length, persist) (0)
+#define audio_extn_gef_store_audio_cal(adev, acdb_dev_id, acdb_device_type,\
+    app_type, topology_id, sample_rate, module_id, param_id, data, length) (0)
+#define audio_extn_gef_retrieve_audio_cal(adev, acdb_dev_id, acdb_device_type,\
+    app_type, topology_id, sample_rate, module_id, param_id, data, length) (0)
+
+#else
+
+void audio_extn_gef_init(struct audio_device *adev);
+void audio_extn_gef_deinit();
+
+void audio_extn_gef_notify_device_config(audio_devices_t audio_device,
+    audio_channel_mask_t channel_mask, int acdb_id);
+int audio_extn_gef_send_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int length, bool persist);
+int audio_extn_gef_get_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int* length, bool persist);
+int audio_extn_gef_store_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int length);
+int audio_extn_gef_retrieve_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int* length);
+
+#endif /* AUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED */
+
+typedef void (* snd_mon_cb)(void * stream, struct str_parms * parms);
+#ifndef SND_MONITOR_ENABLED
+#define audio_extn_snd_mon_init()           (0)
+#define audio_extn_snd_mon_deinit()         (0)
+#define audio_extn_snd_mon_register_listener(stream, cb) (0)
+#define audio_extn_snd_mon_unregister_listener(stream) (0)
+#else
+int audio_extn_snd_mon_init();
+int audio_extn_snd_mon_deinit();
+int audio_extn_snd_mon_register_listener(void *stream, snd_mon_cb cb);
+int audio_extn_snd_mon_unregister_listener(void *stream);
+#endif
 
 #endif /* AUDIO_EXTN_H */
diff --git a/hal/audio_extn/dev_arbi.c b/hal/audio_extn/dev_arbi.c
index d7ab5ff..69d8568 100644
--- a/hal/audio_extn/dev_arbi.c
+++ b/hal/audio_extn/dev_arbi.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014, 2016 The Linux Foundation. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are
@@ -131,6 +131,7 @@
         {SND_DEVICE_OUT_VOICE_HANDSET, AUDIO_DEVICE_OUT_EARPIECE},
         {SND_DEVICE_OUT_SPEAKER, AUDIO_DEVICE_OUT_SPEAKER},
         {SND_DEVICE_OUT_VOICE_SPEAKER, AUDIO_DEVICE_OUT_SPEAKER},
+        {SND_DEVICE_OUT_VOICE_SPEAKER_2, AUDIO_DEVICE_OUT_SPEAKER},
         {SND_DEVICE_OUT_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
         {SND_DEVICE_OUT_VOICE_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
         {SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
diff --git a/hal/audio_extn/gef.c b/hal/audio_extn/gef.c
new file mode 100644
index 0000000..d0ccd8c
--- /dev/null
+++ b/hal/audio_extn/gef.c
@@ -0,0 +1,290 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ *   * Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ *   * Redistributions in binary form must reproduce the above
+ *     copyright notice, this list of conditions and the following
+ *     disclaimer in the documentation and/or other materials provided
+ *     with the distribution.
+ *   * Neither the name of The Linux Foundation nor the names of its
+ *     contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "audio_hw_generic_effect"
+//#define LOG_NDEBUG 0
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <math.h>
+#include <cutils/log.h>
+#include <fcntl.h>
+#include <dirent.h>
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <sys/stat.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <math.h>
+#include <cutils/properties.h>
+#include "audio_extn.h"
+#include "audio_hw.h"
+
+#ifdef AUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED
+
+#define GEF_LIBRARY "/system/vendor/lib/libqtigef.so"
+
+typedef void* (*gef_init_t)(void*);
+typedef void (*gef_device_config_cb_t)(void*, audio_devices_t,
+    audio_channel_mask_t, int);
+
+typedef struct {
+    void* handle;
+    void* gef_ptr;
+    gef_init_t init;
+    gef_device_config_cb_t device_config_cb;
+} gef_data;
+
+static gef_data gef_hal_handle;
+
+typedef enum {
+    ASM = 0,
+    ADM
+} gef_calibration_type;
+
+typedef enum {
+    AUDIO_DEVICE_CAL_TYPE = 0,
+    AUDIO_STREAM_CAL_TYPE,
+} acdb_device_type;
+
+
+static acdb_device_type make_acdb_device_type_from_gef_cal_type
+                            (gef_calibration_type gef_cal_type)
+{
+    int acdb_device_type = 0;
+
+    switch (gef_cal_type) {
+        case ASM:
+            acdb_device_type = AUDIO_STREAM_CAL_TYPE;
+            break;
+        case ADM:
+            acdb_device_type = AUDIO_DEVICE_CAL_TYPE;
+            break;
+        default:
+            acdb_device_type = -1;
+            break;
+    }
+
+    return ((int)acdb_device_type);
+}
+
+void audio_extn_gef_init(struct audio_device *adev)
+{
+    int ret = 0;
+    const char* error = NULL;
+
+    ALOGV("%s: Enter with error", __func__);
+
+    memset(&gef_hal_handle, 0, sizeof(gef_data));
+
+    ret = access(GEF_LIBRARY, R_OK);
+    if (ret == 0) {
+        //: check error for dlopen
+        gef_hal_handle.handle = dlopen(GEF_LIBRARY, RTLD_LAZY);
+        if (gef_hal_handle.handle == NULL) {
+            ALOGE("%s: DLOPEN failed for %s with error %s",
+                __func__, GEF_LIBRARY, dlerror());
+            goto ERROR_RETURN;
+        } else {
+            ALOGV("%s: DLOPEN successful for %s", __func__, GEF_LIBRARY);
+
+            //call dlerror to clear the error
+            dlerror();
+            gef_hal_handle.init =
+                (gef_init_t)dlsym(gef_hal_handle.handle, "gef_init");
+            error = dlerror();
+
+            if(error != NULL) {
+                ALOGE("%s: dlsym of %s failed with error %s",
+                     __func__, "gef_init", error);
+                goto ERROR_RETURN;
+            }
+
+            //call dlerror to clear the error
+            error = dlerror();
+            gef_hal_handle.device_config_cb =
+                 (gef_device_config_cb_t)dlsym(gef_hal_handle.handle,
+                 "gef_device_config_cb");
+            error = dlerror();
+
+            if(error != NULL) {
+                ALOGE("%s: dlsym of %s failed with error %s",
+                     __func__, "gef_device_config_cb", error);
+                goto ERROR_RETURN;
+            }
+
+            gef_hal_handle.gef_ptr = gef_hal_handle.init((void*)adev);
+        }
+    } else {
+        ALOGE("%s: %s access failed", __func__, GEF_LIBRARY);
+    }
+
+ERROR_RETURN:
+    ALOGV("%s: Exit with error %d", __func__, ret);
+    return;
+}
+
+
+//this will be called from GEF to exchange calibration using acdb
+int audio_extn_gef_send_audio_cal(void* dev, int acdb_dev_id,
+    int gef_cal_type, int app_type, int topology_id, int sample_rate,
+    uint32_t module_id, uint32_t param_id, void* data, int length, bool persist)
+{
+    int ret = 0;
+    struct audio_device *adev = (struct audio_device*)dev;
+    int acdb_device_type =
+        make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+    ALOGV("%s: Enter", __func__);
+
+    //lock adev
+    pthread_mutex_lock(&adev->lock);
+
+    //send cal
+    ret = platform_send_audio_cal(adev->platform, acdb_dev_id,
+        acdb_device_type, app_type, topology_id, sample_rate,
+        module_id, param_id, data, length, persist);
+
+    pthread_mutex_unlock(&adev->lock);
+
+    ALOGV("%s: Exit with error %d", __func__, ret);
+
+    return ret;
+}
+
+//this will be called from GEF to exchange calibration using acdb
+int audio_extn_gef_get_audio_cal(void* dev, int acdb_dev_id,
+    int gef_cal_type, int app_type, int topology_id, int sample_rate,
+    uint32_t module_id, uint32_t param_id, void* data, int* length, bool persist)
+{
+    int ret = 0;
+    struct audio_device *adev = (struct audio_device*)dev;
+    int acdb_device_type =
+        make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+    ALOGV("%s: Enter", __func__);
+
+    //lock adev
+    pthread_mutex_lock(&adev->lock);
+
+    ret = platform_get_audio_cal(adev->platform, acdb_dev_id,
+        acdb_device_type, app_type, topology_id, sample_rate,
+        module_id, param_id, data, length, persist);
+
+    pthread_mutex_unlock(&adev->lock);
+
+    ALOGV("%s: Exit with error %d", __func__, ret);
+
+    return ret;
+}
+
+//this will be called from GEF to store into acdb
+int audio_extn_gef_store_audio_cal(void* dev, int acdb_dev_id,
+    int gef_cal_type, int app_type, int topology_id, int sample_rate,
+    uint32_t module_id, uint32_t param_id, void* data, int length)
+{
+    int ret = 0;
+    struct audio_device *adev = (struct audio_device*)dev;
+    int acdb_device_type =
+        make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+    ALOGV("%s: Enter", __func__);
+
+    //lock adev
+    pthread_mutex_lock(&adev->lock);
+
+    ret = platform_store_audio_cal(adev->platform, acdb_dev_id,
+        acdb_device_type, app_type, topology_id, sample_rate,
+        module_id, param_id, data, length);
+
+    pthread_mutex_unlock(&adev->lock);
+
+    ALOGV("%s: Exit with error %d", __func__, ret);
+
+    return ret;
+}
+
+//this will be called from GEF to retrieve calibration using acdb
+int audio_extn_gef_retrieve_audio_cal(void* dev, int acdb_dev_id,
+    int gef_cal_type, int app_type, int topology_id, int sample_rate,
+    uint32_t module_id, uint32_t param_id, void* data, int* length)
+{
+    int ret = 0;
+    struct audio_device *adev = (struct audio_device*)dev;
+    int acdb_device_type =
+        make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+    ALOGV("%s: Enter", __func__);
+
+    //lock adev
+    pthread_mutex_lock(&adev->lock);
+
+    ret = platform_retrieve_audio_cal(adev->platform, acdb_dev_id,
+        acdb_device_type, app_type, topology_id, sample_rate,
+        module_id, param_id, data, length);
+
+    pthread_mutex_unlock(&adev->lock);
+
+    ALOGV("%s: Exit with error %d", __func__, ret);
+
+    return ret;
+}
+
+//this will be called from HAL to notify GEF of new device configuration
+void audio_extn_gef_notify_device_config(audio_devices_t audio_device,
+    audio_channel_mask_t channel_mask, int acdb_id)
+{
+    ALOGV("%s: Enter", __func__);
+
+    //call into GEF to share channel mask and device info
+    if (gef_hal_handle.handle && gef_hal_handle.device_config_cb) {
+        gef_hal_handle.device_config_cb(gef_hal_handle.gef_ptr, audio_device, channel_mask,
+            acdb_id);
+    }
+
+    ALOGV("%s: Exit", __func__);
+
+    return;
+}
+
+void audio_extn_gef_deinit()
+{
+    ALOGV("%s: Enter", __func__);
+
+    if (gef_hal_handle.handle) {
+        dlclose(gef_hal_handle.handle);
+    }
+
+    memset(&gef_hal_handle, 0, sizeof(gef_data));
+
+    ALOGV("%s: Exit", __func__);
+}
+
+#endif
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 5a45b80..243d48d 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -302,6 +302,26 @@
         return false;
 }
 
+int hfp_set_mic_mute(struct audio_device *adev, bool state)
+{
+    struct mixer_ctl *ctl;
+    const char *mixer_ctl_name = "HFP TX Mute";
+    uint32_t set_values[ ] = {0};
+
+    ALOGI("%s: enter, state=%d", __func__, state);
+
+    set_values[0] = state;
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+               __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+    mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+    ALOGV("%s: exit", __func__);
+    return 0;
+}
+
 audio_usecase_t audio_extn_hfp_get_usecase()
 {
     return hfpmod.ucid;
diff --git a/hal/audio_extn/keep_alive.c b/hal/audio_extn/keep_alive.c
index 1a4f135..1df6d08 100644
--- a/hal/audio_extn/keep_alive.c
+++ b/hal/audio_extn/keep_alive.c
@@ -36,9 +36,7 @@
 #include "platform_api.h"
 #include <platform.h>
 
-#define SILENCE_MIXER_PATH "silence-playback hdmi"
-#define SILENCE_DEV_ID 32           /* index into machine driver */
-#define SILENCE_INTERVAL_US 2000000
+#define SILENCE_INTERVAL 2 /*In secs*/
 
 typedef enum {
     STATE_DEINIT = -1,
@@ -52,7 +50,9 @@
 
 typedef struct {
     pthread_mutex_t lock;
+    pthread_mutex_t sleep_lock;
     pthread_cond_t  cond;
+    pthread_cond_t  wake_up_cond;
     pthread_t thread;
     state_t state;
     struct listnode cmd_list;
@@ -88,6 +88,8 @@
     ka.pcm = NULL;
     pthread_mutex_init(&ka.lock, (const pthread_mutexattr_t *) NULL);
     pthread_cond_init(&ka.cond, (const pthread_condattr_t *) NULL);
+    pthread_cond_init(&ka.wake_up_cond, (const pthread_condattr_t *) NULL);
+    pthread_mutex_init(&ka.sleep_lock, (const pthread_mutexattr_t *) NULL);
     list_init(&ka.cmd_list);
     if (pthread_create(&ka.thread,  (const pthread_attr_t *) NULL,
                        keep_alive_loop, NULL) < 0) {
@@ -127,11 +129,14 @@
     if (ka.pcm)
         return -EEXIST;
 
-    ALOGD("opening silence device %d", SILENCE_DEV_ID);
+    int silence_pcm_dev_id = platform_get_pcm_device_id(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE,
+                                                        PCM_PLAYBACK);
+
+    ALOGD("opening silence device %d", silence_pcm_dev_id);
     struct audio_device * adev = (struct audio_device *)ka.userdata;
-    ka.pcm = pcm_open(adev->snd_card, SILENCE_DEV_ID,
+    ka.pcm = pcm_open(adev->snd_card, silence_pcm_dev_id,
                       flags, &silence_config);
-    ALOGD("opened silence device %d", SILENCE_DEV_ID);
+    ALOGD("opened silence device %d", silence_pcm_dev_id);
     if (ka.pcm == NULL || !pcm_is_ready(ka.pcm)) {
         ALOGE("%s: %s", __func__, pcm_get_error(ka.pcm));
         if (ka.pcm != NULL) {
@@ -143,6 +148,27 @@
     return 0;
 }
 
+
+static int set_mixer_control(struct mixer *mixer,
+                             const char * mixer_ctl_name,
+                             const char *mixer_val)
+{
+    struct mixer_ctl *ctl;
+    if ((mixer == NULL) || (mixer_ctl_name == NULL) || (mixer_val == NULL)) {
+       ALOGE("%s: Invalid input", __func__);
+       return -EINVAL;
+    }
+    ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
+    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+
+    return mixer_ctl_set_enum_by_string(ctl, mixer_val);
+}
+
 /* must be called with adev lock held */
 void audio_extn_keep_alive_start()
 {
@@ -151,18 +177,20 @@
     int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT], len = 0, rc;
     struct mixer_ctl *ctl;
     int acdb_dev_id, snd_device;
+    struct listnode *node;
+    struct audio_usecase *usecase;
     int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
 
     pthread_mutex_lock(&ka.lock);
 
     if (ka.state == STATE_DEINIT) {
         ALOGE(" %s : Invalid state ",__func__);
-        return;
+        goto exit;
     }
 
     if (audio_extn_passthru_is_active()) {
         ALOGE(" %s : Pass through is already active", __func__);
-        return;
+        goto exit;
     }
 
     if (ka.state == STATE_ACTIVE) {
@@ -170,11 +198,21 @@
         goto exit;
     }
 
+    /* Dont start keep_alive if any other PCM session is routed to HDMI*/
+    list_for_each(node, &adev->usecase_list) {
+         usecase = node_to_item(node, struct audio_usecase, list);
+         if (usecase->type == PCM_PLAYBACK &&
+                 usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+             goto exit;
+    }
+
     ka.done = false;
 
     /*configure app type */
+    int silence_pcm_dev_id = platform_get_pcm_device_id(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE,
+                                                        PCM_PLAYBACK);
     snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
-             "Audio Stream %d App Type Cfg",SILENCE_DEV_ID);
+             "Audio Stream %d App Type Cfg", silence_pcm_dev_id);
 
     ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
     if (!ctl) {
@@ -184,7 +222,28 @@
         goto exit;
     }
 
-    snd_device = SND_DEVICE_OUT_HDMI;
+    /* Configure HDMI/DP Backend with default values, this as well
+     * helps reconfigure HDMI/DP backend after passthrough.
+     */
+    int ext_disp_type = platform_get_ext_disp_type(adev->platform);
+    switch(ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            snd_device = SND_DEVICE_OUT_HDMI;
+            set_mixer_control(adev->mixer, "HDMI RX Format", "LPCM");
+            set_mixer_control(adev->mixer, "HDMI_RX SampleRate", "KHZ_48");
+            set_mixer_control(adev->mixer, "HDMI_RX Channels", "Two");
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            snd_device = SND_DEVICE_OUT_DISPLAY_PORT;
+            set_mixer_control(adev->mixer, "Display Port Format", "LPCM");
+            set_mixer_control(adev->mixer, "Display Port RX SampleRate", "KHZ_48");
+            set_mixer_control(adev->mixer, "Display Port RX Channels", "Two");
+            break;
+        default:
+            ALOGE("%s: Invalid external display type:%d", __func__, ext_disp_type);
+            goto exit;
+    }
+
     acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
     if (acdb_dev_id < 0) {
         ALOGE("%s: Couldn't get the acdb dev id", __func__);
@@ -204,15 +263,25 @@
     mixer_ctl_set_array(ctl, app_type_cfg, len);
 
     /*send calibration*/
-    struct audio_usecase *usecase = calloc(1, sizeof(struct audio_usecase));
+    usecase = calloc(1, sizeof(struct audio_usecase));
     usecase->type = PCM_PLAYBACK;
-    usecase->out_snd_device = SND_DEVICE_OUT_HDMI;
+    usecase->out_snd_device = snd_device;
 
     platform_send_audio_calibration(adev->platform, usecase,
                 platform_get_default_app_type(adev->platform), sample_rate);
 
     /*apply audio route */
-    audio_route_apply_and_update_path(adev->audio_route, SILENCE_MIXER_PATH);
+    switch(ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            audio_route_apply_and_update_path(adev->audio_route, "silence-playback hdmi");
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            audio_route_apply_and_update_path(adev->audio_route, "silence-playback display-port");
+            break;
+        default:
+            ALOGE("%s: Invalid external display type:%d", __func__, ext_disp_type);
+            goto exit;
+    }
 
     if (open_silence_stream() == 0) {
         send_cmd_l(REQUEST_WRITE);
@@ -232,18 +301,30 @@
 
     pthread_mutex_lock(&ka.lock);
 
-    if (ka.state == STATE_DEINIT)
-        return;
-
-    if (ka.state == STATE_IDLE)
+    if ((ka.state == STATE_DEINIT) || (ka.state == STATE_IDLE))
         goto exit;
 
+    pthread_mutex_lock(&ka.sleep_lock);
     ka.done = true;
+    pthread_cond_signal(&ka.wake_up_cond);
+    pthread_mutex_unlock(&ka.sleep_lock);
     while (ka.state != STATE_IDLE) {
         pthread_cond_wait(&ka.cond, &ka.lock);
     }
     close_silence_stream();
-    audio_route_reset_and_update_path(adev->audio_route, SILENCE_MIXER_PATH);
+
+    /*apply audio route */
+    int ext_disp_type = platform_get_ext_disp_type(adev->platform);
+    switch(ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            audio_route_reset_and_update_path(adev->audio_route, "silence-playback hdmi");
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            audio_route_reset_and_update_path(adev->audio_route, "silence-playback display-port");
+            break;
+        default:
+            ALOGE("%s: Invalid external display type:%d", __func__, ext_disp_type);
+    }
 
 exit:
     pthread_mutex_unlock(&ka.lock);
@@ -290,6 +371,7 @@
     struct listnode *item;
     uint8_t * silence = NULL;
     int32_t bytes = 0;
+    struct timespec ts;
 
     while (true) {
         pthread_mutex_lock(&ka.lock);
@@ -328,9 +410,17 @@
              * Just something to keep the connection alive is sufficient.
              * Hence a short burst of silence periodically.
              */
-            usleep(SILENCE_INTERVAL_US);
-        }
+            pthread_mutex_lock(&ka.sleep_lock);
+            clock_gettime(CLOCK_REALTIME, &ts);
+            ts.tv_sec += SILENCE_INTERVAL;
+            ts.tv_nsec = 0;
 
+            if (!ka.done)
+              pthread_cond_timedwait(&ka.wake_up_cond,
+                            &ka.sleep_lock, &ts);
+
+            pthread_mutex_unlock(&ka.sleep_lock);
+        }
         pthread_mutex_lock(&ka.lock);
         ka.state = STATE_IDLE;
         pthread_cond_signal(&ka.cond);
diff --git a/hal/audio_extn/passthru.c b/hal/audio_extn/passthru.c
index e6ac4dd..eaa8c0a 100644
--- a/hal/audio_extn/passthru.c
+++ b/hal/audio_extn/passthru.c
@@ -82,8 +82,14 @@
  */
 bool audio_extn_passthru_should_drop_data(struct stream_out * out)
 {
-
-    if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+    /*Drop data only
+     *stream is routed to HDMI and
+     *stream has PCM format or
+     *if a compress offload (DSP decode) session
+     */
+    if ((out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) &&
+        (((out->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) ||
+        ((out->compr_config.codec != NULL) && (out->compr_config.codec->compr_passthr == LEGACY_PCM)))) {
         if (android_atomic_acquire_load(&compress_passthru_active) > 0) {
             ALOGI("drop data as pass thru is active");
             return true;
@@ -112,9 +118,6 @@
     ALOGV("inc pass thru count to notify other streams");
     android_atomic_inc(&compress_passthru_active);
 
