diff --git a/configs/msm8998/audio_policy_configuration.xml b/configs/msm8998/audio_policy_configuration.xml
index e084bba..e2ec755 100644
--- a/configs/msm8998/audio_policy_configuration.xml
+++ b/configs/msm8998/audio_policy_configuration.xml
@@ -77,6 +77,7 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
+                <mixPort name="hifi_playback" role="source" />
                 <mixPort name="compress_passthrough" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
                     <profile name="" format="dynamic"
@@ -200,6 +201,7 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3"/>
                 </mixPort>
+                <mixPort name="hifi_input" role="sink" />
             </mixPorts>
 
             <devicePorts>
@@ -329,7 +331,7 @@
                 <route type="mix" sink="BT SCO All"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
                 <route type="mix" sink="USB Device Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out,hifi_playback"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
                 <route type="mix" sink="voice_rx"
@@ -348,6 +350,7 @@
                        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"/>
+                <route type="mix" sink="hifi_input" sources="USB Device In" />
             </routes>
 
         </module>
diff --git a/configs/msm8998/mixer_paths_skuk.xml b/configs/msm8998/mixer_paths_skuk.xml
index e4b3785..f2e9c9a 100644
--- a/configs/msm8998/mixer_paths_skuk.xml
+++ b/configs/msm8998/mixer_paths_skuk.xml
@@ -165,6 +165,7 @@
     <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="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
@@ -204,6 +205,7 @@
     <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 MultiMedia2" 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" />
@@ -2594,4 +2596,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/msm8998/mixer_paths_tasha.xml b/configs/msm8998/mixer_paths_tasha.xml
index 93656e6..4ea726b 100644
--- a/configs/msm8998/mixer_paths_tasha.xml
+++ b/configs/msm8998/mixer_paths_tasha.xml
@@ -212,6 +212,7 @@
     <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="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
@@ -389,6 +390,7 @@
     <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 MultiMedia2" 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" />
@@ -2935,4 +2937,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/msm8998/mixer_paths_tavil.xml b/configs/msm8998/mixer_paths_tavil.xml
index ba37799..fc5a440 100644
--- a/configs/msm8998/mixer_paths_tavil.xml
+++ b/configs/msm8998/mixer_paths_tavil.xml
@@ -173,6 +173,7 @@
     <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="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
@@ -212,6 +213,7 @@
     <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 MultiMedia2" 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" />
@@ -2801,4 +2803,24 @@
        <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm660/audio_policy_configuration.xml b/configs/sdm660/audio_policy_configuration.xml
index 9a67a32..418cf42 100644
--- a/configs/sdm660/audio_policy_configuration.xml
+++ b/configs/sdm660/audio_policy_configuration.xml
@@ -73,6 +73,7 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
+                <mixPort name="hifi_playback" role="source" />
                 <mixPort name="compress_passthrough" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
                     <profile name="" format="dynamic"
@@ -191,6 +192,7 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
                 </mixPort>
+                <mixPort name="hifi_input" role="sink" />
             </mixPorts>
 
             <devicePorts>
@@ -320,7 +322,7 @@
                 <route type="mix" sink="BT SCO All"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="USB Device Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,hifi_playback"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
                 <route type="mix" sink="voice_rx"
@@ -337,6 +339,7 @@
                        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"/>
+                <route type="mix" sink="hifi_input" sources="USB Device In" />
             </routes>
 
         </module>
diff --git a/configs/sdm660/mixer_paths.xml b/configs/sdm660/mixer_paths.xml
index 459ffa1..8cca459 100644
--- a/configs/sdm660/mixer_paths.xml
+++ b/configs/sdm660/mixer_paths.xml
@@ -147,6 +147,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -175,6 +176,7 @@
     <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 MultiMedia2" 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" />
@@ -2312,4 +2314,25 @@
     <path name="mmap-record usb-headset-mic">
        <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm660/mixer_paths_mtp.xml b/configs/sdm660/mixer_paths_mtp.xml
index bba3cb9..4f61224 100644
--- a/configs/sdm660/mixer_paths_mtp.xml
+++ b/configs/sdm660/mixer_paths_mtp.xml
@@ -146,6 +146,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -174,6 +175,7 @@
     <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 MultiMedia2" 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" />
@@ -2373,4 +2375,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm660/mixer_paths_skus.xml b/configs/sdm660/mixer_paths_skus.xml
index 8ecc1a3..bdd0d61 100644
--- a/configs/sdm660/mixer_paths_skus.xml
+++ b/configs/sdm660/mixer_paths_skus.xml
@@ -210,6 +210,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -391,6 +392,7 @@
     <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 MultiMedia2" 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" />
@@ -2849,4 +2851,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm660/mixer_paths_skush.xml b/configs/sdm660/mixer_paths_skush.xml
index cf0838f..aa81e7b 100644
--- a/configs/sdm660/mixer_paths_skush.xml
+++ b/configs/sdm660/mixer_paths_skush.xml
@@ -141,6 +141,7 @@
     <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="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia6 Mixer INT3_MI2S_TX" value="0" />
@@ -166,6 +167,7 @@
     <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 MultiMedia2" 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" />
@@ -2423,4 +2425,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm660/mixer_paths_wcd9326.xml b/configs/sdm660/mixer_paths_wcd9326.xml
index 0745180..9e6623d 100644
--- a/configs/sdm660/mixer_paths_wcd9326.xml
+++ b/configs/sdm660/mixer_paths_wcd9326.xml
@@ -209,6 +209,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -390,6 +391,7 @@
     <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 MultiMedia2" 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" />
@@ -2794,4 +2796,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm660/mixer_paths_wcd9335.xml b/configs/sdm660/mixer_paths_wcd9335.xml
index 8cc6f65..d4c098f 100644
--- a/configs/sdm660/mixer_paths_wcd9335.xml
+++ b/configs/sdm660/mixer_paths_wcd9335.xml
@@ -209,6 +209,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -390,6 +391,7 @@
     <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 MultiMedia2" 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" />
@@ -2871,4 +2873,25 @@
     <path name="mmap-record usb-headset-mic">
        <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm660/mixer_paths_wcd9340.xml b/configs/sdm660/mixer_paths_wcd9340.xml
index 213f5c3..6c6f2cf 100644
--- a/configs/sdm660/mixer_paths_wcd9340.xml
+++ b/configs/sdm660/mixer_paths_wcd9340.xml
@@ -170,6 +170,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -213,6 +214,7 @@
     <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 MultiMedia2" 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" />
@@ -2619,4 +2621,24 @@
        <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm670/audio_policy_configuration.xml b/configs/sdm670/audio_policy_configuration.xml
index 09a4bb9..8e51ba5 100644
--- a/configs/sdm670/audio_policy_configuration.xml
+++ b/configs/sdm670/audio_policy_configuration.xml
@@ -73,6 +73,7 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
+                <mixPort name="hifi_playback" role="source" />
                 <mixPort name="compress_passthrough" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
                     <profile name="" format="dynamic"
@@ -191,6 +192,7 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
                 </mixPort>
+                <mixPort name="hifi_input" role="sink" />
             </mixPorts>
 
             <devicePorts>
@@ -320,7 +322,7 @@
                 <route type="mix" sink="BT SCO All"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="USB Device Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,hifi_playback"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
                 <route type="mix" sink="voice_rx"
@@ -337,6 +339,7 @@
                        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"/>
+                <route type="mix" sink="hifi_input" sources="USB Device In" />
             </routes>
 
         </module>
diff --git a/configs/sdm670/mixer_paths.xml b/configs/sdm670/mixer_paths.xml
index 1b07f80..5f81e65 100644
--- a/configs/sdm670/mixer_paths.xml
+++ b/configs/sdm670/mixer_paths.xml
@@ -146,6 +146,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -174,6 +175,7 @@
     <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 MultiMedia2" 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" />
@@ -2256,4 +2258,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm670/mixer_paths_mtp.xml b/configs/sdm670/mixer_paths_mtp.xml
index b08059c..9b462a4 100644
--- a/configs/sdm670/mixer_paths_mtp.xml
+++ b/configs/sdm670/mixer_paths_mtp.xml
@@ -132,6 +132,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -159,6 +160,7 @@
     <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 MultiMedia2" 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" />
@@ -2119,4 +2121,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm670/mixer_paths_tasha.xml b/configs/sdm670/mixer_paths_tasha.xml
index be2f56b..cbb74a5 100644
--- a/configs/sdm670/mixer_paths_tasha.xml
+++ b/configs/sdm670/mixer_paths_tasha.xml
@@ -195,6 +195,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -375,6 +376,7 @@
     <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 MultiMedia2" 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" />
@@ -2751,4 +2753,24 @@
         <path name="audio-ull-playback" />
     </path>
 