-    ALOGV("keep_alive_stop");
-    audio_extn_keep_alive_stop();
-
     while (true) {
         /* find max period time among active playback use cases */
         list_for_each(node, &adev->usecase_list) {
diff --git a/hal/audio_extn/qaf.c b/hal/audio_extn/qaf.c
new file mode 100644
index 0000000..d631275
--- /dev/null
+++ b/hal/audio_extn/qaf.c
@@ -0,0 +1,1129 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ *     * Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *     * Redistributions in binary form must reproduce the above
+ *       copyright notice, this list of conditions and the following
+ *       disclaimer in the documentation and/or other materials provided
+ *       with the distribution.
+ *     * Neither the name of The Linux Foundation nor the names of its
+ *       contributors may be used to endorse or promote products derived
+ *       from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "audio_hw_qaf"
+/*#define LOG_NDEBUG 0*/
+/*#define VERY_VERY_VERBOSE_LOGGING*/
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define QAF_DEFAULT_COMPR_AUDIO_HANDLE 1001
+#define QAF_DEFAULT_COMPR_PASSTHROUGH_HANDLE 1002
+
+#include <stdlib.h>
+#include <pthread.h>
+#include <errno.h>
+#include <dlfcn.h>
+#include <sys/resource.h>
+#include <sys/prctl.h>
+#include <cutils/properties.h>
+#include <cutils/str_parms.h>
+#include <cutils/log.h>
+#include <cutils/atomic.h>
+#include "audio_utils/primitives.h"
+#include "audio_hw.h"
+#include "platform_api.h"
+#include <platform.h>
+#include <system/thread_defs.h>
+#include <cutils/sched_policy.h>
+#include "audio_extn.h"
+#include <qti_audio.h>
+#include "sound/compress_params.h"
+
+#define QAF_OUTPUT_SAMPLING_RATE 48000
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
+#define QAF_PLAYBACK_LATENCY 30
+
+#define QAF_LATENCY (COMPRESS_OFFLOAD_PLAYBACK_LATENCY + QAF_PLAYBACK_LATENCY)
+
+#ifdef QAF_DUMP_ENABLED
+FILE *fp_output_writer_hdmi = NULL;
+#endif
+
+typedef enum {
+AUDIO_OUTPUT_FLAG_MAIN = 0x4000, // Flag for Main Input Stream
+AUDIO_OUTPUT_FLAG_ASSOCIATED = 0x8000, // Flag for Assocated Input Stream
+} qaf_audio_output_flags_t;
+
+struct qaf {
+    struct audio_device *adev;
+    audio_session_handle_t session_handle;
+    void *qaf_lib;
+    int (*qaf_audio_session_open)(audio_session_handle_t* session_handle, void *p_data, void* license_data);
+    int (*qaf_audio_session_close)(audio_session_handle_t session_handle);
+    int (*qaf_audio_stream_open)(audio_session_handle_t session_handle, audio_stream_handle_t* stream_handle,
+         audio_stream_config_t input_config, audio_devices_t devices, stream_type_t flags);
+    int (*qaf_audio_stream_close)(audio_stream_handle_t stream_handle);
+    int (*qaf_audio_stream_set_param)(audio_stream_handle_t stream_handle, const char* kv_pairs);
+    int (*qaf_audio_session_set_param)(audio_session_handle_t handle, const char* kv_pairs);
+    char* (*qaf_audio_stream_get_param)(audio_stream_handle_t stream_handle, const char* key);
+    char* (*qaf_audio_session_get_param)(audio_session_handle_t handle, const char* key);
+    int (*qaf_audio_stream_start)(audio_stream_handle_t handle);
+    int (*qaf_audio_stream_stop)(audio_stream_handle_t stream_handle);
+    int (*qaf_audio_stream_pause)(audio_stream_handle_t stream_handle);
+    int (*qaf_audio_stream_flush)(audio_stream_handle_t stream_handle);
+    int (*qaf_audio_stream_write)(audio_stream_handle_t stream_handle, const void* buf, int size);
+    void (*qaf_register_event_callback)(audio_session_handle_t session_handle, void *priv_data,
+          notify_event_callback_t event_callback, audio_event_id_t event_id);
+    pthread_mutex_t lock;
+    struct stream_out *stream_drain_main;
+    struct stream_out *qaf_compr_offload_out;
+    struct stream_out *qaf_compr_passthrough_out;
+    int passthrough_enabled;
+    int hdmi_sink_channels;
+    bool multi_ch_out_enabled;
+    bool main_output_active;
+    bool assoc_output_active;
+};
+
+static struct qaf *qaf_mod = NULL;
+
+static void lock_output_stream(struct stream_out *out)
+{
+    pthread_mutex_lock(&out->pre_lock);
+    pthread_mutex_lock(&out->lock);
+    pthread_mutex_unlock(&out->pre_lock);
+}
+
+static int qaf_send_offload_cmd_l(struct stream_out* out, int command)
+{
+    struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
+
+    if (!cmd) {
+        ALOGE("failed to allocate mem for command 0x%x", command);
+        return -ENOMEM;
+    }
+
+    ALOGV("%s %d", __func__, command);
+
+    cmd->cmd = command;
+    list_add_tail(&out->qaf_offload_cmd_list, &cmd->node);
+    pthread_cond_signal(&out->qaf_offload_cond);
+    return 0;
+}
+
+static int audio_extn_qaf_stream_stop(struct stream_out *out)
+{
+    ALOGV("%s: %d start", __func__, __LINE__);
+    if (!qaf_mod->qaf_audio_stream_stop)
+        return -EINVAL;
+
+    return qaf_mod->qaf_audio_stream_stop(out->qaf_stream_handle);
+}
+
+static int qaf_out_standby(struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int status = 0;
+
+    ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
+          stream, out->usecase, use_case_table[out->usecase]);
+
+    lock_output_stream(out);
+    if (!out->standby) {
+        out->standby = true;
+        status = audio_extn_qaf_stream_stop(out);
+    }
+    pthread_mutex_unlock(&out->lock);
+    out->written = 0;
+    return status;
+}
+
+static int qaf_stream_set_param(struct stream_out *out, const char *kv_pair)
+{
+    ALOGV("%s %d kvpair: %s", __func__, __LINE__, kv_pair);
+    if (!qaf_mod->qaf_audio_stream_set_param)
+        return -EINVAL;
+
+    return qaf_mod->qaf_audio_stream_set_param(out->qaf_stream_handle, kv_pair);
+}
+
+static int qaf_out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int ret = 0;
+
+    ALOGV("%s: enter: usecase(%d: %s) kvpairs: %s",
+          __func__, out->usecase, use_case_table[out->usecase], kvpairs);
+    lock_output_stream(out);
+    ret = qaf_stream_set_param(out, kvpairs);
+    pthread_mutex_unlock(&out->lock);
+    if ((NULL != qaf_mod->qaf_compr_offload_out)) {
+        qaf_mod->qaf_compr_offload_out->stream.common.set_parameters((struct audio_stream *) qaf_mod->qaf_compr_offload_out, kvpairs);
+    }
+    return ret;
+}
+
+static int qaf_write_input_buffer(struct stream_out *out, const void *buffer, int bytes)
+{
+    int ret = 0;
+    ALOGVV("%s bytes = %d [%p]", __func__, bytes, out->qaf_stream_handle);
+    if (!qaf_mod->qaf_audio_stream_write)
+        return -EINVAL;
+
+    if (out->qaf_stream_handle)
+        ret = qaf_mod->qaf_audio_stream_write(out->qaf_stream_handle, buffer, bytes);
+    return ret;
+}
+
+static int qaf_out_set_volume(struct audio_stream_out *stream __unused, float left,
+                          float right)
+{
+    if (qaf_mod->qaf_compr_offload_out != NULL) {
+        return qaf_mod->qaf_compr_offload_out->stream.set_volume(
+            (struct audio_stream_out *)qaf_mod->qaf_compr_offload_out, left, right);
+    }
+    return -ENOSYS;
+}
+
+static int qaf_stream_start(struct stream_out *out)
+{
+    if (!qaf_mod->qaf_audio_stream_start)
+        return -EINVAL;
+
+    return qaf_mod->qaf_audio_stream_start(out->qaf_stream_handle);
+}
+
+static int qaf_start_output_stream(struct stream_out *out)
+{
+    int ret = 0;
+    struct audio_device *adev = out->dev;
+    int snd_card_status = get_snd_card_state(adev);
+
+    if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) {
+        ret = -EINVAL;
+        usleep(50000);
+        return ret;
+    }
+
+    ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)",
+          __func__, &out->stream, out->usecase, use_case_table[out->usecase],
+          out->devices);
+
+    if (SND_CARD_STATE_OFFLINE == snd_card_status) {
+        ALOGE("%s: sound card is not active/SSR returning error", __func__);
+        ret = -EIO;
+        usleep(50000);
+        return ret;
+    }
+
+    return qaf_stream_start(out);
+}
+
+static ssize_t qaf_out_write(struct audio_stream_out *stream, const void *buffer,
+                         size_t bytes)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    struct audio_device *adev = out->dev;
+    ssize_t ret = 0;
+
+    ALOGV("qaf_out_write bytes = %d, usecase[%d] and flags[%x] for handle[%p]",(int)bytes, out->usecase, out->flags, out);
+    lock_output_stream(out);
+
+    if (out->standby) {
+        out->standby = false;
+        pthread_mutex_lock(&adev->lock);
+        ret = qaf_start_output_stream(out);
+        pthread_mutex_unlock(&adev->lock);
+        /* ToDo: If use case is compress offload should return 0 */
+        if (ret != 0) {
+            out->standby = true;
+            goto exit;
+        }
+    }
+
+    if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
+        audio_utils_set_hdmi_channel_status(out, (char *)buffer, bytes);
+        adev->is_channel_status_set = true;
+    }
+
+    ret = qaf_write_input_buffer(out, buffer, bytes);
+    ALOGV("%s, ret [%d] ", __func__, (int)ret);
+    if (ret < 0) {
+        goto exit;
+    }
+    out->written += bytes / ((popcount(out->channel_mask) * sizeof(short)));
+
+exit:
+
+    pthread_mutex_unlock(&out->lock);
+
+    if (ret < 0) {
+        if (ret == -EAGAIN) {
+            ALOGV("No space available in ms12 driver, post msg to cb thread");
+            lock_output_stream(out);
+            ret = qaf_send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
+            pthread_mutex_unlock(&out->lock);
+            bytes = 0;
+        }
+        if(ret == -ENOMEM || ret == -EPERM){
+            if (out->pcm)
+                ALOGE("%s: error %d, %s", __func__, (int)ret, pcm_get_error(out->pcm));
+            qaf_out_standby(&out->stream.common);
+            usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
+                            out->stream.common.get_sample_rate(&out->stream.common));
+        }
+    }
+    return bytes;
+}
+
+static int qaf_get_timestamp(struct stream_out *out, uint64_t *frames, struct timespec *timestamp)
+{
+    int ret = 0;
+    struct str_parms *parms;
+    int value = 0;
+    int signed_frames = 0;
+    const char* kvpairs = NULL;
+
+    ALOGV("%s out->format %d", __func__, out->format);
+    if(out->format & AUDIO_FORMAT_PCM_16_BIT) {
+       *frames = out->written;
+       signed_frames = out->written - (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+       // It would be unusual for this value to be negative, but check just in case ...
+       if (signed_frames >= 0) {
+           *frames = signed_frames;
+       }
+       clock_gettime(CLOCK_MONOTONIC, timestamp);
+    } else if (qaf_mod->qaf_audio_stream_get_param) {
+        kvpairs = qaf_mod->qaf_audio_stream_get_param(out->qaf_stream_handle, "position");
+        if (kvpairs) {
+            parms = str_parms_create_str(kvpairs);
+            ret = str_parms_get_int(parms, "position", &value);
+            if (ret >= 0) {
+                *frames = value;
+                signed_frames = value - (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+                // It would be unusual for this value to be negative, but check just in case ...
+                if (signed_frames >= 0) {
+                    *frames = signed_frames;
+                }
+                clock_gettime(CLOCK_MONOTONIC, timestamp);
+            }
+            str_parms_destroy(parms);
+        }
+    } else {
+        ret = -EINVAL;
+    }
+    return ret;
+}
+
+static int qaf_out_get_presentation_position(const struct audio_stream_out *stream,
+                                   uint64_t *frames, struct timespec *timestamp)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int ret = -1;
+    lock_output_stream(out);
+    ret = qaf_get_timestamp(out, frames, timestamp);
+    pthread_mutex_unlock(&out->lock);
+
+    return ret;
+}
+
+static int qaf_stream_pause(struct stream_out *out)
+{
+    ALOGV("%s: %d start", __func__, __LINE__);
+    if (!qaf_mod->qaf_audio_stream_pause)
+        return -EINVAL;
+
+    return qaf_mod->qaf_audio_stream_pause(out->qaf_stream_handle);
+}
+
+static int qaf_out_pause(struct audio_stream_out* stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int status = -ENOSYS;
+    ALOGE("%s", __func__);
+    lock_output_stream(out);
+    status = qaf_stream_pause(out);
+    pthread_mutex_unlock(&out->lock);
+    return status;
+}
+
+static int qaf_out_resume(struct audio_stream_out* stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int status = -ENOSYS;
+    ALOGD("%s", __func__);
+    lock_output_stream(out);
+    status = qaf_stream_start(out);
+    pthread_mutex_unlock(&out->lock);
+    ALOGD("%s Exit", __func__);
+    return status;
+}
+
+static int qaf_out_drain(struct audio_stream_out* stream, audio_drain_type_t type __unused )
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int status = 0;
+    ALOGV("%s stream_handle = %p , format = %x", __func__, out->qaf_stream_handle, out->format);
+    lock_output_stream(out);
+    if (out->offload_callback && out->qaf_stream_handle) {
+        /* Stream stop will trigger EOS and on EOS_EVENT received
+           from callback DRAIN_READY command is sent */
+        status = audio_extn_qaf_stream_stop(out);
+        if (out->format != AUDIO_FORMAT_PCM_16_BIT)
+            qaf_mod->stream_drain_main = out;
+    }
+    pthread_mutex_unlock(&out->lock);
+    return status;
+}
+
+static int audio_extn_qaf_stream_flush(struct stream_out *out)
+{
+    ALOGV("%s: %d exit", __func__, __LINE__);
+    if (!qaf_mod->qaf_audio_stream_flush)
+        return -EINVAL;
+
+    return qaf_mod->qaf_audio_stream_flush(out->qaf_stream_handle);
+}
+
+static int qaf_out_flush(struct audio_stream_out* stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    ALOGV("%s", __func__);
+    int status = -ENOSYS;
+    lock_output_stream(out);
+    status = audio_extn_qaf_stream_flush(out);
+    pthread_mutex_unlock(&out->lock);
+    ALOGV("%s Exit", __func__);
+    return status;
+}
+
+static uint32_t qaf_out_get_latency(const struct audio_stream_out *stream __unused)
+{
+    uint32_t latency = 0;
+
+    latency = QAF_LATENCY;
+    ALOGV("%s: Latency %d", __func__, latency);
+    return latency;
+}
+
+static void notify_event_callback(audio_session_handle_t session_handle __unused, void *prv_data, void *buf, audio_event_id_t event_id, int size, int device)
+{
+
+/*
+ For SPKR:
+ 1. Open pcm device if device_id passed to it SPKR and write the data to pcm device
+
+ For HDMI
+ 1.Open compress device for HDMI(PCM or AC3) based on current_hdmi_output_format
+ 2.create offload_callback thread to receive async events
+ 3.Write the data to compress device. If not all the data is consumed by the driver,
+   add a command to offload_callback thread.
+*/
+    int ret;
+    audio_output_flags_t flags;
+    struct qaf* qaf_module = (struct qaf* ) prv_data;
+    ALOGV("%s device 0x%X, %d in event = %d", __func__, device, __LINE__, event_id);
+
+    if (event_id == AUDIO_DATA_EVENT) {
+        ALOGVV("Device id %x %s %d, bytes to written %d", device, __func__,__LINE__, size);
+
+        pthread_mutex_lock(&qaf_module->lock);
+        if ((device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DD)) ||
+            (device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DDP))) {
+
+            if (NULL == qaf_mod->qaf_compr_passthrough_out) {
+                struct audio_config config;
+                audio_devices_t devices;
+
+                if (qaf_mod->qaf_compr_offload_out) {
+                    adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+                                                 (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+                    qaf_mod->qaf_compr_offload_out = NULL;
+                }
+
+                config.sample_rate = config.offload_info.sample_rate = QAF_OUTPUT_SAMPLING_RATE;
+                config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+                config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+
+                if (device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DDP))
+                    config.format = config.offload_info.format = AUDIO_FORMAT_E_AC3;
+                else
+                    config.format = config.offload_info.format = AUDIO_FORMAT_AC3;
+
+                config.offload_info.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+                config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+                flags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+                devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+
+                ret = adev_open_output_stream((struct audio_hw_device *) qaf_mod->adev, QAF_DEFAULT_COMPR_PASSTHROUGH_HANDLE, devices,
+                                           flags, &config, (struct audio_stream_out **) &(qaf_mod->qaf_compr_passthrough_out), NULL);
+                if (ret < 0) {
+                    ALOGE("%s: adev_open_output_stream failed with ret = %d!", __func__, ret);
+                    pthread_mutex_unlock(&qaf_module->lock);
+                    return;
+                }
+            }
+
+            if (!qaf_mod->passthrough_enabled)
+                qaf_mod->passthrough_enabled = 1;
+
+            ret = qaf_mod->qaf_compr_passthrough_out->stream.write((struct audio_stream_out *) qaf_mod->qaf_compr_passthrough_out, buf, size);
+        } else {
+            if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL && !qaf_mod->multi_ch_out_enabled) {
+                if (qaf_mod->qaf_compr_offload_out) {
+                    adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+                                                 (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+                    qaf_mod->qaf_compr_offload_out = NULL;
+                }
+                qaf_mod->multi_ch_out_enabled = 1;
+            } else if (device == AUDIO_DEVICE_OUT_SPEAKER && qaf_mod->multi_ch_out_enabled) {
+                if (qaf_mod->qaf_compr_offload_out) {
+                    adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+                                                 (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+                    qaf_mod->qaf_compr_offload_out = NULL;
+                }
+                qaf_mod->multi_ch_out_enabled = 0;
+            }
+
+            if (NULL == qaf_mod->qaf_compr_offload_out) {
+                struct audio_config config;
+                audio_devices_t devices;
+
+                config.sample_rate = config.offload_info.sample_rate = QAF_OUTPUT_SAMPLING_RATE;
+                config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+                config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+                config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT_OFFLOAD;
+                config.offload_info.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+                config.format = AUDIO_FORMAT_PCM_16_BIT_OFFLOAD;
+                devices = AUDIO_DEVICE_NONE;
+
+                if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+                    if (qaf_mod->hdmi_sink_channels == 8) {
+                        config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_7POINT1;
+                    } else if (qaf_mod->hdmi_sink_channels == 6) {
+                        config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+                    } else {
+                        config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+                    }
+                    devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+                } else {
+                    config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+                    qaf_mod->multi_ch_out_enabled = 0;
+                }
+                flags = AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+
+                /* TODO:: Need to Propagate errors to framework */
+                ret = adev_open_output_stream((struct audio_hw_device *) qaf_mod->adev, QAF_DEFAULT_COMPR_AUDIO_HANDLE, devices,
+                                             flags, &config, (struct audio_stream_out **) &(qaf_mod->qaf_compr_offload_out), NULL);
+                if (ret < 0) {
+                    ALOGE("%s: adev_open_output_stream failed with ret = %d!", __func__, ret);
+                    pthread_mutex_unlock(&qaf_module->lock);
+                    return;
+                }
+            }
+
+            if (qaf_mod->passthrough_enabled) {
+                qaf_mod->passthrough_enabled = 0;
+                if (qaf_mod->qaf_compr_passthrough_out) {
+                    adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+                                                 (struct audio_stream_out *) (qaf_mod->qaf_compr_passthrough_out));
+                    qaf_mod->qaf_compr_passthrough_out = NULL;
+                }
+            }
+
+            /*
+             * TODO:: Since this is mixed data,
+             * need to identify to which stream the error should be sent
+             */
+            ret = qaf_mod->qaf_compr_offload_out->stream.write((struct audio_stream_out *) qaf_mod->qaf_compr_offload_out, buf, size);
+        }
+
+        ALOGVV("%s:%d stream write ret = %d for out handle[%p]", __func__, __LINE__, ret, qaf_mod->qaf_compr_offload_out);
+        pthread_mutex_unlock(&qaf_module->lock);
+    } else if (event_id == AUDIO_EOS_MAIN_DD_DDP_EVENT || event_id == AUDIO_EOS_MAIN_AAC_EVENT) {
+        /* TODO:: Only MAIN Stream EOS Event is added, need to add ASSOC stream EOS Event */
+        struct stream_out *out = qaf_module->stream_drain_main;
+        if (out != NULL) {
+            lock_output_stream(out);
+            out->offload_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out->offload_cookie);
+            pthread_mutex_unlock(&out->lock);
+            qaf_module->stream_drain_main = NULL;
+            ALOGV("%s %d sent DRAIN_READY", __func__, __LINE__);
+        }
+    }
+    ALOGV("%s %d", __func__, __LINE__);
+}
+
+static int qaf_session_close()
+{
+    ALOGV("%s %d", __func__, __LINE__);
+    if (qaf_mod != NULL) {
+        if (!qaf_mod->qaf_audio_session_close)
+            return -EINVAL;
+
+        qaf_mod->qaf_audio_session_close(qaf_mod->session_handle);
+        qaf_mod->session_handle = NULL;
+        pthread_mutex_destroy(&qaf_mod->lock);
+    }
+    return 0;
+}
+
+static int qaf_stream_close(struct stream_out *out)
+{
+    int ret = 0;
+    ALOGV( "%s %d", __func__, __LINE__);
+    if (!qaf_mod->qaf_audio_stream_close)
+        return -EINVAL;
+    if (out->qaf_stream_handle) {
+        ALOGV( "%s %d output active flag is %x and stream handle %p", __func__, __LINE__, out->flags, out->qaf_stream_handle);
+        if ((out->flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) && (out->flags & AUDIO_OUTPUT_FLAG_MAIN)) { /* Close for Stream with Main and Associated Content*/
+            qaf_mod->main_output_active = false;
+            qaf_mod->assoc_output_active = false;
+        } else if (out->flags & AUDIO_OUTPUT_FLAG_MAIN) {/*Close for Main Stream*/
+            qaf_mod->main_output_active = false;
+            qaf_mod->assoc_output_active = false; /* TODO to remove resetting associated stream active flag when main stream is closed*/
+        } else if (out->flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) { /*Close for Associated Stream*/
+            qaf_mod->assoc_output_active = false;
+        } else { /*Close for Local Playback*/
+            qaf_mod->main_output_active = false;
+        }
+        ret = qaf_mod->qaf_audio_stream_close(out->qaf_stream_handle);
+        out->qaf_stream_handle = NULL;
+    }
+    ALOGV( "%s %d", __func__, __LINE__);
+    return ret;
+}
+
+static int qaf_stream_open(struct stream_out *out, struct audio_config *config, audio_output_flags_t flags, audio_devices_t devices)
+{
+    int status = 0;
+    ALOGV("%s %d", __func__, __LINE__);
+
+    if (!qaf_mod->qaf_audio_stream_open)
+        return -EINVAL;
+
+    audio_stream_config_t input_config;
+    input_config.sample_rate = config->sample_rate;
+    input_config.channel_mask = config->channel_mask;
+    input_config.format = config->format;
+
+    if ((config->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) {
+        input_config.format = AUDIO_FORMAT_AAC;
+    } else if((config->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS) {
+        input_config.format = AUDIO_FORMAT_AAC_ADTS;
+    }
+
+    ALOGV("%s %d audio_stream_open sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x) format(%#x)\
+      ",__func__, __LINE__, input_config.sample_rate, input_config.channel_mask, devices, flags, input_config.format);
+
+    /* TODO to send appropriated flags when support for system tones is added */
+    if (input_config.format == AUDIO_FORMAT_PCM_16_BIT) {
+        status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_SYSTEM_TONE);
+    } else if (input_config.format == AUDIO_FORMAT_AC3 ||
+               input_config.format == AUDIO_FORMAT_E_AC3 ||
+               input_config.format == AUDIO_FORMAT_AAC ||
+               input_config.format == AUDIO_FORMAT_AAC_ADTS) {
+        if (qaf_mod->main_output_active == false) {
+            if ((flags & AUDIO_OUTPUT_FLAG_MAIN) && (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
+                status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+                if (status == 0) {
+                    ALOGV("%s %d Open stream for Input with both Main and Associated stream contents with flag [%x] and stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+                    qaf_mod->main_output_active = true;
+                    qaf_mod->assoc_output_active = true;
+                }
+            } else if (flags & AUDIO_OUTPUT_FLAG_MAIN) {
+                status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+                if (status == 0) {
+                    ALOGV("%s %d Open stream for Input with only Main flag [%x] stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+                    qaf_mod->main_output_active = true;
+                }
+            } else if (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) {
+                ALOGE("%s %d Error main input is not active", __func__, __LINE__);
+                return -EINVAL;
+            } else {
+                status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+                if (status == 0) {
+                    ALOGV("%s %d Open stream for Local playback with flag [%x] stream handle [%p] ", __func__, __LINE__, flags, out->qaf_stream_handle);
+                    qaf_mod->main_output_active = true;
+                }
+            }
+        } else {
+            if (flags & AUDIO_OUTPUT_FLAG_MAIN) {
+                ALOGE("%s %d Error main input is already active", __func__, __LINE__);
+                return -EINVAL;
+            } else if (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) {
+                if (qaf_mod->assoc_output_active) {
+                    ALOGE("%s %d Error assoc input is already active", __func__, __LINE__);
+                    return -EINVAL;
+                } else {
+                    status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_ASSOCIATED);
+                    if (status == 0) {
+                        ALOGV("%s %d Open stream for Input with only Associated flag [%x] stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+                        qaf_mod->assoc_output_active = true;
+                    }
+                }
+            } else {
+                ALOGE("%s %d Error main input is already active", __func__, __LINE__);
+                return -EINVAL;
+            }
+        }
+    }
+
+    return status;
+}
+
+static int qaf_deinit()
+{
+    ALOGV("%s %d", __func__, __LINE__);
+    if (qaf_mod != NULL) {
+        if (qaf_mod->qaf_compr_offload_out != NULL)
+            adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev, (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+        if (qaf_mod->qaf_compr_passthrough_out != NULL)
+            adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev, (struct audio_stream_out *) (qaf_mod->qaf_compr_passthrough_out));
+
+        if (qaf_mod->qaf_lib != NULL) {
+            dlclose(qaf_mod->qaf_lib);
+            qaf_mod->qaf_lib = NULL;
+        }
+        free(qaf_mod);
+        qaf_mod = NULL;
+    }
+    return 0;
+}
+
+static void *qaf_offload_thread_loop(void *context)
+{
+    struct stream_out *out = (struct stream_out *) context;
+    struct listnode *item;
+    int ret = 0;
+    struct str_parms *parms = NULL;
+    int value = 0;
+    char* kvpairs = NULL;
+
+    setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+    set_sched_policy(0, SP_FOREGROUND);
+    prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
+
+    ALOGE("%s", __func__);
+    lock_output_stream(out);
+    for (;;) {
+        struct offload_cmd *cmd = NULL;
+        stream_callback_event_t event;
+        bool send_callback = false;
+
+        ALOGV("%s qaf_offload_cmd_list %d",
+              __func__, list_empty(&out->qaf_offload_cmd_list));
+        if (list_empty(&out->qaf_offload_cmd_list)) {
+            ALOGV("%s SLEEPING", __func__);
+            pthread_cond_wait(&out->qaf_offload_cond, &out->lock);
+            ALOGV("%s RUNNING", __func__);
+            continue;
+        }
+
+        item = list_head(&out->qaf_offload_cmd_list);
+        cmd = node_to_item(item, struct offload_cmd, node);
+        list_remove(item);
+
+        if (cmd->cmd == OFFLOAD_CMD_EXIT) {
+            free(cmd);
+            break;
+        }
+
+        pthread_mutex_unlock(&out->lock);
+        send_callback = false;
+        switch(cmd->cmd) {
+        case OFFLOAD_CMD_WAIT_FOR_BUFFER:
+            ALOGV("wait for ms12 buffer availability");
+            while (1) {
+                kvpairs = qaf_mod->qaf_audio_stream_get_param(out->qaf_stream_handle, "buf_available");
+                if (kvpairs) {
+                    parms = str_parms_create_str(kvpairs);
+                    ret = str_parms_get_int(parms, "buf_available", &value);
+                    if (ret >= 0) {
+                        if (value >= (int)out->compr_config.fragment_size) {
+                            ALOGV("%s buffer available", __func__);
+                            str_parms_destroy(parms);
+                            parms = NULL;
+                            break;
+                        } else {
+                            ALOGV("%s sleep", __func__);
+                            str_parms_destroy(parms);
+                            parms = NULL;
+                            usleep(10000);
+                        }
+                    }
+                    free(kvpairs);
+                    kvpairs = NULL;
+                }
+            }
+            send_callback = true;
+            event = STREAM_CBK_EVENT_WRITE_READY;
+            break;
+        default:
+            ALOGV("%s unknown command received: %d", __func__, cmd->cmd);
+            break;
+        }
+        lock_output_stream(out);
+        if (send_callback && out->offload_callback) {
+            out->offload_callback(event, NULL, out->offload_cookie);
+        }
+        free(cmd);
+    }
+
+    while (!list_empty(&out->qaf_offload_cmd_list)) {
+        item = list_head(&out->qaf_offload_cmd_list);
+        list_remove(item);
+        free(node_to_item(item, struct offload_cmd, node));
+    }
+    pthread_mutex_unlock(&out->lock);
+
+    return NULL;
+}
+
+static int qaf_create_offload_callback_thread(struct stream_out *out)
+{
+    ALOGV("%s", __func__);
+    pthread_cond_init(&out->qaf_offload_cond, (const pthread_condattr_t *) NULL);
+    list_init(&out->qaf_offload_cmd_list);
+    pthread_create(&out->qaf_offload_thread, (const pthread_attr_t *) NULL,
+                    qaf_offload_thread_loop, out);
+    return 0;
+}
+
+static int qaf_destroy_offload_callback_thread(struct stream_out *out)
+{
+    ALOGV("%s", __func__);
+    lock_output_stream(out);
+    qaf_send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
+    pthread_mutex_unlock(&out->lock);
+
+    pthread_join(out->qaf_offload_thread, (void **) NULL);
+    pthread_cond_destroy(&out->qaf_offload_cond);
+
+    return 0;
+}
+
+int audio_extn_qaf_open_output_stream(struct audio_hw_device *dev,
+                                   audio_io_handle_t handle,
+                                   audio_devices_t devices,
+                                   audio_output_flags_t flags,
+                                   struct audio_config *config,
+                                   struct audio_stream_out **stream_out,
+                                   const char *address __unused)
+{
+    int ret = 0;
+    struct stream_out *out;
+
+    ret = adev_open_output_stream(dev, handle, devices, flags, config, stream_out, address);
+    if (*stream_out == NULL) {
+        goto error_open;
+    }
+
+    out = (struct stream_out *) *stream_out;
+
+    /* Override function pointers based on qaf definitions */
+    out->stream.set_volume = qaf_out_set_volume;
+    out->stream.pause = qaf_out_pause;
+    out->stream.resume = qaf_out_resume;
+    out->stream.drain = qaf_out_drain;
+    out->stream.flush = qaf_out_flush;
+
+    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.write = qaf_out_write;
+    out->stream.get_presentation_position = qaf_out_get_presentation_position;
+
+    ret = qaf_stream_open(out, config, flags, devices);
+    if (ret < 0) {
+        ALOGE("%s, Error opening QAF stream err[%d]!", __func__, ret);
+        adev_close_output_stream(dev, *stream_out);
+        goto error_open;
+    }
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY) {
+        out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
+        out->config.period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE;
+        out->config.period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT;
+        out->config.start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+        out->config.avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+    }
+
+    *stream_out = &out->stream;
+    if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+        qaf_create_offload_callback_thread(out);
+    }
+    ALOGV("%s: exit", __func__);
+    return 0;
+error_open:
+    *stream_out = NULL;
+    ALOGD("%s: exit: ret %d", __func__, ret);
+    return ret;
+}
+
+void audio_extn_qaf_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    ALOGV("%s: enter:stream_handle(%p) format = %x", __func__, out, out->format);
+    if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+        qaf_destroy_offload_callback_thread(out);
+    }
+    qaf_mod->stream_drain_main = NULL;
+    lock_output_stream(out);
+    qaf_stream_close(out);
+    pthread_mutex_unlock(&out->lock);
+
+    adev_close_output_stream(dev, stream);
+    ALOGV("%s: exit", __func__);
+}
+
+bool audio_extn_qaf_is_enabled()
+{
+    bool prop_enabled = false;
+    char value[PROPERTY_VALUE_MAX] = {0};
+    property_get("audio.qaf.enabled", value, NULL);
+    prop_enabled = atoi(value) || !strncmp("true", value, 4);
+    return (prop_enabled);
+}
+
+int audio_extn_qaf_session_open(struct qaf *qaf_mod,
+                                device_license_config_t* lic_config)
+{
+    ALOGV("%s %d", __func__, __LINE__);
+    int status = -ENOSYS;
+
+    pthread_mutex_init(&qaf_mod->lock, (const pthread_mutexattr_t *) NULL);
+
+    if (!qaf_mod->qaf_audio_session_open)
+       return -EINVAL;
+
+    status = qaf_mod->qaf_audio_session_open(&qaf_mod->session_handle,
+                                             (void *)(qaf_mod), (void *)lic_config);
+    if(status < 0)
+        return status;
+
+    if (qaf_mod->session_handle == NULL) {
+        ALOGE("%s %d QAF wrapper session handle is NULL", __func__, __LINE__);
+        return -ENOMEM;
+    }
+    if (qaf_mod->qaf_register_event_callback)
+        qaf_mod->qaf_register_event_callback(qaf_mod->session_handle,
+                                             qaf_mod, &notify_event_callback,
+                                             AUDIO_DATA_EVENT);
+    return status;
+}
+
+char* audio_extn_qaf_stream_get_param(struct stream_out *out __unused, const char *kv_pair __unused)
+{
+   return NULL;
+}
+
+int audio_extn_qaf_set_parameters(struct audio_device *adev, struct str_parms *parms)
+{
+    int status = 0, val = 0, channels = 0;
+    char *format_params, *kv_parirs;
+    struct str_parms *qaf_params;
+    char value[32];
+    bool passth_support = false;
+
+    ALOGV("%s %d ", __func__, __LINE__);
+    if (!qaf_mod || !qaf_mod->qaf_audio_session_set_param) {
+        return -EINVAL;
+    }
+
+    status = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
+    if (status >= 0) {
+        val = atoi(value);
+        if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+            if (property_get_bool("audio.offload.passthrough", false) &&
+                property_get_bool("audio.qaf.reencode", false)) {
+
+                qaf_params = str_parms_create();
+                if (platform_is_edid_supported_format(adev->platform, AUDIO_FORMAT_E_AC3)) {
+                    passth_support = true;
+                    if (qaf_params) {
+                        str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+                                              AUDIO_QAF_PARAMETER_VALUE_REENCODE_EAC3);
+                    }
+                } else if (platform_is_edid_supported_format(adev->platform, AUDIO_FORMAT_AC3)) {
+                    passth_support = true;
+                    if (qaf_params) {
+                        str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+                                              AUDIO_QAF_PARAMETER_VALUE_REENCODE_AC3);
+                    }
+                }
+
+                if (passth_support) {
+                    str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+                                          AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+                    format_params = str_parms_to_str(qaf_params);
+
+                    qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+                }
+                str_parms_destroy(qaf_params);
+            }
+
+            if (!passth_support) {
+                channels = platform_edid_get_max_channels(adev->platform);
+
+                qaf_params = str_parms_create();
+                switch (channels) {
+                    case 8:
+                          ALOGV("%s: Switching Qaf output to 7.1 channels", __func__);
+                          str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_CHANNELS,
+                                                AUDIO_QAF_PARAMETER_VALUE_8_CHANNELS);
+                          str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+                                                AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+                          qaf_mod->hdmi_sink_channels = channels;
+                          break;
+                    case 6:
+                          ALOGV("%s: Switching Qaf output to 5.1 channels", __func__);
+                          str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_CHANNELS,
+                                                AUDIO_QAF_PARAMETER_VALUE_6_CHANNELS);
+                          str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+                                                AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+                          qaf_mod->hdmi_sink_channels = channels;
+                          break;
+                    default:
+                          str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+                                                AUDIO_QAF_PARAMETER_VALUE_DEVICE_SPEAKER);
+                          qaf_mod->hdmi_sink_channels = 2;
+                        break;
+                }
+
+                format_params = str_parms_to_str(qaf_params);
+                qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+                str_parms_destroy(qaf_params);
+            }
+        }
+    }
+
+    status = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
+    if (status >= 0) {
+        val = atoi(value);
+        if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+            qaf_params = str_parms_create();
+            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);
+            qaf_mod->hdmi_sink_channels = 0;
+
+            format_params = str_parms_to_str(qaf_params);
+            qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+            str_parms_destroy(qaf_params);
+        }
+    }
+
+    kv_parirs = str_parms_to_str(parms);
+    qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, kv_parirs);
+
+    return status;
+}
+
+char* audio_extn_qaf_get_param(struct audio_device *adev __unused, const char *kv_pair __unused)
+{
+    return 0;
+}
+
+int audio_extn_qaf_init(struct audio_device *adev)
+{
+    char value[PROPERTY_VALUE_MAX] = {0};
+    char lib_name[PROPERTY_VALUE_MAX] = {0};
+    unsigned char* license_data = NULL;
+    device_license_config_t* lic_config = NULL;
+    ALOGV("%s %d", __func__, __LINE__);
+    int ret = 0, size = 0;
+
+    qaf_mod = malloc(sizeof(struct qaf));
+    if(qaf_mod == NULL) {
+        ALOGE("%s, out of memory", __func__);
+        ret = -ENOMEM;
+        goto done;
+    }
+    memset(qaf_mod, 0, sizeof(struct qaf));
+    lic_config = (device_license_config_t*) calloc(1, sizeof(device_license_config_t));
+    if(lic_config == NULL) {
+        ALOGE("%s, out of memory", __func__);
+        ret = -ENOMEM;
+        goto done;
+    }
+    qaf_mod->adev = adev;
+    property_get("audio.qaf.library", value, NULL);
+    snprintf(lib_name, PROPERTY_VALUE_MAX, "%s", value);
+
+    license_data = platform_get_license((struct audio_hw_device *)(qaf_mod->adev->platform), &size);
+    if (!license_data) {
+        ALOGE("License is not present");
+        ret = -EINVAL;
+        goto done;
+    }
+    lic_config->p_license = (unsigned char* ) calloc(1, size);
+     if(lic_config->p_license == NULL) {
+        ALOGE("%s, out of memory", __func__);
+        ret = -ENOMEM;
+        goto done;
+    }
+    lic_config->l_size = size;
+    memcpy(lic_config->p_license, license_data, size);
+
+    if (property_get("audio.qaf.manufacturer", value, "") && atoi(value)) {
+        lic_config->manufacturer_id = (unsigned long) atoi (value);
+    } else {
+        ALOGE("audio.qaf.manufacturer id is not set");
+        ret = -EINVAL;
+        goto done;
+    }
+
+    ret = audio_extn_qaf_session_open(qaf_mod, lic_config);
+done:
+    if (license_data != NULL) {
+        free(license_data);
+        license_data = NULL;
+    }
+    if (lic_config->p_license != NULL) {
+        free(lic_config->p_license);
+        lic_config->p_license = NULL;
+    }
+    if (lic_config != NULL) {
+        free(lic_config);
+        lic_config = NULL;
+    }
+    if (ret != 0) {
+        if (qaf_mod != NULL) {
+            free(qaf_mod);
+            qaf_mod = NULL;
+        }
+    }
+    return ret;
+}
+
+void audio_extn_qaf_deinit()
+{
+    qaf_session_close();
+    qaf_deinit();
+}
diff --git a/hal/audio_extn/sndmonitor.c b/hal/audio_extn/sndmonitor.c
new file mode 100644
index 0000000..eecc448
--- /dev/null
+++ b/hal/audio_extn/sndmonitor.c
@@ -0,0 +1,684 @@
+/*
+* Copyright (c) 2016, The Linux Foundation. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions are
+* met:
+*     * Redistributions of source code must retain the above copyright
+*       notice, this list of conditions and the following disclaimer.
+*     * Redistributions in binary form must reproduce the above
+*       copyright notice, this list of conditions and the following
+*       disclaimer in the documentation and/or other materials provided
+*       with the distribution.
+*     * Neither the name of The Linux Foundation nor the names of its
+*       contributors may be used to endorse or promote products derived
+*       from this software without specific prior written permission.
+*
+* THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+* ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+* BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+* OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#define LOG_TAG "audio_hw_sndmonitor"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+/* monitor sound card, cpe state
+
+   audio_dev registers for a callback from this module in adev_open
+   Each stream in audio_hal registers for a callback in
+   adev_open_*_stream.
+
+   A thread is spawned to poll() on sound card state files in /proc.
+   On observing a sound card state change, this thread invokes the
+   callbacks registered.
+
+   Callbacks are deregistered in adev_close_*_stream and adev_close
+*/
+#include <stdlib.h>
+#include <dirent.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/stat.h>
+#include <sys/poll.h>
+#include <cutils/list.h>
+#include <cutils/hashmap.h>
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <ctype.h>
+
+#include "audio_hw.h"
+#include "audio_extn.h"
+
+//#define MONITOR_DEVICE_EVENTS
+#define CPE_MAGIC_NUM 0x2000
+#define MAX_CPE_SLEEP_RETRY 2
+#define CPE_SLEEP_WAIT 100
+
+#define MAX_SLEEP_RETRY 100
+#define AUDIO_INIT_SLEEP_WAIT 100 /* 100 ms */
+
+#define AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE "ext_audio_device"
+#define INIT_MAP_SIZE 5
+
+typedef enum {
+    audio_event_on,
+    audio_event_off
+} audio_event_status;
+
+typedef struct {
+    int card;
+    int fd;
+    struct listnode node; // membership in sndcards list
+    card_status_t status;
+} sndcard_t;
+
+typedef struct {
+    char *dev;
+    int fd;
+    int status;
+    struct listnode node; // membership in deviceevents list;
+} dev_event_t;
+
+typedef void (*notifyfn)(const void *target, const char *msg);
+
+typedef struct {
+    const void *target;
+    notifyfn notify;
+    struct listnode cards;
+    unsigned int num_cards;
+    struct listnode dev_events;
+    unsigned int num_dev_events;
+    pthread_t monitor_thread;
+    int intpipe[2];
+    Hashmap *listeners; // from stream * -> callback func
+    bool initcheck;
+} sndmonitor_state_t;
+
+static sndmonitor_state_t sndmonitor;
+
+static char *read_state(int fd)
+{
+    struct stat buf;
+    if (fstat(fd, &buf) < 0)
+        return NULL;
+
+    off_t pos = lseek(fd, 0, SEEK_CUR);
+    off_t avail = buf.st_size - pos;
+    if (avail <= 0) {
+        ALOGE("avail %ld", avail);
+        return NULL;
+    }
+
+    char *state = (char *)calloc(avail+1, sizeof(char));
+    if (!state)
+        return NULL;
+
+    ssize_t bytes = read(fd, state, avail);
+    if (bytes <= 0)
+        return NULL;
+
+    // trim trailing whitespace
+    while (bytes && isspace(*(state+bytes-1))) {
+        *(state + bytes - 1) = '\0';
+        --bytes;
+    }
+    lseek(fd, 0, SEEK_SET);
+    return state;
+}
+
+static int add_new_sndcard(int card, int fd)
+{
+    sndcard_t *s = (sndcard_t *)calloc(sizeof(sndcard_t), 1);
+
+    if (!s)
+        return -1;
+
+    s->card = card;
+    s->fd = fd; // dup?
+
+    char *state = read_state(fd);
+
+    if (!state)
+        return -1;
+
+    bool online = state && !strcmp(state, "ONLINE");
+
+    ALOGV("card %d initial state %s %d", card, state, online);
+
+    if (state)
+        free(state);
+
+    s->status = online ? CARD_STATUS_ONLINE : CARD_STATUS_OFFLINE;
+    list_add_tail(&sndmonitor.cards, &s->node);
+    return 0;
+}
+
+static int validate_snd_card(const char *id)
+{
+    return !strncasecmp(id, "msm", 3) ? 0 : -1;
+}
+
+static int enum_sndcards()
+{
+    const char *cards = "/proc/asound/cards";
+    int tries = 10;
+    char *line = NULL;
+    size_t len = 0;
+    ssize_t bytes_read = -1;
+    char path[128] = {0};
+    char *ptr = NULL, *saveptr = NULL, *card_id = NULL;
+    int line_no=0;
+    unsigned int num_cards=0, num_cpe=0;
+    FILE *fp = NULL;
+    int fd = -1, ret = -1;
+
+    while (--tries) {
+        if ((fp = fopen(cards, "r")) == NULL) {
+            ALOGE("Cannot open %s file to get list of sound cards", cards);
+            usleep(100000);
+            continue;
+        }
+        break;
+    }
+
+    if (!tries)
+        return -ENODEV;
+
+    while ((bytes_read = getline(&line, &len, fp) != -1)) {
+        // skip every other line to to match
+        // the output format of /proc/asound/cards
+        if (line_no++ % 2)
+            continue;
+
+        ptr = strtok_r(line, " [", &saveptr);
+        if (!ptr)
+            continue;
+
+        card_id = strtok_r(saveptr+1, "]", &saveptr);
+        if (!card_id)
+            continue;
+
+        // Only consider sound cards associated with ADSP
+        if (validate_snd_card((const char *)card_id) < 0) {
+            ALOGW("Skip over non-ADSP snd card %s", card_id);
+            continue;
+        }
+
+        snprintf(path, sizeof(path), "/proc/asound/card%s/state", ptr);
+        ALOGV("Opening sound card state : %s", path);
+
+        fd = open(path, O_RDONLY);
+        if (fd == -1) {
+            ALOGE("Open %s failed : %s", path, strerror(errno));
+            continue;
+        }
+
+        ret = add_new_sndcard(atoi(ptr), fd);
+        if (ret != 0)
+            continue;
+
+        num_cards++;
+
+        // query cpe state for this card as well
+        tries = MAX_CPE_SLEEP_RETRY;
+        snprintf(path, sizeof(path), "/proc/asound/card%s/cpe0_state", ptr);
+
+        if (access(path, R_OK) < 0) {
+            ALOGW("access %s failed w/ err %s", path, strerror(errno));
+            continue;
+        }
+
+        ALOGV("Open cpe state card state %s", path);
+        while (--tries) {
+            if ((fd = open(path, O_RDONLY)) < 0) {
+                ALOGW("Open cpe state card state failed, retry : %s", path);
+                usleep(CPE_SLEEP_WAIT*1000);
+                continue;
+            }
+            break;
+        }
+
+        if (!tries)
+            continue;
+
+        ret = add_new_sndcard(CPE_MAGIC_NUM+num_cpe, fd);
+        if (ret != 0)
+            continue;
+
+        num_cpe++;
+        num_cards++;
+    }
+    if (line)
+        free(line);
+    fclose(fp);
+    ALOGV("sndmonitor registerer num_cards %d", num_cards);
+    sndmonitor.num_cards = num_cards;
+    return num_cards ? 0 : -1;
+}
+
+static void free_sndcards()
+{
+    while (!list_empty(&sndmonitor.cards)) {
+        struct listnode *n = list_head(&sndmonitor.cards);
+        sndcard_t *s = node_to_item(n, sndcard_t, node);
+        list_remove(n);
+        close(s->fd);
+        free(s);
+    }
+}
+
+#ifdef MONITOR_DEVICE_EVENTS
+static int add_new_dev_event(char *d_name, int fd)
+{
+    dev_event_t *d = (dev_event_t *)calloc(sizeof(dev_event_t), 1);
+
+    if (!d)
+        return -1;
+
+    d->dev = strdup(d_name);
+    d->fd = fd;
+    list_add_tail(&sndmonitor.dev_events, &d->node);
+    return 0;
+}
+
+static int enum_dev_events()
+{
+    const char *events_dir = "/sys/class/switch/";
+    DIR *dp;
+    struct dirent *in_file;
+    int fd;
+    char path[128] = {0};
+    unsigned int num_dev_events = 0;
+
+    if ((dp = opendir(events_dir)) == NULL) {
+        ALOGE("Cannot open switch directory %s err %s",
+              events_dir, strerror(errno));
+        return -1;
+    }
+
+    while ((in_file = readdir(dp)) != NULL) {
+        if (!strstr(in_file->d_name, "qc_"))
+            continue;
+
+        snprintf(path, sizeof(path), "%s/%s/state",
+                 events_dir, in_file->d_name);
+
+        ALOGV("Opening audio dev event state : %s ", path);
+        fd = open(path, O_RDONLY);
+        if (fd == -1) {
+            ALOGE("Open %s failed : %s", path, strerror(errno));
+        } else {
+            if (!add_new_dev_event(in_file->d_name, fd))
+                num_dev_events++;
+        }
+    }
+    closedir(dp);
+    sndmonitor.num_dev_events = num_dev_events;
+    return num_dev_events ? 0 : -1;
+}
+#endif
+
+static void free_dev_events()
+{
+    while (!list_empty(&sndmonitor.dev_events)) {
+        struct listnode *n = list_head(&sndmonitor.dev_events);
+        dev_event_t *d = node_to_item(n, dev_event_t, node);
+        list_remove(n);
+        close(d->fd);
+        free(d->dev);
+        free(d);
+    }
+}
+
+static int notify(const struct str_parms *params)
+{
+    if (!params)
+        return -1;
+
+    char *str = str_parms_to_str((struct str_parms *)params);
+
+    if (!str)
+        return -1;
+
+    if (sndmonitor.notify)
+        sndmonitor.notify(sndmonitor.target, str);
+
+    ALOGV("%s", str);
+    free(str);
+    return 0;
+}
+
+int on_dev_event(dev_event_t *dev_event)
+{
+    char state_buf[2];
+    if (read(dev_event->fd, state_buf, 1) <= 0)
+        return -1;
+
+    lseek(dev_event->fd, 0, SEEK_SET);
+    state_buf[1]='\0';
+    if (atoi(state_buf) == dev_event->status)
+        return 0;
+
+    dev_event->status = atoi(state_buf);
+
+    struct str_parms *params = str_parms_create();
+
+    if (!params)
+        return -1;
+
+    char val[32] = {0};
+    snprintf(val, sizeof(val), "%s,%s", dev_event->dev,
+             dev_event->status ? "ON" : "OFF");
+
+    if (str_parms_add_str(params, AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE, val) < 0)
+        return -1;
+
+    int ret = notify(params);
+    str_parms_destroy(params);
+    return ret;
+}
+
+bool on_sndcard_state_update(sndcard_t *s)
+{
+    char rd_buf[9]={0};
+    card_status_t status;
+
+    if (read(s->fd, rd_buf, 8) <= 0)
+        return -1;
+
+    rd_buf[8] = '\0';
+    lseek(s->fd, 0, SEEK_SET);
+
+    ALOGV("card num %d, new state %s", s->card, rd_buf);
+
+    bool is_cpe = (s->card >= CPE_MAGIC_NUM);
+    if (strstr(rd_buf, "OFFLINE"))
+        status = CARD_STATUS_OFFLINE;
+    else if (strstr(rd_buf, "ONLINE"))
+        status = CARD_STATUS_ONLINE;
+    else {
+        ALOGE("unknown state");
+        return 0;
+    }
+
+    if (status == s->status) // no change
+        return 0;
+
+    s->status = status;
+
+    struct str_parms *params = str_parms_create();
+
+    if (!params)
+        return -1;
+
+    char val[32] = {0};
+    // cpe actual card num is (card - MAGIC_NUM). so subtract accordingly
+    snprintf(val, sizeof(val), "%d,%s", s->card - (is_cpe ? CPE_MAGIC_NUM : 0),
+                 status == CARD_STATUS_ONLINE ? "ONLINE" : "OFFLINE");
+
+    if (str_parms_add_str(params, is_cpe ? "CPE_STATUS" : "SND_CARD_STATUS",
+                          val) < 0)
+        return -1;
+
+    int ret = notify(params);
+    str_parms_destroy(params);
+    return ret;
+}
+
+void *monitor_thread_loop(void *args __unused)
+{
+    ALOGV("Start threadLoop()");
+    unsigned int num_poll_fds = sndmonitor.num_cards +
+                                sndmonitor.num_dev_events + 1/*pipe*/;
+    struct pollfd *pfd = (struct pollfd *)calloc(sizeof(struct pollfd),
+                                                  num_poll_fds);
+    if (!pfd)
+        return NULL;
+
+    pfd[0].fd = sndmonitor.intpipe[0];
+    pfd[0].events = POLLPRI|POLLIN;
+
+    int i = 1;
+    struct listnode *node;
+    list_for_each(node, &sndmonitor.cards) {
+        sndcard_t *s = node_to_item(node, sndcard_t, node);
+        pfd[i].fd = s->fd;
+        pfd[i].events = POLLPRI;
+        ++i;
+    }
+
+    list_for_each(node, &sndmonitor.dev_events) {
+        dev_event_t *d = node_to_item(node, dev_event_t, node);
+        pfd[i].fd = d->fd;
+        pfd[i].events = POLLPRI;
+        ++i;
+    }
+
+    while (1) {
+        if (poll(pfd, num_poll_fds, -1) < 0) {
+            int errno_ = errno;
+            ALOGE("poll() failed w/ err %s", strerror(errno_));
+            switch (errno_) {
+                case EINTR:
+                case ENOMEM:
+                    sleep(2);
+                    continue;
+                default:
+                    /* above errors can be caused due to current system
+                       state .. any other error is not expected */
+                    LOG_ALWAYS_FATAL("unxpected poll() system call failure");
+                    break;
+            }
+        }
+        ALOGV("out of poll()");
+
+#define READY_TO_READ(p) ((p)->revents & (POLLIN|POLLPRI))
+#define ERROR_IN_FD(p) ((p)->revents & (POLLERR|POLLHUP|POLLNVAL))
+
+        // check if requested to exit
+        if (READY_TO_READ(&pfd[0])) {
+            char buf[2]={0};
+            read(pfd[0].fd, buf, 1);
+            if (!strcmp(buf, "Q"))
+                break;
+        } else if (ERROR_IN_FD(&pfd[0])) {
+            // do not consider for poll again
+            // POLLERR - can this happen?
+            // POLLHUP - adev must not close pipe
+            // POLLNVAL - fd is valid
+            LOG_ALWAYS_FATAL("unxpected error in pipe poll fd 0x%x",
+                             pfd[0].revents);
+            // FIXME: If not fatal, then need some logic to close
+            // these fds on error
+            pfd[0].fd *= -1;
+        }
+
+        i = 1;
+        list_for_each(node, &sndmonitor.cards) {
+            sndcard_t *s = node_to_item(node, sndcard_t, node);
+            if (READY_TO_READ(&pfd[i]))
+                on_sndcard_state_update(s);
+            else if (ERROR_IN_FD(&pfd[i])) {
+                // do not consider for poll again
+                // POLLERR - can this happen as we are reading from a fs?
+                // POLLHUP - not valid for cardN/state
+                // POLLNVAL - fd is valid
+                LOG_ALWAYS_FATAL("unxpected error in card poll fd 0x%x",
+                                 pfd[i].revents);
+                // FIXME: If not fatal, then need some logic to close
+                // these fds on error
+                pfd[i].fd *= -1;
+            }
+            ++i;
+        }
+
+        list_for_each(node, &sndmonitor.dev_events) {
+            dev_event_t *d = node_to_item(node, dev_event_t, node);
+            if (READY_TO_READ(&pfd[i]))
+                on_dev_event(d);
+            else if (ERROR_IN_FD(&pfd[i])) {
+                // do not consider for poll again
+                // POLLERR - can this happen as we are reading from a fs?
+                // POLLHUP - not valid for switch/state
+                // POLLNVAL - fd is valid
+                LOG_ALWAYS_FATAL("unxpected error in dev poll fd 0x%x",
+                                 pfd[i].revents);
+                // FIXME: If not fatal, then need some logic to close
+                // these fds on error
+                pfd[i].fd *= -1;
+            }
+            ++i;
+        }
+    }
+
+    return NULL;
+}
+
+// ---- listener static APIs ---- //
+static int hashfn(void *key)
+{
+    return (int)key;
+}
+
+static bool hasheq(void *key1, void *key2)
+{
+    return key1 == key2;
+}
+
+static bool snd_cb(void *key, void *value, void *context)
+{
+    snd_mon_cb cb = (snd_mon_cb)value;
+    cb(key, context);
+    return true;
+}
+
+static void snd_mon_update(const void *target __unused, const char *msg)
+{
+    // target can be used to check if this message is intended for the
+    // recipient or not. (using some statically saved state)
+
+    struct str_parms *parms = str_parms_create_str(msg);
+
+    if (!parms)
+        return;
+
+    hashmapLock(sndmonitor.listeners);
+    hashmapForEach(sndmonitor.listeners, snd_cb, parms);
+    hashmapUnlock(sndmonitor.listeners);
+
+    str_parms_destroy(parms);
+}
+
+static int listeners_init()
+{
+    sndmonitor.listeners = hashmapCreate(INIT_MAP_SIZE, hashfn, hasheq);
+    if (!sndmonitor.listeners)
+        return -1;
+    return 0;
+}
+
+static int listeners_deinit()
+{
+    // XXX TBD
+    return -1;
+}
+
+static int add_listener(void *stream, snd_mon_cb cb)
+{
+    Hashmap *map = sndmonitor.listeners;
+    hashmapLock(map);
+    hashmapPut(map, stream, cb);
+    hashmapUnlock(map);
+    return 0;
+}
+
+static int del_listener(void * stream)
+{
+    Hashmap *map = sndmonitor.listeners;
+    hashmapLock(map);
+    hashmapRemove(map, stream);
+    hashmapUnlock(map);
+    return 0;
+}
+
+// --- public APIs --- //
+
+int audio_extn_snd_mon_deinit()
+{
+    if (!sndmonitor.initcheck)
+        return -1;
+
+    write(sndmonitor.intpipe[1], "Q", 1);
+    pthread_join(sndmonitor.monitor_thread, (void **) NULL);
+    listeners_deinit();
+    free_sndcards();
+    free_dev_events();
+    sndmonitor.initcheck = 0;
+    return 0;
+}
+
+int audio_extn_snd_mon_init()
+{
+    sndmonitor.notify = snd_mon_update;
+    sndmonitor.target = NULL; // unused for now
+    list_init(&sndmonitor.cards);
+    list_init(&sndmonitor.dev_events);
+    sndmonitor.initcheck = false;
+
+    if (pipe(sndmonitor.intpipe) < 0)
+        return -ENODEV;
+
+    if (enum_sndcards() < 0)
+        return -ENODEV;
+
+    if (listeners_init() < 0)
+        return -ENODEV;
+
+#ifdef MONITOR_DEVICE_EVENTS
+    enum_dev_events(); // failure here isn't fatal
+#endif
+
+    int ret = pthread_create(&sndmonitor.monitor_thread,
+                             (const pthread_attr_t *) NULL,
+                             monitor_thread_loop, NULL);
+
+    if (ret) {
+        free_sndcards();
+        free_dev_events();
+        close(sndmonitor.intpipe[0]);
+        close(sndmonitor.intpipe[1]);
+        return -ENODEV;
+    }
+    sndmonitor.initcheck = true;
+    return 0;
+}
+
+int audio_extn_snd_mon_register_listener(void *stream, snd_mon_cb cb)
+{
+    if (!sndmonitor.initcheck) {
+        ALOGW("sndmonitor initcheck failed, cannot register");
+        return -1;
+    }
+
+    return add_listener(stream, cb);
+}
+
+int audio_extn_snd_mon_unregister_listener(void *stream)
+{
+    if (!sndmonitor.initcheck) {
+        ALOGW("sndmonitor initcheck failed, cannot deregister");
+        return -1;
+    }
+
+    ALOGV("deregister listener for stream %p ", stream);
+    return del_listener(stream);
+}
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 7e37efc..6142e86 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -98,9 +98,9 @@
             status = -ENOMEM;
             break;
         }
-        memcpy(&st_ses_info->st_ses, &config->st_ses, sizeof (config->st_ses));
-        ALOGV("%s: add capture_handle %d pcm %p", __func__,
-              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+        memcpy(&st_ses_info->st_ses, &config->st_ses, sizeof (struct sound_trigger_session_info));
+        ALOGV("%s: add capture_handle %d st session opaque ptr %p", __func__,
+              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.p_ses);
         list_add_tail(&st_dev->st_ses_list, &st_ses_info->list);
         break;
 
@@ -112,12 +112,12 @@
         }
         st_ses_info = get_sound_trigger_info(config->st_ses.capture_handle);
         if (!st_ses_info) {
-            ALOGE("%s: pcm %p not in the list!", __func__, config->st_ses.pcm);
+            ALOGE("%s: st session opaque ptr %p not in the list!", __func__, config->st_ses.p_ses);
             status = -EINVAL;
             break;
         }
-        ALOGV("%s: remove capture_handle %d pcm %p", __func__,
-              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+        ALOGV("%s: remove capture_handle %d st session opaque ptr %p", __func__,
+              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.p_ses);
         list_remove(&st_ses_info->list);
         free(st_ses_info);
         break;
@@ -181,7 +181,7 @@
     pthread_mutex_unlock(&st_dev->lock);
     if (st_ses_info) {
         event.u.ses_info = st_ses_info->st_ses;
-        ALOGV("%s: AUDIO_EVENT_STOP_LAB pcm %p", __func__, st_ses_info->st_ses.pcm);
+        ALOGV("%s: AUDIO_EVENT_STOP_LAB st sess %p", __func__, st_ses_info->st_ses.p_ses);
         st_dev->st_callback(AUDIO_EVENT_STOP_LAB, &event);
         in->is_st_session_active = false;
     }
@@ -201,7 +201,6 @@
     list_for_each(node, &st_dev->st_ses_list) {
         st_ses_info = node_to_item(node, struct sound_trigger_info , list);
         if (st_ses_info->st_ses.capture_handle == in->capture_handle) {
-            in->pcm = st_ses_info->st_ses.pcm;
             in->config = st_ses_info->st_ses.config;
             in->channel_mask = audio_channel_in_mask_from_count(in->config.channels);
             in->is_st_session = true;
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 1f88c71..008130f 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -117,6 +117,11 @@
     SPKR_PROTECTION_MODE_CALIBRATE = 1,
 };
 
+struct spkr_prot_r0t0 {
+    int r0[SP_V2_NUM_MAX_SPKRS];
+    int t0[SP_V2_NUM_MAX_SPKRS];
+};
+
 struct speaker_prot_session {
     int spkr_prot_mode;
     int spkr_processing_state;
@@ -142,6 +147,7 @@
     bool spkr_prot_enable;
     bool spkr_in_use;
     struct timespec spkr_last_time_used;
+    struct spkr_prot_r0t0 sp_r0t0_cal;
     bool wsa_found;
     int spkr_1_tzn;
     int spkr_2_tzn;
@@ -340,6 +346,7 @@
     int ret = 0;
     struct audio_cal_fb_spk_prot_cfg    cal_data;
     char value[PROPERTY_VALUE_MAX];
+    static int cal_done = 0;
 
     if (cal_fd < 0) {
         ALOGE("%s: Error: cal_fd = %d", __func__, cal_fd);
@@ -382,6 +389,13 @@
         ret = -ENODEV;
         goto done;
     }
+    if (protCfg->mode == MSM_SPKR_PROT_CALIBRATED  && !cal_done) {
+        handle.sp_r0t0_cal.r0[SP_V2_SPKR_1] = protCfg->r0[SP_V2_SPKR_1];
+        handle.sp_r0t0_cal.r0[SP_V2_SPKR_2] = protCfg->r0[SP_V2_SPKR_2];
+        handle.sp_r0t0_cal.t0[SP_V2_SPKR_1] = protCfg->t0[SP_V2_SPKR_1];
+        handle.sp_r0t0_cal.t0[SP_V2_SPKR_2] = protCfg->t0[SP_V2_SPKR_2];
+        cal_done = 1;
+    }
 done:
     return ret;
 }
@@ -1347,12 +1361,48 @@
     }
 }
 
+int audio_extn_select_spkr_prot_cal_data(snd_device_t snd_device)
+{
+    struct audio_cal_info_spk_prot_cfg protCfg;
+    int acdb_fd = -1;
+    int ret = 0;
+
+    acdb_fd = open("/dev/msm_audio_cal", O_RDWR | O_NONBLOCK);
+    if (acdb_fd < 0) {
+        ALOGE("%s: open msm_acdb failed", __func__);
+        return -ENODEV;
+    }
+    switch(snd_device) {
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+            protCfg.r0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_2];
+            protCfg.r0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_1];
+            protCfg.t0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_2];
+            protCfg.t0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_1];
+            break;
+        default:
+            protCfg.r0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_1];
+            protCfg.r0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_2];
+            protCfg.t0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_1];
+            protCfg.t0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_2];
+            break;
+    }
+    protCfg.mode = MSM_SPKR_PROT_CALIBRATED;
+    ret = set_spkr_prot_cal(acdb_fd, &protCfg);
+    if (ret)
+        ALOGE("%s: speaker protection cal data swap failed", __func__);
+
+    close(acdb_fd);
+    return ret;
+}
+
 int audio_extn_spkr_prot_start_processing(snd_device_t snd_device)
 {
     struct audio_usecase *uc_info_tx;
     struct audio_device *adev = handle.adev_handle;
     int32_t pcm_dev_tx_id = -1, ret = 0;
     bool disable_tx = false;
+    snd_device_t in_snd_device;
 
     ALOGV("%s: Entry", __func__);
     /* cancel speaker calibration */
@@ -1361,6 +1411,15 @@
        return -EINVAL;
     }
     snd_device = platform_get_spkr_prot_snd_device(snd_device);
+    if (handle.spkr_prot_mode == MSM_SPKR_PROT_CALIBRATED) {
+        ret = audio_extn_select_spkr_prot_cal_data(snd_device);
+        if (ret) {
+            ALOGE("%s: Setting speaker protection cal data failed", __func__);
+            return ret;
+        }
+    }
+
+    in_snd_device = platform_get_vi_feedback_snd_device(snd_device);
     spkr_prot_set_spkrstatus(true);
     uc_info_tx = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
     if (!uc_info_tx) {
@@ -1375,12 +1434,12 @@
     if (handle.spkr_processing_state == SPKR_PROCESSING_IN_IDLE) {
         uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
         uc_info_tx->type = PCM_CAPTURE;
-        uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+        uc_info_tx->in_snd_device = in_snd_device;
         uc_info_tx->out_snd_device = SND_DEVICE_NONE;
         handle.pcm_tx = NULL;
         list_add_tail(&adev->usecase_list, &uc_info_tx->list);
         disable_tx = true;
-        enable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+        enable_snd_device(adev, in_snd_device);
         enable_audio_route(adev, uc_info_tx);
 
         pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
@@ -1420,9 +1479,9 @@
         list_remove(&uc_info_tx->list);
         uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
         uc_info_tx->type = PCM_CAPTURE;
-        uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+        uc_info_tx->in_snd_device = in_snd_device;
         uc_info_tx->out_snd_device = SND_DEVICE_NONE;
-        disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+        disable_snd_device(adev, in_snd_device);
         disable_audio_route(adev, uc_info_tx);
         free(uc_info_tx);
     } else
@@ -1436,17 +1495,20 @@
 {
     struct audio_usecase *uc_info_tx;
     struct audio_device *adev = handle.adev_handle;
+    snd_device_t in_snd_device;
 