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm670/mixer_paths_tashalite.xml b/configs/sdm670/mixer_paths_tashalite.xml
index 65dec6a..e4ea674 100644
--- a/configs/sdm670/mixer_paths_tashalite.xml
+++ b/configs/sdm670/mixer_paths_tashalite.xml
@@ -195,6 +195,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -375,6 +376,7 @@
     <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 MultiMedia2" 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" />
@@ -2728,4 +2730,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm670/mixer_paths_tavil.xml b/configs/sdm670/mixer_paths_tavil.xml
index 8a8783b..f822423 100644
--- a/configs/sdm670/mixer_paths_tavil.xml
+++ b/configs/sdm670/mixer_paths_tavil.xml
@@ -157,6 +157,7 @@
     <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_RX Channels" value="Two" />
     <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+    <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
@@ -199,6 +200,7 @@
     <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 MultiMedia2" 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" />
@@ -2386,4 +2388,25 @@
         <path name="audio-ull-playback bt-a2dp" />
         <path name="audio-ull-playback" />
     </path>
+
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/configs/sdm845/audio_policy_configuration.xml b/configs/sdm845/audio_policy_configuration.xml
index d775641..a19151c 100644
--- a/configs/sdm845/audio_policy_configuration.xml
+++ b/configs/sdm845/audio_policy_configuration.xml
@@ -77,6 +77,7 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
+                <mixPort name="hifi_playback" role="source" />
                 <mixPort name="compress_passthrough" role="source"
                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
                     <profile name="" format="dynamic"
@@ -200,6 +201,7 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3"/>
                 </mixPort>
+                <mixPort name="hifi_input" role="sink" />
             </mixPorts>
 
             <devicePorts>
@@ -329,7 +331,7 @@
                 <route type="mix" sink="BT SCO All"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="USB Device Out"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out,hifi_playback"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="voice_tx"/>
                 <route type="mix" sink="voice_rx"
@@ -348,6 +350,7 @@
                        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"/>
+                <route type="mix" sink="hifi_input" sources="USB Device In" />
             </routes>
 
         </module>
diff --git a/configs/sdm845/mixer_paths_tavil.xml b/configs/sdm845/mixer_paths_tavil.xml
index b073fb2..7b2e638 100644
--- a/configs/sdm845/mixer_paths_tavil.xml
+++ b/configs/sdm845/mixer_paths_tavil.xml
@@ -154,6 +154,7 @@
     <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="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
     <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="0" />
@@ -193,6 +194,7 @@
     <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 MultiMedia2" 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" />
@@ -2658,4 +2660,24 @@
        <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="hifi-playback display-port">
+        <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback afe-proxy">
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headset">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-playback usb-headphones">
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+    </path>
+
+    <path name="hifi-record usb-headset-mic">
+        <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+    </path>
+
 </mixer>
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index c62226a..1f350aa 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -51,7 +51,7 @@
 #endif
 
 #ifndef INCALL_MUSIC_ENABLED
-#define AUDIO_OUTPUT_FLAG_INCALL_MUSIC 0x8000
+#define AUDIO_OUTPUT_FLAG_INCALL_MUSIC 0x80000000 //0x8000
 #endif
 
 #ifndef AUDIO_DEVICE_OUT_FM_TX
@@ -213,6 +213,9 @@
 #define audio_extn_usb_enable_sidetone(device, enable)                 (0)
 #define audio_extn_usb_set_sidetone_gain(parms, value, len)            (0)
 #define audio_extn_usb_is_capture_supported()                          (0)
+#define audio_extn_usb_get_max_channels(p)                             (0)
+#define audio_extn_usb_get_max_bit_width(p)                            (0)
+#define audio_extn_usb_get_sup_sample_rates(t, s, l)                   (0)
 #else
 void audio_extn_usb_init(void *adev);
 void audio_extn_usb_deinit();
@@ -226,6 +229,9 @@
 int audio_extn_usb_set_sidetone_gain(struct str_parms *parms,
                                      char *value, int len);
 bool audio_extn_usb_is_capture_supported();
+int audio_extn_usb_get_max_channels(bool playback);
+int audio_extn_usb_get_max_bit_width(bool playback);
+int audio_extn_usb_get_sup_sample_rates(int type, uint32_t *sr, uint32_t l);
 #endif
 
 #ifndef SPLIT_A2DP_ENABLED
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index 7b0309b..60216b1 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -53,10 +53,14 @@
 #define SAMPLE_RATE_11025         11025
 // Supported sample rates for USB
 static uint32_t supported_sample_rates[] =
-    {8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000};
+    {384000, 352800, 192000, 176400, 96000, 88200, 64000, 48000, 44100, 32000, 22050, 16000, 11025, 8000};
+static uint32_t supported_sample_rates_mask[2];
 
 #define  MAX_SAMPLE_RATE_SIZE  sizeof(supported_sample_rates)/sizeof(supported_sample_rates[0])
 
+#define _MAX(x, y) (((x) >= (y)) ? (x) : (y))
+#define _MIN(x, y) (((x) <= (y)) ? (x) : (y))
+
 enum usb_usecase_type{
     USB_PLAYBACK = 0,
     USB_CAPTURE,
@@ -246,7 +250,7 @@
     return 0;
 }
 
-static int usb_get_sample_rates(char *rates_str,
+static int usb_get_sample_rates(int type, char *rates_str,
                                 struct usb_device_config *config)
 {
     uint32_t i;
@@ -277,6 +281,7 @@
             if (supported_sample_rates[i] >= min_sr &&
                 supported_sample_rates[i] <= max_sr) {
                 config->rates[sr_size++] = supported_sample_rates[i];
+                supported_sample_rates_mask[type] |= (1<<i);
                 ALOGI_IF(usb_audio_debug_enable,
                     "%s: continuous sample rate supported_sample_rates[%d] %d",
                     __func__, i, supported_sample_rates[i]);
@@ -291,6 +296,7 @@
                         "%s: sr %d, supported_sample_rates[%d] %d -> matches!!",
                         __func__, sr, i, supported_sample_rates[i]);
                     config->rates[sr_size++] = supported_sample_rates[i];
+                    supported_sample_rates_mask[type] |= (1<<i);
                 }
             }
             next_sr_string = strtok_r(NULL, " ,.-", &temp_ptr);
@@ -455,7 +461,7 @@
         }
         memcpy(rates_str, rates_str_start, size);
         rates_str[size] = '\0';
-        ret = usb_get_sample_rates(rates_str, usb_device_info);
+        ret = usb_get_sample_rates(type, rates_str, usb_device_info);
         if (rates_str)
             free(rates_str);
         if (ret < 0) {
@@ -551,13 +557,24 @@
     return;
 }
 
-static bool usb_valid_device(int device)
+static inline bool usb_output_device(audio_devices_t device) {
+    // ignore accessory for now
+    if (device == AUDIO_DEVICE_OUT_USB_ACCESSORY)
+        return false;
+    return audio_is_usb_out_device(device);
+}
+
+static inline bool usb_input_device(audio_devices_t device) {
+    // ignore accessory for now
+    if (device == AUDIO_DEVICE_IN_USB_ACCESSORY)
+        return false;
+    return audio_is_usb_in_device(device);
+}
+
+static bool usb_valid_device(audio_devices_t device)
 {
-    bool is_usb_device = false;
-    if ((device & AUDIO_DEVICE_OUT_USB_DEVICE) ||
-        (device & AUDIO_DEVICE_IN_USB_DEVICE))
-        is_usb_device = true;
-    return is_usb_device;
+    return usb_output_device(device) ||
+           usb_input_device(device);
 }
 
 static void usb_print_active_device(void){
@@ -852,7 +869,7 @@
         card_info = node_to_item(node_i, struct usb_card_config, list);
         ALOGV("%s: card_dev_type (0x%x), card_no(%d)",
                __func__,  card_info->usb_device_type, card_info->usb_card);
-        if (card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) {
+        if (usb_output_device(card_info->usb_device_type)) {
             if ((i = card_info->usb_sidetone_index[USB_SIDETONE_ENABLE_INDEX]) != -1) {
                 struct mixer_ctl *ctl = mixer_get_ctl_by_name(
                                 card_info->usb_snd_mixer,
@@ -898,8 +915,8 @@
                  "%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 ((is_playback && card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) ||
-            ((!is_playback) && card_info->usb_device_type == AUDIO_DEVICE_IN_USB_DEVICE)){
+        if ((is_playback && usb_output_device(card_info->usb_device_type)) ||
+            (!is_playback && usb_input_device(card_info->usb_device_type))){
             is_usb_supported = usb_audio_backend_apply_policy(
                                            &card_info->usb_device_conf_list,
                                            bit_width,
@@ -914,6 +931,70 @@
     return is_usb_supported;
 }
 