     ALOGV("%s: Entry", __func__);
     snd_device = platform_get_spkr_prot_snd_device(snd_device);
     spkr_prot_set_spkrstatus(false);
+    in_snd_device = platform_get_vi_feedback_snd_device(snd_device);
+
     pthread_mutex_lock(&handle.mutex_spkr_prot);
     if (adev && handle.spkr_processing_state == SPKR_PROCESSING_IN_PROGRESS) {
         uc_info_tx = get_usecase_from_list(adev, USECASE_AUDIO_SPKR_CALIB_TX);
         if (handle.pcm_tx)
             pcm_close(handle.pcm_tx);
         handle.pcm_tx = NULL;
-        disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+        disable_snd_device(adev, in_snd_device);
         if (uc_info_tx) {
             list_remove(&uc_info_tx->list);
             disable_audio_route(adev, uc_info_tx);
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index 9e19eba..f936f99 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -47,7 +47,7 @@
 #define SAMPLE_RATE_11025         11025
 // Supported sample rates for USB
 static uint32_t supported_sample_rates[] =
-    {44100, 48000, 64000, 88200, 96000, 176400, 192000};
+    {44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000};
 
 #define  MAX_SAMPLE_RATE_SIZE  sizeof(supported_sample_rates)/sizeof(supported_sample_rates[0])
 
@@ -93,7 +93,7 @@
 
 static const char * const usb_sidetone_enable_str[] = {
     "Sidetone Playback Switch",
-    "Mic Playback Switchs",
+    "Mic Playback Switch",
 };
 
 static const char * const usb_sidetone_volume_str[] = {
@@ -190,33 +190,6 @@
     }
 }
 
-static int usb_set_channel_mixer_ctl(int channel,
-                                     char *ch_mixer_ctl_name)
-{
-    struct mixer_ctl *ctl;
-
-    ctl = mixer_get_ctl_by_name(usbmod->adev->mixer, ch_mixer_ctl_name);
-    if (!ctl) {
-       ALOGE("%s: Could not get ctl for mixer cmd - %s",
-             __func__, ch_mixer_ctl_name);
-       return -EINVAL;
-    }
-    switch (channel) {
-       case 1:
-           mixer_ctl_set_enum_by_string(ctl, "One");
-           break;
-       case 2:
-           mixer_ctl_set_enum_by_string(ctl, "Two");
-           break;
-       default:
-           ALOGV("%s: channel(%d) not supported, set as default 2 channels",
-                 __func__, channel);
-           mixer_ctl_set_enum_by_string(ctl, "Two");
-           break;
-    }
-    return 0;
-}
-
 static int usb_set_dev_id_mixer_ctl(unsigned int usb_usecase_type, int card,
                                     char *dev_mixer_ctl_name)
 {
@@ -300,25 +273,34 @@
                               struct usb_card_config *usb_card_info,
                               int card)
 {
-    int32_t err = 1;
     int32_t size = 0;
     int32_t fd=-1;
-    int32_t altset_index = 1;
     int32_t channels_no;
-    char *str_start, *channel_start, *bit_width_start, *rates_str_start,
-         *target;
+    char *str_start = NULL;
+    char *str_end = NULL;
+    char *channel_start = NULL;
+    char *bit_width_start = NULL;
+    char *rates_str_start = NULL;
+    char *target = NULL;
     char *read_buf = NULL;
     char *rates_str = NULL;
-    char path[128], altset[9];
+    char path[128];
     int ret = 0;
     char *bit_width_str = NULL;
     struct usb_device_config * usb_device_info;
+    bool check = false;
 
+    memset(path, 0, sizeof(path));
     ALOGV("%s: for %s", __func__, (type == USB_PLAYBACK) ?
           PLAYBACK_PROFILE_STR : CAPTURE_PROFILE_STR);
 
-    snprintf(path, sizeof(path), "/proc/asound/card%u/stream0",
+    ret = snprintf(path, sizeof(path), "/proc/asound/card%u/stream0",
              card);
+    if(ret < 0) {
+        ALOGE("%s: failed on snprintf (%d) to path %s\n",
+          __func__, ret, path);
+        goto done;
+    }
 
     fd = open(path, O_RDONLY);
     if (fd <0) {
@@ -336,7 +318,10 @@
         goto done;
     }
 
-    err = read(fd, read_buf, USB_BUFF_SIZE);
+    if(read(fd, read_buf, USB_BUFF_SIZE) < 0) {
+        ALOGE("file read error\n");
+        goto done;
+    }
     str_start = strstr(read_buf, ((type == USB_PLAYBACK) ?
                        PLAYBACK_PROFILE_STR : CAPTURE_PROFILE_STR));
     if (str_start == NULL) {
@@ -346,21 +331,21 @@
         ret = -EINVAL;
         goto done;
     }
-    ALOGV("%s: usb_config = %s\n", __func__, str_start);
+    str_end = strstr(read_buf, ((type == USB_PLAYBACK) ?
+                       CAPTURE_PROFILE_STR : PLAYBACK_PROFILE_STR));
+    if (str_end > str_start)
+        check = true;
+
+    ALOGV("%s: usb_config = %s, check %d\n", __func__, str_start, check);
 
     while (str_start != NULL) {
-        sprintf(altset, "Altset %d", altset_index);
-        ALOGV("%s: altset_index %d\n", __func__, altset_index);
-        str_start = strstr(str_start, altset);
-        if (str_start == NULL) {
-            if (altset_index == 1) {
-                ALOGE("%s: error %s section not found in usb config file",
-                       __func__, (type == USB_PLAYBACK) ?
-                      PLAYBACK_PROFILE_STR : CAPTURE_PROFILE_STR);
-                ret = -EINVAL;
-            }
+        str_start = strstr(str_start, "Altset");
+        if ((str_start == NULL) || (check  && (str_start >= str_end))) {
+            ALOGV("%s: done parsing %s\n", __func__, str_start);
             break;
         }
+        ALOGV("%s: remaining string %s\n", __func__, str_start);
+        str_start += sizeof("Altset");
         usb_device_info = calloc(1, sizeof(struct usb_device_config));
         if (usb_device_info == NULL) {
             ALOGE("%s: error unable to allocate memory",
@@ -368,7 +353,6 @@
             ret = -ENOMEM;
             break;
         }
-        altset_index++;
         /* Bit bit_width parsing */
         bit_width_start = strstr(str_start, "Format: ");
         if (bit_width_start == NULL) {
@@ -461,8 +445,6 @@
                                     int card)
 {
     int ret;
-    struct listnode *node_d;
-    struct usb_device_config *dev_info;
 
     /* get capabilities */
     if ((ret = usb_get_capability(USB_PLAYBACK, usb_card_info, card))) {
@@ -470,14 +452,6 @@
                __func__);
         goto exit;
     }
-    /* Currently only use the first profile using to configure channel for simplification */
-    list_for_each(node_d, &usb_card_info->usb_device_conf_list) {
-        dev_info = node_to_item(node_d, struct usb_device_config, list);
-        if (dev_info != NULL) {
-            usb_set_channel_mixer_ctl(dev_info->channels, "USB_AUDIO_RX Channels");
-            break;
-        }
-    }
     usb_set_dev_id_mixer_ctl(USB_PLAYBACK, card, "USB_AUDIO_RX dev_token");
 
 exit:
@@ -489,8 +463,6 @@
                                       int card)
 {
     int ret;
-    struct listnode *node_d;
-    struct usb_device_config *dev_info;
 
     /* get capabilities */
     if ((ret = usb_get_capability(USB_CAPTURE, usb_card_info, card))) {
@@ -498,14 +470,6 @@
                __func__);
         goto exit;
     }
-    /* Currently only use the first profile using to configure channel for simplification */
-    list_for_each(node_d, &usb_card_info->usb_device_conf_list) {
-        dev_info = node_to_item(node_d, struct usb_device_config, list);
-        if (dev_info != NULL) {
-            usb_set_channel_mixer_ctl(dev_info->channels, "USB_AUDIO_TX Channels");
-            break;
-        }
-    }
     usb_set_dev_id_mixer_ctl(USB_CAPTURE, card, "USB_AUDIO_TX dev_token");
 
 exit:
@@ -739,7 +703,7 @@
                  "%s: USB ch(%d)bw(%d), stm ch(%d)bw(%d)sr(%d), candidate(%d)",
                  __func__, dev_info->channels, dev_info->bit_width,
                  ch, bit_width, stream_sample_rate, candidate);
-        if ((dev_info->bit_width != bit_width) && dev_info->channels != ch)
+        if ((dev_info->bit_width != bit_width) || dev_info->channels != ch)
             continue;
 
         candidate = 0;
@@ -786,34 +750,32 @@
 static bool usb_audio_backend_apply_policy(struct listnode *dev_list,
                                            unsigned int *bit_width,
                                            unsigned int *sample_rate,
-                                           unsigned int ch)
+                                           unsigned int *ch)
 {
-    unsigned int channel;
     bool is_usb_supported = true;
 
     ALOGV("%s: from stream: bit-width(%d) sample_rate(%d) channels (%d)",
-           __func__, *bit_width, *sample_rate, ch);
+           __func__, *bit_width, *sample_rate, *ch);
     if (list_empty(dev_list)) {
         *sample_rate = 48000;
         *bit_width = 16;
-        channel = 2;
+        *ch = 2;
         ALOGI("%s: list is empty,fall back to default setting", __func__);
         goto exit;
     }
     usb_get_best_match_for_bit_width(dev_list, *bit_width, bit_width);
     usb_get_best_match_for_channels(dev_list,
                                     *bit_width,
-                                    ch,
-                                    &channel);
+                                    *ch,
+                                    ch);
     usb_get_best_match_for_sample_rate(dev_list,
                                        *bit_width,
-                                       channel,
+                                       *ch,
                                        *sample_rate,
                                        sample_rate);
 exit:
     ALOGV("%s: Updated sample rate per profile: bit-width(%d) rate(%d) chs(%d)",
-           __func__, *bit_width, *sample_rate, channel);
-    usb_set_channel_mixer_ctl(channel, "USB_AUDIO_RX Channels");
+           __func__, *bit_width, *sample_rate, *ch);
     return is_usb_supported;
 }
 
@@ -885,21 +847,23 @@
 
 bool audio_extn_usb_is_config_supported(unsigned int *bit_width,
                                         unsigned int *sample_rate,
-                                        unsigned int ch)
+                                        unsigned int *ch,
+                                        bool is_playback)
 {
     struct listnode *node_i;
     struct usb_card_config *card_info;
     bool is_usb_supported = false;
 
     ALOGV("%s: from stream: bit-width(%d) sample_rate(%d) ch(%d)",
-           __func__, *bit_width, *sample_rate, ch);
+           __func__, *bit_width, *sample_rate, *ch);
     list_for_each(node_i, &usbmod->usb_card_conf_list) {
         card_info = node_to_item(node_i, struct usb_card_config, list);
         ALOGI_IF(usb_audio_debug_enable,
                  "%s: card_dev_type (0x%x), card_no(%d)",
                  __func__,  card_info->usb_device_type, card_info->usb_card);
         /* Currently only apply the first playback sound card configuration */
-        if (card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) {
+        if ((is_playback && card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) ||
+            ((!is_playback) && card_info->usb_device_type == AUDIO_DEVICE_IN_USB_DEVICE)){
             is_usb_supported = usb_audio_backend_apply_policy(
                                            &card_info->usb_device_conf_list,
                                            bit_width,
@@ -908,8 +872,8 @@
             break;
         }
     }
-    ALOGV("%s: updated: bit-width(%d) sample_rate(%d)",
-           __func__, *bit_width, *sample_rate);
+    ALOGV("%s: updated: bit-width(%d) sample_rate(%d) channels (%d)",
+           __func__, *bit_width, *sample_rate, *ch);
 
     return is_usb_supported;
 }
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index e3f1b6c..9542fbd 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -90,9 +90,7 @@
 #ifdef INCALL_MUSIC_ENABLED
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_INCALL_MUSIC),
 #endif
-#ifdef HDMI_PASSTHROUGH_ENABLED
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH),
-#endif
 };
 
 const struct string_to_enum s_format_name_to_enum_table[] = {
@@ -133,6 +131,7 @@
     STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_LC),
     STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V1),
     STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V2),
+    STRING_TO_ENUM(AUDIO_FORMAT_DSD),
 #endif
 };
 
@@ -496,7 +495,7 @@
     struct stream_format *sf_info;
     char value[PROPERTY_VALUE_MAX] = {0};
 
-    if ((24 == bit_width) &&
+    if ((bit_width >= 24) &&
         (devices & AUDIO_DEVICE_OUT_SPEAKER)) {
         int32_t bw = platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
         if (-ENOSYS != bw)
@@ -515,6 +514,28 @@
                                __func__, sample_rate);
         }
     }
+
+    /* Set sampling rate to 176.4 for DSD64
+     * and 352.8Khz for DSD128.
+     * Set Bit Width to 16. output will be 16 bit
+     * post DoP in ASM.
+     */
+    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH) &&
+        (format == AUDIO_FORMAT_DSD)) {
+        bit_width = 16;
+        if (sample_rate == INPUT_SAMPLING_RATE_DSD64)
+            sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+        else if (sample_rate == INPUT_SAMPLING_RATE_DSD128)
+            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",
            __func__, flags, format, sample_rate);
     list_for_each(node_i, streams_output_cfg_list) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index a8ebb6b..3816748 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -36,6 +36,7 @@
  */
 
 #define LOG_TAG "audio_hw_primary"
+#define ATRACE_TAG (ATRACE_TAG_AUDIO|ATRACE_TAG_HAL)
 /*#define LOG_NDEBUG 0*/
 /*#define VERY_VERY_VERBOSE_LOGGING*/
 #ifdef VERY_VERY_VERBOSE_LOGGING
@@ -55,6 +56,7 @@
 #include <sys/prctl.h>
 
 #include <cutils/log.h>
+#include <cutils/trace.h>
 #include <cutils/str_parms.h>
 #include <cutils/properties.h>
 #include <cutils/atomic.h>
@@ -80,6 +82,7 @@
 /* ToDo: Check and update a proper value in msec */
 #define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
 #define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define DSD_VOLUME_MIN_DB (-110)
 
 #define PROXY_OPEN_RETRY_COUNT           100
 #define PROXY_OPEN_WAIT_TIME             20
@@ -92,6 +95,8 @@
 #define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_deep_buffer
 #endif
 
+#define ULL_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000)
+
 static unsigned int configured_low_latency_capture_period_size =
         LOW_LATENCY_CAPTURE_PERIOD_SIZE;
 
@@ -117,6 +122,20 @@
     .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
 };
 
+static int af_period_multiplier = 4;
+struct pcm_config pcm_config_rt = {
+    .channels = 2,
+    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+    .period_size = ULL_PERIOD_SIZE, //1 ms
+    .period_count = 512, //=> buffer size is 512ms
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = ULL_PERIOD_SIZE*8, //8ms
+    .stop_threshold = INT_MAX,
+    .silence_threshold = 0,
+    .silence_size = 0,
+    .avail_min = ULL_PERIOD_SIZE, //1 ms
+};
+
 struct pcm_config pcm_config_hdmi_multi = {
     .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
     .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
@@ -134,6 +153,19 @@
     .format = PCM_FORMAT_S16_LE,
 };
 
+struct pcm_config pcm_config_audio_capture_rt = {
+    .channels = 2,
+    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+    .period_size = ULL_PERIOD_SIZE,
+    .period_count = 512,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+    .stop_threshold = INT_MAX,
+    .silence_threshold = 0,
+    .silence_size = 0,
+    .avail_min = ULL_PERIOD_SIZE, //1 ms
+};
+
 #define AFE_PROXY_CHANNEL_COUNT 2
 #define AFE_PROXY_SAMPLING_RATE 48000
 
@@ -223,6 +255,7 @@
 
     [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
     [USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
+    [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = "silence-playback",
 };
 
 static const audio_usecase_t offload_usecases[] = {
@@ -284,6 +317,103 @@
 //cache last MBDRC cal step level
 static int last_known_cal_step = -1 ;
 
+static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
+                               int flags __unused)
+{
+    int dir = 0;
+    switch (uc_id) {
+        case USECASE_AUDIO_RECORD_LOW_LATENCY:
+            dir = 1;
+        case USECASE_AUDIO_PLAYBACK_ULL:
+            break;
+        default:
+            return false;
+    }
+
+    int dev_id = platform_get_pcm_device_id(uc_id, dir == 0 ?
+                                            PCM_PLAYBACK : PCM_CAPTURE);
+    if (adev->adm_is_noirq_avail)
+        return adev->adm_is_noirq_avail(adev->adm_data,
+                                        adev->snd_card, dev_id, dir);
+    return false;
+}
+
+static void register_out_stream(struct stream_out *out)
+{
+    struct audio_device *adev = out->dev;
+    if (is_offload_usecase(out->usecase) ||
+        !adev->adm_register_output_stream)
+        return;
+
+    // register stream first for backward compatibility
+    adev->adm_register_output_stream(adev->adm_data,
+                                     out->handle,
+                                     out->flags);
+
+    if (!adev->adm_set_config)
+        return;
+
+    if (out->realtime)
+        adev->adm_set_config(adev->adm_data,
+                             out->handle,
+                             out->pcm, &out->config);
+}
+
+static void register_in_stream(struct stream_in *in)
+{
+    struct audio_device *adev = in->dev;
+    if (!adev->adm_register_input_stream)
+        return;
+
+    adev->adm_register_input_stream(adev->adm_data,
+                                    in->capture_handle,
+                                    in->flags);
+
+    if (!adev->adm_set_config)
+        return;
+
+    if (in->realtime)
+        adev->adm_set_config(adev->adm_data,
+                             in->capture_handle,
+                             in->pcm,
+                             &in->config);
+}
+
+static void request_out_focus(struct stream_out *out, long ns)
+{
+    struct audio_device *adev = out->dev;
+
+    if (adev->adm_request_focus_v2)
+        adev->adm_request_focus_v2(adev->adm_data, out->handle, ns);
+    else if (adev->adm_request_focus)
+        adev->adm_request_focus(adev->adm_data, out->handle);
+}
+
+static void request_in_focus(struct stream_in *in, long ns)
+{
+    struct audio_device *adev = in->dev;
+
+    if (adev->adm_request_focus_v2)
+        adev->adm_request_focus_v2(adev->adm_data, in->capture_handle, ns);
+    else if (adev->adm_request_focus)
+        adev->adm_request_focus(adev->adm_data, in->capture_handle);
+}
+
+static void release_out_focus(struct stream_out *out)
+{
+    struct audio_device *adev = out->dev;
+
+    if (adev->adm_abandon_focus)
+        adev->adm_abandon_focus(adev->adm_data, out->handle);
+}
+
+static void release_in_focus(struct stream_in *in)
+{
+    struct audio_device *adev = in->dev;
+    if (adev->adm_abandon_focus)
+        adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+}
+
 __attribute__ ((visibility ("default")))
 bool audio_hw_send_gain_dep_calibration(int level) {
     bool ret_val = false;
@@ -360,6 +490,7 @@
         format == AUDIO_FORMAT_FLAC ||
         format == AUDIO_FORMAT_ALAC ||
         format == AUDIO_FORMAT_APE ||
+        format == AUDIO_FORMAT_DSD ||
         format == AUDIO_FORMAT_VORBIS ||
         format == AUDIO_FORMAT_WMA ||
         format == AUDIO_FORMAT_WMA_PRO)
@@ -368,6 +499,12 @@
     return false;
 }
 
+static inline bool is_mmap_usecase(audio_usecase_t uc_id)
+{
+    return (uc_id == USECASE_AUDIO_RECORD_AFE_PROXY) ||
+           (uc_id == USECASE_AUDIO_PLAYBACK_AFE_PROXY);
+}
+
 static int get_snd_codec_id(audio_format_t format)
 {
     int id = 0;
@@ -394,6 +531,9 @@
     case AUDIO_FORMAT_APE:
         id = SND_AUDIOCODEC_APE;
         break;
+    case AUDIO_FORMAT_DSD:
+        id = SND_AUDIOCODEC_DSD;
+        break;
     case AUDIO_FORMAT_VORBIS:
         id = SND_AUDIOCODEC_VORBIS;
         break;
@@ -469,6 +609,82 @@
     return 0;
 }
 
+static void enable_asrc_mode(struct audio_device *adev)
+{
+    ALOGV("%s", __func__);
+    audio_route_apply_and_update_path(adev->audio_route,
+                                  "asrc-mode");
+    adev->asrc_mode_enabled = true;
+}
+
+static void disable_asrc_mode(struct audio_device *adev)
+{
+    ALOGV("%s", __func__);
+    audio_route_reset_and_update_path(adev->audio_route,
+                                  "asrc-mode");
+    adev->asrc_mode_enabled = false;
+}
+
+/*
+ * - Enable ASRC mode for incoming mix path use case(Headphone backend)if Headphone
+ *   44.1 or Native DSD backends are enabled for any of current use case.
+ *   e.g. 48-> + (Naitve DSD or Headphone 44.1)
+ * - Disable current mix path use case(Headphone backend) and re-enable it with
+ *   ASRC mode for incoming Headphone 44.1 or Native DSD use case.
+ *   e.g. Naitve DSD or Headphone 44.1 -> + 48
+ */
+static void check_and_set_asrc_mode(struct audio_device *adev, snd_device_t snd_device)
+{
+    ALOGV("%s snd device %d", __func__, snd_device);
+    int new_backend_idx = platform_get_backend_index(snd_device);
+
+    if (((new_backend_idx == HEADPHONE_BACKEND) ||
+            (new_backend_idx == HEADPHONE_44_1_BACKEND) ||
+            (new_backend_idx == DSD_NATIVE_BACKEND)) &&
+            !adev->asrc_mode_enabled) {
+        struct listnode *node = NULL;
+        struct audio_usecase *uc = NULL;
+        struct stream_out *curr_out = NULL;
+        int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
+
+        list_for_each(node, &adev->usecase_list) {
+            uc = node_to_item(node, struct audio_usecase, list);
+            curr_out = (struct stream_out*) uc->stream.out;
+            if (curr_out && PCM_PLAYBACK == uc->type) {
+                usecase_backend_idx = platform_get_backend_index(uc->out_snd_device);
+
+                if((new_backend_idx == HEADPHONE_BACKEND) &&
+                       ((usecase_backend_idx == HEADPHONE_44_1_BACKEND) ||
+                       (usecase_backend_idx == DSD_NATIVE_BACKEND))) {
+                    ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware",
+                          __func__);
+                    enable_asrc_mode(adev);
+                    break;
+                } else if(((new_backend_idx == HEADPHONE_44_1_BACKEND) ||
+                          (new_backend_idx == DSD_NATIVE_BACKEND)) &&
+                          (usecase_backend_idx == HEADPHONE_BACKEND)) {
+                    ALOGD("%s:48K stream detected, disabling and enabling it with asrcmode in hardware",
+                          __func__);
+                    disable_audio_route(adev, uc);
+                    disable_snd_device(adev, uc->out_snd_device);
+                    // Apply true-high-quality-mode if DSD or > 44.1KHz or >=24-bit
+                    if (new_backend_idx == DSD_NATIVE_BACKEND)
+                        audio_route_apply_and_update_path(adev->audio_route,
+                                                "hph-true-highquality-mode");
+                    else if ((new_backend_idx == HEADPHONE_44_1_BACKEND) &&
+                             (curr_out->bit_width >= 24))
+                        audio_route_apply_and_update_path(adev->audio_route,
+                                                     "hph-highquality-mode");
+                    enable_asrc_mode(adev);
+                    enable_snd_device(adev, uc->out_snd_device);
+                    enable_audio_route(adev, uc);
+                    break;
+                }
+            }
+        }
+    }
+}
+
 int pcm_ioctl(struct pcm *pcm, int request, ...)
 {
     va_list ap;
@@ -568,7 +784,6 @@
     if (audio_extn_spkr_prot_is_enabled())
          audio_extn_spkr_prot_calib_cancel(adev);
 
-
     if (platform_can_enable_spkr_prot_on_device(snd_device) &&
          audio_extn_spkr_prot_is_enabled()) {
        if (platform_get_spkr_prot_acdb_id(snd_device) < 0) {
@@ -588,6 +803,13 @@
         }
     } else {
         ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
+
+       if ((SND_DEVICE_OUT_BT_A2DP == snd_device) &&
+           (audio_extn_a2dp_start_playback() < 0)) {
+           ALOGE(" fail to configure A2dp control path ");
+           return -EINVAL;
+       }
+
         /* due to the possibility of calibration overwrite between listen
             and audio, notify listen hal before audio calibration is sent */
         audio_extn_sound_trigger_update_device_status(snd_device,
@@ -644,6 +866,7 @@
 
     if (adev->snd_dev_ref_cnt[snd_device] == 0) {
         ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
+
         if (platform_can_enable_spkr_prot_on_device(snd_device) &&
              audio_extn_spkr_prot_is_enabled()) {
             audio_extn_spkr_prot_stop_processing(snd_device);
@@ -656,7 +879,10 @@
             audio_route_reset_and_update_path(adev->audio_route, device_name);
         }
 
-        if (snd_device == SND_DEVICE_OUT_HDMI)
+        if (SND_DEVICE_OUT_BT_A2DP == snd_device)
+            audio_extn_a2dp_stop_playback();
+
+        if (snd_device == SND_DEVICE_OUT_HDMI || snd_device == SND_DEVICE_OUT_DISPLAY_PORT)
             adev->is_channel_status_set = false;
         else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
                  adev->native_playback_enabled) {
@@ -665,6 +891,11 @@
             audio_route_reset_and_update_path(adev->audio_route,
                                               "true-native-mode");
             adev->native_playback_enabled = false;
+        } else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
+                 adev->asrc_mode_enabled) {
+            ALOGD("%s: %d: disabling asrc mode in hardware", __func__, __LINE__);
+            disable_asrc_mode(adev);
+            audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
         }
 
         audio_extn_dev_arbi_release(snd_device);
@@ -685,7 +916,7 @@
     struct audio_usecase *usecase;
     bool switch_device[AUDIO_USECASE_MAX];
     int i, num_uc_to_switch = 0;
-
+    bool force_restart_session = false;
     /*
      * This function is to make sure that all the usecases that are active on
      * the hardware codec backend are always routed to any one device that is
@@ -705,7 +936,15 @@
      */
     bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info,
                          snd_device);
-
+    /* For a2dp device reconfigure all active sessions
+     * with new AFE encoder format based on a2dp state
+     */
+    if ((SND_DEVICE_OUT_BT_A2DP == snd_device ||
+         SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) &&
+         audio_extn_a2dp_is_force_device_switch()) {
+         force_routing = true;
+         force_restart_session = true;
+    }
     ALOGD("%s:becf: force routing %d", __func__, force_routing);
 
     /* Disable all the usecases on the shared backend other than the
@@ -724,9 +963,13 @@
               platform_check_backends_match(snd_device, usecase->out_snd_device));
         if (usecase->type != PCM_CAPTURE &&
             usecase != uc_info &&
-            (usecase->out_snd_device != snd_device || force_routing)  &&
-            usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND &&
-            platform_check_backends_match(snd_device, usecase->out_snd_device)) {
+            (usecase->out_snd_device != snd_device || force_routing) &&
+            ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
+             (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
+             (usecase->devices & AUDIO_DEVICE_OUT_USB_DEVICE) ||
+             (force_restart_session)) &&
+             (platform_check_backends_match(snd_device, usecase->out_snd_device))) {
+
                 ALOGD("%s:becf: check_usecases (%s) is active on (%s) - disabling ..",
                     __func__, use_case_table[usecase->id],
                       platform_get_snd_device_name(usecase->out_snd_device));
@@ -874,6 +1117,12 @@
 
     reset_hdmi_sink_caps(out);
 
+    /* Cache ext disp type */
+    if (platform_get_ext_disp_type(adev->platform) <= 0) {
+        ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
+        return -EINVAL;
+    }
+
     switch (channels) {
     case 8:
         ALOGV("%s: HDMI supports 7.1 channels", __func__);
@@ -997,6 +1246,29 @@
     return active;
 }
 
+/*
+ * if native DSD playback active
+ */
+bool audio_is_dsd_native_stream_active(struct audio_device *adev)
+{
+    bool active = false;
+    struct listnode *node = NULL;
+    struct audio_usecase *uc = NULL;
+    struct stream_out *curr_out = NULL;
+
+    list_for_each(node, &adev->usecase_list) {
+        uc = node_to_item(node, struct audio_usecase, list);
+        curr_out = (struct stream_out*) uc->stream.out;
+
+        if (curr_out && PCM_PLAYBACK == uc->type &&
+               (DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) {
+            active = true;
+            ALOGV("%s:DSD playback is active", __func__);
+            break;
+        }
+    }
+    return active;
+}
 
 static bool force_device_switch(struct audio_usecase *usecase)
 {
@@ -1016,6 +1288,14 @@
         }
     }
 
+    // Force all a2dp output devices to reconfigure for proper AFE encode format
+    if((usecase->stream.out) &&
+       (usecase->stream.out->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
+       audio_extn_a2dp_is_force_device_switch()) {
+         ALOGD("Force a2dp device switch to update new encoder config");
+         ret = true;
+     }
+
     return ret;
 }
 
@@ -1060,6 +1340,8 @@
                                                get_usecase_id_from_usecase_type(adev, VOICE_CALL));
             if ((vc_usecase) && (((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
                                  (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) ||
+                                 ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+                                 (usecase->devices & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) ||
                                 (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
                 in_snd_device = vc_usecase->in_snd_device;
                 out_snd_device = vc_usecase->out_snd_device;
@@ -1067,7 +1349,8 @@
         } else if (voice_extn_compress_voip_is_active(adev)) {
             voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
             if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
-                (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+                ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
+                  ((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) &&
                  (voip_usecase->stream.out != adev->primary_output))) {
                     in_snd_device = voip_usecase->in_snd_device;
                     out_snd_device = voip_usecase->out_snd_device;
@@ -1169,6 +1452,8 @@
     /* Enable new sound devices */
     if (out_snd_device != SND_DEVICE_NONE) {
         check_usecases_codec_backend(adev, usecase, out_snd_device);
+        if (platform_check_codec_asrc_support(adev->platform))
+            check_and_set_asrc_mode(adev, out_snd_device);
         enable_snd_device(adev, out_snd_device);
     }
 
@@ -1182,10 +1467,6 @@
                                                         out_snd_device,
                                                         in_snd_device);
         enable_audio_route_for_voice_usecases(adev, usecase);
-        /* Enable sidetone only if voice/voip call already exists */
-        if (voice_is_call_state_active(adev) ||
-            voice_extn_compress_voip_is_started(adev))
-            voice_set_sidetone(adev, out_snd_device, true);
     }
 
     usecase->in_snd_device = in_snd_device;
@@ -1202,10 +1483,23 @@
                                                 usecase->stream.out->channel_mask,
                                                 &usecase->stream.out->app_type_cfg);
         ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
+
+        /* Notify device change info to effect clients registered */
+        audio_extn_gef_notify_device_config(
+                usecase->stream.out->devices,
+                usecase->stream.out->channel_mask,
+                platform_get_snd_device_acdb_id(usecase->out_snd_device));
     }
 
     enable_audio_route(adev, usecase);
 
+    if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
+        /* Enable sidetone only if other voice/voip call already exists */
+        if (voice_is_call_state_active(adev) ||
+            voice_extn_compress_voip_is_started(adev))
+            voice_set_sidetone(adev, out_snd_device, true);
+    }
+
     /* Applicable only on the targets that has external modem.
      * Enable device command should be sent to modem only after
      * enabling voice call mixer controls
@@ -1324,6 +1618,8 @@
     if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
         flags |= PCM_MMAP | PCM_NOIRQ;
         pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+    } else if (in->realtime) {
+        flags |= PCM_MMAP | PCM_NOIRQ;
     }
 
     while (1) {
@@ -1354,6 +1650,13 @@
         goto error_open;
     }
 
+    register_in_stream(in);
+    if (in->realtime) {
+        ret = pcm_start(in->pcm);
+        if (ret < 0)
+            goto error_open;
+    }
+
     audio_extn_perf_lock_release(&adev->perf_lock_handle);
     ALOGD("%s: exit", __func__);
 
@@ -1614,193 +1917,6 @@
     return 0;
 }
 
-static bool allow_hdmi_channel_config(struct audio_device *adev,
-                                      bool enable_passthru)
-{
-    struct listnode *node;
-    struct audio_usecase *usecase;
-    bool ret = true;
-
-    if (enable_passthru && !audio_extn_passthru_is_enabled()) {
-        ret = false;
-        goto exit;
-    }
-
-    if (audio_extn_passthru_is_active()) {
-        ALOGI("%s: Compress audio passthrough is active,"
-              "no HDMI config change allowed", __func__);
-        ret = false;
-        goto exit;
-    }
-
-    list_for_each(node, &adev->usecase_list) {
-        usecase = node_to_item(node, struct audio_usecase, list);
-        if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-            /*
-             * If voice call is already existing, do not proceed further to avoid
-             * disabling/enabling both RX and TX devices, CSD calls, etc.
-             * Once the voice call done, the HDMI channels can be configured to
-             * max channels of remaining use cases.
-             */
-            if (usecase->id == USECASE_VOICE_CALL) {
-                ALOGV("%s: voice call is active, no change in HDMI channels",
-                      __func__);
-                ret = false;
-                break;
-            } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
-                if (!enable_passthru) {
-                    ALOGV("%s: multi channel playback is active, "
-                          "no change in HDMI channels", __func__);
-                    ret = false;
-                    break;
-                }
-            } else if (is_offload_usecase(usecase->id) &&
-                       audio_channel_count_from_out_mask(usecase->stream.out->channel_mask) > 2) {
-                if (!enable_passthru) {
-                    ALOGD("%s:multi-channel(%x) compress offload playback is active"
-                        ", no change in HDMI channels", __func__,
-                        usecase->stream.out->channel_mask);
-                    ret = false;
-                    break;
-                }
-            }
-        }
-    }
-    ALOGV("allow hdmi config %d", ret);
-exit:
-    return ret;
-}
-
-static int check_and_set_hdmi_config(struct audio_device *adev,
-                                     uint32_t channels,
-                                     uint32_t sample_rate,
-                                     audio_format_t format,
-                                     bool enable_passthru)
-{
-    struct listnode *node;
-    struct audio_usecase *usecase;
-    int32_t factor = 1;
-    bool config = false;
-
-    ALOGV("%s channels %d sample_rate %d format:%x enable_passthru:%d",
-         __func__, channels, sample_rate, format, enable_passthru);
-
-    if (channels != adev->cur_hdmi_channels) {
-        ALOGV("channel does not match current hdmi channels");
-        config = true;
-    }
-
-    if (sample_rate != adev->cur_hdmi_sample_rate) {
-        ALOGV("sample rate does not match current hdmi sample rate");
-        config = true;
-    }
-
-    if (format != adev->cur_hdmi_format) {
-        ALOGV("format does not match current hdmi format");
-        config = true;
-    }
-
-    /* TBD - add check for bit width */
-    if (!config) {
-        ALOGV("No need to config hdmi");
-        return 0;
-    }
-
-    if (enable_passthru &&
-        (format == AUDIO_FORMAT_E_AC3)) {
-        ALOGV("factor 4 for E_AC3 passthru");
-        factor = 4;
-    }
-
-    platform_set_hdmi_config(adev->platform, channels, factor * sample_rate,
-                             enable_passthru);
-    adev->cur_hdmi_channels = channels;
-    adev->cur_hdmi_format = format;
-    adev->cur_hdmi_sample_rate = sample_rate;
-
-    /*
-     * Deroute all the playback streams routed to HDMI so that
-     * the back end is deactivated. Note that backend will not
-     * be deactivated if any one stream is connected to it.
-     */
-    list_for_each(node, &adev->usecase_list) {
-        usecase = node_to_item(node, struct audio_usecase, list);
-        if (usecase->type == PCM_PLAYBACK &&
-                usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-            disable_audio_route(adev, usecase);
-        }
-    }
-
-    bool was_active = audio_extn_keep_alive_is_active();
-    if (was_active)
-        audio_extn_keep_alive_stop();
-
-    /*
-     * Enable all the streams disabled above. Now the HDMI backend
-     * will be activated with new channel configuration
-     */
-    list_for_each(node, &adev->usecase_list) {
-        usecase = node_to_item(node, struct audio_usecase, list);
-        if (usecase->type == PCM_PLAYBACK &&
-                usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-            enable_audio_route(adev, usecase);
-        }
-    }
-
-    if (was_active)
-        audio_extn_keep_alive_start();
-
-    return 0;
-}
-
-/* called with out lock taken */
-static int check_and_set_hdmi_backend(struct stream_out *out)
-{
-    struct audio_device *adev = out->dev;
-    int ret;
-    bool enable_passthru = false;
-
-    if (!(out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL))
-        return -1;
-
-    ALOGV("%s usecase %s out->format:%x out->bit_width:%d", __func__, use_case_table[out->usecase],out->format,out->bit_width);
-
-    if (is_offload_usecase(out->usecase) &&
-        audio_extn_passthru_is_passthrough_stream(out)) {
-        enable_passthru = true;
-        ALOGV("%s : enable_passthru is set to true", __func__);
-    }
-
-    /* Check if change in HDMI channel config is allowed */
-    if (!allow_hdmi_channel_config(adev, enable_passthru)) {
-        return -EPERM;
-    }
-
-    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
-        uint32_t channels;
-        ALOGV("Offload usecase, enable passthru %d", enable_passthru);
-
-        if (enable_passthru) {
-            audio_extn_passthru_on_start(out);
-            audio_extn_passthru_update_stream_configuration(adev, out);
-        }
-
-        /* For pass through case, the backend should be configured as stereo */
-        channels = enable_passthru ? DEFAULT_HDMI_OUT_CHANNELS :
-                                     out->compr_config.codec->ch_in;
-
-        ret = check_and_set_hdmi_config(adev, channels,
-                                        out->sample_rate, out->format,
-                                        enable_passthru);
-    } else
-        ret = check_and_set_hdmi_config(adev, out->config.channels,
-                                        out->config.rate,
-                                        out->format,
-                                        false);
-    return ret;
-}
-
-
 static int stop_output_stream(struct stream_out *out)
 {
     int ret = 0;
@@ -1841,17 +1957,14 @@
         ALOGV("Disable passthrough , reset mixer to pcm");
         /* NO_PASSTHROUGH */
         out->compr_config.codec->compr_passthr = 0;
-
         audio_extn_passthru_on_stop(out);
         audio_extn_dolby_set_dap_bypass(adev, DAP_STATE_ON);
     }
 
     /* Must be called after removing the usecase from list */
     if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
-        check_and_set_hdmi_config(adev, DEFAULT_HDMI_OUT_CHANNELS,
-                                  DEFAULT_HDMI_OUT_SAMPLE_RATE,
-                                  DEFAULT_HDMI_OUT_FORMAT,
-                                  false);
+        audio_extn_keep_alive_start();
+
     ALOGV("%s: exit: status(%d)", __func__, ret);
     return ret;
 }
@@ -1893,12 +2006,6 @@
         goto error_config;
     }
 
-    /* This must be called before adding this usecase to the list */
-    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-        /* This call can fail if compress pass thru is already active */
-        check_and_set_hdmi_backend(out);
-    }
-
     uc_info->id = out->usecase;
     uc_info->type = PCM_PLAYBACK;
     uc_info->stream.out = out;
@@ -1910,6 +2017,16 @@
     audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
                                  adev->perf_lock_opts,
                                  adev->perf_lock_opts_size);
+
+    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+        audio_extn_keep_alive_stop();
+        if (audio_extn_passthru_is_enabled() &&
+            audio_extn_passthru_is_passthrough_stream(out)) {
+            audio_extn_passthru_on_start(out);
+            audio_extn_passthru_update_stream_configuration(adev, out);
+        }
+    }
+
     select_devices(adev, out->usecase);
 
     ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
@@ -1920,6 +2037,8 @@
         if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
             flags |= PCM_MMAP | PCM_NOIRQ;
             pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+        } else if (out->realtime) {
+            flags |= PCM_MMAP | PCM_NOIRQ;
         } else
             flags |= PCM_MONOTONIC;
 
@@ -2000,10 +2119,20 @@
             audio_extn_check_and_set_dts_hpx_state(adev);
         }
     }
+
+    if (ret == 0) {
+        register_out_stream(out);
+        if (out->realtime) {
+            ret = pcm_start(out->pcm);
+            if (ret < 0)
+                goto error_open;
+        }
+    }
+
     audio_extn_perf_lock_release(&adev->perf_lock_handle);
     ALOGD("%s: exit", __func__);
 
-    return 0;
+    return ret;
 error_open:
     audio_extn_perf_lock_release(&adev->perf_lock_handle);
     stop_output_stream(out);
@@ -2086,6 +2215,35 @@
     return size;
 }
 
+static size_t get_output_period_size(uint32_t sample_rate,
+                                    audio_format_t format,
+                                    int channel_count,
+                                    int duration /*in millisecs*/)
+{
+    size_t size = 0;
+    uint32_t bytes_per_sample = audio_bytes_per_sample(format);
+
+    if ((duration == 0) || (sample_rate == 0) ||
+        (bytes_per_sample == 0) || (channel_count == 0)) {
+        ALOGW("Invalid config duration %d sr %d bps %d ch %d", duration, sample_rate,
+               bytes_per_sample, channel_count);
+        return -EINVAL;
+    }
+
+    size = (sample_rate *
+            duration *
+            bytes_per_sample *
+            channel_count) / 1000;
+    /*
+     * To have same PCM samples for all channels, the buffer size requires to
+     * be multiple of (number of channels * bytes per sample)
+     * For writes to succeed, the buffer must be written at address which is multiple of 32
+     */
+    size = ALIGN(size, (bytes_per_sample * channel_count * 32));
+
+    return (size/(channel_count * bytes_per_sample));
+}
+
 static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
 {
     uint64_t actual_frames_rendered = 0;
@@ -2141,7 +2299,7 @@
     else if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)
         return out->hal_fragment_size;
 
-    return out->config.period_size *
+    return out->config.period_size * out->af_period_multiplier *
                 audio_stream_out_frame_size((const struct audio_stream_out *)stream);
 }
 
@@ -2172,13 +2330,6 @@
 
     ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
           stream, out->usecase, use_case_table[out->usecase]);
-    if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
-        /* Ignore standby in case of voip call because the voip output
-         * stream is closed in adev_close_output_stream()
-         */
-        ALOGD("%s: Ignore Standby in VOIP call", __func__);
-        return 0;
-    }
 
     lock_output_stream(out);
     if (!out->standby) {
@@ -2190,7 +2341,13 @@
 
         pthread_mutex_lock(&adev->lock);
         out->standby = true;
-        if (!is_offload_usecase(out->usecase)) {
+        if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
+            voice_extn_compress_voip_close_output_stream(stream);
+            pthread_mutex_unlock(&adev->lock);
+            pthread_mutex_unlock(&out->lock);
+            ALOGD("VOIP output entered standby");
+            return 0;
+        } else if (!is_offload_usecase(out->usecase)) {
             if (out->pcm) {
                 pcm_close(out->pcm);
                 out->pcm = NULL;
@@ -2284,6 +2441,17 @@
                 (platform_get_edid_info(adev->platform) != 0) /* HDMI disconnected */) {
             val = AUDIO_DEVICE_OUT_SPEAKER;
         }
+        /*
+         * When A2DP is disconnected the
+         * music playback is paused and the policy manager sends routing=0
+         * But the audioflingercontinues to write data until standby time
+         * (3sec). As BT is turned off, the write gets blocked.
+         * Avoid this by routing audio to speaker until standby.
+         */
+        if ((out->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
+                (val == AUDIO_DEVICE_NONE)) {
+                val = AUDIO_DEVICE_OUT_SPEAKER;
+        }
 
         /*
          * select_devices() call below switches all the usecases on the same
@@ -2304,15 +2472,9 @@
          *       playback to headset.
          */
         if (val != 0) {
-            out->devices = val;
-
-            if (!out->standby) {
-                audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
-                                             adev->perf_lock_opts,
-                                             adev->perf_lock_opts_size);
-                select_devices(adev, out->usecase);
-                audio_extn_perf_lock_release(&adev->perf_lock_handle);
-            }
+            audio_devices_t new_dev = val;
+            bool same_dev = out->devices == new_dev;
+            out->devices = new_dev;
 
             if (output_drives_call(adev, out)) {
                 if(!voice_is_in_call(adev)) {
@@ -2325,6 +2487,21 @@
                     voice_update_devices_for_all_voice_usecases(adev);
                 }
             }
+
+            if (!out->standby) {
+                if (!same_dev) {
+                    ALOGV("update routing change");
+                    audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
+                                                 adev->perf_lock_opts,
+                                                 adev->perf_lock_opts_size);
+                    if (adev->adm_on_routing_change)
+                        adev->adm_on_routing_change(adev->adm_data,
+                                                    out->handle);
+                }
+                select_devices(adev, out->usecase);
+                if (!same_dev)
+                    audio_extn_perf_lock_release(&adev->perf_lock_handle);
+            }
         }
 
         pthread_mutex_unlock(&adev->lock);
@@ -2478,11 +2655,21 @@
 
 static uint32_t out_get_latency(const struct audio_stream_out *stream)
 {
+    uint32_t period_ms;
     struct stream_out *out = (struct stream_out *)stream;
     uint32_t latency = 0;
 
     if (is_offload_usecase(out->usecase)) {
         latency = COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
+    } else if (out->realtime) {
+        // since the buffer won't be filled up faster than realtime,
+        // return a smaller number
+        if (out->config.rate)
+            period_ms = (out->af_period_multiplier * out->config.period_size *
+                         1000) / (out->config.rate);
+        else
+            period_ms = 0;
+        latency = period_ms + platform_render_latency(out->usecase)/1000;
     } else {
         latency = (out->config.period_count * out->config.period_size * 1000) /
            (out->config.rate);
@@ -2492,6 +2679,17 @@
     return latency;
 }
 