+int audio_extn_usb_get_max_channels(bool is_playback)
+{
+    struct listnode *node_i, *node_j;
+    struct usb_device_config *dev_info;
+    struct usb_card_config *card_info;
+    unsigned int max_ch = 1;
+    list_for_each(node_i, &usbmod->usb_card_conf_list) {
+            card_info = node_to_item(node_i, struct usb_card_config, list);
+            if (usb_output_device(card_info->usb_device_type) && !is_playback)
+                continue;
+            else if (usb_input_device(card_info->usb_device_type) && is_playback)
+                continue;
+
+            list_for_each(node_j, &card_info->usb_device_conf_list) {
+                dev_info = node_to_item(node_j, struct usb_device_config, list);
+                max_ch = _MAX(max_ch, dev_info->channels);
+            }
+    }
+
+    return max_ch;
+}
+
+int audio_extn_usb_get_max_bit_width(bool is_playback)
+{
+    struct listnode *node_i, *node_j;
+    struct usb_device_config *dev_info;
+    struct usb_card_config *card_info;
+    unsigned int max_bw = 16;
+    list_for_each(node_i, &usbmod->usb_card_conf_list) {
+            card_info = node_to_item(node_i, struct usb_card_config, list);
+            if (usb_output_device(card_info->usb_device_type) && !is_playback)
+                continue;
+            else if (usb_input_device(card_info->usb_device_type) && is_playback)
+                continue;
+
+            list_for_each(node_j, &card_info->usb_device_conf_list) {
+                dev_info = node_to_item(node_j, struct usb_device_config, list);
+                max_bw = _MAX(max_bw, dev_info->bit_width);
+            }
+    }
+
+    return max_bw;
+}
+
+int audio_extn_usb_get_sup_sample_rates(bool is_playback,
+                                        uint32_t *sample_rates,
+                                        uint32_t sample_rate_size)
+{
+    int type = is_playback ? USB_PLAYBACK : USB_CAPTURE;
+
+    ALOGV("%s supported_sample_rates_mask 0x%x", __func__, supported_sample_rates_mask[type]);
+    uint32_t bm = supported_sample_rates_mask[type];
+    uint32_t tries = _MIN(sample_rate_size, (uint32_t)__builtin_popcount(bm));
+
+    int i = 0;
+    while (tries--) {
+        int idx = __builtin_ffs(bm) - 1;
+        sample_rates[i++] = supported_sample_rates[idx];
+        bm &= ~(1<<idx);
+    }
+
+    return i;
+}
+
 bool audio_extn_usb_is_capture_supported()
 {
     if (usbmod == NULL) {
@@ -968,18 +1049,18 @@
         goto exit;
     }
     list_init(&usb_card_info->usb_device_conf_list);
-    if (device & AUDIO_DEVICE_OUT_USB_DEVICE) {
+    if (usb_output_device(device)) {
         if (!usb_get_device_pb_config(usb_card_info, card)){
             usb_card_info->usb_card = card;
-            usb_card_info->usb_device_type = AUDIO_DEVICE_OUT_USB_DEVICE;
+            usb_card_info->usb_device_type = device;
             usb_get_sidetone_mixer(usb_card_info);
             list_add_tail(&usbmod->usb_card_conf_list, &usb_card_info->list);
             goto exit;
         }
-    } else if (device & AUDIO_DEVICE_IN_USB_DEVICE) {
+    } else if (usb_input_device(device)) {
         if (!usb_get_device_cap_config(usb_card_info, card)) {
             usb_card_info->usb_card = card;
-            usb_card_info->usb_device_type = AUDIO_DEVICE_IN_USB_DEVICE;
+            usb_card_info->usb_device_type = device;
             usbmod->is_capture_supported = true;
             list_add_tail(&usbmod->usb_card_conf_list, &usb_card_info->list);
             goto exit;
@@ -1034,7 +1115,12 @@
             free(node_to_item(node_i, struct usb_card_config, list));
         }
     }
-    usbmod->is_capture_supported = false;
+    if (audio_is_usb_in_device(device)) { // XXX not sure if we need to check for card
+        usbmod->is_capture_supported = false;
+        supported_sample_rates_mask[USB_CAPTURE] = 0;
+    } else
+        supported_sample_rates_mask[USB_PLAYBACK] = 0;
+
 exit:
     if (usb_audio_debug_enable)
         usb_print_active_device();
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index d0c0cfa..6cf2231 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -63,6 +63,7 @@
 #include <cutils/sched_policy.h>
 
 #include <hardware/audio_effect.h>
+#include <hardware/audio_alsaops.h>
 #include <system/thread_defs.h>
 #include <tinyalsa/asoundlib.h>
 #include <audio_effects/effect_aec.h>
@@ -116,6 +117,10 @@
     .format = PCM_FORMAT_S16_LE,
 };
 
+#define MIN_CHANNEL_COUNT                1
+#define DEFAULT_CHANNEL_COUNT            2
+#define MAX_HIFI_CHANNEL_COUNT           8
+
 static unsigned int configured_low_latency_capture_period_size =
         LOW_LATENCY_CAPTURE_PERIOD_SIZE;
 
@@ -184,6 +189,17 @@
     .avail_min = MMAP_PERIOD_SIZE, //1 ms
 };
 
+struct pcm_config pcm_config_hifi = {
+    .channels = DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
+    .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
+    .period_size = HIFI_BUFFER_OUTPUT_PERIOD_SIZE, /* change #define */
+    .period_count = HIFI_BUFFER_OUTPUT_PERIOD_COUNT,
+    .format = PCM_FORMAT_S24_3LE,
+    .start_threshold = 0,
+    .stop_threshold = INT_MAX,
+    .avail_min = 0,
+};
+
 struct pcm_config pcm_config_audio_capture = {
     .channels = 2,
     .period_count = AUDIO_CAPTURE_PERIOD_COUNT,
@@ -276,6 +292,7 @@
     [USECASE_AUDIO_PLAYBACK_OFFLOAD9] = "compress-offload-playback9",
     [USECASE_AUDIO_PLAYBACK_FM] = "play-fm",
     [USECASE_AUDIO_PLAYBACK_MMAP] = "mmap-playback",
+    [USECASE_AUDIO_PLAYBACK_HIFI] = "hifi-playback",
 
     [USECASE_AUDIO_RECORD] = "audio-record",
     [USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress",
@@ -285,6 +302,7 @@
     [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
     [USECASE_AUDIO_RECORD_FM_VIRTUAL] = "fm-virtual-record",
     [USECASE_AUDIO_RECORD_MMAP] = "mmap-record",
+    [USECASE_AUDIO_RECORD_HIFI] = "hifi-record",
 
     [USECASE_AUDIO_HFP_SCO] = "hfp-sco",
     [USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb",
@@ -361,7 +379,7 @@
     uint32_t value;
 };
 
-static const struct string_to_enum out_channels_name_to_enum_table[] = {
+static const struct string_to_enum channels_name_to_enum_table[] = {
     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_2POINT1),
     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
@@ -370,9 +388,23 @@
     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_6POINT1),
     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_2),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_3),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_4),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_5),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_6),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_7),
+    STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_8),
 };
 
-static const struct string_to_enum out_formats_name_to_enum_table[] = {
+static const struct string_to_enum formats_name_to_enum_table[] = {
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
     STRING_TO_ENUM(AUDIO_FORMAT_AC3),
     STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
     STRING_TO_ENUM(AUDIO_FORMAT_E_AC3_JOC),
@@ -387,7 +419,7 @@
     32000, 44100, 48000, 88200, 96000, 176400, 192000,
 };
 
-static const struct string_to_enum out_hdmi_sample_rates_name_to_enum_table[] = {
+static const struct string_to_enum out_sample_rates_name_to_enum_table[] = {
     STRING_TO_ENUM(32000),
     STRING_TO_ENUM(44100),
     STRING_TO_ENUM(48000),
@@ -1444,6 +1476,116 @@
     return ret;
 }
 