+static float AmpToDb(float amplification)
+{
+    float db = DSD_VOLUME_MIN_DB;
+    if (amplification > 0) {
+        db = 20 * log10(amplification);
+        if(db < DSD_VOLUME_MIN_DB)
+            return DSD_VOLUME_MIN_DB;
+    }
+    return db;
+}
+
 static int out_set_volume(struct audio_stream_out *stream, float left,
                           float right)
 {
@@ -2510,6 +2708,20 @@
              * Mute is 0 and unmute 1
              */
             audio_extn_passthru_set_volume(out, (left == 0.0f));
+        } else if (out->format == AUDIO_FORMAT_DSD){
+            char mixer_ctl_name[128] =  "DSD Volume";
+            struct audio_device *adev = out->dev;
+            struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+
+            if (!ctl) {
+                ALOGE("%s: Could not get ctl for mixer cmd - %s",
+                      __func__, mixer_ctl_name);
+                return -EINVAL;
+            }
+            volume[0] = (int)(AmpToDb(left));
+            volume[1] = (int)(AmpToDb(right));
+            mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
+            return 0;
         } else {
             char mixer_ctl_name[128];
             struct audio_device *adev = out->dev;
@@ -2556,8 +2768,11 @@
             /* increase written size during SSR to avoid mismatch
              * with the written frames count in AF
              */
-            if (audio_bytes_per_sample(out->format) != 0)
-                out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
+            // bytes per frame
+            size_t bpf = audio_bytes_per_sample(out->format) *
+                         audio_channel_count_from_out_mask(out->channel_mask);
+            if (bpf != 0)
+                out->written += bytes / bpf;
             ALOGD(" %s: sound card is not active/SSR state", __func__);
             ret= -EIO;
             goto exit;
@@ -2565,9 +2780,10 @@
     }
 
     if (audio_extn_passthru_should_drop_data(out)) {
-        ALOGD(" %s : Drop data as compress passthrough session is going on", __func__);
-        usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
-                        out_get_sample_rate(&out->stream.common));
+        ALOGV(" %s : Drop data as compress passthrough session is going on", __func__);
+        if (audio_bytes_per_sample(out->format) != 0)
+            out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
+        ret = -EIO;
         goto exit;
     }
 
@@ -2589,9 +2805,6 @@
             ALOGD("%s: retry previous failed cal level set", __func__);
             audio_hw_send_gain_dep_calibration(last_known_cal_step);
         }
-
-        if (!is_offload_usecase(out->usecase) && adev->adm_register_output_stream)
-            adev->adm_register_output_stream(adev->adm_data, out->handle, out->flags);
     }
 
     if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
@@ -2647,7 +2860,8 @@
         if (ret < 0)
             ret = -errno;
         ALOGVV("%s: writing buffer (%zu bytes) to compress device returned %zd", __func__, bytes, ret);
-        if (ret >= 0 && ret < (ssize_t)bytes) {
+        /*msg to cb thread only if non blocking write is enabled*/
+        if (ret >= 0 && ret < (ssize_t)bytes && out->non_blocking) {
             ALOGD("No space available in compress driver, post msg to cb thread");
             send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
         } else if (-ENETRESET == ret) {
@@ -2660,8 +2874,14 @@
         if ( ret == (ssize_t)bytes && !out->non_blocking)
             out->written += bytes;
 
-        if (!out->playback_started && ret >= 0) {
-            compress_start(out->compr);
+        /* Call compr start only when non-zero bytes of data is there to be rendered */
+        if (!out->playback_started && ret > 0) {
+            int status = compress_start(out->compr);
+            if (status < 0) {
+                ret = status;
+                ALOGE("%s: compr start failed with err %d", __func__, errno);
+                goto exit;
+            }
             audio_extn_dts_eagle_fade(adev, true, out);
             out->playback_started = 1;
             out->offload_state = OFFLOAD_STATE_PLAYING;
@@ -2679,12 +2899,19 @@
 
             ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes);
 
-            if (adev->adm_request_focus)
-                adev->adm_request_focus(adev->adm_data, out->handle);
+            long ns = 0;
 
-            if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
+            if (out->config.rate)
+                ns = pcm_bytes_to_frames(out->pcm, bytes)*1000000000LL/
+                                                     out->config.rate;
+
+            bool use_mmap = is_mmap_usecase(out->usecase) || out->realtime;
+
+            request_out_focus(out, ns);
+
+            if (use_mmap)
                 ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
-            } else if (out->hal_op_format != out->hal_ip_format &&
+            else if (out->hal_op_format != out->hal_ip_format &&
                        out->convert_buffer != NULL) {
 
                 memcpy_by_audio_format(out->convert_buffer,
@@ -2701,15 +2928,14 @@
                 ret = pcm_write(out->pcm, (void *)buffer, bytes);
             }
 
+            release_out_focus(out);
+
             if (ret < 0)
                 ret = -errno;
             else if (ret == 0 && (audio_bytes_per_sample(out->format) != 0))
                 out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
             else
                 ret = -EINVAL;
-
-            if (adev->adm_abandon_focus)
-                adev->adm_abandon_focus(adev->adm_data, out->handle);
         }
     }
 
@@ -2732,8 +2958,9 @@
             out->standby = true;
         }
         out_standby(&out->stream.common);
-        usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
-                        out_get_sample_rate(&out->stream.common));
+        if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
+            usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
+                            out_get_sample_rate(&out->stream.common));
     }
     return bytes;
 }
@@ -2939,7 +3166,6 @@
                 if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
                     pthread_mutex_lock(&out->dev->lock);
                     ALOGV("offload resume, check and set hdmi backend again");
-                    check_and_set_hdmi_backend(out);
                     pthread_mutex_unlock(&out->dev->lock);
                 }
                 status = compress_resume(out->compr);
@@ -2979,8 +3205,12 @@
     if (is_offload_usecase(out->usecase)) {
         ALOGD("copl(%p):calling compress flush", out);
         lock_output_stream(out);
-        stop_compressed_output_l(out);
-        out->written = 0;
+        if (out->offload_state == OFFLOAD_STATE_PAUSED) {
+            stop_compressed_output_l(out);
+            out->written = 0;
+        } else {
+            ALOGW("%s called in invalid state %d", __func__, out->offload_state);
+        }
         pthread_mutex_unlock(&out->lock);
         ALOGD("copl(%p):out of compress flush", out);
         return 0;
@@ -3011,8 +3241,8 @@
     else if(audio_extn_compr_cap_usecase_supported(in->usecase))
         return audio_extn_compr_cap_get_buffer_size(in->config.format);
 
-    return in->config.period_size *
-                audio_stream_in_frame_size((const struct audio_stream_in *)stream);
+    return in->config.period_size * in->af_period_multiplier *
+        audio_stream_in_frame_size((const struct audio_stream_in *)stream);
 }
 
 static uint32_t in_get_channels(const struct audio_stream *stream)
@@ -3043,14 +3273,6 @@
     ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
           stream, in->usecase, use_case_table[in->usecase]);
 
-    if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
-        /* Ignore standby in case of voip call because the voip input
-         * stream is closed in adev_close_input_stream()
-         */
-        ALOGV("%s: Ignore Standby in VOIP call", __func__);
-        return status;
-    }
-
     lock_input_stream(in);
     if (!in->standby && in->is_st_session) {
         ALOGD("%s: sound trigger pcm stop lab", __func__);
@@ -3064,11 +3286,16 @@
 
         pthread_mutex_lock(&adev->lock);
         in->standby = true;
-        if (in->pcm) {
-            pcm_close(in->pcm);
-            in->pcm = NULL;
+        if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
+            voice_extn_compress_voip_close_input_stream(stream);
+            ALOGD("VOIP input entered standby");
+        } else {
+            if (in->pcm) {
+                pcm_close(in->pcm);
+                in->pcm = NULL;
+            }
+            status = stop_input_stream(in);
         }
-        status = stop_input_stream(in);
         pthread_mutex_unlock(&adev->lock);
     }
     pthread_mutex_unlock(&in->lock);
@@ -3125,8 +3352,13 @@
         if (((int)in->device != val) && (val != 0)) {
             in->device = val;
             /* If recording is in progress, change the tx device to new device */
-            if (!in->standby && !in->is_st_session)
+            if (!in->standby && !in->is_st_session) {
+                ALOGV("update input routing change");
+                if (adev->adm_on_routing_change)
+                        adev->adm_on_routing_change(adev->adm_data,
+                                                    in->capture_handle);
                 ret = select_devices(adev, in->usecase);
+            }
         }
     }
 
@@ -3212,19 +3444,24 @@
             goto exit;
         }
         in->standby = 0;
-        if (adev->adm_register_input_stream)
-            adev->adm_register_input_stream(adev->adm_data, in->capture_handle, in->flags);
     }
 
-    if (adev->adm_request_focus)
-        adev->adm_request_focus(adev->adm_data, in->capture_handle);
+    // what's the duration requested by the client?
+    long ns = 0;
+
+    if (in->config.rate)
+        ns = pcm_bytes_to_frames(in->pcm, bytes)*1000000000LL/
+                                             in->config.rate;
+
+    request_in_focus(in, ns);
+    bool use_mmap = is_mmap_usecase(in->usecase) || in->realtime;
 
     if (in->pcm) {
         if (audio_extn_ssr_get_stream() == in) {
             ret = audio_extn_ssr_read(stream, buffer, bytes);
         } else if (audio_extn_compr_cap_usecase_supported(in->usecase)) {
             ret = audio_extn_compr_cap_read(in, buffer, bytes);
-        } else if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
+        } else if (use_mmap) {
             ret = pcm_mmap_read(in->pcm, buffer, bytes);
         } else {
             ret = pcm_read(in->pcm, buffer, bytes);
@@ -3246,8 +3483,7 @@
         }
     }
 
-    if (adev->adm_abandon_focus)
-        adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+    release_in_focus(in);
 
     /*
      * Instead of writing zeroes here, we could trust the hardware
@@ -3333,7 +3569,7 @@
     return add_remove_audio_effect(stream, effect, false);
 }
 
-static int adev_open_output_stream(struct audio_hw_device *dev,
+int adev_open_output_stream(struct audio_hw_device *dev,
                                    audio_io_handle_t handle,
                                    audio_devices_t devices,
                                    audio_output_flags_t flags,
@@ -3514,7 +3750,12 @@
         else if (config->channel_mask) {
             out->channel_mask = config->channel_mask;
             config->offload_info.channel_mask = config->channel_mask;
+        } else {
+            ALOGE("out->channel_mask not set for OFFLOAD/DIRECT_PCM");
+            ret = -EINVAL;
+            goto error_open;
         }
+
         format = out->format = config->offload_info.format;
         out->sample_rate = config->offload_info.sample_rate;
 
@@ -3531,7 +3772,7 @@
         out->compr_config.codec->bit_rate =
                     config->offload_info.bit_rate;
         out->compr_config.codec->ch_in =
-                audio_channel_count_from_out_mask(config->channel_mask);
+                audio_channel_count_from_out_mask(out->channel_mask);
         out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
         out->bit_width = AUDIO_OUTPUT_BIT_WIDTH;
         /*TODO: Do we need to change it for passthrough */
@@ -3609,19 +3850,30 @@
 
         audio_extn_dts_create_state_notifier_node(out->usecase);
 
-        create_offload_callback_thread(out);
         ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
                 __func__, config->offload_info.version,
                 config->offload_info.bit_rate);
 
+        /* Check if DSD audio format is supported in codec
+         * and there is no active native DSD use case
+         */
+
+        if ((config->format == AUDIO_FORMAT_DSD) &&
+                (!platform_check_codec_dsd_support(adev->platform) ||
+                audio_is_dsd_native_stream_active(adev))) {
+            ret = -EINVAL;
+            goto error_open;
+        }
+
         /* Disable gapless if any of the following is true
          * passthrough playback
          * AV playback
          * Direct PCM playback
          */
         if (audio_extn_passthru_is_passthrough_stream(out) ||
-            config->offload_info.has_video ||
-            out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+                (config->format == AUDIO_FORMAT_DSD) ||
+                config->offload_info.has_video ||
+                out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
             check_and_set_gapless_mode(adev, false);
         } else
             check_and_set_gapless_mode(adev, true);
@@ -3629,6 +3881,13 @@
         if (audio_extn_passthru_is_passthrough_stream(out)) {
             out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
         }
+        if (config->format == AUDIO_FORMAT_DSD) {
+            out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
+            out->compr_config.codec->compr_passthr = PASSTHROUGH_DSD;
+        }
+
+        create_offload_callback_thread(out);
+
     } else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
         ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
         if (ret != 0) {
@@ -3659,15 +3918,35 @@
         out->config = pcm_config_afe_proxy_playback;
         adev->voice_tx_output = out;
     } else {
+        unsigned int channels = 0;
+        /*Update config params to default if not set by the caller*/
+        if (config->sample_rate == 0)
+            config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        if (config->channel_mask == AUDIO_CHANNEL_NONE)
+            config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+        if (config->format == AUDIO_FORMAT_DEFAULT)
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
+
+        channels = audio_channel_count_from_out_mask(out->channel_mask);
+
         if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
             out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
-            out->config = pcm_config_low_latency;
+            out->realtime = may_use_noirq_mode(adev, USECASE_AUDIO_PLAYBACK_ULL,
+                                               out->flags);
+            out->config = out->realtime ? pcm_config_rt : pcm_config_low_latency;
         } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
             out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
             out->config = pcm_config_low_latency;
         } else if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
             out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
             out->config = pcm_config_deep_buffer;
+            out->config.period_size = get_output_period_size(config->sample_rate, out->format,
+                                                 channels, DEEP_BUFFER_OUTPUT_PERIOD_DURATION);
+            if (out->config.period_size <= 0) {
+                ALOGE("Invalid configuration period size is not valid");
+                ret = -EINVAL;
+                goto error_open;
+            }
         } else {
             /* primary path is the default path selected if no other outputs are available/suitable */
             out->usecase = USECASE_AUDIO_PLAYBACK_PRIMARY;
@@ -3679,7 +3958,7 @@
         out->bit_width = format_to_bitwidth_table[out->hal_op_format] << 3;
         out->config.rate = config->sample_rate;
         out->sample_rate = out->config.rate;
-        out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
+        out->config.channels = channels;
         if (out->hal_ip_format != out->hal_op_format) {
             uint32_t buffer_size = out->config.period_size *
                                    format_to_bitwidth_table[out->hal_op_format] *
@@ -3748,6 +4027,7 @@
     out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
     out->stream.get_presentation_position = out_get_presentation_position;
 
+    out->af_period_multiplier  = out->realtime ? af_period_multiplier : 1;
     out->standby = 1;
     /* out->muted = false; by calloc() */
     /* out->written = 0; by calloc() */
@@ -3776,7 +4056,7 @@
     return ret;
 }
 
-static void adev_close_output_stream(struct audio_hw_device *dev __unused,
+void adev_close_output_stream(struct audio_hw_device *dev __unused,
                                      struct audio_stream_out *stream)
 {
     struct stream_out *out = (struct stream_out *)stream;
@@ -3942,10 +4222,16 @@
     if (ret >= 0) {
         val = atoi(value);
         if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-            ALOGV("cache new edid");
+            ALOGV("cache new ext disp type and edid");
+            ret = platform_get_ext_disp_type(adev->platform);
+            if (ret < 0) {
+                ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
+                status = ret;
+                goto done;
+            }
             platform_cache_edid(adev->platform);
         } else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
-                   (val & AUDIO_DEVICE_IN_USB_DEVICE)) {
+                   !(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.
@@ -3969,7 +4255,7 @@
             ALOGV("invalidate cached edid");
             platform_invalidate_hdmi_config(adev->platform);
         } else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
-                   (val & AUDIO_DEVICE_IN_USB_DEVICE)) {
+                   !(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
             ret = str_parms_get_str(parms, "card", value, sizeof(value));
             if (ret >= 0) {
                 audio_extn_usb_remove_device(val, atoi(value));
@@ -3979,8 +4265,26 @@
         }
     }
 
+    ret = str_parms_get_str(parms,"reconfigA2dp", value, sizeof(value));
+    if (ret >= 0) {
+        struct audio_usecase *usecase;
+        struct listnode *node;
+        list_for_each(node, &adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            if ((usecase->type == PCM_PLAYBACK) &&
+                (usecase->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP)){
+                ALOGD("reconfigure a2dp... forcing device switch");
+                lock_output_stream(usecase->stream.out);
+                audio_extn_a2dp_set_handoff_mode(true);
+                //force device switch to re configure encoder
+                select_devices(adev, usecase->id);
+                audio_extn_a2dp_set_handoff_mode(false);
+                pthread_mutex_unlock(&usecase->stream.out->lock);
+                break;
+            }
+        }
+    }
     audio_extn_set_parameters(adev, parms);
-
 done:
     str_parms_destroy(parms);
     pthread_mutex_unlock(&adev->lock);
@@ -4188,12 +4492,21 @@
 #if LOW_LATENCY_CAPTURE_USE_CASE
         in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
 #endif
+        in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags);
     }
-    in->config = pcm_config_audio_capture;
-    in->config.rate = config->sample_rate;
+
     in->format = config->format;
+    if (in->realtime) {
+        in->config = pcm_config_audio_capture_rt;
+        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;
-    in->sample_rate = config->sample_rate;
 
     if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
         if (adev->mode != AUDIO_MODE_IN_CALL) {
@@ -4269,14 +4582,17 @@
             }
         }
 
-        in->format = config->format;
         in->config.channels = channel_count;
-        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;
+        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;
+        }
+
         if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
                (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
                (voice_extn_compress_voip_is_format_supported(in->format)) &&
@@ -4357,8 +4673,11 @@
     if ((--audio_device_ref_count) == 0) {
         audio_extn_sound_trigger_deinit(adev);
         audio_extn_listen_deinit(adev);
+        if (audio_extn_qaf_is_enabled())
+            audio_extn_qaf_deinit();
         audio_extn_utils_release_streams_output_cfg_list(&adev->streams_output_cfg_list);
         audio_route_free(adev->audio_route);
+        audio_extn_gef_deinit();
         free(adev->snd_dev_ref_cnt);
         platform_deinit(adev->platform);
         if (adev->adm_deinit)
@@ -4392,6 +4711,8 @@
 static int adev_open(const hw_module_t *module, const char *name,
                      hw_device_t **device)
 {
+    int ret;
+
     ALOGD("%s: enter", __func__);
     if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
 
@@ -4466,9 +4787,26 @@
         ALOGE("%s: Failed to init platform data, aborting.", __func__);
         *device = NULL;
         pthread_mutex_unlock(&adev_init_lock);
+        pthread_mutex_destroy(&adev->lock);
+        pthread_mutex_destroy(&adev->snd_card_status.lock);
         return -EINVAL;
     }
 
+    if (audio_extn_qaf_is_enabled()) {
+        ret = audio_extn_qaf_init(adev);
+        if (ret < 0) {
+            free(adev);
+            ALOGE("%s: Failed to init platform data, aborting.", __func__);
+            *device = NULL;
+            pthread_mutex_unlock(&adev_init_lock);
+            pthread_mutex_destroy(&adev->lock);
+            return ret;
+        }
+
+        adev->device.open_output_stream = audio_extn_qaf_open_output_stream;
+        adev->device.close_output_stream = audio_extn_qaf_close_output_stream;
+    }
+
     adev->snd_card_status.state = SND_CARD_STATE_ONLINE;
 
     if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
@@ -4487,6 +4825,7 @@
     }
     audio_extn_listen_init(adev, adev->snd_card);
     audio_extn_sound_trigger_init(adev);
+    audio_extn_gef_init(adev);
 
     if (access(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, R_OK) == 0) {
         adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW);
@@ -4535,6 +4874,14 @@
                                     dlsym(adev->adm_lib, "adm_request_focus");
             adev->adm_abandon_focus = (adm_abandon_focus_t)
                                     dlsym(adev->adm_lib, "adm_abandon_focus");
+            adev->adm_set_config = (adm_set_config_t)
+                                    dlsym(adev->adm_lib, "adm_set_config");
+            adev->adm_request_focus_v2 = (adm_request_focus_v2_t)
+                                    dlsym(adev->adm_lib, "adm_request_focus_v2");
+            adev->adm_is_noirq_avail = (adm_is_noirq_avail_t)
+                                    dlsym(adev->adm_lib, "adm_is_noirq_avail");
+            adev->adm_on_routing_change = (adm_on_routing_change_t)
+                                    dlsym(adev->adm_lib, "adm_on_routing_change");
         }
     }
 
@@ -4566,6 +4913,16 @@
         }
     }
 
+    if (property_get("audio_hal.period_multiplier", value, NULL) > 0) {
+        af_period_multiplier = atoi(value);
+        if (af_period_multiplier < 0)
+            af_period_multiplier = 2;
+        else if (af_period_multiplier > 4)
+            af_period_multiplier = 4;
+
+        ALOGV("new period_multiplier = %d", af_period_multiplier);
+    }
+
     adev->multi_offload_enable = property_get_bool("audio.offload.multiple.enabled", false);
     pthread_mutex_unlock(&adev_init_lock);
 
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 8197fec..83ad54d 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -79,6 +79,11 @@
 
 #define MAX_PERF_LOCK_OPTS 20
 
+typedef enum card_status_t {
+    CARD_STATUS_OFFLINE,
+    CARD_STATUS_ONLINE
+} card_status_t;
+
 /* These are the supported use cases by the hardware.
  * Each usecase is mapped to a specific PCM device.
  * Refer to pcm_device_table[].
@@ -141,6 +146,8 @@
     USECASE_AUDIO_PLAYBACK_AFE_PROXY,
     USECASE_AUDIO_RECORD_AFE_PROXY,
 
+    USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE,
+
     AUDIO_USECASE_MAX
 };
 
@@ -162,6 +169,7 @@
     OFFLOAD_CMD_DRAIN,              /* send a full drain request to DSP */
     OFFLOAD_CMD_PARTIAL_DRAIN,      /* send a partial drain request to DSP */
     OFFLOAD_CMD_WAIT_FOR_BUFFER,    /* wait for buffer released by DSP */
+    OFFLOAD_CMD_ERROR,              /* offload playback hit some error */
 };
 
 enum {
@@ -228,7 +236,13 @@
     audio_format_t hal_op_format;
     void *convert_buffer;
 
+    bool realtime;
+    int af_period_multiplier;
     struct audio_device *dev;
+    void* qaf_stream_handle;
+    pthread_cond_t qaf_offload_cond;
+    pthread_t qaf_offload_thread;
+    struct listnode qaf_offload_cmd_list;
 };
 
 struct stream_in {
@@ -252,6 +266,8 @@
     bool is_st_session_active;
     int sample_rate;
     int bit_width;
+    bool realtime;
+    int af_period_multiplier;
     struct audio_device *dev;
 };
 
@@ -308,6 +324,12 @@
 typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t);
 typedef void (*adm_request_focus_t)(void *, audio_io_handle_t);
 typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t);
+typedef void (*adm_set_config_t)(void *, audio_io_handle_t,
+                                         struct pcm *,
+                                         struct pcm_config *);
+typedef void (*adm_request_focus_v2_t)(void *, audio_io_handle_t, long);
+typedef bool (*adm_is_noirq_avail_t)(void *, int, int, int);
+typedef void (*adm_on_routing_change_t)(void *, audio_io_handle_t);
 
 struct audio_device {
     struct audio_hw_device device;
@@ -361,6 +383,10 @@
     adm_deregister_stream_t adm_deregister_stream;
     adm_request_focus_t adm_request_focus;
     adm_abandon_focus_t adm_abandon_focus;
+    adm_set_config_t adm_set_config;
+    adm_request_focus_v2_t adm_request_focus_v2;
+    adm_is_noirq_avail_t adm_is_noirq_avail;
+    adm_on_routing_change_t adm_on_routing_change;
 
     void (*offload_effects_get_parameters)(struct str_parms *,
                                            struct str_parms *);
@@ -371,6 +397,7 @@
     int perf_lock_opts[MAX_PERF_LOCK_OPTS];
     int perf_lock_opts_size;
     bool native_playback_enabled;
+    bool asrc_mode_enabled;
 };
 
 int select_devices(struct audio_device *adev,
@@ -392,12 +419,24 @@
 
 bool audio_is_true_native_stream_active(struct audio_device *adev);
 
+bool audio_is_dsd_native_stream_active(struct audio_device *adev);
+
 int pcm_ioctl(struct pcm *pcm, int request, ...);
 
 int get_snd_card_state(struct audio_device *adev);
 audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
                                                  usecase_type_t type);
 
+int adev_open_output_stream(struct audio_hw_device *dev,
+                            audio_io_handle_t handle,
+                            audio_devices_t devices,
+                            audio_output_flags_t flags,
+                            struct audio_config *config,
+                            struct audio_stream_out **stream_out,
+                            const char *address __unused);
+void adev_close_output_stream(struct audio_hw_device *dev __unused,
+                              struct audio_stream_out *stream);
+
 #define LITERAL_TO_STRING(x) #x
 #define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
             __FILE__ ":" LITERAL_TO_STRING(__LINE__)\
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
index d9add29..28b2397 100644
--- a/hal/msm8916/hw_info.c
+++ b/hal/msm8916/hw_info.c
@@ -324,7 +324,8 @@
         strstr(snd_card_name, "msm8909") || strstr(snd_card_name, "msm8952") ||
         strstr(snd_card_name, "msm8976") || strstr(snd_card_name, "msm8953") ||
         strstr(snd_card_name, "msm8937") || strstr(snd_card_name, "msm8917") ||
-        strstr(snd_card_name, "msm8940") || strstr(snd_card_name, "msmfalcon")) {
+        strstr(snd_card_name, "msm8940") || strstr(snd_card_name, "msm8920") ||
+        strstr(snd_card_name, "msmfalcon")) {
         ALOGV("8x16 - variant soundcard");
         update_hardware_info_8x16(hw_info, snd_card_name);
     } else {
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 500a28d..a5cc804 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -68,7 +68,6 @@
 #define PLATFORM_INFO_XML_PATH_EXTCODEC  "/system/etc/audio_platform_info_extcodec.xml"
 
 #define LIB_ACDB_LOADER "libacdbloader.so"
-#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
 #define CVD_VERSION_MIXER_CTL "CVD Version"
 
 #define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
@@ -119,6 +118,8 @@
 #define AUDIO_PARAMETER_KEY_AUD_CALDATA   "cal_data"
 #define AUDIO_PARAMETER_KEY_AUD_CALRESULT "cal_result"
 
+#define AUDIO_PARAMETER_KEY_MONO_SPEAKER "mono_speaker"
+
 /* Reload ACDB files from specified path */
 #define AUDIO_PARAMETER_KEY_RELOAD_ACDB "reload_acdb"
 
@@ -192,8 +193,10 @@
 typedef struct codec_backend_cfg {
     uint32_t sample_rate;
     uint32_t bit_width;
+    uint32_t channels;
     char     *bitwidth_mixer_ctl;
     char     *samplerate_mixer_ctl;
+    char     *channels_mixer_ctl;
 } codec_backend_cfg_t;
 
 static native_audio_prop na_props = {0, 0, 0};
@@ -220,6 +223,7 @@
     /* Vbat monitor related flags */
     bool is_vbat_speaker;
     bool gsm_mode_enabled;
+    int mono_speaker;
     /* Audio calibration related functions */
     void                       *acdb_handle;
     int                        voice_feature_set;
@@ -241,8 +245,8 @@
     struct csd_data *csd;
     void *edid_info;
     bool edid_valid;
+    int ext_disp_type;
     codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
-    codec_backend_cfg_t current_tx_backend_cfg[MAX_CODEC_TX_BACKENDS];
     char ec_ref_mixer_path[64];
     char codec_version[CODEC_VERSION_MAX_LENGTH];
     int hw_dep_fd;
@@ -316,6 +320,7 @@
                                           AFE_PROXY_RECORD_PCM_DEVICE},
     [USECASE_AUDIO_RECORD_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
                                         AFE_PROXY_RECORD_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = {MULTIMEDIA9_PCM_DEVICE, -1},
 };
 
 /* Array to store sound devices */
@@ -340,12 +345,19 @@
     [SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
     [SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = "wsa-voice-speaker",
     [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "vbat-voice-speaker",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2] = "voice-speaker-2",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA] = "wsa-voice-speaker-2",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = "vbat-voice-speaker-2",
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
     [SND_DEVICE_OUT_VOICE_LINE] = "voice-line",
     [SND_DEVICE_OUT_HDMI] = "hdmi",
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
+    [SND_DEVICE_OUT_DISPLAY_PORT] = "display-port",
+    [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = "speaker-and-display-port",
     [SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
     [SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+    [SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
@@ -362,8 +374,10 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = "voice-speaker-2-protected",
     [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = "voice-speaker-2-protected-vbat",
 #ifdef RECORD_PLAY_CONCURRENCY
     [SND_DEVICE_OUT_VOIP_HANDSET] = "voip-handset",
     [SND_DEVICE_OUT_VOIP_SPEAKER] = "voip-speaker",
@@ -416,6 +430,8 @@
     [SND_DEVICE_IN_HANDSET_STEREO_DMIC] = "handset-stereo-dmic-ef",
     [SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = "speaker-stereo-dmic-ef",
     [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = "vi-feedback",
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = "vi-feedback-mono-1",
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = "vi-feedback-mono-2",
     [SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = "voice-speaker-dmic-broadside",
     [SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = "speaker-dmic-broadside",
     [SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = "speaker-dmic-broadside",
@@ -458,13 +474,20 @@
     [SND_DEVICE_OUT_VOICE_HANDSET] = 7,
     [SND_DEVICE_OUT_VOICE_LINE] = 10,
     [SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2] = 14,
     [SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = 135,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA] = 135,
     [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 135,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = 135,
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
     [SND_DEVICE_OUT_HDMI] = 18,
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
+    [SND_DEVICE_OUT_DISPLAY_PORT] = 18,
+    [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = 14,
     [SND_DEVICE_OUT_BT_SCO] = 22,
     [SND_DEVICE_OUT_BT_SCO_WB] = 39,
+    [SND_DEVICE_OUT_BT_A2DP] = 20,
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
@@ -481,8 +504,10 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = 103,
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = 101,
     [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = 101,
 #ifdef RECORD_PLAY_CONCURRENCY
     [SND_DEVICE_OUT_VOIP_HANDSET] = 133,
     [SND_DEVICE_OUT_VOIP_SPEAKER] = 132,
@@ -534,6 +559,8 @@
     [SND_DEVICE_IN_HANDSET_STEREO_DMIC] = 34,
     [SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = 35,
     [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = 102,
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = 102,
     [SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = 12,
     [SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = 12,
     [SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = 119,
@@ -580,12 +607,19 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_WSA)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_LINE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_DISPLAY_PORT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
@@ -602,8 +636,10 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT)},
 #ifdef RECORD_PLAY_CONCURRENCY
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_SPEAKER)},
@@ -654,6 +690,8 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2)},
     {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_FLUENCE_DMIC_AANC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE)},
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE)},
@@ -704,6 +742,7 @@
     {TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
     {TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO)},
     {TO_NAME_INDEX(USECASE_AUDIO_SPKR_CALIB_TX)},
+    {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE)},
 };
 
 #define NO_COLS 2
@@ -767,6 +806,7 @@
 #define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
 #define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
 #define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+#define ULL_PLATFORM_DELAY (6*1000LL)
 
 static void update_codec_type(const char *snd_card_name) {
 
@@ -1194,6 +1234,8 @@
     backend_tag_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
     backend_tag_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
     backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
+    backend_tag_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("display-port");
+    backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("speaker-and-display-port");
     backend_tag_table[SND_DEVICE_OUT_VOICE_TX] = strdup("afe-proxy");
     backend_tag_table[SND_DEVICE_IN_VOICE_RX] = strdup("afe-proxy");
     backend_tag_table[SND_DEVICE_OUT_AFE_PROXY] = strdup("afe-proxy");
@@ -1205,9 +1247,16 @@
     backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
     backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
     backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("vbat-voice-speaker");
+    backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("vbat-voice-speaker-2");
+    backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
+    backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
 
     hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
+    hw_interface_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("DISPLAY_PORT_RX");
+    hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("SLIMBUS_0_RX-and-DISPLAY_PORT_RX");
+    hw_interface_table[SND_DEVICE_OUT_USB_HEADSET] = strdup("USB_AUDIO_RX");
+    hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = strdup("SLIMBUS_0_RX-and-USB_AUDIO_RX");
     hw_interface_table[SND_DEVICE_OUT_VOICE_TX] = strdup("AFE_PCM_RX");
 
     my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
@@ -1661,8 +1710,10 @@
     my_data->slowtalk = false;
     my_data->hd_voice = false;
     my_data->edid_info = NULL;
+    my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
     my_data->is_wsa_speaker = false;
     my_data->hw_dep_fd = -1;
+    my_data->mono_speaker = SPKR_1;
 
     property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
     if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
@@ -1847,6 +1898,9 @@
     /* init usb */
     audio_extn_usb_init(adev);
 
+    /*init a2dp*/
+    audio_extn_a2dp_init(adev);
+
     /* Read one time ssr property */
     audio_extn_ssr_update_enabled();
     audio_extn_spkr_prot_init(adev);
@@ -1867,13 +1921,14 @@
         if (idx == HEADPHONE_44_1_BACKEND)
             my_data->current_backend_cfg[idx].sample_rate = OUTPUT_SAMPLING_RATE_44100;
         my_data->current_backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_CHANNELS;
+        if (idx > MAX_RX_CODEC_BACKENDS)
+            my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+        my_data->current_backend_cfg[idx].bitwidth_mixer_ctl = NULL;
+        my_data->current_backend_cfg[idx].samplerate_mixer_ctl = NULL;
+        my_data->current_backend_cfg[idx].channels_mixer_ctl = NULL;
     }
 
-    my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].sample_rate =
-                                               CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-    my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bit_width =
-                                               CODEC_BACKEND_DEFAULT_BIT_WIDTH;
-
     if (is_external_codec) {
         my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
             strdup("SLIM_0_RX Format");
@@ -1890,9 +1945,9 @@
         my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
             strdup("SLIM_6_RX SampleRate");
 
-        my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+        my_data->current_backend_cfg[SLIMBUS_0_TX].bitwidth_mixer_ctl =
             strdup("SLIM_0_TX Format");
-        my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+        my_data->current_backend_cfg[SLIMBUS_0_TX].samplerate_mixer_ctl =
             strdup("SLIM_0_TX SampleRate");
     } else {
         my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
@@ -1900,21 +1955,38 @@
         my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
             strdup("MI2S_RX SampleRate");
 
-        my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+        my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
             strdup("MI2S_TX Format");
-        my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+        my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
             strdup("MI2S_TX SampleRate");
     }
+    my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
+        strdup("USB_AUDIO_TX Format");
+    my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
+        strdup("USB_AUDIO_TX SampleRate");
+    my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].channels_mixer_ctl =
+            strdup("USB_AUDIO_TX Channels");
 
     my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].bitwidth_mixer_ctl =
         strdup("USB_AUDIO_RX Format");
     my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].samplerate_mixer_ctl =
         strdup("USB_AUDIO_RX SampleRate");
+    my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].channels_mixer_ctl =
+        strdup("USB_AUDIO_RX Channels");
 
     my_data->current_backend_cfg[HDMI_RX_BACKEND].bitwidth_mixer_ctl =
         strdup("HDMI_RX Bit Format");
     my_data->current_backend_cfg[HDMI_RX_BACKEND].samplerate_mixer_ctl =
         strdup("HDMI_RX SampleRate");
+    my_data->current_backend_cfg[HDMI_RX_BACKEND].channels_mixer_ctl =
+        strdup("HDMI_RX Channels");
+
+    my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bitwidth_mixer_ctl =
+        strdup("Display Port RX Bit Format");
+    my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].samplerate_mixer_ctl =
+        strdup("Display Port RX SampleRate");
+    my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels_mixer_ctl =
+        strdup("Display Port RX Channels");
 
     ret = audio_extn_utils_get_codec_version(snd_card_name,
                                              my_data->adev->snd_card,
@@ -2032,7 +2104,8 @@
         return;
     }
 
-    if((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+    if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
         !(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
         ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
         return;
@@ -2420,12 +2493,23 @@
     return ret;
 }
 
+int codec_device_supports_native_playback(audio_devices_t out_device)
+{
+    int ret = false;
 
-static int platform_get_backend_index(snd_device_t snd_device)
+    if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+        out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+        out_device & AUDIO_DEVICE_OUT_LINE)
+        ret = true;
+
+    return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
 {
     int32_t port = DEFAULT_CODEC_BACKEND;
 
-    if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+    if (snd_device >= SND_DEVICE_OUT_BEGIN && snd_device < SND_DEVICE_OUT_END) {
         if (backend_tag_table[snd_device] != NULL) {
                 if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
                             sizeof("headphones-44.1")) == 0)
@@ -2435,15 +2519,22 @@
                         port = HEADPHONE_BACKEND;
                 else if (strcmp(backend_tag_table[snd_device], "hdmi") == 0)
                         port = HDMI_RX_BACKEND;
+                else if (strcmp(backend_tag_table[snd_device], "display-port") == 0)
+                        port = DISP_PORT_RX_BACKEND;
                 else if (strcmp(backend_tag_table[snd_device], "usb-headphones") == 0)
                         port = USB_AUDIO_RX_BACKEND;
         }
+    } else if (snd_device >= SND_DEVICE_IN_BEGIN && snd_device < SND_DEVICE_IN_END) {
+        port = DEFAULT_CODEC_TX_BACKEND;
+        if (backend_tag_table[snd_device] != NULL) {
+                if (strcmp(backend_tag_table[snd_device], "usb-headset-mic") == 0)
+                        port = USB_AUDIO_TX_BACKEND;
+        }
     } else {
-        ALOGV("%s:napb: Invalid device - %d ", __func__, snd_device);
+        ALOGW("%s:napb: Invalid device - %d ", __func__, snd_device);
     }
 
-    ALOGV("%s:napb: backend port - %d device - %d ", __func__, port,
-        snd_device);
+    ALOGV("%s:napb: backend port - %d device - %d ", __func__, port, snd_device);
     return port;
 }
 
@@ -2526,7 +2617,9 @@
         return ret;
 
     if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
-         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
          audio_extn_spkr_prot_is_enabled()) {
         if (my_data->is_vbat_speaker)
             acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2561,9 +2654,18 @@
     if (my_data->acdb_send_voice_cal == NULL) {
         ALOGE("%s: dlsym error for acdb_send_voice_call", __func__);
     } else {
-        if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
-            audio_extn_spkr_prot_is_enabled())
-            out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+        if (audio_extn_spkr_prot_is_enabled()) {
+            if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+                out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+            else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+                out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
+            else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+            else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
+        }
 
         acdb_rx_id = acdb_device_table[out_snd_device];
         acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2590,7 +2692,9 @@
         return ret;
 
     if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
-         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
          audio_extn_spkr_prot_is_enabled()) {
         if (my_data->is_vbat_speaker)
             acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2797,12 +2901,29 @@
 
         new_snd_devices[1] = SND_DEVICE_OUT_HDMI;
         status = true;
+    } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT &&
+               !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_DISPLAY_PORT)) {
+        *num_devices = 2;
+
+        if (my_data->is_vbat_speaker)
+            new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_VBAT;
+        else if (my_data->is_wsa_speaker)
+            new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_WSA;
+        else
+            new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+
+        new_snd_devices[1] = SND_DEVICE_OUT_DISPLAY_PORT;
+        status = true;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_USB_HEADSET)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_USB_HEADSET;
         status = true;
+    } else if (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) {
+        *num_devices = 2;
+        new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+        new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
     }
 
     ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
@@ -2811,6 +2932,42 @@
     return status;
 }
 
+int platform_get_ext_disp_type(void *platform)
+{
+    int disp_type;
+    struct platform_data *my_data = (struct platform_data *)platform;
+
+    if (my_data->ext_disp_type != EXT_DISPLAY_TYPE_NONE) {
+         ALOGD("%s: Returning cached ext disp type:%s",
+               __func__, (my_data->ext_disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+         return my_data->ext_disp_type;
+    }
+
+#ifdef DISPLAY_PORT_ENABLED
+    struct audio_device *adev = my_data->adev;
+    struct mixer_ctl *ctl;
+    char *mixer_ctl_name = "External Display Type";
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+
+    disp_type = mixer_ctl_get_value(ctl, 0);
+    if (disp_type == EXT_DISPLAY_TYPE_NONE) {
+         ALOGE("%s: Invalid external display type: %d", __func__, disp_type);
+         return -EINVAL;
+    }
+#else
+    disp_type = EXT_DISPLAY_TYPE_HDMI;
+#endif
+    my_data->ext_disp_type = disp_type;
+    ALOGD("%s: ext disp type:%s", __func__, (disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+    return disp_type;
+}
+
 snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
@@ -2868,13 +3025,26 @@
                 snd_device = SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES;
         } else if (devices == (AUDIO_DEVICE_OUT_AUX_DIGITAL |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
-            snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+            switch(my_data->ext_disp_type) {
+                case EXT_DISPLAY_TYPE_HDMI:
+                     snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+                     break;
+                case EXT_DISPLAY_TYPE_DP:
+                     snd_device = SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT;
+                     break;
+                default:
+                     ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+                     goto exit;
+            }
         } else if (devices == (AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
             snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
         } else if (devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
             snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
+        } else if ((devices & AUDIO_DEVICE_OUT_SPEAKER) &&
+                   (devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
+            snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP;
         } else {
             ALOGE("%s: Invalid combo device(%#x)", __func__, devices);
             goto exit;
@@ -2925,13 +3095,25 @@
                 snd_device = SND_DEVICE_OUT_BT_SCO_WB;
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
+        } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+                snd_device = SND_DEVICE_OUT_BT_A2DP;
         } else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
-                if (my_data->is_vbat_speaker)
-                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
-                else if (my_data->is_wsa_speaker)
-                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
-                else
-                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+                if (my_data->is_vbat_speaker) {
+                    if (my_data->mono_speaker == SPKR_1)
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+                    else
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT;
+                } else if (my_data->is_wsa_speaker) {
+                    if (my_data->mono_speaker == SPKR_1)
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
+                    else
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA;
+                } else {
+                    if (my_data->mono_speaker == SPKR_1)
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+                    else
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2;
+                }
         } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
                    devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
             snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -3007,7 +3189,19 @@
         else
             snd_device = SND_DEVICE_OUT_BT_SCO;
     } else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-        snd_device = SND_DEVICE_OUT_HDMI ;
+            switch(my_data->ext_disp_type) {
+                case EXT_DISPLAY_TYPE_HDMI:
+                    snd_device = SND_DEVICE_OUT_HDMI;
+                    break;
+                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_ALL_A2DP) {
+        snd_device = SND_DEVICE_OUT_BT_A2DP;
     } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
                devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
         ALOGD("%s: setting USB hadset channel capability(2) for Proxy", __func__);
@@ -3430,7 +3624,7 @@
     struct audio_device *adev = my_data->adev;
     struct mixer_ctl *ctl;
     const char *channel_cnt_str = NULL;
-    const char *mixer_ctl_name = "HDMI_RX Channels";
+    char *mixer_ctl_name;
     switch (channel_count) {
     case 8:
         channel_cnt_str = "Eight"; break;
@@ -3447,13 +3641,25 @@
     default:
         channel_cnt_str = "Two"; break;
     }
+
+    switch(my_data->ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            mixer_ctl_name = "HDMI_RX Channels";
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            mixer_ctl_name = "Display Port RX Channels";
+            break;
+        default:
+            ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+            return -EINVAL;
+    }
     ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
     if (!ctl) {
         ALOGE("%s: Could not get ctl for mixer cmd - %s",
               __func__, mixer_ctl_name);
         return -EINVAL;
     }
-    ALOGV("HDMI channel count: %s", channel_cnt_str);
+    ALOGV("Ext disp channel count: %s", channel_cnt_str);
     mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
     return 0;
 }
@@ -3549,6 +3755,8 @@
     char *kv_pairs = NULL;
 
     kv_pairs = str_parms_to_str(parms);
+    if(!kv_pairs)
+        return ret;
     len = strlen(kv_pairs);
     ALOGV("%s: enter: - %s", __func__, kv_pairs);
     free(kv_pairs);
@@ -3611,6 +3819,16 @@
 
     }
 
+    err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER, value, len);
+    if (err >= 0) {
+        if (!strncmp("left", value, sizeof("left")))
+            my_data->mono_speaker = SPKR_1;
+        else if (!strncmp("right", value, sizeof("right")))
+            my_data->mono_speaker = SPKR_2;
+
+        str_parms_del(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER);
+    }
+
 #ifdef RECORD_PLAY_CONCURRENCY
     err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_REC_PLAY_CONC, value, sizeof(value));
     if (err >= 0) {
@@ -3787,8 +4005,8 @@
                 !strncmp("true", propValue, 4);
         }
 
-        if (prop_playback_enabled && (voice_is_in_call(my_data->adev) ||
-             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev)))) {
+        if ((prop_playback_enabled && (voice_is_in_call(my_data->adev))) ||
+             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev))) {
             char *decoder_mime_type = value;
 
             //check if unsupported mime type or not
@@ -3814,6 +4032,13 @@
     free(kv_pairs);
 }
 
+unsigned char* platform_get_license(void *platform __unused, int *size __unused)
+{
+    ALOGE("%s: Not implemented", __func__);
+    return NULL;
+}
+
+
 /* Delay in Us, only to be used for PCM formats */
 int64_t platform_render_latency(audio_usecase_t usecase)
 {
@@ -3825,6 +4050,8 @@
         case USECASE_AUDIO_PLAYBACK_OFFLOAD:
         case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
             return PCM_OFFLOAD_PLATFORM_DELAY;
+        case USECASE_AUDIO_PLAYBACK_ULL:
+            return ULL_PLATFORM_DELAY;
         default:
             return 0;
     }
@@ -3845,7 +4072,9 @@
     if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
         (snd_device < SND_DEVICE_IN_END) &&
         (snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
-        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
         needs_event = true;
 
     return needs_event;
@@ -3908,7 +4137,9 @@
     if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
         (snd_device < SND_DEVICE_IN_END) &&
         (snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
-        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
         needs_event = true;
 
     return needs_event;
@@ -4009,27 +4240,55 @@
     return fragment_size;
 }
 
+/*
+ * return backend_idx on which voice call is active
+ */
+static int platform_get_voice_call_backend(struct audio_device* adev)
+{
+   struct audio_usecase *uc = NULL;
+   struct listnode *node;
+   snd_device_t out_snd_device = SND_DEVICE_NONE;
+
+   int backend_idx = -1;
+
+   if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
+       list_for_each(node, &adev->usecase_list) {
+           uc =  node_to_item(node, struct audio_usecase, list);
+           if (uc && (uc->type == VOICE_CALL || uc->type == VOIP_CALL) && uc->stream.out) {
+               out_snd_device = platform_get_output_snd_device(adev->platform, uc->stream.out);
+               backend_idx = platform_get_backend_index(out_snd_device);
+               break;
+           }
+       }
+   }
+   return backend_idx;
+}
 
 /*
  * configures afe with bit width and Sample Rate
  */
 static int platform_set_codec_backend_cfg(struct audio_device* adev,
-                         snd_device_t snd_device, unsigned int bit_width,
-                         unsigned int sample_rate, audio_format_t format)
+                         snd_device_t snd_device, struct audio_backend_cfg backend_cfg)
 {
     int ret = 0;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
+    unsigned int bit_width = backend_cfg.bit_width;
+    unsigned int sample_rate = backend_cfg.sample_rate;
+    unsigned int channels = backend_cfg.channels;
+    audio_format_t format = backend_cfg.format;
+    bool passthrough_enabled = backend_cfg.passthrough_enabled;
 
     backend_idx = platform_get_backend_index(snd_device);
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
-          __func__, bit_width, sample_rate, backend_idx,
+
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d, backend_idx %d device (%s)",
+          __func__, bit_width, sample_rate, channels,backend_idx,
           platform_get_snd_device_name(snd_device));
 
     if (bit_width !=
         my_data->current_backend_cfg[backend_idx].bit_width) {
 
-        struct  mixer_ctl *ctl;
+        struct  mixer_ctl *ctl = NULL;
         ctl = mixer_get_ctl_by_name(adev->mixer,
                         my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
         if (!ctl) {
@@ -4045,7 +4304,7 @@
             else
                 mixer_ctl_set_enum_by_string(ctl, "S24_LE");
         } else if (bit_width == 32) {
-            mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+            mixer_ctl_set_enum_by_string(ctl, "S32_LE");
         } else {
             mixer_ctl_set_enum_by_string(ctl, "S16_LE");
         }
@@ -4086,14 +4345,24 @@
                 rate_str = "KHZ_44P1";
                 break;
             case 64000:
-            case 88200:
             case 96000:
                 rate_str = "KHZ_96";
                 break;
+            case 88200:
+                rate_str = "KHZ_88P2";
+            break;
             case 176400:
+                rate_str = "KHZ_176P4";
+                break;
             case 192000:
                 rate_str = "KHZ_192";
                 break;
+            case 352800:
+                rate_str = "KHZ_352P8";
+                break;
+            case 384000:
+                rate_str = "KHZ_384";
+                break;
             default:
                 rate_str = "KHZ_48";
                 break;
@@ -4115,19 +4384,163 @@
             mixer_ctl_set_enum_by_string(ctl, rate_str);
             my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
     }
+    if ((my_data->current_backend_cfg[backend_idx].channels_mixer_ctl) &&
+        (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+        struct  mixer_ctl *ctl;
+        char *channel_cnt_str = NULL;
+
+        switch (channels) {
+        case 8:
+            channel_cnt_str = "Eight"; break;
+        case 7:
+            channel_cnt_str = "Seven"; break;
+        case 6:
+            channel_cnt_str = "Six"; break;
+        case 5:
+            channel_cnt_str = "Five"; break;
+        case 4:
+            channel_cnt_str = "Four"; break;
+        case 3:
+            channel_cnt_str = "Three"; break;
+        case 1:
+            channel_cnt_str = "One"; break;
+        case 2:
+        default:
+            channel_cnt_str = "Two"; break;
+        }
+
+        ctl = mixer_get_ctl_by_name(adev->mixer,
+           my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                   __func__,
+                   my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+            return -EINVAL;
+        }
+        mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
+        my_data->current_backend_cfg[backend_idx].channels = channels;
+
+        if (backend_idx == HDMI_RX_BACKEND)
+            platform_set_edid_channels_configuration(adev->platform, channels);
+
+        ALOGD("%s:becf: afe: %s set to %s", __func__,
+               my_data->current_backend_cfg[backend_idx].channels_mixer_ctl, channel_cnt_str);
+    }
+
+    bool set_ext_disp_format = false;
+    char *ext_disp_format = NULL;
+
+    if (backend_idx == HDMI_RX_BACKEND) {
+        ext_disp_format = "HDMI RX Format";
+        set_ext_disp_format = true;
+    } else if (backend_idx == DISP_PORT_RX_BACKEND) {
+        ext_disp_format = "Display Port Rx Format";
+        set_ext_disp_format = true;
+    } else {
+        ALOGV("%s: Format doesnt have to be set", __func__);
+    }
+
+    if (set_ext_disp_format) {
+        struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, ext_disp_format);
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                   __func__, ext_disp_format);
+            return -EINVAL;
+        }
+
+        if (passthrough_enabled) {
+            ALOGD("%s:Ext display compress format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "Compr");
+        } else {
+            ALOGD("%s: Ext display PCM format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "LPCM");
+        }
+    }
 
     return ret;
 }
 
 /*
+ *Validate the selected bit_width, sample_rate and channels using the edid
+ *of the connected sink device.
+ */
+static void platform_check_hdmi_backend_cfg(struct audio_device* adev,
+                                   struct audio_usecase* usecase,
+                                   int backend_idx,
+                                   struct audio_backend_cfg *hdmi_backend_cfg)
+{
+    unsigned int bit_width;
+    unsigned int sample_rate;
+    unsigned int channels, max_supported_channels = 0;
+    struct platform_data *my_data = (struct platform_data *)adev->platform;
+    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool passthrough_enabled = false;
+
+    bit_width = hdmi_backend_cfg->bit_width;
+    sample_rate = hdmi_backend_cfg->sample_rate;
+    channels = hdmi_backend_cfg->channels;
+
+
+    ALOGI("%s:becf: HDMI: bitwidth %d, samplerate %d, channels %d"
+          ", usecase = %d", __func__, bit_width,
+          sample_rate, channels, usecase->id);
+
+    if (audio_extn_passthru_is_enabled() && audio_extn_passthru_is_active()
+        && (usecase->stream.out->compr_config.codec->compr_passthr != 0)) {
+        passthrough_enabled = true;
+        ALOGI("passthrough is enabled for this stream");
+    }
+
+    // For voice calls use default configuration i.e. 16b/48K, only applicable to
+    // default backend
+    if (!passthrough_enabled) {
+
+        max_supported_channels = platform_edid_get_max_channels(my_data);
+
+        //Check EDID info for supported samplerate
+        if (!edid_is_supported_sr(edid_info,sample_rate)) {
+            //reset to current sample rate
+            sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
+        }
+
+        //Check EDID info for supported bit width
+        if (!edid_is_supported_bps(edid_info,bit_width)) {
+            //reset to current sample rate
+            bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
+        }
+
+        if (channels > max_supported_channels)
+            channels = max_supported_channels;
+
+    } else {
+        /*During pass through set default bit width and channels*/
+        channels = DEFAULT_HDMI_OUT_CHANNELS;
+        if ((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+            (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
+            sample_rate = sample_rate * 4 ;
+
+        bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        /* We force route so that the BE format can be set to Compr */
+    }
+
+    ALOGI("%s:becf: afe: HDMI backend: passthrough %d updated bit width: %d and sample rate: %d"
+           "channels %d", __func__, passthrough_enabled , bit_width,
+           sample_rate, channels);
+
+    hdmi_backend_cfg->bit_width = bit_width;
+    hdmi_backend_cfg->sample_rate = sample_rate;
+    hdmi_backend_cfg->channels = channels;
+    hdmi_backend_cfg->passthrough_enabled = passthrough_enabled;
+}
+
+/*
  * goes through all the current usecases and picks the highest
  * bitwidth & samplerate
  */
 static bool platform_check_codec_backend_cfg(struct audio_device* adev,
                                    struct audio_usecase* usecase,
                                    snd_device_t snd_device,
-                                   unsigned int* new_bit_width,
-                                   unsigned int* new_sample_rate)
+                                   struct audio_backend_cfg *backend_cfg)
 {
     bool backend_change = false;
     struct listnode *node;
@@ -4135,24 +4548,25 @@
     char value[PROPERTY_VALUE_MAX] = {0};
     unsigned int bit_width;
     unsigned int sample_rate;
+    unsigned int channels;
+    bool passthrough_enabled = false;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
-    int na_mode = platform_get_native_support();
-    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool channels_updated = false;
 
     backend_idx = platform_get_backend_index(snd_device);
 
-    bit_width = *new_bit_width;
-    sample_rate = *new_sample_rate;
+    bit_width = backend_cfg->bit_width;
+    sample_rate = backend_cfg->sample_rate;
+    channels = backend_cfg->channels;
 
-    ALOGI("%s:becf: afe: Codec selected backend: %d current bit width: %d and sample rate: %d",
-          __func__, backend_idx, bit_width, sample_rate);
+    ALOGI("%s:becf: afe: Codec selected backend: %d current bit width: %d sample rate: %d channels: %d",
+          __func__, backend_idx, bit_width, sample_rate, channels);
 
     // For voice calls use default configuration i.e. 16b/48K, only applicable to
     // default backend
     // force routing is not required here, caller will do it anyway
-    if ((voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
-        backend_idx == DEFAULT_CODEC_BACKEND) {
+    if (backend_idx == platform_get_voice_call_backend(adev)) {
         ALOGW("%s:becf: afe:Use default bw and sr for voice/voip calls ",
               __func__);
         bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
@@ -4173,11 +4587,12 @@
             uc = node_to_item(node, struct audio_usecase, list);
             struct stream_out *out = (struct stream_out*) uc->stream.out;
             if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+                unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
 
                 ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
-                      "(%d) device %s", __func__, i++, use_case_table[uc->id],
+                      "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
                       uc->id, out->sample_rate,
-                      out->bit_width,
+                      out->bit_width, out_channels,
                       platform_get_snd_device_name(uc->out_snd_device));
 
                 if (platform_check_backends_match(snd_device, uc->out_snd_device)) {
@@ -4187,75 +4602,87 @@
                             sample_rate = out->sample_rate;
                         if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
                             sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                        if (channels < out_channels)
+                            channels = out_channels;
                 }
             }
         }
     }
 
-    if (audio_is_true_native_stream_active(adev)) {
-        if (check_hdset_combo_device(snd_device)) {
-        /*
-         * In true native mode Tasha has a limitation that one port at 44.1 khz
-         * cannot drive both spkr and hdset, to simiplify the solution lets
-         * move the AFE to 48khzwhen a ring tone selects combo device.
-         */
-            sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-            bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
-            ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
-                  __func__);
-        } else {
-        /*
-         * in single BE mode, if native audio playback
-         * is active then it will take priority
-         */
-            sample_rate = OUTPUT_SAMPLING_RATE_44100;
-            ALOGD("%s:becf: afe: napb active set rate to 44.1 khz",
-                  __func__);
+    /* Native playback is preferred for Headphone/HS device over 192Khz */
+    if (codec_device_supports_native_playback(usecase->devices)) {
+        if (audio_is_true_native_stream_active(adev)) {
+            if (check_hdset_combo_device(snd_device)) {
+                /*
+                 * In true native mode Tasha has a limitation that one port at 44.1 khz
+                 * cannot drive both spkr and hdset, to simiplify the solution lets
+                 * move the AFE to 48khzwhen a ring tone selects combo device.
+                 * or if NATIVE playback is not enabled.
+                 */
+                    sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                    bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+                    ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
+                          __func__);
+            } else {
+             /*
+              * in single BE mode, if native audio playback
+              * is active then it will take priority
+              */
+                 sample_rate = OUTPUT_SAMPLING_RATE_44100;
+                 ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
+                       __func__);
+            }
+        } else if (OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
+                 sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                 ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
+                       __func__);
         }
-    }
-
-    /*
-     * hifi playback not supported on spkr devices, limit the Sample Rate
-     * to 48 khz.
-     */
-    if (SND_DEVICE_OUT_SPEAKER == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_WSA == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_VBAT == snd_device) {
+    } else if ((usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) ||
+               (usecase->devices & AUDIO_DEVICE_OUT_EARPIECE) ) {
         sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-        ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+
+        if (bit_width >= 24) {
+            bit_width = platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
+            ALOGD("%s:becf: afe: reset bitwidth to %d (based on supported"
+                  " value for this platform)", __func__, bit_width);
+        }
+
+        ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
             "default Sample Rate(48k)", __func__);
     }
 
     /*
-     * native playback is not enabled.Configure afe to default Sample Rate(48k)
-     */
-    if (NATIVE_AUDIO_MODE_INVALID == na_mode &&
-            OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
-        sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-        ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
-              __func__);
-    }
-
-    /*
      * reset the sample rate to default value(48K), if hifi audio is not supported
      */
-    if (!my_data->hifi_audio) {
+    if (!my_data->hifi_audio && (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) {
                ALOGD("%s:becf: afe: only 48KHZ sample rate is supported "
                       "Configure afe to default Sample Rate(48k)", __func__);
                sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
     }
 
     if (backend_idx == HDMI_RX_BACKEND) {
-        //Check EDID info for supported samplerate
-        if (!edid_is_supported_sr(edid_info,sample_rate)) {
-            //reset to current sample rate
-            sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
-        }
-        //Check EDID info for supported bit widhth
-        if (!edid_is_supported_bps(edid_info,bit_width)) {
-            //reset to current sample rate
-            bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
-        }
+        struct audio_backend_cfg hdmi_backend_cfg;
+        hdmi_backend_cfg.bit_width = bit_width;
+        hdmi_backend_cfg.sample_rate = sample_rate;
+        hdmi_backend_cfg.channels = channels;
+        hdmi_backend_cfg.passthrough_enabled = false;
+
+        /*HDMI does not support 384Khz/32bit playback hence configure BE to 24b/192Khz*/
+        /* TODO: Instead have the validation against edid return the next best match*/
+        if (bit_width > 24)
+            hdmi_backend_cfg.bit_width = 24;
+        if (sample_rate > 192000)
+            hdmi_backend_cfg.sample_rate = 192000;
+
+        platform_check_hdmi_backend_cfg(adev, usecase, backend_idx, &hdmi_backend_cfg);
+
+        bit_width = hdmi_backend_cfg.bit_width;
+        sample_rate = hdmi_backend_cfg.sample_rate;
+        channels = hdmi_backend_cfg.channels;
+        passthrough_enabled = hdmi_backend_cfg.passthrough_enabled;
+
+        if (channels != my_data->current_backend_cfg[backend_idx].channels)
+            channels_updated = true;
     }
 
     //check if mulitchannel clip needs to be down sampled to 48k
@@ -4274,10 +4701,11 @@
     }
 
     if (backend_idx == USB_AUDIO_RX_BACKEND) {
-        unsigned int channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
-        audio_extn_usb_is_config_supported(&bit_width, &sample_rate, channels);
+        audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, true);
         ALOGV("%s: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
                    __func__, bit_width, sample_rate, channels);
+        if (channels != my_data->current_backend_cfg[backend_idx].channels)
+            channels_updated = true;
     }
 
     ALOGI("%s:becf: afe: Codec selected backend: %d updated bit width: %d and sample rate: %d",
@@ -4286,13 +4714,16 @@
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
     if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
-        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
-        *new_bit_width = bit_width;
-        *new_sample_rate = sample_rate;
+        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+         passthrough_enabled || channels_updated) {
+        backend_cfg->bit_width = bit_width;
+        backend_cfg->sample_rate = sample_rate;
+        backend_cfg->channels = channels;
+        backend_cfg->passthrough_enabled = passthrough_enabled;
         backend_change = true;
-        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d new sample rate: %d",
-              __func__,
-             *new_bit_width, *new_sample_rate);
+        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d"
+              " new sample rate: %d new channels %d",__func__,
+               backend_cfg->bit_width, backend_cfg->sample_rate, backend_cfg->channels);
     }
 
     return backend_change;
@@ -4301,23 +4732,24 @@
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
     struct audio_usecase *usecase, snd_device_t snd_device)
 {
-    unsigned int new_bit_width;
-    unsigned int new_sample_rate;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     int new_snd_devices[SND_DEVICE_OUT_END];
     int i, num_devices = 1;
+    struct audio_backend_cfg backend_cfg;
     bool ret = false;
-    audio_format_t format;
 
     backend_idx = platform_get_backend_index(snd_device);
 
-    new_bit_width = usecase->stream.out->bit_width;
-    new_sample_rate = usecase->stream.out->sample_rate;
-    format = usecase->stream.out->format;
+    backend_cfg.bit_width = usecase->stream.out->bit_width;
+    backend_cfg.sample_rate = usecase->stream.out->sample_rate;
+    backend_cfg.format = usecase->stream.out->format;
+    backend_cfg.channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
+    /*this is populated by check_codec_backend_cfg hence set default value to false*/
+    backend_cfg.passthrough_enabled = false;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
-          new_sample_rate, backend_idx, usecase->id,
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+          ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
+          backend_cfg.sample_rate,  backend_cfg.channels, backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
 
     if (!platform_can_split_snd_device(adev->platform, snd_device,
@@ -4328,9 +4760,9 @@
         ALOGI("%s: becf: new_snd_devices[%d] is %s", __func__, i,
             platform_get_snd_device_name(new_snd_devices[i]));
         if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
-                                             &new_bit_width, &new_sample_rate)) {
+                                             &backend_cfg)) {
                 platform_set_codec_backend_cfg(adev, new_snd_devices[i],
-                                               new_bit_width, new_sample_rate, format);
+                                               backend_cfg);
                 ret = true;
         }
     }
@@ -4339,143 +4771,26 @@
 }
 
 /*
- * configures afe with bit width and Sample Rate
- */
-
-int platform_set_capture_codec_backend_cfg(struct audio_device* adev,
-                         snd_device_t snd_device,
-                         unsigned int bit_width, unsigned int sample_rate,
-                         audio_format_t format)
-{
-    int ret = 0;
-    int backend_idx = DEFAULT_CODEC_BACKEND;
-    struct platform_data *my_data = (struct platform_data *)adev->platform;
-
-    ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
-          __func__, bit_width, sample_rate, backend_idx,
-          platform_get_snd_device_name(snd_device));
-
-    if (bit_width !=
-        my_data->current_tx_backend_cfg[backend_idx].bit_width) {
-
-        struct  mixer_ctl *ctl = NULL;
-        ctl = mixer_get_ctl_by_name(adev->mixer,
-                        my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
-        if (!ctl) {
-            ALOGE("%s:txbecf: afe: Could not get ctl for mixer command - %s",
-                  __func__,
-                  my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
-            return -EINVAL;
-        }
-
-        if (bit_width == 24) {
-            if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
-                ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
-            else
-                ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
-        } else {
-            ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
-        }
-
-        if (ret < 0) {
-            ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
-                  __func__,
-                  my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
-            return -EINVAL;
-        }
-
-        my_data->current_tx_backend_cfg[backend_idx].bit_width = bit_width;
-        ALOGD("%s:txbecf: afe: %s mixer set to %d bit", __func__,
-              my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
-    }
-
-    /*
-     * Backend sample rate configuration follows:
-     * 16 bit record - 48khz for streams at any valid sample rate
-     * 24 bit record - 48khz for stream sample rate less than 48khz
-     * 24 bit record - 96khz for sample rate range of 48khz to 96khz
-     * 24 bit record - 192khz for sample rate range of 96khz to 192 khz
-     * Upper limit is inclusive in the sample rate range.
-     */
-    // TODO: This has to be more dynamic based on policy file
-
-    if (sample_rate != my_data->current_tx_backend_cfg[(int)backend_idx].sample_rate) {
-            /*
-             * sample rate update is needed only for hifi audio enabled platforms
-             */
-            char *rate_str = NULL;
-            struct  mixer_ctl *ctl = NULL;
-
-            switch (sample_rate) {
-            case 8000:
-            case 11025:
-            case 16000:
-            case 22050:
-            case 32000:
-            case 44100:
-            case 48000:
-                rate_str = "KHZ_48";
-                break;
-            case 64000:
-            case 88200:
-            case 96000:
-                rate_str = "KHZ_96";
-                break;
-            case 176400:
-            case 192000:
-                rate_str = "KHZ_192";
-                break;
-            default:
-                rate_str = "KHZ_48";
-                break;
-            }
-
-            ctl = mixer_get_ctl_by_name(adev->mixer,
-                my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-
-            if (ctl < 0) {
-                ALOGE("%s:txbecf: afe: Could not get ctl to set the Sample Rate for mixer command - %s",
-                      __func__,
-                      my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-                return -EINVAL;
-            }
-
-            ALOGD("%s:txbecf: afe: %s set to %s", __func__,
-                  my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl,
-                  rate_str);
-            ret = mixer_ctl_set_enum_by_string(ctl, rate_str);
-            if (ret < 0) {
-                ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
-                      __func__,
-                      my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-                return -EINVAL;
-            }
-
-            my_data->current_tx_backend_cfg[backend_idx].sample_rate = sample_rate;
-    }
-
-    return ret;
-}
-
-/*
  * goes through all the current usecases and picks the highest
  * bitwidth & samplerate
  */
-bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
-                                   unsigned int* new_bit_width,
-                                   unsigned int* new_sample_rate)
+static bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
+                                          int backend_idx,
+                                          struct audio_backend_cfg *backend_cfg)
 {
     bool backend_change = false;
     unsigned int bit_width;
     unsigned int sample_rate;
-    int backend_idx = DEFAULT_CODEC_BACKEND;
+    unsigned int channels;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
 
-    bit_width = *new_bit_width;
-    sample_rate = *new_sample_rate;
+    bit_width = backend_cfg->bit_width;
+    sample_rate = backend_cfg->sample_rate;
+    channels = backend_cfg->channels;
 
     ALOGI("%s:txbecf: afe: Codec selected backend: %d current bit width: %d and "
-          "sample rate: %d",__func__,backend_idx, bit_width, sample_rate);
+          "sample rate: %d, channels %d",__func__,backend_idx, bit_width,
+          sample_rate, channels);
 
     // For voice calls use default configuration i.e. 16b/48K, only applicable to
     // default backend
@@ -4487,18 +4802,27 @@
         bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
         sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
     }
+    if (backend_idx == USB_AUDIO_TX_BACKEND) {
+        audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, false);
+        ALOGV("%s: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
+              __func__, bit_width, sample_rate, channels);
+    }
 
     ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
           "sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
-    if ((bit_width != my_data->current_tx_backend_cfg[backend_idx].bit_width) ||
-        (sample_rate != my_data->current_tx_backend_cfg[backend_idx].sample_rate)) {
-        *new_bit_width = bit_width;
-        *new_sample_rate = sample_rate;
+    if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+        (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+        backend_cfg->bit_width = bit_width;
+        backend_cfg->sample_rate= sample_rate;
+        backend_cfg->channels = channels;
         backend_change = true;
         ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
-              "new sample rate: %d", __func__, *new_bit_width, *new_sample_rate);
+              "new sample rate: %d new channel: %d",
+              __func__, backend_cfg->bit_width,
+              backend_cfg->sample_rate, backend_cfg->channels);
     }
 
     return backend_change;
@@ -4507,29 +4831,34 @@
 bool platform_check_and_set_capture_codec_backend_cfg(struct audio_device* adev,
     struct audio_usecase *usecase, snd_device_t snd_device)
 {
-    unsigned int new_bit_width;
-    unsigned int new_sample_rate;
-    audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
-    int backend_idx = DEFAULT_CODEC_BACKEND;
+    int backend_idx = platform_get_backend_index(snd_device);
     int ret = 0;
+    struct audio_backend_cfg backend_cfg;
 
+    backend_cfg.passthrough_enabled = false;
     if(usecase->type == PCM_CAPTURE) {
-        new_sample_rate = usecase->stream.in->sample_rate;
-        new_bit_width = usecase->stream.in->bit_width;
-        format = usecase->stream.in->format;
+        backend_cfg.sample_rate= usecase->stream.in->sample_rate;
+        backend_cfg.bit_width= usecase->stream.in->bit_width;
+        backend_cfg.format= usecase->stream.in->format;
+        backend_cfg.channels = audio_channel_count_from_in_mask(usecase->stream.in->channel_mask);
     } else {
-        new_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
-        new_sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        backend_cfg.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        backend_cfg.sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        backend_cfg.format = AUDIO_FORMAT_PCM_16_BIT;
+        backend_cfg.channels = 1;
     }
 
-    ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
-          new_sample_rate, backend_idx, usecase->id,
+    ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, channel %d"
+          ", backend_idx %d usecase = %d device (%s)", __func__,
+          backend_cfg.bit_width,
+          backend_cfg.sample_rate,
+          backend_cfg.channels,
+          backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
-    if (platform_check_capture_codec_backend_cfg(adev, &new_bit_width,
-                                       &new_sample_rate)) {
-        ret = platform_set_capture_codec_backend_cfg(adev, snd_device,
-                                       new_bit_width, new_sample_rate, format);
+    if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
+                                                 &backend_cfg)) {
+        ret = platform_set_codec_backend_cfg(adev, snd_device,
+                                             backend_cfg);
         if(!ret)
             return true;
     }
@@ -4692,7 +5021,7 @@
     struct audio_device *adev = my_data->adev;
     char block[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE];
     int ret, count;
-
+    char *mix_ctl_name;
     struct mixer_ctl *ctl;
     char edid_data[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE + 1] = {0};
     edid_audio_info *info;
@@ -4702,17 +5031,28 @@
         return 0;
     }
 
+    switch(my_data->ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            mix_ctl_name = "HDMI EDID";
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            mix_ctl_name = "Display Port EDID";
+            break;
+        default:
+            ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+            return -EINVAL;
+    }
+
     if (my_data->edid_info == NULL) {
         my_data->edid_info =
             (struct edid_audio_info *)calloc(1, sizeof(struct edid_audio_info));
     }
 
     info = my_data->edid_info;
-
-    ctl = mixer_get_ctl_by_name(adev->mixer, AUDIO_DATA_BLOCK_MIXER_CTL);
+    ctl = mixer_get_ctl_by_name(adev->mixer, mix_ctl_name);
     if (!ctl) {
         ALOGE("%s: Could not get ctl for mixer cmd - %s",
-              __func__, AUDIO_DATA_BLOCK_MIXER_CTL);
+              __func__, mix_ctl_name);
         goto fail;
     }
 
@@ -4731,8 +5071,9 @@
     }
     edid_data[0] = count;
     memcpy(&edid_data[1], block, count);
+
     if (!edid_get_sink_caps(info, edid_data)) {
-        ALOGE("%s: Failed to get HDMI sink capabilities", __func__);
+        ALOGE("%s: Failed to get extn disp sink capabilities", __func__);
         goto fail;
     }
     my_data->edid_valid = true;
@@ -4751,16 +5092,28 @@
 int platform_set_channel_allocation(void *platform, int channel_alloc)
 {
     struct mixer_ctl *ctl;
-    const char *mixer_ctl_name = "HDMI RX CA";
+    char *mixer_ctl_name;
     int ret;
     struct platform_data *my_data = (struct platform_data *)platform;
     struct audio_device *adev = my_data->adev;
 
+    switch(my_data->ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            mixer_ctl_name = "HDMI RX CA";
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            mixer_ctl_name = "Display Port RX CA";
+            break;
+        default:
+            ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+            return -EINVAL;
+    }
+
     ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
     if (!ctl) {
         ALOGE("%s: Could not get ctl for mixer cmd - %s",
               __func__, mixer_ctl_name);
-        ret = EINVAL;
+        return -EINVAL;
     }
     ALOGD(":%s channel allocation = 0x%x", __func__, channel_alloc);
     ret = mixer_ctl_set_value(ctl, 0, channel_alloc);
@@ -4775,7 +5128,7 @@
 int platform_set_channel_map(void *platform, int ch_count, char *ch_map, int snd_id)
 {
     struct mixer_ctl *ctl;
-    char mixer_ctl_name[44]; // max length of name is 44 as defined
+    char mixer_ctl_name[44] = {0}; // max length of name is 44 as defined
     int ret;
     unsigned int i;
     int set_values[8] = {0};
@@ -4991,15 +5344,25 @@
 
 void platform_invalidate_hdmi_config(void * platform)
 {
+    //reset ext display EDID info
     struct platform_data *my_data = (struct platform_data *)platform;
     my_data->edid_valid = false;
     if (my_data->edid_info) {
         memset(my_data->edid_info, 0, sizeof(struct edid_audio_info));
     }
 
-    //reset HDMI_RX_BACKEND to default values
-    my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-    my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+    if (my_data->ext_disp_type == EXT_DISPLAY_TYPE_HDMI) {
+        //reset HDMI_RX_BACKEND to default values
+        my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        my_data->current_backend_cfg[HDMI_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+        my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+    } else {
+        //reset Display port BACKEND to default values
+        my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+        my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+    }
+    my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
 }
 
 int platform_set_mixer_control(struct stream_out *out, const char * mixer_ctl_name,
@@ -5018,91 +5381,6 @@
     return mixer_ctl_set_enum_by_string(ctl, mixer_val);
 }
 
-static int set_mixer_control(struct mixer *mixer,
-                             const char * mixer_ctl_name,
-                             const char *mixer_val)
-{
-    struct mixer_ctl *ctl;
-    ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
-    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
-    if (!ctl) {
-        ALOGE("%s: could not get ctl for mixer cmd - %s",
-              __func__, mixer_ctl_name);
-        return -EINVAL;
-    }
-
-    return mixer_ctl_set_enum_by_string(ctl, mixer_val);
-}
-
-int platform_set_hdmi_config(void *platform, uint32_t channel_count,
-                             uint32_t sample_rate, bool enable_passthrough)
-{
-    struct platform_data *my_data = (struct platform_data *)platform;
-    struct audio_device *adev = my_data->adev;
-    const char *hdmi_format_ctrl = "HDMI RX Format";
-    const char *hdmi_rate_ctrl   = "HDMI_RX SampleRate";
-    const char *hdmi_chans_ctrl  = "HDMI_RX Channels";
-    const char *channel_cnt_str  = NULL;
-
-    ALOGI("%s ch[%d] sr[%d], pthru[%d]", __func__,
-        channel_count, sample_rate, enable_passthrough);
-
-    switch (channel_count) {
-    case 8:
-        channel_cnt_str = "Eight"; break;
-    case 7:
-        channel_cnt_str = "Seven"; break;
-    case 6:
-        channel_cnt_str = "Six"; break;
-    case 5:
-        channel_cnt_str = "Five"; break;
-    case 4:
-        channel_cnt_str = "Four"; break;
-    case 3:
-        channel_cnt_str = "Three"; break;
-    default:
-        channel_cnt_str = "Two"; break;
-    }
-    ALOGV("%s: HDMI channel count: %s", __func__, channel_cnt_str);
-    set_mixer_control(adev->mixer, hdmi_chans_ctrl, channel_cnt_str);
-
-    if (enable_passthrough) {
-        ALOGD("%s:HDMI compress format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "Compr");
-    } else {
-        ALOGD("%s: HDMI PCM format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "LPCM");
-    }
-
-    switch (sample_rate) {
-    case 32000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_32");
-        break;
-    case 44100:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_44P1");
-        break;
-    case 96000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_96");
-        break;
-    case 128000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_128");
-        break;
-    case 176400:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_176_4");
-        break;
-    case 192000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_192");
-        break;
-    default:
-    case 48000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_48");
-        break;
-    }
-
-    return 0;
-}
-
-
 int platform_set_device_params(struct stream_out *out, int param, int value)
 {
     struct audio_device *adev = out->dev;
@@ -5234,8 +5512,11 @@
         snd_device == SND_DEVICE_OUT_SPEAKER_WSA ||
         snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
         snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT ||
         snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
-        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA) {
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA) {
         ret = true;
     }
 
@@ -5255,12 +5536,19 @@
         case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:
              acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
              break;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA:
+             acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED);
+             break;
         case SND_DEVICE_OUT_SPEAKER_VBAT:
              acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
              break;
         case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
              acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
              break;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+             acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT);
+             break;
         default:
              acdb_id = -EINVAL;
              break;
@@ -5280,15 +5568,37 @@
         case SND_DEVICE_OUT_VOICE_SPEAKER:
         case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:
              return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA:
+             return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
         case SND_DEVICE_OUT_SPEAKER_VBAT:
              return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
         case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
              return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+             return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
         default:
              return snd_device;
     }
 }
 
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device)
+{
+    switch(snd_device) {
+        case SND_DEVICE_OUT_SPEAKER_PROTECTED:
+        case SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED:
+        case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2;
+        default:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+    }
+}
+
 int platform_set_sidetone(struct audio_device *adev,
                           snd_device_t out_snd_device,
                           bool enable,
@@ -5311,3 +5621,53 @@
     }
     return 0;
 }
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+    return false;
+}
+
+bool platform_check_codec_asrc_support(void *platform __unused)
+{
+    return false;
+}
+
+int platform_send_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused, uint32_t module_id __unused,
+        uint32_t param_id __unused, void* data __unused,
+        int length __unused, bool persist __unused)
+{
+    return -ENOSYS;
+}
+
+int platform_get_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused, uint32_t module_id __unused,
+        uint32_t param_id __unused, void* data __unused,
+        int* length __unused, bool persist __unused)
+{
+    return -ENOSYS;
+}
+
+int platform_store_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused, uint32_t module_id __unused,
+        uint32_t param_id __unused,  void* data __unused,
+        int length __unused)
+{
+     return -ENOSYS;
+}
+
+int platform_retrieve_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused, uint32_t module_id __unused,
+        uint32_t param_id __unused, void* data __unused,
+        int* length __unused)
+{
+    return -ENOSYS;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 756c749..cba9068 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -39,6 +39,11 @@
     SOURCE_QUAD_MIC  = 0x8,            /* Target contains 4 mics */
 };
 
+enum {
+    SPKR_1,
+    SPKR_2
+};
+
 #define PLATFORM_IMAGE_NAME "modem"
 
 /*
@@ -92,12 +97,19 @@
     SND_DEVICE_OUT_VOICE_SPEAKER,
     SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
     SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT,
     SND_DEVICE_OUT_VOICE_HEADPHONES,
     SND_DEVICE_OUT_VOICE_LINE,
     SND_DEVICE_OUT_HDMI,
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+    SND_DEVICE_OUT_DISPLAY_PORT,
+    SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
     SND_DEVICE_OUT_BT_SCO,
     SND_DEVICE_OUT_BT_SCO_WB,
+    SND_DEVICE_OUT_BT_A2DP,
+    SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -114,8 +126,10 @@
     SND_DEVICE_OUT_ANC_HANDSET,
     SND_DEVICE_OUT_SPEAKER_PROTECTED,
     SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED,
     SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
     SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT,
 #ifdef RECORD_PLAY_CONCURRENCY
     SND_DEVICE_OUT_VOIP_HANDSET,
     SND_DEVICE_OUT_VOIP_SPEAKER,
@@ -174,6 +188,8 @@
     SND_DEVICE_IN_HANDSET_STEREO_DMIC,
     SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
     SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+    SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1,
+    SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2,
     SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,
     SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,
     SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,
@@ -196,24 +212,35 @@
     SND_DEVICE_MAX = SND_DEVICE_IN_END,
 
 };
-
+#define INPUT_SAMPLING_RATE_DSD64       2822400
+#define INPUT_SAMPLING_RATE_DSD128      5644800
 #define DEFAULT_OUTPUT_SAMPLING_RATE 48000
 #define OUTPUT_SAMPLING_RATE_44100      44100
+#define OUTPUT_SAMPLING_RATE_DSD64       176400
+#define OUTPUT_SAMPLING_RATE_DSD128      352800
 #define MAX_PORT                        6
 #define ALL_CODEC_BACKEND_PORT          0
 #define HEADPHONE_44_1_BACKEND_PORT     5
-#define MAX_CODEC_TX_BACKENDS           1
 enum {
     DEFAULT_CODEC_BACKEND,
     SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+    DSD_NATIVE_BACKEND,
+    SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
     HEADPHONE_44_1_BACKEND,
     SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
     HEADPHONE_BACKEND,
     SLIMBUS_6_RX = HEADPHONE_BACKEND,
     HDMI_RX_BACKEND,
+    DISP_PORT_RX_BACKEND,
     USB_AUDIO_RX_BACKEND,
+    MAX_RX_CODEC_BACKENDS = USB_AUDIO_RX_BACKEND,
+    /* TX BE follows RX BE */
+    SLIMBUS_0_TX,
+    DEFAULT_CODEC_TX_BACKEND = SLIMBUS_0_TX,
+    USB_AUDIO_TX_BACKEND,
     MAX_CODEC_BACKENDS
 };
+
 #define AUDIO_PARAMETER_KEY_NATIVE_AUDIO "audio.nat.codec.enabled"
 #define AUDIO_PARAMETER_KEY_NATIVE_AUDIO_MODE "native_audio_mode"
 
@@ -240,6 +267,7 @@
  * the buffer size of an input/output stream
  */
 #define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_DURATION 40 /* 40 millisecs */
 #define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
 #define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
 #define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
@@ -274,6 +302,7 @@
 #define AUDIO_RECORD_PCM_DEVICE 0
 #define MULTIMEDIA2_PCM_DEVICE 1
 #define MULTIMEDIA3_PCM_DEVICE 4
+#define MULTIMEDIA9_PCM_DEVICE 32
 #define FM_PLAYBACK_PCM_DEVICE 5
 #define FM_CAPTURE_PCM_DEVICE  6
 #define HFP_PCM_RX 5
@@ -354,7 +383,8 @@
 enum {
     LEGACY_PCM = 0,
     PASSTHROUGH,
-    PASSTHROUGH_CONVERT
+    PASSTHROUGH_CONVERT,
+    PASSTHROUGH_DSD
 };
 /*
  * ID for setting mute and lateny on the device side
@@ -370,4 +400,13 @@
     char device_name[100];
     char interface_name[100];
 };
+
+struct audio_backend_cfg {
+    unsigned int   sample_rate;
+    unsigned int   channels;
+    unsigned int   bit_width;
+    bool           passthrough_enabled;
+    audio_format_t format;
+};
+
 #endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 2b6a1d7..b5a4f11 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -667,6 +667,11 @@
     return -ENOSYS;
 }
 
+int platform_get_ext_disp_type(void *platform)
+{
+    return EXT_DISPLAY_TYPE_HDMI;
+}
+
 snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
@@ -1034,6 +1039,12 @@
     return -ENOSYS;
 }
 
+unsigned char* platform_get_license(void *platform, int *size)
+{
+    ALOGE("%s: Not implemented", __func__);
+    return NULL;
+}
+
 /* Delay in Us */
 int64_t platform_render_latency(audio_usecase_t usecase)
 {
@@ -1218,6 +1229,11 @@
     return -ENOSYS;
 }
 
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device __unused)
+{
+    return -ENOSYS;
+}
+
 int platform_spkr_prot_is_wsa_analog_mode(void *adev __unused)
 {
     return 0;
@@ -1232,7 +1248,48 @@
 bool platform_check_backends_match(snd_device_t snd_device1 __unused,
                                    snd_device_t snd_device2 __unused)
 {
-    return true;
+    return -ENOSYS;
+}
+
+int platform_send_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused, uint32_t module_id,
+        uint32_t param_id, void* data __unused,
+        int length __unused, bool persist __unused)
+{
+    return -ENOSYS;
+}
+
+int platform_get_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused, uint32_t module_id,
+        uint32_t param_id, void* data __unused,
+        int* length __unused, bool persist __unused)
+{
+    return -ENOSYS;
+}
+
+int platform_store_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused,  uint32_t module_id,
+        uint32_t param_id, void* data __unused,
+        int length __unused)
+{
+     return -ENOSYS;
+}
+
+
+int platform_retrieve_audio_cal(void* platform __unused,
+        int acdb_dev_id __unused, int acdb_device_type __unused,
+        int app_type __unused, int topology_id __unused,
+        int sample_rate __unused, uint32_t module_id,
+        uint32_t param_id, void* data __unused,
+        int* length __unused)
+{
+    return -ENOSYS;
 }
 
 int platform_set_sidetone(struct audio_device *adev,
@@ -1257,3 +1314,19 @@
     }
     return 0;
 }
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+    return false;
+}
+
+int platform_get_backend_index(snd_device_t snd_device __unused);
+{
+    return 0;
+}
+
+bool platform_check_codec_asrc_support(void *platform __unused)
+{
+    return false;
+}
+
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index aab5436..c9ac74a 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2013, 2015 The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013, 2016 The Linux Foundation. All rights reserved.
  * Not a contribution.
  *
  * Copyright (C) 2013 The Android Open Source Project
@@ -63,7 +63,11 @@
     SND_DEVICE_OUT_VOICE_HEADPHONES,
     SND_DEVICE_OUT_HDMI,
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+    SND_DEVICE_OUT_DISPLAY_PORT,
+    SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
     SND_DEVICE_OUT_BT_SCO,
+    SND_DEVICE_OUT_BT_A2DP,
+    SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
     SND_DEVICE_OUT_BT_SCO_WB,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
@@ -110,6 +114,12 @@
 #define SOUND_CARD 0
 
 #define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+#define INPUT_SAMPLING_RATE_DSD64       2822400
+#define INPUT_SAMPLING_RATE_DSD128      5644800
+#define OUTPUT_SAMPLING_RATE_DSD64       176400
+#define OUTPUT_SAMPLING_RATE_DSD128      352800
+#define DSD_NATIVE_BACKEND 1
+#define PASSTHROUGH_DSD 3
 
 #define ALL_SESSION_VSID                0xFFFFFFFF
 #define DEFAULT_MUTE_RAMP_DURATION_MS   20
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 9143d35..dae7ff3 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -135,6 +135,10 @@
     SND_DEVICE_OUT_SPEAKER
 };
 
+static const snd_device_t tasha_sbc_variant_devices[] = {
+    SND_DEVICE_IN_HANDSET_MIC
+};
+
 static const snd_device_t taiko_apq8084_sbc_variant_devices[] = {
     SND_DEVICE_IN_HANDSET_MIC,
     SND_DEVICE_IN_SPEAKER_MIC,
@@ -301,6 +305,12 @@
         hw_info->snd_devices = (snd_device_t *)tasha_DB_variant_devices;
         hw_info->num_snd_devices = ARRAY_SIZE(tasha_DB_variant_devices);
         strlcpy(hw_info->dev_extn, "-db", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "msm8996-tasha-sbc-snd-card")) {
+        strlcpy(hw_info->type, " sbc", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)tasha_sbc_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(tasha_sbc_variant_devices);
+        strlcpy(hw_info->dev_extn, "-sbc", sizeof(hw_info->dev_extn));
     } else {
         ALOGW("%s: Not a 8996 device", __func__);
     }
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index b98cc73..fa67342 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -42,6 +42,7 @@
 #include "edid.h"
 #include "sound/compress_params.h"
 #include "sound/msmcal-hwdep.h"
+#include <linux/msm_audio_calibration.h>
 
 #define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
 #define MIXER_XML_DEFAULT_PATH "/system/etc/mixer_paths.xml"
@@ -57,10 +58,10 @@
 #include <linux/msm_audio.h>
 
 #define LIB_ACDB_LOADER "libacdbloader.so"
-#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
 #define CVD_VERSION_MIXER_CTL "CVD Version"
 
-#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024 * 1024)
 #define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
@@ -108,6 +109,8 @@
 #define AUDIO_PARAMETER_KEY_AUD_CALDATA   "cal_data"
 #define AUDIO_PARAMETER_KEY_AUD_CALRESULT "cal_result"
 
+#define AUDIO_PARAMETER_KEY_MONO_SPEAKER "mono_speaker"
+
 #define AUDIO_PARAMETER_KEY_PERF_LOCK_OPTS "perf_lock_opts"
 
 /* Reload ACDB files from specified path */
@@ -188,8 +191,10 @@
 typedef struct codec_backend_cfg {
     uint32_t sample_rate;
     uint32_t bit_width;
+    uint32_t channels;
     char     *bitwidth_mixer_ctl;
     char     *samplerate_mixer_ctl;
+    char     *channels_mixer_ctl;
 } codec_backend_cfg_t;
 
 static native_audio_prop na_props = {0, 0, 0};
@@ -216,6 +221,7 @@
     /* Vbat monitor related flags */
     bool is_vbat_speaker;
     bool gsm_mode_enabled;
+    int mono_speaker;
     /* Audio calibration related functions */
     void                       *acdb_handle;
     int                        voice_feature_set;
@@ -235,9 +241,9 @@
     struct csd_data *csd;
     void *edid_info;
     bool edid_valid;
+    int ext_disp_type;
     char ec_ref_mixer_path[64];
     codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
-    codec_backend_cfg_t current_tx_backend_cfg[MAX_CODEC_TX_BACKENDS];
     char codec_version[CODEC_VERSION_MAX_LENGTH];
     int hw_dep_fd;
     char cvd_version[MAX_CVD_VERSION_STRING_SIZE];
@@ -245,6 +251,8 @@
     int metainfo_key;
     int source_mic_type;
     int max_mic_count;
+    bool is_dsd_supported;
+    bool is_asrc_supported;
 };
 
 static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -318,6 +326,7 @@
                                           AFE_PROXY_RECORD_PCM_DEVICE},
     [USECASE_AUDIO_RECORD_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
                                         AFE_PROXY_RECORD_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = {MULTIMEDIA9_PCM_DEVICE, -1},
 
 };
 
@@ -332,6 +341,7 @@
     [SND_DEVICE_OUT_SPEAKER_VBAT] = "speaker-vbat",
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
     [SND_DEVICE_OUT_HEADPHONES] = "headphones",
+    [SND_DEVICE_OUT_HEADPHONES_DSD] = "headphones-dsd",
     [SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
     [SND_DEVICE_OUT_LINE] = "line",
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
@@ -341,12 +351,18 @@
     [SND_DEVICE_OUT_VOICE_HANDSET] = "voice-handset",
     [SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
     [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "voice-speaker-vbat",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2] = "voice-speaker-2",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = "voice-speaker-2-vbat",
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
     [SND_DEVICE_OUT_VOICE_LINE] = "voice-line",
     [SND_DEVICE_OUT_HDMI] = "hdmi",
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
+    [SND_DEVICE_OUT_DISPLAY_PORT] = "display-port",
+    [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = "speaker-and-display-port",
     [SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
     [SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+    [SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
@@ -363,8 +379,10 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = "voice-speaker-2-protected",
     [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = "voice-speaker-2-protected-vbat",
 
     /* Capture sound devices */
     [SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -414,6 +432,8 @@
     [SND_DEVICE_IN_HANDSET_STEREO_DMIC] = "handset-stereo-dmic-ef",
     [SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = "speaker-stereo-dmic-ef",
     [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = "vi-feedback",
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = "vi-feedback-mono-1",
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = "vi-feedback-mono-2",
     [SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = "voice-speaker-dmic-broadside",
     [SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = "speaker-dmic-broadside",
     [SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = "speaker-dmic-broadside",
@@ -423,6 +443,7 @@
     [SND_DEVICE_IN_SPEAKER_QMIC_AEC] = "quad-mic",
     [SND_DEVICE_IN_SPEAKER_QMIC_NS] = "quad-mic",
     [SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = "quad-mic",
+    [SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = "quad-mic",
     [SND_DEVICE_IN_THREE_MIC] = "three-mic",
     [SND_DEVICE_IN_HANDSET_TMIC] = "three-mic",
     [SND_DEVICE_IN_UNPROCESSED_MIC] = "unprocessed-mic",
@@ -446,6 +467,7 @@
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
     [SND_DEVICE_OUT_LINE] = 10,
     [SND_DEVICE_OUT_HEADPHONES] = 10,
+    [SND_DEVICE_OUT_HEADPHONES_DSD] = 10,
     [SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
     [SND_DEVICE_OUT_SPEAKER_AND_LINE] = 10,
@@ -454,12 +476,18 @@
     [SND_DEVICE_OUT_VOICE_HANDSET] = 7,
     [SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
     [SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 14,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2] = 14,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = 14,
     [SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
     [SND_DEVICE_OUT_VOICE_LINE] = 10,
     [SND_DEVICE_OUT_HDMI] = 18,
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
+    [SND_DEVICE_OUT_DISPLAY_PORT] = 18,
+    [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = 14,
     [SND_DEVICE_OUT_BT_SCO] = 22,
     [SND_DEVICE_OUT_BT_SCO_WB] = 39,
+    [SND_DEVICE_OUT_BT_A2DP] = 20,
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
@@ -476,8 +504,10 @@
     [SND_DEVICE_OUT_ANC_HANDSET] = 103,
     [SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = 101,
     [SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
     [SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
+    [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = 101,
 
     [SND_DEVICE_IN_HANDSET_MIC] = 4,
     [SND_DEVICE_IN_HANDSET_MIC_EXTERNAL] = 4,
@@ -526,6 +556,8 @@
     [SND_DEVICE_IN_HANDSET_STEREO_DMIC] = 34,
     [SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = 35,
     [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = 102,
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = 102,
     [SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = 12,
     [SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = 12,
     [SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = 119,
@@ -535,6 +567,7 @@
     [SND_DEVICE_IN_SPEAKER_QMIC_AEC] = 126,
     [SND_DEVICE_IN_SPEAKER_QMIC_NS] = 127,
     [SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = 129,
+    [SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = 125,
     [SND_DEVICE_IN_THREE_MIC] = 46, /* for APSS Surround Sound Recording */
     [SND_DEVICE_IN_HANDSET_TMIC] = 125, /* for 3mic recording with fluence */
     [SND_DEVICE_IN_UNPROCESSED_MIC] = 143,
@@ -560,6 +593,7 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_DSD)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_LINE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
@@ -569,12 +603,18 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_LINE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_DISPLAY_PORT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
@@ -590,8 +630,10 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_EXTERNAL)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC)},
@@ -637,6 +679,8 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2)},
     {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE)},
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE)},
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE)},
@@ -646,6 +690,7 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_AEC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_NS)},
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE)},
     {TO_NAME_INDEX(SND_DEVICE_IN_THREE_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_MIC)},
@@ -691,6 +736,7 @@
     {TO_NAME_INDEX(USECASE_AUDIO_SPKR_CALIB_TX)},
     {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_AFE_PROXY)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_AFE_PROXY)},
+    {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE)},
 };
 
 #define NO_COLS 2
@@ -783,6 +829,7 @@
 #define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
 #define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
 #define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+#define ULL_PLATFORM_DELAY         (6*1000LL)
 
 bool platform_send_gain_dep_cal(void *platform, int level) {
     bool ret_val = false;
@@ -1051,7 +1098,8 @@
                  sizeof("apq8084-taiko-i2s-cdp-snd-card"))) {
         plat_data->is_i2s_ext_modem = true;
     }
-    ALOGV("%s, is_i2s_ext_modem:%d",__func__, plat_data->is_i2s_ext_modem);
+    ALOGV("%s, is_i2s_ext_modem:%d soundcard name is %s",__func__,
+           plat_data->is_i2s_ext_modem, snd_card_name);
 
     return plat_data->is_i2s_ext_modem;
 }
@@ -1080,6 +1128,8 @@
     backend_tag_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
     backend_tag_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
     backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
+    backend_tag_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("display-port");
+    backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("speaker-and-display-port");
     backend_tag_table[SND_DEVICE_OUT_VOICE_TX] = strdup("afe-proxy");
     backend_tag_table[SND_DEVICE_IN_VOICE_RX] = strdup("afe-proxy");
     backend_tag_table[SND_DEVICE_OUT_AFE_PROXY] = strdup("afe-proxy");
@@ -1089,12 +1139,21 @@
     backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
     backend_tag_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
     backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
+    backend_tag_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("headphones-dsd");
     backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
     backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
+    backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("voice-speaker-2-vbat");
+    backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
+    backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
 
+    hw_interface_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("SLIMBUS_2_RX");
     hw_interface_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("SLIMBUS_5_RX");
     hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
+    hw_interface_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("DISPLAY_PORT_RX");
+    hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("SLIMBUS_0_RX-and-DISPLAY_PORT_RX");
+    hw_interface_table[SND_DEVICE_OUT_USB_HEADSET] = strdup("USB_AUDIO_RX");
+    hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = strdup("SLIMBUS_0_RX-and-USB_AUDIO_RX");
     hw_interface_table[SND_DEVICE_OUT_VOICE_TX] = strdup("AFE_PCM_RX");
 
     my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
@@ -1505,7 +1564,9 @@
     my_data->slowtalk = false;
     my_data->hd_voice = false;
     my_data->edid_info = NULL;
+    my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
     my_data->hw_dep_fd = -1;
+    my_data->mono_speaker = SPKR_1;
 
     property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
     if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
@@ -1672,6 +1733,9 @@
     /* init usb */
     audio_extn_usb_init(adev);
 
+    /*init a2dp*/
+    audio_extn_a2dp_init(adev);
+
     /* init dap hal */
     audio_extn_dap_hal_init(adev->snd_card);
 
@@ -1690,6 +1754,12 @@
         if (idx == HEADPHONE_44_1_BACKEND)
             my_data->current_backend_cfg[idx].sample_rate = OUTPUT_SAMPLING_RATE_44100;
         my_data->current_backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_CHANNELS;
+        if (idx > MAX_RX_CODEC_BACKENDS)
+            my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+        my_data->current_backend_cfg[idx].bitwidth_mixer_ctl = NULL;
+        my_data->current_backend_cfg[idx].samplerate_mixer_ctl = NULL;
+        my_data->current_backend_cfg[idx].channels_mixer_ctl = NULL;
     }
 
     my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
@@ -1697,16 +1767,28 @@
     my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
         strdup("SLIM_0_RX SampleRate");
 
+    my_data->current_backend_cfg[DSD_NATIVE_BACKEND].bitwidth_mixer_ctl =
+        strdup("SLIM_2_RX Format");
+    my_data->current_backend_cfg[DSD_NATIVE_BACKEND].samplerate_mixer_ctl =
+        strdup("SLIM_2_RX SampleRate");
+
     my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].bitwidth_mixer_ctl =
         strdup("SLIM_5_RX Format");
     my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
         strdup("SLIM_5_RX SampleRate");
 
-    my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+    my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
         strdup("SLIM_0_TX Format");
-    my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+    my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
         strdup("SLIM_0_TX SampleRate");
 
+    my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
+        strdup("USB_AUDIO_TX Format");
+    my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
+        strdup("USB_AUDIO_TX SampleRate");
+    my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].channels_mixer_ctl =
+        strdup("USB_AUDIO_TX Channels");
+
     ret = audio_extn_utils_get_codec_version(snd_card_name,
                                              my_data->adev->snd_card,
                                              my_data->codec_version);
@@ -1727,6 +1809,13 @@
         }
     }
 
+    if(strstr(snd_card_name, "tavil")) {
+        ALOGD("%s:DSD playback is supported", __func__);
+        my_data->is_dsd_supported = true;
+        my_data->is_asrc_supported = true;
+        platform_set_native_support(NATIVE_AUDIO_MODE_MULTIPLE_44_1);
+    }
+
     my_data->current_backend_cfg[HEADPHONE_BACKEND].bitwidth_mixer_ctl =
         strdup("SLIM_6_RX Format");
     my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
@@ -1735,11 +1824,21 @@
         strdup("HDMI_RX Bit Format");
     my_data->current_backend_cfg[HDMI_RX_BACKEND].samplerate_mixer_ctl =
         strdup("HDMI_RX SampleRate");
+    my_data->current_backend_cfg[HDMI_RX_BACKEND].channels_mixer_ctl =
+        strdup("HDMI_RX Channels");
+    my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bitwidth_mixer_ctl =
+        strdup("Display Port RX Bit Format");
+    my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].samplerate_mixer_ctl =
+        strdup("Display Port RX SampleRate");
+    my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels_mixer_ctl =
+        strdup("Display Port RX Channels");
 
     my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].bitwidth_mixer_ctl =
         strdup("USB_AUDIO_RX Format");
     my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].samplerate_mixer_ctl =
         strdup("USB_AUDIO_RX SampleRate");
+    my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].channels_mixer_ctl =
+        strdup("USB_AUDIO_RX Channels");
 
     my_data->edid_info = NULL;
     free(snd_card_name);
@@ -1839,7 +1938,8 @@
         return;
     }
 
-    if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+    if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
         !(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
         ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
         return;
@@ -2065,14 +2165,15 @@
 {
     if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
         ALOGE("%s: Invalid snd_device = %d", __func__, snd_device);
-        return DEFAULT_OUTPUT_SAMPLING_RATE;
+        return CODEC_BACKEND_DEFAULT_BIT_WIDTH;
     }
     return backend_bit_width_table[snd_device];
 }
 
 int platform_set_native_support(int na_mode)
 {
-    if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode) {
+    if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode
+        || NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode) {
         na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled = true;
         na_props.na_mode = na_mode;
         ALOGD("%s:napb: native audio playback enabled in (%s) mode v2.0", __func__,
@@ -2087,6 +2188,18 @@
     return 0;
 }
 
+bool platform_check_codec_dsd_support(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    return my_data->is_dsd_supported;
+}
+
+bool platform_check_codec_asrc_support(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    return my_data->is_asrc_supported;
+}
+
 int platform_get_native_support()
 {
     int ret = NATIVE_AUDIO_MODE_INVALID;
@@ -2139,6 +2252,8 @@
             mode = NATIVE_AUDIO_MODE_SRC;
         else if (value && !strncmp(value, "true", sizeof("true")))
             mode = NATIVE_AUDIO_MODE_TRUE_44_1;
+        else if (value && !strncmp(value, "multiple", sizeof("multiple")))
+            mode = NATIVE_AUDIO_MODE_MULTIPLE_44_1;
         else {
             mode = NATIVE_AUDIO_MODE_INVALID;
             ALOGE("%s:napb:native_audio_mode in platform info xml,invalid mode string",
@@ -2205,28 +2320,52 @@
 
     return ret;
 }
-static int platform_get_backend_index(snd_device_t snd_device)
+
+int codec_device_supports_native_playback(audio_devices_t out_device)
+{
+    int ret = false;
+
+    if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+        out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+        out_device & AUDIO_DEVICE_OUT_LINE)
+        ret = true;
+
+    return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
 {
     int32_t port = DEFAULT_CODEC_BACKEND;
 
-    if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+    if (snd_device >= SND_DEVICE_OUT_BEGIN && snd_device < SND_DEVICE_OUT_END) {
         if (backend_tag_table[snd_device] != NULL) {
                 if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
                             sizeof("headphones-44.1")) == 0)
                         port = HEADPHONE_44_1_BACKEND;
+                else if (strncmp(backend_tag_table[snd_device], "headphones-dsd",
+                            sizeof("headphones-dsd")) == 0)
+                        port = DSD_NATIVE_BACKEND;
                 else if (strncmp(backend_tag_table[snd_device], "headphones",
                             sizeof("headphones")) == 0)
                         port = HEADPHONE_BACKEND;
                 else if (strcmp(backend_tag_table[snd_device], "hdmi") == 0)
                         port = HDMI_RX_BACKEND;
+                else if (strcmp(backend_tag_table[snd_device], "display-port") == 0)
+                        port = DISP_PORT_RX_BACKEND;
                 else if (strcmp(backend_tag_table[snd_device], "usb-headphones") == 0)
                         port = USB_AUDIO_RX_BACKEND;
         }
+    } else if (snd_device >= SND_DEVICE_IN_BEGIN && snd_device < SND_DEVICE_IN_END) {
+        port = DEFAULT_CODEC_TX_BACKEND;
+        if (backend_tag_table[snd_device] != NULL) {
+                if (strcmp(backend_tag_table[snd_device], "usb-headset-mic") == 0)
+                        port = USB_AUDIO_TX_BACKEND;
+        }
     } else {
-        ALOGV("%s:napb: Invalid device - %d ", __func__, snd_device);
+        ALOGW("%s:napb: Invalid device - %d ", __func__, snd_device);
     }
 
-    ALOGV("%s:napb: backend port - %d snd_device %d", __func__, port, snd_device);
+    ALOGV("%s:napb: backend port - %d device - %d ", __func__, port, snd_device);
     return port;
 }
 
@@ -2311,7 +2450,9 @@
         return ret;
 
     if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
-         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
          audio_extn_spkr_prot_is_enabled()) {
         if (my_data->is_vbat_speaker)
             acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2346,9 +2487,16 @@
     if (my_data->acdb_send_voice_cal == NULL) {
         ALOGE("%s: dlsym error for acdb_send_voice_call", __func__);
     } else {
-        if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
-            audio_extn_spkr_prot_is_enabled())
-            out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+        if (audio_extn_spkr_prot_is_enabled()) {
+            if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+            else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+            else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
+            else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)
+                out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
+        }
 
         acdb_rx_id = acdb_device_table[out_snd_device];
         acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2375,7 +2523,9 @@
         return ret;
 
     if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
-         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+         out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
          audio_extn_spkr_prot_is_enabled()) {
         if (my_data->is_vbat_speaker)
             acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2576,20 +2726,68 @@
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_HDMI;
         status = true;
+    } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT &&
+               !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_DISPLAY_PORT)) {
+        *num_devices = 2;
+        new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+        new_snd_devices[1] = SND_DEVICE_OUT_DISPLAY_PORT;
+        status = true;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_USB_HEADSET)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_USB_HEADSET;
         status = true;
+    } else if (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) {
+        *num_devices = 2;
+        new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+        new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
     }
 
+
     ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
         snd_device, *num_devices, *new_snd_devices);
 
     return status;
 }
 
+int platform_get_ext_disp_type(void *platform)
+{
+    int disp_type;
+    struct platform_data *my_data = (struct platform_data *)platform;
+
+    if (my_data->ext_disp_type != EXT_DISPLAY_TYPE_NONE) {
+         ALOGD("%s: Returning cached ext disp type:%s",
+               __func__, (my_data->ext_disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+         return my_data->ext_disp_type;
+    }
+
+#ifdef DISPLAY_PORT_ENABLED
+    struct audio_device *adev = my_data->adev;
+    struct mixer_ctl *ctl;
+    char *mixer_ctl_name = "External Display Type";
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+
+    disp_type = mixer_ctl_get_value(ctl, 0);
+    if (disp_type == EXT_DISPLAY_TYPE_NONE) {
+         ALOGE("%s: Invalid external display type: %d", __func__, disp_type);
+         return -EINVAL;
+    }
+#else
+    disp_type = EXT_DISPLAY_TYPE_HDMI;
+#endif
+
+    my_data->ext_disp_type = disp_type;
+    ALOGD("%s: ext disp type:%s", __func__, (disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+    return disp_type;
+}
+
 snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
@@ -2635,13 +2833,26 @@
             snd_device = SND_DEVICE_OUT_SPEAKER_AND_LINE;
         } else if (devices == (AUDIO_DEVICE_OUT_AUX_DIGITAL |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
-            snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+            switch(my_data->ext_disp_type) {
+                case EXT_DISPLAY_TYPE_HDMI:
+                    snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+                    break;
+                case EXT_DISPLAY_TYPE_DP:
+                    snd_device = SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT;
+                    break;
+                default:
+                    ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+                    goto exit;
+            }
         } else if (devices == (AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
             snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
         } else if (devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
             snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
+        } else if ((devices & AUDIO_DEVICE_OUT_SPEAKER) &&
+                   (devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
+            snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP;
         } else {
             ALOGE("%s: Invalid combo device(%#x)", __func__, devices);
             goto exit;
@@ -2693,10 +2904,19 @@
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
         } else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
-                if (my_data->is_vbat_speaker)
-                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
-                else
-                    snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+                if (my_data->is_vbat_speaker) {
+                    if (my_data->mono_speaker == SPKR_1)
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+                    else
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT;
+                } else {
+                    if (my_data->mono_speaker == SPKR_1)
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+                    else
+                        snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2;
+                }
+        } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+            snd_device = SND_DEVICE_OUT_BT_A2DP;
         } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
                    devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
             snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -2716,7 +2936,8 @@
     }
 
     if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
-        devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+        devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+        devices & AUDIO_DEVICE_OUT_LINE) {
         if (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET
             && audio_extn_get_anc_enabled()) {
             if (audio_extn_should_use_fb_anc())
@@ -2726,6 +2947,12 @@
         } else if (NATIVE_AUDIO_MODE_SRC == na_mode &&
                    OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
                 snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+        } else if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode &&
+                   (sample_rate % OUTPUT_SAMPLING_RATE_44100 == 0) &&
+                   (out->format != AUDIO_FORMAT_DSD)) {
+                snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+        } else if (out->format == AUDIO_FORMAT_DSD) {
+                snd_device = SND_DEVICE_OUT_HEADPHONES_DSD;
         } else
             snd_device = SND_DEVICE_OUT_HEADPHONES;
     } else if (devices & AUDIO_DEVICE_OUT_LINE) {
@@ -2746,8 +2973,20 @@
             snd_device = SND_DEVICE_OUT_BT_SCO_WB;
         else
             snd_device = SND_DEVICE_OUT_BT_SCO;
+    } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+        snd_device = SND_DEVICE_OUT_BT_A2DP;
     } else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-        snd_device = SND_DEVICE_OUT_HDMI ;
+            switch(my_data->ext_disp_type) {
+                case EXT_DISPLAY_TYPE_HDMI:
+                    snd_device = SND_DEVICE_OUT_HDMI;
+                    break;
+                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_ANLG_DOCK_HEADSET ||
                devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
         ALOGD("%s: setting USB hadset channel capability(2) for Proxy", __func__);
@@ -2896,7 +3135,7 @@
             if (my_data->fluence_in_voice_rec && channel_count == 1) {
                 if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
                     (my_data->source_mic_type & SOURCE_QUAD_MIC)) {
-                     snd_device = SND_DEVICE_IN_HANDSET_QMIC;
+                     snd_device = SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE;
                 } else if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
                     (my_data->source_mic_type & SOURCE_THREE_MIC)) {
                     snd_device = SND_DEVICE_IN_HANDSET_TMIC;
@@ -3153,7 +3392,7 @@
     struct audio_device *adev = my_data->adev;
     struct mixer_ctl *ctl;
     const char *channel_cnt_str = NULL;
-    const char *mixer_ctl_name = "HDMI_RX Channels";
+    char *mixer_ctl_name;
     switch (channel_count) {
     case 8:
         channel_cnt_str = "Eight"; break;
@@ -3170,13 +3409,26 @@
     default:
         channel_cnt_str = "Two"; break;
     }
+
+    switch(my_data->ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            mixer_ctl_name = "HDMI_RX Channels";
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            mixer_ctl_name = "Display Port RX Channels";
+            break;
+        default:
+            ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+            return -EINVAL;
+    }
+
     ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
     if (!ctl) {
         ALOGE("%s: Could not get ctl for mixer cmd - %s",
               __func__, mixer_ctl_name);
         return -EINVAL;
     }
-    ALOGV("HDMI channel count: %s", channel_cnt_str);
+    ALOGV("Ext disp channel count: %s", channel_cnt_str);
     mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
     return 0;
 }
@@ -3558,6 +3810,16 @@
 
     }
 
+    err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER, value, len);
+    if (err >= 0) {
+        if (!strncmp("left", value, sizeof("left")))
+            my_data->mono_speaker = SPKR_1;
+        else if (!strncmp("right", value, sizeof("right")))
+            my_data->mono_speaker = SPKR_2;
+
+        str_parms_del(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER);
+    }
+
     err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE,
                             value, len);
     if (err >= 0) {
@@ -3850,8 +4112,8 @@
                 !strncmp("true", propValue, 4);
         }
 
-        if (prop_playback_enabled && (voice_is_in_call(my_data->adev) ||
-             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev)))) {
+        if ((prop_playback_enabled && (voice_is_in_call(my_data->adev))) ||
+             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev))) {
             char *decoder_mime_type = value;
 
             //check if unsupported mime type or not
@@ -3875,6 +4137,54 @@
     free(kv_pairs);
 }
 
+unsigned char* platform_get_license(void *platform, int *size)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    char value[PROPERTY_VALUE_MAX] = {0};
+    acdb_audio_cal_cfg_t cal;
+    unsigned char *dptr = NULL;
+    int ret=0;
+    uint32_t param_len;
+
+    if (platform == NULL) {
+        ALOGE("[%s] received null pointer %d ",__func__, __LINE__);
+        ret = -EINVAL;
+        goto done;
+    }
+    memset(&cal, 0, sizeof(cal));
+    cal.persist = 1;
+    cal.cal_type = AUDIO_CORE_METAINFO_CAL_TYPE;
+    if (!property_get("audio.qaf.acdbid", value , "") && !atoi(value)) {
+        ALOGE("[%s] audio.qaf.acdbid is not set %d ",__func__, __LINE__);
+        ret = -EINVAL;
+        goto done;
+    }
+    cal.acdb_dev_id = (uint32_t) atoi (value);
+    param_len = MAX_SET_CAL_BYTE_SIZE;
+    dptr = (unsigned char*) calloc(param_len, sizeof(unsigned char*));
+    if (dptr == NULL) {
+        ALOGE("[%s] Memory allocation failed for length %d",__func__,param_len);
+        ret = -ENOMEM;
+        goto done;
+    }
+    if (my_data->acdb_get_audio_cal != NULL) {
+        ret = my_data->acdb_get_audio_cal((void*)&cal, (void*)dptr, &param_len);
+        ALOGE("%s, ret[%d], param_len[%d] line %d", __func__, ret, param_len, __LINE__);
+        if (ret == 0) {
+            *size = param_len;
+            return dptr;
+        } else {
+            *size = 0;
+        }
+    }
+done:
+    if (dptr != NULL)
+        free(dptr);
+
+    return NULL;
+}
+
+/* Delay in Us */
 /* Delay in Us, only to be used for PCM formats */
 int64_t platform_render_latency(audio_usecase_t usecase)
 {
@@ -3886,6 +4196,8 @@
         case USECASE_AUDIO_PLAYBACK_OFFLOAD:
         case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
              return PCM_OFFLOAD_PLATFORM_DELAY;
+        case USECASE_AUDIO_PLAYBACK_ULL:
+             return ULL_PLATFORM_DELAY;
         default:
             return 0;
     }
@@ -3906,7 +4218,9 @@
     if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
         (snd_device < SND_DEVICE_IN_END) &&
         (snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
-        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
         needs_event = true;
 
     return needs_event;
@@ -3924,7 +4238,9 @@
     if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
         (snd_device < SND_DEVICE_IN_END) &&
         (snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
-        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
         needs_event = true;
 
     return needs_event;
@@ -3956,18 +4272,21 @@
         fragment_size = info->offload_buffer_size;
     }
 
-    // For FLAC use max size since it is loss less, and has sampling rates
-    // upto 192kHZ
-    if (info != NULL && !info->has_video &&
-        info->format == AUDIO_FORMAT_FLAC) {
-       fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
-       ALOGV("FLAC fragment size %d", fragment_size);
-    }
-
-    if (info != NULL && info->has_video && info->is_streaming) {
-        fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
-        ALOGV("%s: offload fragment size reduced for AV streaming to %d",
-               __func__, fragment_size);
+    if (info != NULL && !info->has_video) {
+        if (info->is_streaming) {
+            fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
+            ALOGV("%s: offload fragment size reduced for AV streaming to %d",
+                   __func__, fragment_size);
+        } else if (info->format == AUDIO_FORMAT_FLAC) {
+            fragment_size = FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+            ALOGV("FLAC fragment size %d", fragment_size);
+        } else if (info->format == AUDIO_FORMAT_DSD) {
+            fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+            if((property_get("audio.native.dsd.buffer.size.kb", value, "")) &&
+                    atoi(value))
+                fragment_size =  atoi(value) * 1024;
+            ALOGV("DSD fragment size %d", fragment_size);
+        }
     }
 
     fragment_size = ALIGN( fragment_size, 1024);
@@ -3981,26 +4300,54 @@
 }
 
 /*
+ * return backend_idx on which voice call is active
+ */
+static int platform_get_voice_call_backend(struct audio_device* adev)
+{
+   struct audio_usecase *uc = NULL;
+   struct listnode *node;
+   snd_device_t out_snd_device = SND_DEVICE_NONE;
+
+   int backend_idx = -1;
+
+   if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
+       list_for_each(node, &adev->usecase_list) {
+           uc =  node_to_item(node, struct audio_usecase, list);
+           if (uc && (uc->type == VOICE_CALL || uc->type == VOIP_CALL) && uc->stream.out) {
+               out_snd_device = platform_get_output_snd_device(adev->platform, uc->stream.out);
+               backend_idx = platform_get_backend_index(out_snd_device);
+               break;
+           }
+       }
+   }
+   return backend_idx;
+}
+
+/*
  * configures afe with bit width and Sample Rate
  */
 static int platform_set_codec_backend_cfg(struct audio_device* adev,
-                         snd_device_t snd_device, unsigned int bit_width,
-                         unsigned int sample_rate, audio_format_t format)
+                         snd_device_t snd_device, struct audio_backend_cfg backend_cfg)
 {
     int ret = 0;
-    int backend_idx = DEFAULT_CODEC_BACKEND;
+    int backend_idx = platform_get_backend_index(snd_device);
     struct platform_data *my_data = (struct platform_data *)adev->platform;
-
     backend_idx = platform_get_backend_index(snd_device);
+    unsigned int bit_width = backend_cfg.bit_width;
+    unsigned int sample_rate = backend_cfg.sample_rate;
+    unsigned int channels = backend_cfg.channels;
+    audio_format_t format = backend_cfg.format;
+    bool passthrough_enabled = backend_cfg.passthrough_enabled;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d device (%s)", __func__,  bit_width, sample_rate, backend_idx,
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+          ", backend_idx %d device (%s)", __func__,  bit_width,
+          sample_rate, channels, backend_idx,
           platform_get_snd_device_name(snd_device));
 
     if (bit_width !=
         my_data->current_backend_cfg[backend_idx].bit_width) {
 
-        struct  mixer_ctl *ctl;
+        struct  mixer_ctl *ctl = NULL;
         ctl = mixer_get_ctl_by_name(adev->mixer,
                     my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
         if (!ctl) {
@@ -4012,31 +4359,30 @@
 
         if (bit_width == 24) {
             if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
-                 mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+                 ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
             else
-                 mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+                 ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
         } else if (bit_width == 32) {
-            mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+            ret = mixer_ctl_set_enum_by_string(ctl, "S32_LE");
         } else {
-            mixer_ctl_set_enum_by_string(ctl, "S16_LE");
+            ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
         }
-        my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
-        ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
-              my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+        if ( ret < 0) {
+            ALOGE("%s:becf: afe: fail for %s mixer set to %d bit for %x format", __func__,
+                  my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+        } else {
+            my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
+            ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+                  my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+        }
+        /* set the ret as 0 and not pass back to upper layer */
+        ret = 0;
     }
 
-    /*
-     * Backend sample rate configuration follows:
-     * 16 bit playback - 48khz for streams at any valid sample rate
-     * 24 bit playback - 48khz for stream sample rate less than 48khz
-     * 24 bit playback - 96khz for sample rate range of 48khz to 96khz
-     * 24 bit playback - 192khz for sample rate range of 96khz to 192 khz
-     * Upper limit is inclusive in the sample rate range.
-     */
     if (sample_rate !=
        my_data->current_backend_cfg[backend_idx].sample_rate) {
             char *rate_str = NULL;
-            struct  mixer_ctl *ctl;
+            struct  mixer_ctl *ctl = NULL;
 
             switch (sample_rate) {
             case 8000:
@@ -4051,14 +4397,24 @@
                 rate_str = "KHZ_44P1";
                 break;
             case 64000:
-            case 88200:
             case 96000:
                 rate_str = "KHZ_96";
                 break;
+            case 88200:
+                rate_str = "KHZ_88P2";
+                break;
             case 176400:
+                rate_str = "KHZ_176P4";
+                break;
             case 192000:
                 rate_str = "KHZ_192";
                 break;
+            case 352800:
+                rate_str = "KHZ_352P8";
+                break;
+            case 384000:
+                rate_str = "KHZ_384";
+                break;
             default:
                 rate_str = "KHZ_48";
                 break;
@@ -4078,48 +4434,208 @@
             mixer_ctl_set_enum_by_string(ctl, rate_str);
             my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
     }
+    if ((my_data->current_backend_cfg[backend_idx].channels_mixer_ctl) &&
+        (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+        struct  mixer_ctl *ctl = NULL;
+        char *channel_cnt_str = NULL;
+
+        switch (channels) {
+        case 8:
+            channel_cnt_str = "Eight"; break;
+        case 7:
+            channel_cnt_str = "Seven"; break;
+        case 6:
+            channel_cnt_str = "Six"; break;
+        case 5:
+            channel_cnt_str = "Five"; break;
+        case 4:
+            channel_cnt_str = "Four"; break;
+        case 3:
+            channel_cnt_str = "Three"; break;
+        case 1:
+            channel_cnt_str = "One"; break;
+        case 2:
+        default:
+            channel_cnt_str = "Two"; break;
+        }
+
+        ctl = mixer_get_ctl_by_name(adev->mixer,
+           my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                   __func__,
+                   my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+            return -EINVAL;
+        }
+        mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
+        my_data->current_backend_cfg[backend_idx].channels = channels;
+
+        if (backend_idx == HDMI_RX_BACKEND)
+            platform_set_edid_channels_configuration(adev->platform, channels);
+
+        ALOGD("%s:becf: afe: %s set to %s", __func__,
+               my_data->current_backend_cfg[backend_idx].channels_mixer_ctl,
+               channel_cnt_str);
+    }
+
+    bool set_ext_disp_format = false;
+    char *ext_disp_format = NULL;
+
+    if (backend_idx == HDMI_RX_BACKEND) {
+        ext_disp_format = "HDMI RX Format";
+        set_ext_disp_format = true;
+    } else if (backend_idx == DISP_PORT_RX_BACKEND) {
+        ext_disp_format = "Display Port Rx Format";
+        set_ext_disp_format = true;
+    } else {
+        ALOGV("%s: Format doesnt have to be set", __func__);
+    }
+
+    if (set_ext_disp_format) {
+        struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, ext_disp_format);
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                  __func__, ext_disp_format);
+            return -EINVAL;
+        }
+
+        if (passthrough_enabled) {
+            ALOGD("%s:Ext display compress format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "Compr");
+        } else {
+            ALOGD("%s: Ext display PCM format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "LPCM");
+        }
+    }
+
+    if (snd_device == SND_DEVICE_OUT_HEADPHONES || snd_device ==
+        SND_DEVICE_OUT_HEADPHONES_44_1) {
+        if (sample_rate > 48000 ||
+            (bit_width >= 24 && (sample_rate == 48000  || sample_rate == 44100))) {
+            ALOGV("%s: apply HPH HQ mode\n", __func__);
+            audio_route_apply_and_update_path(adev->audio_route, "hph-highquality-mode");
+        } else {
+            ALOGV("%s: apply HPH LP mode\n", __func__);
+            audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
+        }
+    }
 
     return ret;
 }
 
 /*
+ *Validate the selected bit_width, sample_rate and channels using the edid
+ *of the connected sink device.
+ */
+static void platform_check_hdmi_backend_cfg(struct audio_device* adev,
+                                   struct audio_usecase* usecase,
+                                   int backend_idx,
+                                   struct audio_backend_cfg *hdmi_backend_cfg)
+{
+    unsigned int bit_width;
+    unsigned int sample_rate;
+    unsigned int channels, max_supported_channels = 0;
+    struct platform_data *my_data = (struct platform_data *)adev->platform;
+    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool passthrough_enabled = false;
+
+    bit_width = hdmi_backend_cfg->bit_width;
+    sample_rate = hdmi_backend_cfg->sample_rate;
+    channels = hdmi_backend_cfg->channels;
+
+
+    ALOGI("%s:becf: HDMI: bitwidth %d, samplerate %d, channels %d"
+          ", usecase = %d", __func__, bit_width,
+          sample_rate, channels, usecase->id);
+
+    if (audio_extn_passthru_is_enabled() && audio_extn_passthru_is_active()
+        && (usecase->stream.out->compr_config.codec->compr_passthr != 0)) {
+        passthrough_enabled = true;
+        ALOGI("passthrough is enabled for this stream");
+    }
+
+    // For voice calls use default configuration i.e. 16b/48K, only applicable to
+    // default backend
+    if (!passthrough_enabled) {
+
+        max_supported_channels = platform_edid_get_max_channels(my_data);
+
+        //Check EDID info for supported samplerate
+        if (!edid_is_supported_sr(edid_info,sample_rate)) {
+            //reset to current sample rate
+            sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
+        }
+
+        //Check EDID info for supported bit width
+        if (!edid_is_supported_bps(edid_info,bit_width)) {
+            //reset to current sample rate
+            bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
+        }
+
+        if (channels > max_supported_channels)
+            channels = max_supported_channels;
+
+    } else {
+        /*During pass through set default bit width and channels*/
+        channels = DEFAULT_HDMI_OUT_CHANNELS;
+        if ((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+            (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
+            sample_rate = sample_rate * 4 ;
+
+        bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        /* We force route so that the BE format can be set to Compr */
+    }
+
+    ALOGI("%s:becf: afe: HDMI backend: passthrough %d updated bit width: %d and sample rate: %d"
+           "channels %d", __func__, passthrough_enabled , bit_width,
+           sample_rate, channels);
+
+    hdmi_backend_cfg->bit_width = bit_width;
+    hdmi_backend_cfg->sample_rate = sample_rate;
+    hdmi_backend_cfg->channels = channels;
+    hdmi_backend_cfg->passthrough_enabled = passthrough_enabled;
+}
+
+/*
  * goes through all the current usecases and picks the highest
  * bitwidth & samplerate
  */
 static bool platform_check_codec_backend_cfg(struct audio_device* adev,
                                    struct audio_usecase* usecase,
                                    snd_device_t snd_device,
-                                   unsigned int* new_bit_width,
-                                   unsigned int* new_sample_rate)
+                                   struct audio_backend_cfg *backend_cfg)
 {
     bool backend_change = false;
     struct listnode *node;
     unsigned int bit_width;
     unsigned int sample_rate;
+    unsigned int channels;
+    bool passthrough_enabled = false;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
     int na_mode = platform_get_native_support();
-    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool channels_updated = false;
 
     backend_idx = platform_get_backend_index(snd_device);
 
-    bit_width = *new_bit_width;
-    sample_rate = *new_sample_rate;
+    bit_width = backend_cfg->bit_width;
+    sample_rate = backend_cfg->sample_rate;
+    channels = backend_cfg->channels;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
           ", backend_idx %d usecase = %d device (%s)", __func__, bit_width,
-          sample_rate, backend_idx, usecase->id,
+          sample_rate, channels, backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
 
     // For voice calls use default configuration i.e. 16b/48K, only applicable to
     // default backend
     // force routing is not required here, caller will do it anyway
-    if ((voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
-        backend_idx == DEFAULT_CODEC_BACKEND) {
+    if (backend_idx == platform_get_voice_call_backend(adev)) {
         ALOGW("%s:becf: afe:Use default bw and sr for voice/voip calls ",
               __func__);
         bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
         sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        channels = CODEC_BACKEND_DEFAULT_CHANNELS;
     } else {
         /*
          * The backend should be configured at highest bit width and/or
@@ -4136,11 +4652,12 @@
             uc = node_to_item(node, struct audio_usecase, list);
             struct stream_out *out = (struct stream_out*) uc->stream.out;
             if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+                unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
 
                 ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
-                      "(%d) device %s", __func__, i++, use_case_table[uc->id],
+                      "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
                       uc->id, out->sample_rate,
-                      out->bit_width,
+                      out->bit_width, out_channels,
                       platform_get_snd_device_name(uc->out_snd_device));
 
                 if (platform_check_backends_match(snd_device, uc->out_snd_device)) {
@@ -4150,86 +4667,124 @@
                             sample_rate = out->sample_rate;
                         if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
                             sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                        if (channels < out_channels)
+                            channels = out_channels;
                 }
             }
         }
     }
 
-    if (audio_is_true_native_stream_active(adev)) {
-        if (check_hdset_combo_device(snd_device)) {
-        /*
-         * In true native mode Tasha has a limitation that one port at 44.1 khz
-         * cannot drive both spkr and hdset, to simiplify the solution lets
-         * move the AFE to 48khzwhen a ring tone selects combo device.
-         */
-            sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-            bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
-            ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
-                  __func__);
-        } else {
-        /*
-         * in single BE mode, if native audio playback
-         * is active then it will take priority
-         */
-            sample_rate = OUTPUT_SAMPLING_RATE_44100;
-            ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
-                  __func__);
+    /* Native playback is preferred for Headphone/HS device over 192Khz */
+    if (codec_device_supports_native_playback(usecase->devices)) {
+        if (audio_is_true_native_stream_active(adev)) {
+            if (check_hdset_combo_device(snd_device)) {
+                /*
+                 * In true native mode Tasha has a limitation that one port at 44.1 khz
+                 * cannot drive both spkr and hdset, to simiplify the solution lets
+                 * move the AFE to 48khzwhen a ring tone selects combo device.
+                 * or if NATIVE playback is not enabled.
+                 */
+                    sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                    bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+                    ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
+                          __func__);
+            } else {
+             /*
+              * in single BE mode, if native audio playback
+              * is active then it will take priority
+              */
+                 sample_rate = OUTPUT_SAMPLING_RATE_44100;
+                 ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
+                       __func__);
+            }
+        } else if ((OUTPUT_SAMPLING_RATE_44100 == sample_rate) &&
+                   (na_mode != NATIVE_AUDIO_MODE_MULTIPLE_44_1)) {
+                 sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                 ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
+                       __func__);
         }
-    }
+    } else if ((usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) ||
+               (usecase->devices & AUDIO_DEVICE_OUT_EARPIECE) ) {
 
-    /*
-     * hifi playback not supported on spkr devices, limit the Sample Rate
-     * to 48 khz.
-     */
-    if (SND_DEVICE_OUT_SPEAKER == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_WSA == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_VBAT == snd_device) {
+        if (bit_width >= 24) {
+            bit_width = platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
+            ALOGD("%s:becf: afe: reset bitwidth to %d (based on supported"
+                   " value for this platform)", __func__, bit_width);
+        }
         sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-        ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+        ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
             "default Sample Rate(48k)", __func__);
     }
 
-    /*
-     * native playback is not enabled.Configure afe to default Sample Rate(48k)
-     */
-    if (NATIVE_AUDIO_MODE_INVALID == na_mode &&
-            OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
-        sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-        ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
-              __func__);
-    }
-
     if (backend_idx == USB_AUDIO_RX_BACKEND) {
-        unsigned int channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
-        audio_extn_usb_is_config_supported(&bit_width, &sample_rate, channels);
+        audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, true);
         ALOGV("%s: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
                    __func__, bit_width, sample_rate, channels);
+        if (channels != my_data->current_backend_cfg[backend_idx].channels)
+            channels_updated = true;
     }
 
-    if (backend_idx == HDMI_RX_BACKEND) {
-        //Check EDID info for supported samplerate
-        if (!edid_is_supported_sr(edid_info,sample_rate)) {
-            //reset to current sample rate
-            sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
-        }
-        //Check EDID info for supported bit widhth
-        if (!edid_is_supported_bps(edid_info,bit_width)) {
-            //reset to current sample rate
-            bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
+    if (backend_idx == HDMI_RX_BACKEND || backend_idx == DISP_PORT_RX_BACKEND) {
+        struct audio_backend_cfg hdmi_backend_cfg;
+        hdmi_backend_cfg.bit_width = bit_width;
+        hdmi_backend_cfg.sample_rate = sample_rate;
+        hdmi_backend_cfg.channels = channels;
+        hdmi_backend_cfg.passthrough_enabled = false;
+
+        /*
+         * HDMI does not support 384Khz/32bit playback hence configure BE to 24b/192Khz
+         * TODO: Instead have the validation against edid return the next best match
+         */
+        if (bit_width > 24)
+            hdmi_backend_cfg.bit_width = 24;
+        if (sample_rate > 192000)
+            hdmi_backend_cfg.sample_rate = 192000;
+
+        platform_check_hdmi_backend_cfg(adev, usecase, backend_idx, &hdmi_backend_cfg);
+
+        bit_width = hdmi_backend_cfg.bit_width;
+        sample_rate = hdmi_backend_cfg.sample_rate;
+        channels = hdmi_backend_cfg.channels;
+        passthrough_enabled = hdmi_backend_cfg.passthrough_enabled;
+
+        if (channels != my_data->current_backend_cfg[backend_idx].channels)
+            channels_updated = true;
+    }
+
+    /*
+     * Map native sampling rates to upper limit range
+     * if multiple of native sampling rates are not supported.
+     */
+    if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 != na_mode) {
+        switch (sample_rate) {
+            case 88200:
+                sample_rate = 96000;
+                break;
+            case 176400:
+                sample_rate = 192000;
+                break;
+            case 352800:
+                sample_rate = 192000;
+                break;
         }
     }
+
     ALOGI("%s:becf: afe: Codec selected backend: %d updated bit width: %d and sample rate: %d",
           __func__, backend_idx , bit_width, sample_rate);
 
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
     if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
-        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
-        *new_bit_width = bit_width;
-        *new_sample_rate = sample_rate;
+        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+         passthrough_enabled || channels_updated) {
+        backend_cfg->bit_width = bit_width;
+        backend_cfg->sample_rate = sample_rate;
+        backend_cfg->channels = channels;
+        backend_cfg->passthrough_enabled = passthrough_enabled;
         backend_change = true;
-        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d new sample rate: %d",
-              __func__, *new_bit_width, *new_sample_rate);
+        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d"
+               "new sample rate: %d new channels: %d",
+              __func__, backend_cfg->bit_width, backend_cfg->sample_rate, backend_cfg->channels);
     }
 
     return backend_change;
@@ -4238,158 +4793,50 @@
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
     struct audio_usecase *usecase, snd_device_t snd_device)
 {
-    unsigned int new_bit_width;
-    unsigned int new_sample_rate;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     int new_snd_devices[SND_DEVICE_OUT_END];
     int i, num_devices = 1;
     bool ret = false;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
-    audio_format_t format;
+    struct audio_backend_cfg backend_cfg;
 
     backend_idx = platform_get_backend_index(snd_device);
 
-    new_bit_width = usecase->stream.out->bit_width;
-    new_sample_rate = usecase->stream.out->sample_rate;
-    format = usecase->stream.out->format;
+    backend_cfg.bit_width = usecase->stream.out->bit_width;
+    backend_cfg.sample_rate = usecase->stream.out->sample_rate;
+    backend_cfg.format = usecase->stream.out->format;
+    backend_cfg.channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
+    /*this is populated by check_codec_backend_cfg hence set default value to false*/
+    backend_cfg.passthrough_enabled = false;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
-          new_sample_rate, backend_idx, usecase->id,
+    /* Set Backend sampling rate to 176.4 for DSD64 and
+     * 352.8Khz for DSD128.
+     * Set Bit Width to 16
+     */
+    if ((backend_idx == DSD_NATIVE_BACKEND) && (backend_cfg.format == AUDIO_FORMAT_DSD)) {
+        backend_cfg.bit_width = 16;
+        if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD64)
+            backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+        else if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD128)
+            backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+    }
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+          ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
+          backend_cfg.sample_rate, backend_cfg.channels, backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
 
-
     if (!platform_can_split_snd_device(my_data, snd_device, &num_devices, new_snd_devices))
         new_snd_devices[0] = snd_device;
 
     for (i = 0; i < num_devices; i++) {
         ALOGI("%s: new_snd_devices[%d] is %d", __func__, i, new_snd_devices[i]);
-        if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
-                                             &new_bit_width, &new_sample_rate)) {
-                platform_set_codec_backend_cfg(adev, new_snd_devices[i],
-                                               new_bit_width, new_sample_rate, format);
-                ret = true;
+        if ((platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
+                                             &backend_cfg))) {
+            platform_set_codec_backend_cfg(adev, new_snd_devices[i],
+                                           backend_cfg);
+            ret = true;
         }
     }
-
-    return ret;
-}
-
-/*
- * configures afe with bit width and Sample Rate
- */
-
-int platform_set_capture_codec_backend_cfg(struct audio_device* adev,
-                         snd_device_t snd_device,
-                         unsigned int bit_width, unsigned int sample_rate,
-                         audio_format_t format)
-{
-    int ret = 0;
-    int backend_idx = DEFAULT_CODEC_BACKEND;
-    struct platform_data *my_data = (struct platform_data *)adev->platform;
-
-    ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
-          __func__, bit_width, sample_rate, backend_idx,
-          platform_get_snd_device_name(snd_device));
-
-    if (bit_width !=
-        my_data->current_tx_backend_cfg[backend_idx].bit_width) {
-
-        struct  mixer_ctl *ctl = NULL;
-        ctl = mixer_get_ctl_by_name(adev->mixer,
-                        my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
-        if (!ctl) {
-            ALOGE("%s:txbecf: afe: Could not get ctl for mixer command - %s",
-                  __func__,
-                  my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
-            return -EINVAL;
-        }
-        if (bit_width == 24) {
-            if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
-                ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
-            else
-                ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
-        } else {
-            ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
-        }
-
-        if (ret < 0) {
-            ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
-                  __func__,
-                  my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
-            return -EINVAL;
-        }
-
-        my_data->current_tx_backend_cfg[backend_idx].bit_width = bit_width;
-        ALOGD("%s:txbecf: afe: %s mixer set to %d bit", __func__,
-              my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
-    }
-
-    /*
-     * Backend sample rate configuration follows:
-     * 16 bit record - 48khz for streams at any valid sample rate
-     * 24 bit record - 48khz for stream sample rate less than 48khz
-     * 24 bit record - 96khz for sample rate range of 48khz to 96khz
-     * 24 bit record - 192khz for sample rate range of 96khz to 192 khz
-     * Upper limit is inclusive in the sample rate range.
-     */
-    // TODO: This has to be more dynamic based on policy file
-
-    if (sample_rate != my_data->current_tx_backend_cfg[(int)backend_idx].sample_rate) {
-            /*
-             * sample rate update is needed only for hifi audio enabled platforms
-             */
-            char *rate_str = NULL;
-            struct  mixer_ctl *ctl = NULL;
-
-            switch (sample_rate) {
-            case 8000:
-            case 11025:
-            case 16000:
-            case 22050:
-            case 32000:
-            case 44100:
-            case 48000:
-                rate_str = "KHZ_48";
-                break;
-            case 64000:
-            case 88200:
-            case 96000:
-                rate_str = "KHZ_96";
-                break;
-            case 176400:
-            case 192000:
-                rate_str = "KHZ_192";
-                break;
-            default:
-                rate_str = "KHZ_48";
-                break;
-            }
-
-            ctl = mixer_get_ctl_by_name(adev->mixer,
-                my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-
-            if (!ctl) {
-                ALOGE("%s:txbecf: afe: Could not get ctl to set the Sample Rate for mixer command - %s",
-                      __func__,
-                      my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-                return -EINVAL;
-            }
-
-            ALOGD("%s:txbecf: afe: %s set to %s", __func__,
-                  my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl,
-                  rate_str);
-            ret = mixer_ctl_set_enum_by_string(ctl, rate_str);
-            if (ret < 0) {
-                ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
-                      __func__,
-                      my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-                return -EINVAL;
-            }
-
-            my_data->current_tx_backend_cfg[backend_idx].sample_rate = sample_rate;
-    }
-
     return ret;
 }
 
@@ -4397,43 +4844,54 @@
  * goes through all the current usecases and picks the highest
  * bitwidth & samplerate
  */
-bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
-                                   unsigned int* new_bit_width,
-                                   unsigned int* new_sample_rate)
+static bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
+                                   int backend_idx,
+                                   struct audio_backend_cfg *backend_cfg)
 {
     bool backend_change = false;
     unsigned int bit_width;
     unsigned int sample_rate;
-    int backend_idx = DEFAULT_CODEC_BACKEND;
+    unsigned int channels;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
 
-    bit_width = *new_bit_width;
-    sample_rate = *new_sample_rate;
+    bit_width = backend_cfg->bit_width;
+    sample_rate = backend_cfg->sample_rate;
+    channels = backend_cfg->channels;
 
     ALOGI("%s:txbecf: afe: Codec selected backend: %d current bit width: %d and "
-          "sample rate: %d",__func__,backend_idx, bit_width, sample_rate);
+          "sample rate: %d, channels %d",__func__,backend_idx, bit_width,
+          sample_rate, channels);
 
     // For voice calls use default configuration i.e. 16b/48K, only applicable to
     // default backend
     // force routing is not required here, caller will do it anyway
     if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
-        ALOGW("%s:txbecf: afe:Use default bw and sr for voice/voip calls and "
+        ALOGW("%s:txbecf: afe: Use default bw and sr for voice/voip calls and "
               "for unprocessed/camera source", __func__);
         bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
         sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
     }
+    if (backend_idx == USB_AUDIO_TX_BACKEND) {
+        audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, false);
+        ALOGV("%s:txbecf: afe: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
+              __func__, bit_width, sample_rate, channels);
+    }
 
     ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
           "sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
-    if ((bit_width != my_data->current_tx_backend_cfg[backend_idx].bit_width) ||
-        (sample_rate != my_data->current_tx_backend_cfg[backend_idx].sample_rate)) {
-        *new_bit_width = bit_width;
-        *new_sample_rate = sample_rate;
+    if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+        (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+        backend_cfg->bit_width = bit_width;
+        backend_cfg->sample_rate= sample_rate;
+        backend_cfg->channels = channels;
         backend_change = true;
         ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
-              "new sample rate: %d", __func__, *new_bit_width, *new_sample_rate);
+              "new sample rate: %d new channel: %d",
+              __func__, backend_cfg->bit_width,
+              backend_cfg->sample_rate, backend_cfg->channels);
     }
 
     return backend_change;
@@ -4442,28 +4900,34 @@
 bool platform_check_and_set_capture_codec_backend_cfg(struct audio_device* adev,
     struct audio_usecase *usecase, snd_device_t snd_device)
 {
-    unsigned int new_bit_width;
-    unsigned int new_sample_rate;
-    audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
-    int backend_idx = DEFAULT_CODEC_BACKEND;
+    int backend_idx = platform_get_backend_index(snd_device);
     int ret = 0;
+    struct audio_backend_cfg backend_cfg;
+
+    backend_cfg.passthrough_enabled = false;
     if(usecase->type == PCM_CAPTURE) {
-        new_sample_rate = usecase->stream.in->sample_rate;
-        new_bit_width = usecase->stream.in->bit_width;
-        format = usecase->stream.in->format;
+        backend_cfg.sample_rate= usecase->stream.in->sample_rate;
+        backend_cfg.bit_width= usecase->stream.in->bit_width;
+        backend_cfg.format= usecase->stream.in->format;
+        backend_cfg.channels = audio_channel_count_from_in_mask(usecase->stream.in->channel_mask);
     } else {
-        new_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
-        new_sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        backend_cfg.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        backend_cfg.sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        backend_cfg.format = AUDIO_FORMAT_PCM_16_BIT;
+        backend_cfg.channels = 1;
     }
 
-    ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
-          new_sample_rate, backend_idx, usecase->id,
+    ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, channel %d"
+          ", backend_idx %d usecase = %d device (%s)", __func__,
+          backend_cfg.bit_width,
+          backend_cfg.sample_rate,
+          backend_cfg.channels,
+          backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
-    if (platform_check_capture_codec_backend_cfg(adev, &new_bit_width,
-                                                 &new_sample_rate)) {
-        ret = platform_set_capture_codec_backend_cfg(adev, snd_device,
-                                       new_bit_width, new_sample_rate, format);
+    if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
+                                                 &backend_cfg)) {
+        ret = platform_set_codec_backend_cfg(adev, snd_device,
+                                             backend_cfg);
         if(!ret)
             return true;
     }
@@ -4626,7 +5090,7 @@
     struct audio_device *adev = my_data->adev;
     char block[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE];
     int ret, count;
-
+    char *mix_ctl_name;
     struct mixer_ctl *ctl;
     char edid_data[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE + 1] = {0};
     edid_audio_info *info;
@@ -4636,17 +5100,28 @@
         return 0;
     }
 
+    switch(my_data->ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            mix_ctl_name = "HDMI EDID";
+            break;
+        case EXT_DISPLAY_TYPE_DP:
+            mix_ctl_name = "Display Port EDID";
+            break;
+        default:
+            ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+            return -EINVAL;
+    }
+
     if (my_data->edid_info == NULL) {
         my_data->edid_info =
             (struct edid_audio_info *)calloc(1, sizeof(struct edid_audio_info));
     }
 
     info = my_data->edid_info;
-
-    ctl = mixer_get_ctl_by_name(adev->mixer, AUDIO_DATA_BLOCK_MIXER_CTL);
+    ctl = mixer_get_ctl_by_name(adev->mixer, mix_ctl_name);
     if (!ctl) {
         ALOGE("%s: Could not get ctl for mixer cmd - %s",
-              __func__, AUDIO_DATA_BLOCK_MIXER_CTL);
+              __func__, mix_ctl_name);
         goto fail;
     }
 
@@ -4667,7 +5142,7 @@
     memcpy(&edid_data[1], block, count);
 
     if (!edid_get_sink_caps(info, edid_data)) {
-        ALOGE("%s: Failed to get HDMI sink capabilities", __func__);
+        ALOGE("%s: Failed to get extn disp sink capabilities", __func__);
         goto fail;
     }
     my_data->edid_valid = true;
@@ -4686,16 +5161,28 @@
 int platform_set_channel_allocation(void *platform, int channel_alloc)
 {
     struct mixer_ctl *ctl;
-    const char *mixer_ctl_name = "HDMI RX CA";
+    char *mixer_ctl_name;
     int ret;
     struct platform_data *my_data = (struct platform_data *)platform;
     struct audio_device *adev = my_data->adev;
 
+    switch(my_data->ext_disp_type) {
+        case EXT_DISPLAY_TYPE_HDMI:
+            mixer_ctl_name = "HDMI RX CA";
+        break;
+        case EXT_DISPLAY_TYPE_DP:
+            mixer_ctl_name = "Display Port RX CA";
+            break;
+        default:
+            ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+            return -EINVAL;
+    }
+
     ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
     if (!ctl) {
         ALOGE("%s: Could not get ctl for mixer cmd - %s",
               __func__, mixer_ctl_name);
-        ret = EINVAL;
+        return -EINVAL;
     }
     ALOGD(":%s channel allocation = 0x%x", __func__, channel_alloc);
     ret = mixer_ctl_set_value(ctl, 0, channel_alloc);
@@ -4927,16 +5414,25 @@
 
 void platform_invalidate_hdmi_config(void * platform)
 {
-    //reset HDMI EDID info
+    //reset ext display EDID info
     struct platform_data *my_data = (struct platform_data *)platform;
     my_data->edid_valid = false;
     if (my_data->edid_info) {
         memset(my_data->edid_info, 0, sizeof(struct edid_audio_info));
     }
 
-    //reset HDMI_RX_BACKEND to default values
-    my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-    my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+    if (my_data->ext_disp_type == EXT_DISPLAY_TYPE_HDMI) {
+        //reset HDMI_RX_BACKEND to default values
+        my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        my_data->current_backend_cfg[HDMI_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+        my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+    } else {
+        //reset Display port BACKEND to default values
+        my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+        my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+    }
+    my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
 }
 
 int platform_set_mixer_control(struct stream_out *out, const char * mixer_ctl_name,
@@ -4955,90 +5451,6 @@
     return mixer_ctl_set_enum_by_string(ctl, mixer_val);
 }
 
-static int set_mixer_control(struct mixer *mixer,
-                             const char * mixer_ctl_name,
-                             const char *mixer_val)
-{
-    struct mixer_ctl *ctl;
-    ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
-    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
-    if (!ctl) {
-        ALOGE("%s: could not get ctl for mixer cmd - %s",
-              __func__, mixer_ctl_name);
-        return -EINVAL;
-    }
-
-    return mixer_ctl_set_enum_by_string(ctl, mixer_val);
-}
-
-int platform_set_hdmi_config(void *platform, uint32_t channel_count,
-                             uint32_t sample_rate, bool enable_passthrough)
-{
-    struct platform_data *my_data = (struct platform_data *)platform;
-    struct audio_device *adev = my_data->adev;
-    const char *hdmi_format_ctrl = "HDMI RX Format";
-    const char *hdmi_rate_ctrl   = "HDMI_RX SampleRate";
-    const char *hdmi_chans_ctrl  = "HDMI_RX Channels";
-    const char *channel_cnt_str  = NULL;
-
-    ALOGI("%s ch[%d] sr[%d], pthru[%d]", __func__,
-        channel_count, sample_rate, enable_passthrough);
-
-    switch (channel_count) {
-    case 8:
-        channel_cnt_str = "Eight"; break;
-    case 7:
-        channel_cnt_str = "Seven"; break;
-    case 6:
-        channel_cnt_str = "Six"; break;
-    case 5:
-        channel_cnt_str = "Five"; break;
-    case 4:
-        channel_cnt_str = "Four"; break;
-    case 3:
-        channel_cnt_str = "Three"; break;
-    default:
-        channel_cnt_str = "Two"; break;
-    }
-    ALOGV("%s: HDMI channel count: %s", __func__, channel_cnt_str);
-    set_mixer_control(adev->mixer, hdmi_chans_ctrl, channel_cnt_str);
-
-    if (enable_passthrough) {
-        ALOGD("%s:HDMI compress format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "Compr");
-    } else {
-        ALOGD("%s: HDMI PCM format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "LPCM");
-    }
-
-    switch (sample_rate) {
-    case 32000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_32");
-        break;
-    case 44100:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_44P1");
-        break;
-    case 96000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_96");
-        break;
-    case 128000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_128");
-        break;
-    case 176400:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_176_4");
-        break;
-    case 192000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_192");
-        break;
-    default:
-    case 48000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_48");
-        break;
-    }
-
-    return 0;
-}
-
 int platform_set_device_params(struct stream_out *out, int param, int value)
 {
     struct audio_device *adev = out->dev;
@@ -5073,7 +5485,9 @@
     if (snd_device == SND_DEVICE_OUT_SPEAKER ||
         snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
         snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
-        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) {
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2) {
         ret = true;
     }
 
@@ -5091,12 +5505,18 @@
         case SND_DEVICE_OUT_VOICE_SPEAKER:
              acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
              break;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+             acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED);
+             break;
         case SND_DEVICE_OUT_SPEAKER_VBAT:
              acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
              break;
         case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
              acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
              break;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+             acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT);
+             break;
         default:
              acdb_id = -EINVAL;
              break;
@@ -5114,14 +5534,34 @@
              return SND_DEVICE_OUT_SPEAKER_PROTECTED;
         case SND_DEVICE_OUT_VOICE_SPEAKER:
              return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+             return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
         case SND_DEVICE_OUT_SPEAKER_VBAT:
              return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
         case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
              return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+             return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
         default:
              return snd_device;
     }
 }
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device)
+{
+    switch(snd_device) {
+        case SND_DEVICE_OUT_SPEAKER_PROTECTED:
+        case SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED:
+        case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1;
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+        case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2;
+        default:
+             return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+    }
+}
 
 int platform_spkr_prot_is_wsa_analog_mode(void *adev __unused)
 {
@@ -5202,3 +5642,153 @@
     }
     return 0;
 }
+
+static void make_cal_cfg(acdb_audio_cal_cfg_t* cal, int acdb_dev_id,
+        int acdb_device_type, int app_type, int topology_id,
+        int sample_rate, uint32_t module_id, uint32_t param_id, bool persist)
+{
+    int persist_send_flags = 1;
+
+    if (!cal) {
+        return;
+    }
+
+    if (persist)
+        persist_send_flags |= 0x2;
+
+    memset(cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+    cal->persist = persist;
+    cal->app_type = app_type;
+    cal->acdb_dev_id = acdb_dev_id;
+    cal->sampling_rate = sample_rate;
+    cal->topo_id = topology_id;
+    //if module and param id is set to 0, the whole blob will be stored
+    //or sent to the DSP
+    cal->module_id = module_id;
+    cal->param_id = param_id;
+    cal->cal_type = acdb_device_type;
+    cal->persist = persist;
+
+}
+
+int platform_send_audio_cal(void* platform, int acdb_dev_id,
+       int acdb_device_type, int app_type, int topology_id, int sample_rate,
+       uint32_t module_id, uint32_t param_id, void* data, int length, bool persist)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+    acdb_audio_cal_cfg_t cal;
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+    if (!my_data) {
+        ret = -EINVAL;
+        goto ERROR_RETURN;
+    }
+
+    make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+        sample_rate, module_id, param_id, true);
+
+    if (my_data->acdb_set_audio_cal) {
+        // persist audio cal in local cache
+        if (persist) {
+            ret = my_data->acdb_set_audio_cal((void*)&cal, data, (uint32_t)length);
+        }
+        // send audio cal to dsp
+        if (ret == 0) {
+            cal.persist = false;
+            ret = my_data->acdb_set_audio_cal((void*)&cal, data, (uint32_t)length);
+            if (persist && (ret != 0)) {
+                ALOGV("[%s] audio cal stored with success, ignore set cal failure", __func__);
+                ret = 0;
+            }
+        }
+    }
+
+ERROR_RETURN:
+    return ret;
+}
+
+int platform_get_audio_cal(void* platform, int acdb_dev_id,
+       int acdb_device_type, int app_type, int topology_id,
+       int sample_rate, uint32_t module_id, uint32_t param_id,
+       void* data, int* length, bool persist)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+    acdb_audio_cal_cfg_t cal;
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+    if (!my_data) {
+        ret = -EINVAL;
+        goto ERROR_RETURN;
+    }
+
+    make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+        sample_rate, module_id, param_id, false);
+
+    if (my_data->acdb_get_audio_cal) {
+        // get cal from dsp
+        ret = my_data->acdb_get_audio_cal((void*)&cal, data, (uint32_t*)length);
+        // get cached cal if prevoius attempt fails and persist flag is set
+        if ((ret != 0) && persist) {
+            cal.persist = true;
+            ret = my_data->acdb_get_audio_cal((void*)&cal, data, (uint32_t*)length);
+        }
+    }
+
+ERROR_RETURN:
+    return ret;
+}
+
+int platform_store_audio_cal(void* platform, int acdb_dev_id,
+       int acdb_device_type, int app_type, int topology_id,
+       int sample_rate, uint32_t module_id, uint32_t param_id,
+       void* data, int length)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+    acdb_audio_cal_cfg_t cal;
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+    if (!my_data) {
+        ret = -EINVAL;
+        goto ERROR_RETURN;
+    }
+
+    make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+        sample_rate, module_id, param_id, true);
+
+    if (my_data->acdb_set_audio_cal) {
+        ret = my_data->acdb_set_audio_cal((void*)&cal, data, (uint32_t)length);
+    }
+
+ERROR_RETURN:
+    return ret;
+}
+
+int platform_retrieve_audio_cal(void* platform, int acdb_dev_id,
+        int acdb_device_type, int app_type, int topology_id,
+        int sample_rate, uint32_t module_id, uint32_t param_id,
+        void* data, int* length)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+    acdb_audio_cal_cfg_t cal;
+    memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+    if (!my_data) {
+        ret = -EINVAL;
+        goto ERROR_RETURN;
+    }
+
+    make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+        sample_rate, module_id, param_id, true);
+
+    if (my_data->acdb_get_audio_cal) {
+        ret = my_data->acdb_get_audio_cal((void*)&cal, data, (uint32_t*)length);
+    }
+
+ERROR_RETURN:
+    return ret;
+}
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 019678a..c231843 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -39,6 +39,11 @@
     SOURCE_QUAD_MIC  = 0x8,            /* Target contains 4 mics */
 };
 
+enum {
+    SPKR_1,
+    SPKR_2
+};
+
 /*
  * Below are the devices for which is back end is same, SLIMBUS_0_RX.
  * All these devices are handled by the internal HW codec. We can
@@ -80,6 +85,7 @@
     SND_DEVICE_OUT_SPEAKER_VBAT,
     SND_DEVICE_OUT_LINE,
     SND_DEVICE_OUT_HEADPHONES,
+    SND_DEVICE_OUT_HEADPHONES_DSD,
     SND_DEVICE_OUT_HEADPHONES_44_1,
     SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
     SND_DEVICE_OUT_SPEAKER_AND_LINE,
@@ -88,12 +94,18 @@
     SND_DEVICE_OUT_VOICE_HANDSET,
     SND_DEVICE_OUT_VOICE_SPEAKER,
     SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT,
     SND_DEVICE_OUT_VOICE_HEADPHONES,
     SND_DEVICE_OUT_VOICE_LINE,
     SND_DEVICE_OUT_HDMI,
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+    SND_DEVICE_OUT_DISPLAY_PORT,
+    SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
     SND_DEVICE_OUT_BT_SCO,
     SND_DEVICE_OUT_BT_SCO_WB,
+    SND_DEVICE_OUT_BT_A2DP,
+    SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -110,10 +122,13 @@
     SND_DEVICE_OUT_ANC_HANDSET,
     SND_DEVICE_OUT_SPEAKER_PROTECTED,
     SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED,
     SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
     SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT,
     SND_DEVICE_OUT_SPEAKER_WSA,
     SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
+    SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA,
     SND_DEVICE_OUT_END,
 
     /*
@@ -167,6 +182,8 @@
     SND_DEVICE_IN_HANDSET_STEREO_DMIC,
     SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
     SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+    SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1,
+    SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2,
     SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,
     SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,
     SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,
@@ -177,6 +194,7 @@
     SND_DEVICE_IN_SPEAKER_QMIC_AEC,
     SND_DEVICE_IN_SPEAKER_QMIC_NS,
     SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS,
+    SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE,
     SND_DEVICE_IN_THREE_MIC,
     SND_DEVICE_IN_HANDSET_TMIC,
     SND_DEVICE_IN_UNPROCESSED_MIC,
@@ -189,19 +207,29 @@
     SND_DEVICE_MAX = SND_DEVICE_IN_END,
 
 };
-
+#define INPUT_SAMPLING_RATE_DSD64       2822400
+#define INPUT_SAMPLING_RATE_DSD128      5644800
 #define DEFAULT_OUTPUT_SAMPLING_RATE    48000
 #define OUTPUT_SAMPLING_RATE_44100      44100
-#define MAX_CODEC_TX_BACKENDS           1
+#define OUTPUT_SAMPLING_RATE_DSD64       176400
+#define OUTPUT_SAMPLING_RATE_DSD128      352800
 enum {
     DEFAULT_CODEC_BACKEND,
     SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+    DSD_NATIVE_BACKEND,
+    SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
     HEADPHONE_44_1_BACKEND,
     SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
     HEADPHONE_BACKEND,
     SLIMBUS_6_RX = HEADPHONE_BACKEND,
     HDMI_RX_BACKEND,
+    DISP_PORT_RX_BACKEND,
     USB_AUDIO_RX_BACKEND,
+    MAX_RX_CODEC_BACKENDS = USB_AUDIO_RX_BACKEND,
+    /* TX BE follows RX BE */
+    SLIMBUS_0_TX,
+    DEFAULT_CODEC_TX_BACKEND = SLIMBUS_0_TX,
+    USB_AUDIO_TX_BACKEND,
     MAX_CODEC_BACKENDS
 };
 
@@ -229,7 +257,14 @@
  * We should take care of returning proper size when AudioFlinger queries for
  * the buffer size of an input/output stream
  */
+
+/* for 384Khz output below period size corresponds to 20ms worth duration of buffer,
+ * current implementation can support buffer size of 40ms duration
+ * for 32b/384Khz/stereo output.
+ */
 #define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_DURATION 40 /* 40 milisecs */
+
 #define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
 #define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
 #define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
@@ -264,6 +299,7 @@
 #define AUDIO_RECORD_PCM_DEVICE 0
 #define MULTIMEDIA2_PCM_DEVICE 1
 #define MULTIMEDIA3_PCM_DEVICE 4
+#define MULTIMEDIA9_PCM_DEVICE 32
 #define FM_PLAYBACK_PCM_DEVICE 5
 #define FM_CAPTURE_PCM_DEVICE  6
 #define HFP_PCM_RX 5
@@ -444,7 +480,8 @@
 enum {
     LEGACY_PCM = 0,
     PASSTHROUGH,
-    PASSTHROUGH_CONVERT
+    PASSTHROUGH_CONVERT,
+    PASSTHROUGH_DSD
 };
 /*
  * ID for setting mute and lateny on the device side
@@ -460,4 +497,13 @@
     char device_name[100];
     char interface_name[100];
 };
+
+struct audio_backend_cfg {
+    unsigned int   sample_rate;
+    unsigned int   channels;
+    unsigned int   bit_width;
+    bool           passthrough_enabled;
+    audio_format_t format;
+};
+
 #endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 0bb73f3..7dcd1b6 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -23,10 +23,14 @@
 
 #define CODEC_BACKEND_DEFAULT_BIT_WIDTH 16
 #define CODEC_BACKEND_DEFAULT_SAMPLE_RATE 48000
+#define CODEC_BACKEND_DEFAULT_CHANNELS 2
+#define CODEC_BACKEND_DEFAULT_TX_CHANNELS 1
+
 
 enum {
     NATIVE_AUDIO_MODE_SRC = 1,
     NATIVE_AUDIO_MODE_TRUE_44_1,
+    NATIVE_AUDIO_MODE_MULTIPLE_44_1,
     NATIVE_AUDIO_MODE_INVALID
 };
 
@@ -36,6 +40,8 @@
     int na_mode;
 } native_audio_prop;
 
+enum card_status_t;
+
 void *platform_init(struct audio_device *adev);
 void platform_deinit(void *platform);
 const char *platform_get_snd_device_name(snd_device_t snd_device);
@@ -139,6 +145,7 @@
 bool platform_can_enable_spkr_prot_on_device(snd_device_t snd_device);
 int platform_get_spkr_prot_acdb_id(snd_device_t snd_device);
 int platform_get_spkr_prot_snd_device(snd_device_t snd_device);
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device);
 int platform_spkr_prot_is_wsa_analog_mode(void *adev);
 bool platform_can_split_snd_device(void *platform,
                                    snd_device_t snd_device,
@@ -151,4 +158,27 @@
                           bool enable,
                           char * str);
 bool platform_supports_true_32bit();
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device, snd_device_t cuurent_snd_device);
+bool platform_check_codec_dsd_support(void *platform);
+bool platform_check_codec_asrc_support(void *platform);
+int platform_get_backend_index(snd_device_t snd_device);
+int platform_get_ext_disp_type(void *platform);
+
+int platform_send_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int length, bool persist);
+
+int platform_get_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int* length, bool persist);
+
+int platform_store_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int length);
+
+int platform_retrieve_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+    int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+    void* data, int* length);
+
+unsigned char* platform_get_license(void* platform, int* size);
 #endif // AUDIO_PLATFORM_API_H
diff --git a/hal/voice.c b/hal/voice.c
index f86483e..b84c7b7 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -417,11 +417,13 @@
     int err = 0;
 
     adev->voice.mic_mute = state;
-    if (adev->mode == AUDIO_MODE_IN_CALL)
+    if (audio_extn_hfp_is_active(adev)) {
+        err = hfp_set_mic_mute(adev, state);
+    } else if (adev->mode == AUDIO_MODE_IN_CALL) {
         err = platform_set_mic_mute(adev->platform, state);
-    if (adev->mode == AUDIO_MODE_IN_COMMUNICATION)
+    } else if (adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
         err = voice_extn_compress_voip_set_mic_mute(adev, state);
-
+    }
     return err;
 }
 
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 7293485..3222e0b 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -244,6 +244,7 @@
 {
     int ret = 0;
     struct audio_usecase *uc_info;
+    struct listnode *node;
 
     ALOGD("%s: enter, out_stream_count=%d, in_stream_count=%d",
            __func__, voip_data.out_stream_count, voip_data.in_stream_count);
@@ -277,6 +278,12 @@
 
         list_remove(&uc_info->list);
         free(uc_info);
+
+        // restore device for other active usecases
+        list_for_each(node, &adev->usecase_list) {
+            uc_info = node_to_item(node, struct audio_usecase, list);
+            select_devices(adev, uc_info->id);
+        }
     } else
         ALOGV("%s: NO-OP because out_stream_count=%d, in_stream_count=%d",
                __func__, voip_data.out_stream_count, voip_data.in_stream_count);
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index b89c82c..d06929c 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -435,6 +435,48 @@
     return BAD_VALUE;
 }
 
+bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(routing_strategy strategy)
+{
+    if (strategy == STRATEGY_MEDIA) {
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<SwAudioOutputDescriptor> newOutputDesc = mOutputs.valueAt(i);
+            if (newOutputDesc->mFormat == AUDIO_FORMAT_DSD)
+                return false;
+        }
+    }
+    return true;
+}
+
+void AudioPolicyManagerCustom::checkOutputForStrategy(routing_strategy strategy)
+{
+    audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+    audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mOutputs);
+    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+    // also take into account external policy-related changes: add all outputs which are
+    // associated with policies in the "before" and "after" output vectors
+    ALOGV("checkOutputForStrategy(): policy related outputs");
+    for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
+        const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+        if (desc != 0 && desc->mPolicyMix != NULL) {
+            srcOutputs.add(desc->mIoHandle);
+            ALOGV(" previous outputs: adding %d", desc->mIoHandle);
+        }
+    }
+    for (size_t i = 0 ; i < mOutputs.size() ; i++) {
+        const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+        if (desc != 0 && desc->mPolicyMix != NULL) {
+            dstOutputs.add(desc->mIoHandle);
+            ALOGV(" new outputs: adding %d", desc->mIoHandle);
+        }
+    }
+
+    if (!vectorsEqual(srcOutputs,dstOutputs) && isInvalidationOfMusicStreamNeeded(strategy)) {
+        AudioPolicyManager::checkOutputForStrategy(strategy);
+    }
+}
+
 // This function checks for the parameters which can be offloaded.
 // This can be enhanced depending on the capability of the DSP and policy
 // of the system.
@@ -464,6 +506,11 @@
          }
     }
 #endif
+    if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
+        isInCall() &&  (offloadInfo.format == AUDIO_FORMAT_DSD)) {
+        ALOGD("blocking DSD compress offload on call mode");
+        return false;
+    }
 #ifdef RECORD_PLAY_CONCURRENCY
     char recConcPropValue[PROPERTY_VALUE_MAX];
     bool prop_rec_play_enabled = false;
@@ -511,6 +558,14 @@
             ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
         return false;
     }
+
+    if ((((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.bit_rate > MAX_BITRATE_WMA)) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_PRO)) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_LOSSLESS))){
+        //Safely choose the min bitrate as threshold and leave the restriction to NT decoder as we can't distinguish wma pro and wma lossless here.
+        ALOGD("offload disabled for WMA/WMA_PRO/WMA_LOSSLESS clips with bit rate over maximum supported value");
+        return false;
+    }
 #endif
     //TODO: enable audio offloading with video when ready
     const bool allowOffloadWithVideo =
@@ -549,6 +604,7 @@
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DSD) ||
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
             return false;
 #endif
@@ -563,16 +619,7 @@
     if (mEffects.isNonOffloadableEffectEnabled()) {
         return false;
     }
-    // Check for soundcard status
-    String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
-    String8("SND_CARD_STATUS"));
-    AudioParameter result = AudioParameter(valueStr);
-    int isonline = 0;
-    if ((result.getInt(String8("SND_CARD_STATUS"), isonline) == NO_ERROR)
-           && !isonline) {
-         ALOGD("copl: soundcard is offline rejecting offload request");
-         return false;
-    }
+
     // See if there is a profile to support this.
     // AUDIO_DEVICE_NONE
     sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
@@ -846,6 +893,26 @@
     }
 
 #endif
+
+    sp<SwAudioOutputDescriptor> outputDesc = NULL;
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        outputDesc = mOutputs.valueAt(i);
+        if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+            ALOGD("voice_conc:ouput desc / profile is NULL");
+            continue;
+        }
+
+        if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
+            (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+            (outputDesc->mFormat == AUDIO_FORMAT_DSD)) {
+            ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output");
+            closeOutput(mOutputs.keyAt(i));
+            // call invalidate for music, so that DSD compress will fallback to deep-buffer.
+            mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+        }
+
+    }
+
 #ifdef RECORD_PLAY_CONCURRENCY
     char recConcPropValue[PROPERTY_VALUE_MAX];
     bool prop_rec_play_enabled = false;
@@ -1073,6 +1140,7 @@
                         outputDesc->sharesHwModuleWith(desc) &&
                         (newDevice != desc->device())) {
                         audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
+                        bool force = desc->device() != dev;
                         uint32_t delayMs;
                         if (dev == prevDevice) {
                             delayMs = 0;
@@ -1081,7 +1149,7 @@
                         }
                         setOutputDevice(desc,
                                     dev,
-                                    true,
+                                    force,
                                     delayMs);
                 }
             }
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index dfda1c9..00da599 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -39,6 +39,10 @@
 #ifndef AUDIO_EXTN_AFE_PROXY_ENABLED
 #define AUDIO_DEVICE_OUT_PROXY 0x1000000
 #endif
+
+#define MAX_BITRATE_WMA          384000
+#define MAX_BITRATE_WMA_PRO      1536000
+#define MAX_BITRATE_WMA_LOSSLESS 1152000
 // ----------------------------------------------------------------------------
 
 class AudioPolicyManagerCustom: public AudioPolicyManager
@@ -92,6 +96,14 @@
         // see getDeviceForStrategy() for the use of fromCache parameter
         audio_devices_t getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
                                            bool fromCache);
+
+        // avoid invalidation for active music stream on  previous outputs
+        // which is supported on the new device.
+        bool isInvalidationOfMusicStreamNeeded(routing_strategy strategy);
+
+        // Must be called before updateDevicesAndOutputs()
+        void checkOutputForStrategy(routing_strategy strategy);
+
         // returns true if given output is direct output
         bool isDirectOutput(audio_io_handle_t output);
 
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index 464bc0d..fd5ee8c 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -852,8 +852,9 @@
         if (pCmdData == NULL ||
             cmdSize < (int)(sizeof(effect_param_t) + sizeof(uint32_t)) ||
             pReplyData == NULL ||
-            *replySize < (int)(sizeof(effect_param_t) + sizeof(uint32_t) +
-                               sizeof(uint16_t))) {
+            *replySize < (int)(sizeof(effect_param_t) + sizeof(uint32_t) + sizeof(uint16_t)) ||
+            // constrain memcpy below
+            ((effect_param_t *)pCmdData)->psize > *replySize - sizeof(effect_param_t)) {
             status = -EINVAL;
             ALOGW("EFFECT_CMD_GET_PARAM invalid command cmdSize %d *replySize %d",
                   cmdSize, *replySize);
diff --git a/qahw_api/Android.mk b/qahw_api/Android.mk
new file mode 100644
index 0000000..8c99c5b
--- /dev/null
+++ b/qahw_api/Android.mk
@@ -0,0 +1,32 @@
+ifeq ($(strip $(BOARD_SUPPORTS_QAHW)),true)
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+libqahw-inc := $(LOCAL_PATH)/inc
+
+LOCAL_MODULE := libqahw
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_OWNER := qti
+LOCAL_C_INCLUDES   := $(libqahw-inc)
+
+LOCAL_SRC_FILES := \
+    src/qahw.c
+
+LOCAL_SHARED_LIBRARIES := \
+    liblog \
+    libcutils \
+    libhardware
+
+LOCAL_COPY_HEADERS_TO   := mm-audio/qahw_api/inc
+LOCAL_COPY_HEADERS      := inc/qahw_api.h
+LOCAL_COPY_HEADERS      += inc/qahw_defs.h
+
+LOCAL_PRELINK_MODULE    := false
+
+include $(BUILD_SHARED_LIBRARY)
+
+#test app compilation
+include $(LOCAL_PATH)/test/Android.mk
+endif
diff --git a/qahw_api/inc/qahw_api.h b/qahw_api/inc/qahw_api.h
new file mode 100644
index 0000000..17f6f5f
--- /dev/null
+++ b/qahw_api/inc/qahw_api.h
@@ -0,0 +1,420 @@
+/*
+ * Copyright (c) 2016, 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_HAL_API_H
+#define QTI_AUDIO_HAL_API_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;
+
+/**************************************/
+/* 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(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(qahw_stream_handle_t *out_handle);
+
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_out_get_sample_rate(const qahw_stream_handle_t *stream);
+
+/*
+ *  use set_parameters with key QAHW_PARAMETER_STREAM_SAMPLING_RATE
+ */
+int qahw_out_set_sample_rate(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(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(const qahw_stream_handle_t *stream);
+
+/*
+ * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
+ */
+audio_format_t qahw_out_get_format(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(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(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(const qahw_stream_handle_t *stream,
+                               const char *keys);
+
+/*
+ * Return the audio hardware driver estimated latency in milliseconds.
+ */
+uint32_t qahw_out_get_latency(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(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(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(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(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(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(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(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(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(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(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(qahw_stream_handle_t *in_handle);
+
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_in_get_sample_rate(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(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(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(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(const qahw_stream_handle_t *in_handle);
+
+/*
+ * currently unused - use set_parameters with key
+ *     QAHW_PARAMETER_STREAM_FORMAT
+ */
+int qahw_in_set_format(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(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(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(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(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(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(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(const char *hw_module_id);
+
+int qahw_unload_module(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(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(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(qahw_module_handle_t *hw_module, audio_mode_t mode);
+
+/* set/get global audio parameters */
+int qahw_set_parameters(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(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(const qahw_module_handle_t *hw_module,
+                                  const struct audio_config *config);
+
+/*returns current QTI HAL version */
+int qahw_get_version();
+
+__END_DECLS
+
+#endif  // QTI_AUDIO_HAL_API_H
diff --git a/qahw_api/inc/qahw_defs.h b/qahw_api/inc/qahw_defs.h
new file mode 100644
index 0000000..4441435
--- /dev/null
+++ b/qahw_api/inc/qahw_defs.h
@@ -0,0 +1,215 @@
+/*
+ * Copyright (c) 2016, 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"
+
+/* Query fm volume */
+#define QAHW_PARAMETER_KEY_FM_VOLUME "fm_volume"
+
+/* Query if a2dp  is supported */
+#define QAHW_PARAMETER_KEY_HANDLE_A2DP_DEVICE "isA2dpDeviceSupported"
+
+/* 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_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;
+
+__END_DECLS
+
+#endif  // QTI_AUDIO_HAL_DEFS_H
+
diff --git a/qahw_api/src/qahw.c b/qahw_api/src/qahw.c
new file mode 100644
index 0000000..06bcee1
--- /dev/null
+++ b/qahw_api/src/qahw.c
@@ -0,0 +1,1513 @@
+/*
+* Copyright (c) 2016, The Linux Foundation. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions are
+* met:
+*     * Redistributions of source code must retain the above copyright
+*       notice, this list of conditions and the following disclaimer.
+*     * Redistributions in binary form must reproduce the above
+*       copyright notice, this list of conditions and the following
+*       disclaimer in the documentation and/or other materials provided
+*       with the distribution.
+*     * Neither the name of The Linux Foundation nor the names of its
+*       contributors may be used to endorse or promote products derived
+*       from this software without specific prior written permission.
+*
+* THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+* ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+* BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+* OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#define LOG_TAG "qahw"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <utils/Log.h>
+#include <stdlib.h>
+#include <cutils/list.h>
+
+#include <hardware/audio.h>
+#include "qahw_api.h"
+
+#define NO_ERROR 0
+#define MAX_MODULE_NAME_LENGTH  100
+
+/*
+ * The current HAL API version.
+ */
+#define QAHW_MODULE_API_VERSION_CURRENT QAHW_MODULE_API_VERSION_0_0
+
+typedef struct {
+    audio_hw_device_t *audio_device;
+    char module_name[MAX_MODULE_NAME_LENGTH];
+    struct listnode module_list;
+    struct listnode in_list;
+    struct listnode out_list;
+    pthread_mutex_t lock;
+    uint32_t ref_count;
+} qahw_module_t;
+
+typedef struct {
+    qahw_module_t *module;
+    struct listnode module_list;
+    pthread_mutex_t lock;
+} qahw_module_instances_t;
+
+typedef struct {
+    audio_stream_out_t *stream;
+    qahw_module_t *module;
+    struct listnode list;
+    pthread_mutex_t lock;
+} qahw_stream_out_t;
+
+typedef struct {
+    audio_stream_in_t *stream;
+    qahw_module_t *module;
+    struct listnode list;
+    pthread_mutex_t lock;
+} qahw_stream_in_t;
+
+typedef enum {
+    STREAM_DIR_IN,
+    STREAM_DIR_OUT,
+} qahw_stream_direction_t;
+
+static struct listnode qahw_module_list;
+static int qahw_list_count;
+static pthread_mutex_t qahw_module_init_lock = PTHREAD_MUTEX_INITIALIZER;
+
+/** Start of internal functions */
+/******************************************************************************/
+
+/* call this function without anylock held */
+static bool is_valid_qahw_stream(void *qahw_stream,
+                                 qahw_stream_direction_t dir)
+{
+
+    int is_valid = false;
+    struct listnode *module_node = NULL;
+    struct listnode *stream_node = NULL;
+    struct listnode *list_node = NULL;
+    void  *stream = NULL;
+    qahw_module_t *qahw_module = NULL;
+
+    if (qahw_stream == NULL) {
+        ALOGE("%s:: Invalid stream", __func__);
+        goto exit;
+    }
+
+    if ((dir != STREAM_DIR_OUT) && (dir != STREAM_DIR_IN)) {
+        ALOGE("%s:: Invalid stream direction %d", __func__, dir);
+        goto exit;
+    }
+
+    /* go through all the modules and check for valid stream */
+    pthread_mutex_lock(&qahw_module_init_lock);
+    list_for_each(module_node, &qahw_module_list) {
+        qahw_module = node_to_item(module_node, qahw_module_t, module_list);
+        pthread_mutex_lock(&qahw_module->lock);
+        if(dir == STREAM_DIR_OUT)
+            list_node = &qahw_module->out_list;
+        else
+            list_node = &qahw_module->in_list;
+        list_for_each(stream_node, list_node) {
+            if(dir == STREAM_DIR_OUT)
+                stream = (void *)node_to_item(stream_node,
+                                              qahw_stream_out_t,
+                                              list);
+            else
+                stream = (void *)node_to_item(stream_node,
+                                              qahw_stream_in_t,
+                                              list);
+            if(stream == qahw_stream) {
+                is_valid = true;
+                break;
+            }
+        }
+        pthread_mutex_unlock(&qahw_module->lock);
+        if(is_valid)
+            break;
+    }
+    pthread_mutex_unlock(&qahw_module_init_lock);
+
+exit:
+    return is_valid;
+}
+
+/* call this fucntion with ahw_module_init_lock held*/
+static qahw_module_t* get_qahw_module_by_ptr(qahw_module_t *qahw_module)
+{
+    struct listnode *node = NULL;
+    qahw_module_t *module = NULL, *module_temp = NULL;
+
+    if (qahw_module == NULL)
+        goto exit;
+
+    list_for_each(node, &qahw_module_list) {
+        module_temp = node_to_item(node, qahw_module_t, module_list);
+        if (module_temp == qahw_module) {
+            module = module_temp;
+            break;
+        }
+    }
+exit:
+    return module;
+}
+
+/* call this function with qahw_module_init_lock held*/
+static qahw_module_t* get_qahw_module_by_name(const char *qahw_name)
+{
+    struct listnode *node = NULL;
+    qahw_module_t *module = NULL, *module_temp = NULL;
+
+    if (qahw_name == NULL)
+        goto exit;
+
+    list_for_each(node, &qahw_module_list) {
+        module_temp = node_to_item(node, qahw_module_t, module_list);
+        if(!strncmp(qahw_name, module_temp->module_name, MAX_MODULE_NAME_LENGTH)) {
+            module = module_temp;
+            break;
+        }
+    }
+exit:
+    return module;
+}
+/* End of of internal functions */
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_out_get_sample_rate(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)) {
+        ALOGV("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.get_sample_rate)
+        rate = out->common.get_sample_rate(&out->common);
+    else
+        ALOGW("%s not supported", __func__);
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+     return rate;
+}
+
+/*
+ * 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)
+{
+    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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.set_sample_rate) {
+        rc = out->common.set_sample_rate(&out->common, rate);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+exit:
+    return rc;
+}
+
+size_t qahw_out_get_buffer_size(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.get_buffer_size) {
+        buf_size = out->common.get_buffer_size(&out->common);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return buf_size;
+}
+
+audio_channel_mask_t qahw_out_get_channels(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.get_channels) {
+        ch_mask = out->common.get_channels(&out->common);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return ch_mask;
+}
+
+audio_format_t qahw_out_get_format(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.get_format) {
+        format = out->common.get_format(&out->common);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return format;
+}
+
+int qahw_out_standby(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.standby) {
+        rc = out->common.standby(&out->common);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+int qahw_out_set_parameters(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        rc = -EINVAL;
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.set_parameters) {
+        rc = out->common.set_parameters(&out->common, kv_pairs);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+char *qahw_out_get_parameters(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->common.get_parameters) {
+        str_param = out->common.get_parameters(&out->common, keys);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return str_param;
+}
+
+uint32_t qahw_out_get_latency(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->get_latency) {
+        latency = out->get_latency(out);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return latency;
+}
+
+int qahw_out_set_volume(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->set_volume) {
+        rc = out->set_volume(out, left, right);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+   return rc;
+}
+
+ssize_t qahw_out_write(qahw_stream_handle_t *out_handle,
+        qahw_out_buffer_t *out_buf)
+{
+    int rc = -EINVAL;
+    qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+    audio_stream_out_t *out = NULL;
+
+    if ((out_buf == NULL) || (out_buf->buffer == NULL)) {
+        ALOGE("%s::Invalid meta data %p", __func__, out_buf);
+        goto exit;
+    }
+
+    if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    /*TBD:: validate other meta data parameters */
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->write) {
+        rc = out->write(out, out_buf->buffer, out_buf->bytes);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+exit:
+    return rc;
+}
+
+int qahw_out_get_render_position(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->get_render_position) {
+        rc = out->get_render_position(out, dsp_frames);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+exit:
+    return rc;
+}
+
+int qahw_out_set_callback(qahw_stream_handle_t *out_handle,
+                          qahw_stream_callback_t callback,
+                          void *cookie)
+{
+    /*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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->set_callback) {
+        rc = out->set_callback(out, (stream_callback_t)callback, cookie);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+int qahw_out_pause(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->pause) {
+        rc = out->pause(out);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+int qahw_out_resume(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->resume) {
+        rc = out->resume(out);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+int qahw_out_drain(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->drain) {
+        rc = out->drain(out,(audio_drain_type_t)type);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+int qahw_out_flush(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->flush) {
+        rc = out->flush(out);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+int qahw_out_get_presentation_position(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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    out = qahw_stream_out->stream;
+    if (out->get_presentation_position) {
+        rc = out->get_presentation_position(out, frames, timestamp);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+    return rc;
+}
+
+/* Input stream specific APIs */
+uint32_t qahw_in_get_sample_rate(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.get_sample_rate) {
+        rate = in->common.get_sample_rate(&in->common);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return rate;
+}
+
+/*
+ * 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 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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.set_sample_rate) {
+        rc = in->common.set_sample_rate(&in->common, rate);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return rc;
+}
+
+size_t qahw_in_get_buffer_size(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.get_sample_rate) {
+        buf_size = in->common.get_buffer_size(&in->common);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return buf_size;
+}
+
+
+audio_channel_mask_t qahw_in_get_channels(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.get_channels) {
+        ch_mask = in->common.get_channels(&in->common);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return ch_mask;
+}
+
+audio_format_t qahw_in_get_format(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.get_format) {
+        format = in->common.get_format(&in->common);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return format;
+}
+
+/*
+ * 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 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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.set_format) {
+        rc = in->common.set_format(&in->common, format);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return rc;
+}
+
+int qahw_in_standby(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.standby) {
+        rc = in->common.standby(&in->common);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return rc;
+}
+
+/*
+ * 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(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.set_parameters) {
+        rc = in->common.set_parameters(&in->common, kv_pairs);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+exit:
+    return rc;
+}
+
+/*
+ * 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,
+                              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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->common.get_parameters) {
+        str_param = in->common.get_parameters(&in->common, keys);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return str_param;
+}
+
+/*
+ * Read audio buffer in from audio driver. Returns number of bytes read, or a
+ *  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,
+                     qahw_in_buffer_t *in_buf)
+{
+    int rc = -EINVAL;
+    qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+    audio_stream_in_t *in = NULL;
+
+    if ((in_buf == NULL) || (in_buf->buffer == NULL)) {
+        ALOGE("%s::Invalid meta data %p", __func__, in_buf);
+        goto exit;
+    }
+
+    if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    /*TBD:: call HAL timestamp read API*/
+    if (in->read) {
+        rc = in->read(in, in_buf->buffer, in_buf->bytes);
+        in_buf->offset = 0;
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return rc;
+}
+
+/*
+ * 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(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    in = qahw_stream_in->stream;
+    if (in->get_input_frames_lost) {
+        rc = in->get_input_frames_lost(in);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+    return rc;
+}
+
+/*
+ * 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(const qahw_stream_handle_t *in_handle __unused,
+                                 int64_t *frames __unused, int64_t *time __unused)
+{
+    /*TBD:: do we need this*/
+    return -ENOSYS;
+}
+
+/*
+ * 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 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);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    if (qahw_module->audio_device->init_check) {
+        rc = qahw_module->audio_device->init_check(qahw_module->audio_device);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+    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 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);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    if (qahw_module->audio_device->set_voice_volume) {
+        rc = qahw_module->audio_device->set_voice_volume(qahw_module->audio_device,
+                                                         volume);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+    return rc;
+}
+
+/*
+ * 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(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);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    if (qahw_module->audio_device->set_mode) {
+        rc = qahw_module->audio_device->set_mode(qahw_module->audio_device,
+                                                 mode);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+    return rc;
+}
+
+/* set/get global audio parameters */
+int qahw_set_parameters(qahw_module_handle_t *hw_module, const char *kv_pairs)
+{
+    int rc = -EINVAL;
+    qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+    qahw_module_t *qahw_module_temp;
+    audio_hw_device_t *audio_device;
+
+    pthread_mutex_lock(&qahw_module_init_lock);
+    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    audio_device = qahw_module->audio_device;
+    if (qahw_module->audio_device->set_parameters) {
+        rc = audio_device->set_parameters(qahw_module->audio_device, kv_pairs);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+    return rc;
+}
+
+/*
+ * 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,
+                           const char *keys)
+{
+    char *str_param = NULL;
+    qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+    qahw_module_t *qahw_module_temp;
+    audio_hw_device_t *audio_device;
+
+    pthread_mutex_lock(&qahw_module_init_lock);
+    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    audio_device = qahw_module->audio_device;
+    if (qahw_module->audio_device->get_parameters) {
+        str_param = audio_device->get_parameters(qahw_module->audio_device, keys);
+    } else {
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+    return str_param;
+}
+
+/* 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(const qahw_module_handle_t *hw_module,
+                                  const struct audio_config *config)
+{
+    size_t rc = 0;
+    qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+    qahw_module_t *qahw_module_temp;
+    audio_hw_device_t *audio_device;
+
+    pthread_mutex_lock(&qahw_module_init_lock);
+    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        goto exit;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    audio_device = qahw_module->audio_device;
+    if (qahw_module->audio_device->get_input_buffer_size) {
+        rc = audio_device->get_input_buffer_size(qahw_module->audio_device,
+                                                 config);
+    } else {
+        rc = -ENOSYS;
+        ALOGW("%s not supported", __func__);
+    }
+    pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+    return rc;
+}
+
+/*
+ * 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(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 rc = -EINVAL;
+    qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+    qahw_module_t *qahw_module_temp = NULL;
+    audio_hw_device_t *audio_device = NULL;
+    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);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        return rc;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    audio_device = qahw_module->audio_device;
+    qahw_stream_out = (qahw_stream_out_t *)calloc(1, sizeof(qahw_stream_out_t));
+    if (qahw_stream_out == NULL) {
+        ALOGE("%s:: calloc failed for out stream_out_t",__func__);
+        rc = -ENOMEM;
+        goto exit;
+    }
+
+    rc = audio_device->open_output_stream(audio_device,
+                                          handle,
+                                          devices,
+                                          flags,
+                                          config,
+                                          &qahw_stream_out->stream,
+                                          address);
+    if (rc) {
+        ALOGE("%s::open output stream failed %d",__func__, rc);
+        free(qahw_stream_out);
+    } else {
+        qahw_stream_out->module = hw_module;
+        *out_handle = (void *)qahw_stream_out;
+        pthread_mutex_init(&qahw_stream_out->lock, (const pthread_mutexattr_t *)NULL);
+        list_add_tail(&qahw_module->out_list, &qahw_stream_out->list);
+    }
+
+exit:
+    pthread_mutex_unlock(&qahw_module->lock);
+    return rc;
+}
+
+int qahw_close_output_stream(qahw_stream_handle_t *out_handle)
+{
+
+    int rc = 0;
+    qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+    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)) {
+        ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+        rc = -EINVAL;
+        goto exit;
+    }
+
+    ALOGV("%s::calling device close_output_stream %p", __func__, out_handle);
+    pthread_mutex_lock(&qahw_stream_out->lock);
+    qahw_module = qahw_stream_out->module;
+    audio_device = qahw_module->audio_device;
+    audio_device->close_output_stream(audio_device,
+                                      qahw_stream_out->stream);
+
+    pthread_mutex_lock(&qahw_module->lock);
+    list_remove(&qahw_stream_out->list);
+    pthread_mutex_unlock(&qahw_module->lock);
+
+    pthread_mutex_unlock(&qahw_stream_out->lock);
+
+    pthread_mutex_destroy(&qahw_stream_out->lock);
+    free(qahw_stream_out);
+
+exit:
+    return rc;
+}
+
+/* This method creates and opens the audio hardware input stream */
+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)
+{
+    int rc = -EINVAL;
+    qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+    qahw_module_t *qahw_module_temp = NULL;
+    audio_hw_device_t *audio_device = NULL;
+    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);
+    pthread_mutex_unlock(&qahw_module_init_lock);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        return rc;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    audio_device = qahw_module->audio_device;
+    qahw_stream_in = (qahw_stream_in_t *)calloc(1, sizeof(qahw_stream_in_t));
+    if (qahw_stream_in == NULL) {
+        ALOGE("%s:: calloc failed for in stream_in_t",__func__);
+        rc = -ENOMEM;
+        goto exit;
+    }
+
+    rc = audio_device->open_input_stream(audio_device,
+                                          handle,
+                                          devices,
+                                          config,
+                                          &qahw_stream_in->stream,
+                                          flags,
+                                          address,
+                                          source);
+    if (rc) {
+        ALOGE("%s::open input stream failed %d",__func__, rc);
+        free(qahw_stream_in);
+    } else {
+        qahw_stream_in->module = hw_module;
+        *in_handle = (void *)qahw_stream_in;
+        pthread_mutex_init(&qahw_stream_in->lock, (const pthread_mutexattr_t *)NULL);
+        list_add_tail(&qahw_module->in_list, &qahw_stream_in->list);
+    }
+
+exit:
+    pthread_mutex_unlock(&qahw_module->lock);
+    return rc;
+}
+
+int qahw_close_input_stream(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)) {
+        ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+        rc = -EINVAL;
+        goto exit;
+    }
+
+    ALOGV("%s:: calling device close_input_stream %p", __func__, in_handle);
+    pthread_mutex_lock(&qahw_stream_in->lock);
+    qahw_module = qahw_stream_in->module;
+    audio_device = qahw_module->audio_device;
+    audio_device->close_input_stream(audio_device,
+                                     qahw_stream_in->stream);
+
+    pthread_mutex_lock(&qahw_module->lock);
+    list_remove(&qahw_stream_in->list);
+    pthread_mutex_unlock(&qahw_module->lock);
+
+    pthread_mutex_unlock(&qahw_stream_in->lock);
+
+    pthread_mutex_destroy(&qahw_stream_in->lock);
+    free(qahw_stream_in);
+
+exit:
+    return rc;
+}
+
+/*returns current QTI HAL verison */
+int qahw_get_version() {
+    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)
+{
+    int rc = -EINVAL;
+    qahw_module_handle_t *qahw_mod_handle = NULL;
+    qahw_module_t *qahw_module = NULL;
+    char *ahal_name = NULL;
+    const hw_module_t* module = NULL;
+    audio_hw_device_t* audio_device = NULL;
+
+    if (hw_module_id == NULL) {
+        ALOGE("%s::module id is NULL",__func__);
+        goto exit;
+    }
+
+    if (!strcmp(hw_module_id, QAHW_MODULE_ID_PRIMARY)) {
+        ahal_name = "primary";
+    } else if (!strcmp(hw_module_id, QAHW_MODULE_ID_A2DP)) {
+        ahal_name = "a2dp";
+    } else if (!strcmp(hw_module_id, QAHW_MODULE_ID_USB)) {
+        ahal_name = "usb";
+    } else {
+        ALOGE("%s::Invalid Module id %s", __func__, hw_module_id);
+        goto exit;
+    }
+
+    /* 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);
+        if(qahw_module != NULL) {
+            qahw_mod_handle = (void *)qahw_module;
+            pthread_mutex_lock(&qahw_module->lock);
+            qahw_module->ref_count++;
+            pthread_mutex_unlock(&qahw_module->lock);
+            goto error_exit;
+        }
+    }
+
+    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, ahal_name, &module);
+    if(rc) {
+        ALOGE("%s::HAL Loading failed %d", __func__, rc);
+        goto error_exit;
+    }
+
+    rc = audio_hw_device_open(module, &audio_device);
+    if(rc) {
+        ALOGE("%s::HAL Device open failed %d", __func__, rc);
+        goto error_exit;
+    }
+
+    qahw_module = (qahw_module_t *)calloc(1, sizeof(qahw_module_t));
+    if(qahw_module == NULL) {
+        ALOGE("%s::calloc failed", __func__);
+        audio_hw_device_close(audio_device);
+        goto error_exit;
+    }
+    ALOGD("%s::Loaded HAL %s module %p", __func__, ahal_name, qahw_module);
+
+    if (!qahw_list_count)
+        list_init(&qahw_module_list);
+    qahw_list_count++;
+
+    pthread_mutex_init(&qahw_module->lock, (const pthread_mutexattr_t *) NULL);
+    pthread_mutex_lock(&qahw_module->lock);
+    qahw_module->ref_count++;
+    pthread_mutex_unlock(&qahw_module->lock);
+
+    list_init(&qahw_module->out_list);
+    list_init(&qahw_module->in_list);
+
+    /* update qahw_module */
+    qahw_module->audio_device = audio_device;
+    strlcpy(&qahw_module->module_name[0], hw_module_id, MAX_MODULE_NAME_LENGTH);
+
+    qahw_mod_handle = (void *)qahw_module;
+
+    /* Add module list to global module list */
+    list_add_tail(&qahw_module_list, &qahw_module->module_list);
+
+
+error_exit:
+    pthread_mutex_unlock(&qahw_module_init_lock);
+
+exit:
+    return qahw_mod_handle;
+}
+
+int qahw_unload_module(qahw_module_handle_t *hw_module)
+{
+    int rc = -EINVAL;
+    bool is_empty = false;
+    qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+    qahw_module_t *qahw_module_temp = NULL;
+
+    /* close HW device if its valid and all the streams on
+     * it is closed
+    */
+    pthread_mutex_lock(&qahw_module_init_lock);
+    qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+    if (qahw_module_temp == NULL) {
+        ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+        goto error_exit;
+    }
+
+    pthread_mutex_lock(&qahw_module->lock);
+    qahw_module->ref_count--;
+    if (qahw_module->ref_count > 0) {
+        rc = 0;
+        ALOGE("%s:: skipping module unload of %p count %d", __func__,
+              qahw_module,
+              qahw_module->ref_count);
+        pthread_mutex_unlock(&qahw_module->lock);
+        goto error_exit;
+    }
+
+    is_empty = (list_empty(&qahw_module->out_list) &&
+             list_empty(&qahw_module->in_list));
+    if (is_empty) {
+        rc = audio_hw_device_close(qahw_module->audio_device);
+        if(rc) {
+            ALOGE("%s::HAL Device close failed Error %d Module %p",__func__,
+                    rc, qahw_module);
+            rc = 0;
+        }
+        qahw_list_count--;
+        list_remove(&qahw_module->module_list);
+        pthread_mutex_unlock(&qahw_module->lock);
+        pthread_mutex_destroy(&qahw_module->lock);
+        free(qahw_module);
+    } else {
+        pthread_mutex_unlock(&qahw_module->lock);
+        ALOGE("%s::failed as all the streams on this module"
+               "is not closed", __func__);
+        rc = -EINVAL;
+    }
+
+error_exit:
+    pthread_mutex_unlock(&qahw_module_init_lock);
+
+exit:
+    return rc;
+}
+
+__END_DECLS
diff --git a/qahw_api/test/Android.mk b/qahw_api/test/Android.mk
new file mode 100644
index 0000000..1688069
--- /dev/null
+++ b/qahw_api/test/Android.mk
@@ -0,0 +1,40 @@
+LOCAL_PATH := $(call my-dir)
+
+# audio_hal_playback_test
+# ==============================================================================
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES := qahw_playback_test.c
+LOCAL_MODULE := hal_play_test
+
+hal-play-inc     = $(TARGET_OUT_HEADERS)/mm-audio/qahw_api/inc
+
+LOCAL_CFLAGS += -Wall -Werror -Wno-sign-compare
+
+LOCAL_SHARED_LIBRARIES := \
+    libaudioutils\
+    libqahw \
+    libutils
+
+LOCAL_32_BIT_ONLY := true
+
+LOCAL_C_INCLUDES += $(hal-play-inc)
+
+include $(BUILD_EXECUTABLE)
+
+# audio_hal_multi_record_test
+# ==============================================================================
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES := qahw_multi_record_test.c
+LOCAL_MODULE := hal_rec_test
+LOCAL_CFLAGS += -Wall -Werror -Wno-sign-compare
+LOCAL_SHARED_LIBRARIES := \
+    libaudioutils \
+    libqahw \
+    libutils
+
+LOCAL_32_BIT_ONLY := true
+
+hal-rec-inc     = $(TARGET_OUT_HEADERS)/mm-audio/qahw_api/inc
+
+LOCAL_C_INCLUDES += $(hal-rec-inc)
+include $(BUILD_EXECUTABLE)
diff --git a/qahw_api/test/qahw_multi_record_test.c b/qahw_api/test/qahw_multi_record_test.c
new file mode 100644
index 0000000..07dede7
--- /dev/null
+++ b/qahw_api/test/qahw_multi_record_test.c
@@ -0,0 +1,488 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2015 The Android Open Source Project *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/* Test app to record multiple audio sessions at the HAL layer */
+
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include <pthread.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include "qahw_api.h"
+#include "qahw_defs.h"
+
+struct audio_config_params {
+    qahw_module_handle_t *qahw_mod_handle;
+    audio_io_handle_t handle;
+    audio_devices_t input_device;
+    audio_config_t config;
+    audio_input_flags_t flags;
+    const char* kStreamName ;
+    audio_source_t kInputSource;
+    char output_filename[256];
+    double loopTime;
+    char profile[50];
+};
+
+#define SOUNDFOCUS_PARAMS "SoundFocus.start_angles;SoundFocus.enable_sectors;" \
+                          "SoundFocus.gain_step"
+#define SOURCETRACK_PARAMS "SourceTrack.vad;SourceTrack.doa_speech;SourceTrack.doa_noise;"\
+                           "SourceTrack.polar_activity;ssr.noise_level;ssr.noise_level_after_ns"
+int sourcetrack_done = 0;
+static pthread_mutex_t glock;
+pthread_cond_t gcond;
+int tests_running;
+bool gerror;
+
+void *read_sourcetrack_data(void* data)
+{
+    char kvpair_soundfocus[200] = SOUNDFOCUS_PARAMS;
+    char kvpair_sourcetrack[200] = SOURCETRACK_PARAMS;
+    char *string = NULL;
+    char *token = NULL;
+    char choice = '\0';
+    int i =0;
+    qahw_module_handle_t *qawh_module_handle =
+                (qahw_module_handle_t *)data;
+
+    while (1) {
+        printf("\nGet SoundFocus Params from app");
+        string = qahw_get_parameters(qawh_module_handle, kvpair_soundfocus);
+        if (!string) {
+            printf("Error.Failed Get SoundFocus Params\n");
+        } else {
+            token = strtok (string , "=");
+            while (token) {
+                if (*token == 'S') {
+                    choice = *(token + 11);
+                    token = strtok (NULL,",;");
+                    i=0;
+                }
+                switch (choice) {
+                    case 'g':
+                        printf ("\nSoundFocus.gain_step=%s",token);
+                        break;
+                    case 'e':
+                        printf ("\nSoundFocus.enable_sectors[%d]=%s",i,token);
+                        i++;
+                        break;
+                    case 's':
+                        printf ("\nSoundFocus.start_angles[%d]=%s",i,token);
+                        i++;
+                        break;
+                }
+                token = strtok (NULL,",;=");
+            }
+        }
+        choice = '\0';
+        printf ("\nGet SourceTracking Params from app");
+        string = qahw_get_parameters(qawh_module_handle, kvpair_sourcetrack);
+        if (!string) {
+            printf ("Error.Failed Get SourceTrack Params\n");
+        } else {
+            token = strtok (string , "=");
+            while (token) {
+                if (*token == 'S') {
+                    choice = *(token + 12);
+                    if (choice == 'd')
+                        choice = *(token + 16);
+                    token = strtok (NULL,",;");
+                    i=0;
+                }
+                switch (choice) {
+                    case 'p':
+                        printf ("\nSourceTrack.polar_activity=%s,",token);
+                        choice = '\0';
+                        break;
+                    case 'v':
+                        printf ("\nSourceTrack.vad[%d]=%s",i,token);
+                        i++;
+                        break;
+                    case 's':
+                        printf ("\nSourceTrack.doa_speech=%s",token);
+                        break;
+                    case 'n':
+                        printf ("\nSourceTrack.doa_noise[%d]=%s",i,token);
+                        i++;
+                        break;
+                    default :
+                        printf ("%s,",token);
+                        break;
+                }
+                token = strtok (NULL,",;=");
+            }
+        }
+        if (sourcetrack_done == 1)
+            return NULL;
+    }
+}
+
+void *start_input(void *thread_param)
+{
+  int rc = 0;
+  struct audio_config_params* params = (struct audio_config_params*) thread_param;
+  qahw_module_handle_t *qahw_mod_handle = params->qahw_mod_handle;
+
+  // Open audio input stream.
+  qahw_stream_handle_t* in_handle = NULL;
+
+  rc = qahw_open_input_stream(qahw_mod_handle,
+                              params->handle, params->input_device,
+                              &params->config, &in_handle,
+                              params->flags, params->kStreamName,
+                              params->kInputSource);
+  if (rc) {
+      printf("ERROR :::: Could not open input stream.\n" );
+      pthread_mutex_lock(&glock);
+      gerror = true;
+      pthread_cond_signal(&gcond);
+      pthread_mutex_unlock(&glock);
+      pthread_exit(0);
+  }
+
+  // Get buffer size to get upper bound on data to read from the HAL.
+  size_t buffer_size;
+      buffer_size = qahw_in_get_buffer_size(in_handle);
+  char *buffer;
+  buffer = (char *)calloc(1, buffer_size);
+  if (buffer == NULL) {
+     printf("calloc failed!!\n");
+     pthread_mutex_lock(&glock);
+     gerror = true;
+     pthread_cond_signal(&gcond);
+     pthread_mutex_unlock(&glock);
+     pthread_exit(0);
+  }
+
+  printf("input opened, buffer = %p, size %zun",
+         buffer, buffer_size);
+
+  int num_channels = audio_channel_count_from_in_mask(params->config.channel_mask);
+
+  time_t start_time = time(0);
+  ssize_t bytes_read = -1;
+  char param[100] = "audio_stream_profile=";
+  qahw_in_buffer_t in_buf;
+
+  // set profile for the recording session
+  strlcat(param, params->profile, sizeof(param));
+  qahw_in_set_parameters(in_handle, param);
+
+  printf("\nPlease speak into the microphone for %lf seconds.\n", params->loopTime);
+
+  FILE *fd = fopen(params->output_filename,"w");
+  if (fd == NULL) {
+     printf("File open failed \n");
+     pthread_mutex_lock(&glock);
+     gerror = true;
+     pthread_cond_signal(&gcond);
+     pthread_mutex_unlock(&glock);
+     pthread_exit(0);
+  }
+  pthread_mutex_lock(&glock);
+  tests_running++;
+  pthread_cond_signal(&gcond);
+  pthread_mutex_unlock(&glock);
+  memset(&in_buf,0, sizeof(qahw_in_buffer_t));
+
+  while(true) {
+      in_buf.buffer = buffer;
+      in_buf.bytes = buffer_size;
+      bytes_read = qahw_in_read(in_handle, &in_buf);
+      fwrite(in_buf.buffer, sizeof(char), buffer_size, fd);
+      if(difftime(time(0), start_time) > params->loopTime) {
+          printf("\nTest completed.\n");
+          break;
+      }
+  }
+
+  printf("closing input");
+
+  // Close output stream and device.
+  rc = qahw_in_standby(in_handle);
+  if (rc) {
+      printf("out standby failed %d \n",rc);
+  }
+
+  rc = qahw_close_input_stream(in_handle);
+  if (rc) {
+      printf("could not close input stream %d \n",rc);
+  }
+
+  // Print instructions to access the file.
+  printf("\nThe audio recording has been saved to %s. Please use adb pull to get "
+         "the file and play it using audacity. The audio data has the "
+         "following characteristics:\nsample rate: %i\nformat: %d\n"
+         "num channels: %i\n",
+         params->output_filename, params->config.sample_rate,
+         params->config.format, num_channels);
+
+  pthread_mutex_lock(&glock);
+  tests_running--;
+  pthread_cond_signal(&gcond);
+  pthread_mutex_unlock(&glock);
+  pthread_exit(0);
+  return NULL;
+}
+
+int read_config_params_from_user(struct audio_config_params *thread_param, int rec_session) {
+    int channels = 0, format = 0, sample_rate = 0,source = 0, device = 0;
+
+    thread_param->kStreamName = "input_stream";
+
+    printf(" \n Enter input device (4->built-in mic, 16->wired_headset .. etc) ::::: ");
+    scanf(" %d", &device);
+    if (device & AUDIO_DEVICE_IN_BUILTIN_MIC)
+        thread_param->input_device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+    else if (device & AUDIO_DEVICE_IN_WIRED_HEADSET)
+        thread_param->input_device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+
+    printf(" \n Enter the channels (1 -mono, 2 -stereo and 4 -quad channels) ::::: ");
+    scanf(" %d", &channels);
+    if (channels == 1) {
+        thread_param->config.channel_mask = AUDIO_CHANNEL_IN_MONO;
+    } else if (channels == 2) {
+        thread_param->config.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+    } else if (channels == 4) {
+        thread_param->config.channel_mask = AUDIO_CHANNEL_INDEX_MASK_4;
+    } else {
+        gerror = true;
+        printf("\nINVALID channels");
+        return -1;
+    }
+
+    printf(" \n Enter the format (16 - 16 bit recording, 24 - 24 bit recording) ::::: ");
+    scanf(" %d", &format);
+    if (format == 16) {
+        thread_param->config.format = AUDIO_FORMAT_PCM_16_BIT;
+    } else if (format == 24) {
+        thread_param->config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+    } else {
+        gerror = true;
+        printf("\n INVALID format");
+        return -1;
+    }
+
+    printf(" \n Enter the sample rate (48000, 16000 etc) :::: ");
+    scanf(" %d", &sample_rate);
+    thread_param->config.sample_rate = sample_rate;
+
+#ifdef MULTIRECORD_SUPPOT
+    printf(" \n Enter profile (none, record_fluence, record_mec, record_unprocessed etc) :::: ");
+    scanf(" %s", thread_param->profile);
+#else
+    thread_param->flags = (audio_input_flags_t)AUDIO_INPUT_FLAG_NONE;
+#endif
+    printf("\n Enter the audio source ( ref: system/media/audio/include/system/audio.h) :::: ");
+    scanf(" %d", &source);
+    thread_param->kInputSource = (audio_source_t)source;
+
+    if (rec_session == 1) {
+        thread_param->handle = 0x999;
+        strcpy(thread_param->output_filename, "/data/rec1.raw");
+    } else if (rec_session == 2) {
+        thread_param->handle = 0x998;
+        strcpy(thread_param->output_filename, "/data/rec2.raw");
+    } else if (rec_session == 3) {
+        thread_param->handle = 0x997;
+        strcpy(thread_param->output_filename, "/data/rec3.raw");
+    } else if (rec_session == 4) {
+        thread_param->handle = 0x996;
+        strcpy(thread_param->output_filename, "/data/rec4.raw");
+    }
+
+    printf("\n Enter the record duration in seconds ::::  ");
+    scanf(" %lf", &thread_param->loopTime);
+    return 0;
+}
+
+int main() {
+    int max_recordings_requested = 0, source_track = 0;
+    int thread_active[4] = {0};
+    qahw_module_handle_t *qahw_mod_handle;
+    const  char *mod_name = "audio.primary";
+
+    pthread_cond_init(&gcond, (const pthread_condattr_t *) NULL);
+
+    qahw_mod_handle = qahw_load_module(mod_name);
+    if(qahw_mod_handle == NULL) {
+        printf(" qahw_load_module failed");
+        return -1;
+    }
+#ifdef MULTIRECORD_SUPPOT
+    printf("Starting audio hal multi recording test. \n");
+    printf(" Enter number of record sessions to be started \n");
+    printf("             (Maximum of 4 record sessions are allowed)::::  ");
+    scanf(" %d", &max_recordings_requested);
+#else
+    max_recordings_requested = 1;
+#endif
+    printf(" \n Source Tracking enabled ??? ( 1 - Enable 0 - Disable)::: ");
+    scanf(" %d", &source_track);
+
+    struct audio_config_params thread1_params, thread2_params;
+    struct audio_config_params thread3_params, thread4_params;
+
+    switch (max_recordings_requested) {
+        case 4:
+            printf(" Enter the config params for fourth record session \n");
+            thread4_params.qahw_mod_handle = qahw_mod_handle;
+            read_config_params_from_user( &thread4_params, 4);
+            thread_active[3] = 1;
+            printf(" \n");
+        case 3:
+            printf(" Enter the config params for third record session \n");
+            thread3_params.qahw_mod_handle = qahw_mod_handle;
+            read_config_params_from_user( &thread3_params, 3);
+            thread_active[2] = 1;
+            printf(" \n");
+        case 2:
+            printf(" Enter the config params for second record session \n");
+            thread2_params.qahw_mod_handle = qahw_mod_handle;
+            read_config_params_from_user( &thread2_params, 2);
+            thread_active[1] = 1;
+            printf(" \n");
+        case 1:
+            printf(" Enter the config params for first record session \n");
+            thread1_params.qahw_mod_handle = qahw_mod_handle;
+            read_config_params_from_user( &thread1_params, 1);
+            thread_active[0] = 1;
+            printf(" \n");
+            break;
+        default:
+            printf(" INVALID input -- Max record sessions supported is 4 -exit \n");
+            gerror = true;
+            break;
+    }
+
+    pthread_t tid[4];
+    pthread_t sourcetrack_thread;
+    int ret = -1;
+
+    if (thread_active[0] == 1) {
+        printf("\n Create first record thread \n");
+        ret = pthread_create(&tid[0], NULL, start_input, (void *)&thread1_params);
+        if (ret) {
+            gerror = true;
+            printf(" Failed to create first record thread \n ");
+            thread_active[0] = 0;
+        }
+    }
+    if (thread_active[1] == 1) {
+        printf("Create second record thread \n");
+        ret = pthread_create(&tid[1], NULL, start_input, (void *)&thread2_params);
+        if (ret) {
+            gerror = true;
+            printf(" Failed to create second record thread \n ");
+            thread_active[1] = 0;
+        }
+    }
+    if (thread_active[2] == 1) {
+        printf("Create third record thread \n");
+        ret = pthread_create(&tid[2], NULL, start_input, (void *)&thread3_params);
+        if (ret) {
+            gerror = true;
+            printf(" Failed to create third record thread \n ");
+            thread_active[2] = 0;
+        }
+    }
+    if (thread_active[3] == 1) {
+        printf("Create fourth record thread \n");
+        ret = pthread_create(&tid[3], NULL, start_input, (void *)&thread4_params);
+        if (ret) {
+            gerror = true;
+            printf(" Failed to create fourth record thread \n ");
+            thread_active[3] = 0;
+        }
+    }
+    if (source_track && max_recordings_requested) {
+        printf("Create source tracking thread \n");
+        ret = pthread_create(&sourcetrack_thread,
+                NULL, read_sourcetrack_data,
+                (void *)qahw_mod_handle);
+        if (ret) {
+            printf(" Failed to create source tracking thread \n ");
+            source_track = 0;
+        }
+    }
+
+    // set bad mic param
+    while (max_recordings_requested && !source_track) {
+        bool test_completed = false;
+
+        pthread_mutex_lock(&glock);
+        if (!tests_running && !gerror)
+            pthread_cond_wait(&gcond, &glock);
+        test_completed = (tests_running == 0);
+        gerror = true;
+        pthread_mutex_unlock(&glock);
+
+        if (test_completed)
+            break;
+#ifdef MULTIRECORD_SUPPOT
+        char ch;
+        printf("\n Bad mic test required (y/n):::");
+        scanf(" %c", &ch);
+        if (ch == 'y' || ch == 'Y') {
+            int bad_mic_ch_index, ret;
+            char param[100] = "bad_mic_channel_index=";
+            printf("\nEnter bad mic channel index (1, 2, 4 ...):::");
+            scanf(" %d", &bad_mic_ch_index);
+            snprintf(param, sizeof(param), "%s%d", param, bad_mic_ch_index);
+            ret = qahw_set_parameters(qahw_mod_handle, param);
+            printf("param %s set to hal with return value %d\n", param, ret);
+        } else {
+            break;
+        }
+#endif
+    }
+
+    printf(" Waiting for threads exit \n");
+    if (thread_active[0] == 1) {
+        pthread_join(tid[0], NULL);
+        printf("after first record thread exit \n");
+    }
+    if (thread_active[1] == 1) {
+        pthread_join(tid[1], NULL);
+        printf("after second record thread exit \n");
+    }
+    if (thread_active[2] == 1) {
+        pthread_join(tid[2], NULL);
+        printf("after third record thread exit \n");
+    }
+    if (thread_active[3] == 1) {
+        pthread_join(tid[3], NULL);
+        printf("after fourth record thread exit \n");
+    }
+    if (source_track) {
+        sourcetrack_done = 1;
+        pthread_join(sourcetrack_thread,NULL);
+        printf("after source tracking thread exit \n");
+    }
+
+    ret = qahw_unload_module(qahw_mod_handle);
+    if (ret) {
+        printf("could not unload hal %d \n",ret);
+    }
+
+
+    printf("Done with hal record test \n");
+    pthread_cond_destroy(&gcond);
+    return 0;
+}
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
new file mode 100644
index 0000000..8426945
--- /dev/null
+++ b/qahw_api/test/qahw_playback_test.c
@@ -0,0 +1,205 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2015 The Android Open Source Project *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/* Test app to play audio at the HAL layer */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <errno.h>
+#include "qahw_api.h"
+#include "qahw_defs.h"
+#define nullptr NULL
+#define WAV 1
+#define MP3 2
+
+
+/* Play audio from a WAV file.
+
+ Parameters:
+   out_stream: A pointer to the output audio stream.
+   in_file: A pointer to a SNDFILE object.
+   config: A pointer to struct that contains audio configuration data.
+
+ Returns: An int which has a non-negative number on success.
+*/
+
+int play_file(qahw_stream_handle_t* out_handle, FILE* in_file) {
+
+  int rc = 0;
+  size_t frames_read = 1;
+  size_t bytes_wanted ;
+  char  *data = NULL;
+  qahw_out_buffer_t out_buf;
+
+  bytes_wanted = qahw_out_get_buffer_size(out_handle);
+  data = (char *) malloc (bytes_wanted);
+  if (data == NULL) {
+      printf("calloc failed!!\n");
+      return -ENOMEM;
+  }
+
+  while(frames_read != 0) {
+      frames_read = fread(data, bytes_wanted , 1, in_file);
+      if (frames_read < 1) {
+          if (feof(in_file))
+              break;
+          else
+              printf("Error in fread --%d\n",ferror(in_file));
+      }
+      memset(&out_buf,0, sizeof(qahw_out_buffer_t));
+      out_buf.buffer = data;
+      out_buf.bytes = frames_read * bytes_wanted;
+      rc = qahw_out_write(out_handle, &out_buf);
+      if (rc < 0) {
+          printf("Writing data to hal failed %d \n",rc);
+          break;
+      }
+  }
+  return rc;
+}
+
+// Prints usage information if input arguments are missing.
+void Usage() {
+    fprintf(stderr, "Usage:hal_play [device] [filename] [filetype]\n"
+            "device: hex value representing the audio device (see "
+            "system/media/audio/include/system/audio.h)\n"
+            "filename must be passed as an argument.\n"
+            "filetype (1:WAV 2:MP3) \n");
+}
+
+int main(int argc, char* argv[]) {
+    if (argc < 4) {
+        Usage();
+        return -1;
+  }
+  // Process command line arguments.
+  FILE *filestream = NULL;
+  char header[44] = {0};
+  int sample_rate = 0;
+  int channels = 0;
+  const int audio_device_base = 16;
+  char* filename = nullptr;
+  int filetype;
+  qahw_module_handle_t *qahw_mod_handle;
+  const  char *mod_name = "audio.primary";
+
+  uint32_t desired_output_device = strtol(
+      argv[1], nullptr /* look at full string*/, audio_device_base);
+
+  filename = argv[2];
+  filetype = atoi (argv[3]);
+
+  printf("Starting audio hal tests.\n");
+  int rc = 0;
+
+  qahw_mod_handle = qahw_load_module(mod_name);
+
+ // Set to a high number so it doesn't interfere with existing stream handles
+  audio_io_handle_t handle = 0x999;
+  audio_devices_t output_device =
+      (audio_devices_t)desired_output_device;
+  audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
+  audio_config_t config;
+
+  memset(&config, 0, sizeof(audio_config_t));
+
+  if (filename) {
+      printf("filename-----%s\n",filename);
+      filestream = fopen (filename,"r");
+      if (filestream == NULL) {
+          printf("failed to open\n");
+          exit(0);
+      }
+  }
+
+  switch (filetype) {
+      case WAV:
+          //Read the wave header
+          rc = fread (header, 44 , 1, filestream);
+          if (rc != 1) {
+              printf("Error .Fread failed\n");
+              exit(0);
+          }
+          if (strncmp (header,"RIFF",4) && strncmp (header+8, "WAVE",4)) {
+              printf("Not a wave format\n");
+              exit (1);
+          }
+          memcpy (&channels, &header[22], 2);
+          memcpy (&sample_rate, &header[24], 4);
+          config.channel_mask = audio_channel_out_mask_from_count(channels);
+          config.offload_info.channel_mask = config.channel_mask;
+          config.offload_info.sample_rate = sample_rate;
+          config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT;
+          break;
+     case MP3:
+          printf("Enter Number of channels:");
+          scanf ("%d",&channels);
+          config.channel_mask = audio_channel_out_mask_from_count(channels);
+          printf("\nEnter Sample Rate:");
+          scanf ("%d",&sample_rate);
+          config.offload_info.channel_mask = config.channel_mask;
+          config.offload_info.sample_rate = sample_rate;
+          config.offload_info.format = AUDIO_FORMAT_MP3;
+          break;
+     default:
+          printf("Does not support given filetype\n");
+          Usage();
+          exit (0);
+  }
+  config.offload_info.version = AUDIO_OFFLOAD_INFO_VERSION_CURRENT;
+  config.offload_info.size = sizeof(audio_offload_info_t);
+
+  printf("Now playing to output_device=%d sample_rate=%d \n",output_device,
+          config.offload_info.sample_rate);
+  const char* stream_name = "output_stream";
+
+  // Open audio output stream.
+  qahw_stream_handle_t* out_handle = nullptr;
+  printf("calling open_out_put_stream:\n");
+  rc = qahw_open_output_stream(qahw_mod_handle, handle, output_device,
+                                        flags, &config, &out_handle,
+                                        stream_name);
+  printf("open output stream is sucess:%d  out_handhle %p\n",rc,out_handle);
+  if (rc) {
+    printf("could not open output stream %d \n",rc);
+    return -1;
+  }
+
+  play_file(out_handle, filestream);
+
+  // Close output stream and device.
+  rc = qahw_out_standby(out_handle);
+  if (rc) {
+      printf("out standby failed %d \n",rc);
+  }
+
+  rc = qahw_close_output_stream(out_handle);
+  if (rc) {
+      printf("could not close output stream %d \n",rc);
+  }
+
+  rc = qahw_unload_module(qahw_mod_handle);
+  if (rc) {
+      printf("could not unload hal  %d \n",rc);
+      return -1;
+  }
+
+  printf("Done with hal tests \n");
+  return 0;
+}
diff --git a/visualizer/Android.mk b/visualizer/Android.mk
index 87d4987..c9795ea 100644
--- a/visualizer/Android.mk
+++ b/visualizer/Android.mk
@@ -21,6 +21,10 @@
 
 LOCAL_CFLAGS+= -O2 -fvisibility=hidden
 
+ifneq ($(filter msmcobalt,$(TARGET_BOARD_PLATFORM)),)
+    LOCAL_CFLAGS += -DPLATFORM_MSMCOBALT
+endif
+
 LOCAL_SHARED_LIBRARIES := \
 	libcutils \
 	liblog \
diff --git a/visualizer/offload_visualizer.c b/visualizer/offload_visualizer.c
index b2f0952..f49c434 100644
--- a/visualizer/offload_visualizer.c
+++ b/visualizer/offload_visualizer.c
@@ -179,7 +179,11 @@
 
 #define MIXER_CARD 0
 #define SOUND_CARD 0
+#ifdef PLATFORM_MSMCOBALT
+#define CAPTURE_DEVICE 7
+#else
 #define CAPTURE_DEVICE 8
+#endif
 
 /* Proxy port supports only MMAP read and those fixed parameters*/
 #define AUDIO_CAPTURE_CHANNEL_COUNT 2
diff --git a/voice_processing/voice_processing.c b/voice_processing/voice_processing.c
index 1e1e123..610bee6 100644
--- a/voice_processing/voice_processing.c
+++ b/voice_processing/voice_processing.c
@@ -565,7 +565,9 @@
             if (pCmdData == NULL ||
                     cmdSize < (int)sizeof(effect_param_t) ||
                     pReplyData == NULL ||
-                    *replySize < (int)sizeof(effect_param_t)) {
+                    *replySize < (int)sizeof(effect_param_t) ||
+                    // constrain memcpy below
+                    ((effect_param_t *)pCmdData)->psize > *replySize - sizeof(effect_param_t)) {
                 ALOGV("fx_command() EFFECT_CMD_GET_PARAM invalid args");
                 return -EINVAL;
             }