+static inline ssize_t read_usb_sup_sample_rates(bool is_playback __unused,
+                                         uint32_t *supported_sample_rates __unused,
+                                         uint32_t max_rates __unused)
+{
+    ssize_t count = audio_extn_usb_get_sup_sample_rates(is_playback,
+                                                        supported_sample_rates,
+                                                        max_rates);
+    for (ssize_t i=0; i<count; i++) {
+        ALOGV("%s %s %d", __func__, is_playback ? "P" : "C",
+              supported_sample_rates[i]);
+    }
+    return count;
+}
+
+static inline int read_usb_sup_channel_masks(bool is_playback,
+                                      audio_channel_mask_t *supported_channel_masks,
+                                      uint32_t max_masks)
+{
+    int channels = audio_extn_usb_get_max_channels(is_playback);
+    int channel_count;
+    uint32_t num_masks = 0;
+    if (channels > MAX_HIFI_CHANNEL_COUNT)
+        channels = MAX_HIFI_CHANNEL_COUNT;
+
+    if (is_playback) {
+        // For playback we never report mono because the framework always outputs stereo
+        channel_count = DEFAULT_CHANNEL_COUNT;
+        // audio_channel_out_mask_from_count() does return positional masks for channel counts
+        // above 2 but we want indexed masks here. So we
+        for ( ; channel_count <= channels && num_masks < max_masks; channel_count++) {
+            supported_channel_masks[num_masks++] = audio_channel_out_mask_from_count(channel_count);
+        }
+        for ( ; channel_count <= channels && num_masks < max_masks; channel_count++) {
+            supported_channel_masks[num_masks++] =
+                    audio_channel_mask_for_index_assignment_from_count(channel_count);
+        }
+    } else {
+        // For capture we report all supported channel masks from 1 channel up.
+        channel_count = MIN_CHANNEL_COUNT;
+        // audio_channel_in_mask_from_count() does the right conversion to either positional or
+        // indexed mask
+        for ( ; channel_count <= channels && num_masks < max_masks; channel_count++) {
+            supported_channel_masks[num_masks++] =
+                    audio_channel_in_mask_from_count(channel_count);
+        }
+    }
+    ALOGV("%s: %s supported ch %d supported_channel_masks[0] %08x num_masks %d", __func__,
+          is_playback ? "P" : "C", channels, supported_channel_masks[0], num_masks);
+    return num_masks;
+}
+
+static inline int read_usb_sup_formats(bool is_playback __unused,
+                                audio_format_t *supported_formats,
+                                uint32_t max_formats __unused)
+{
+    int bitwidth = audio_extn_usb_get_max_bit_width(is_playback);
+    switch (bitwidth) {
+        case 24:
+            // XXX : usb.c returns 24 for s24 and s24_le?
+            supported_formats[0] = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+            break;
+        case 32:
+            supported_formats[0] = AUDIO_FORMAT_PCM_32_BIT;
+            break;
+        case 16:
+        default :
+            supported_formats[0] = AUDIO_FORMAT_PCM_16_BIT;
+            break;
+    }
+    ALOGV("%s: %s supported format %d", __func__,
+          is_playback ? "P" : "C", bitwidth);
+    return 1;
+}
+
+static inline int read_usb_sup_params_and_compare(bool is_playback,
+                                           audio_format_t *format,
+                                           audio_format_t *supported_formats,
+                                           uint32_t max_formats,
+                                           audio_channel_mask_t *mask,
+                                           audio_channel_mask_t *supported_channel_masks,
+                                           uint32_t max_masks,
+                                           uint32_t *rate,
+                                           uint32_t *supported_sample_rates,
+                                           uint32_t max_rates) {
+    int ret = 0;
+    int num_formats;
+    int num_masks;
+    int num_rates;
+    int i;
+
+    num_formats = read_usb_sup_formats(is_playback, supported_formats,
+                                       max_formats);
+    num_masks = read_usb_sup_channel_masks(is_playback, supported_channel_masks,
+                                           max_masks);
+
+    num_rates = read_usb_sup_sample_rates(is_playback,
+                                          supported_sample_rates, max_rates);
+
+#define LUT(table, len, what, dflt)                  \
+    for (i=0; i<len && (table[i] != what); i++);    \
+    if (i==len) { ret |= (what == dflt ? 0 : -1); what=table[0]; }
+
+    LUT(supported_formats, num_formats, *format, AUDIO_FORMAT_DEFAULT);
+    LUT(supported_channel_masks, num_masks, *mask, AUDIO_CHANNEL_NONE);
+    LUT(supported_sample_rates, num_rates, *rate, 0);
+
+#undef LUT
+    return ret < 0 ? -EINVAL : 0; // HACK TBD
+}
+
 audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
                                                  usecase_type_t type)
 {
@@ -2854,7 +2996,8 @@
         (format != AUDIO_FORMAT_PCM_24_BIT_PACKED) && (format != AUDIO_FORMAT_PCM_32_BIT) &&
         (format != AUDIO_FORMAT_PCM_FLOAT)) &&
         !voice_extn_compress_voip_is_format_supported(format) &&
-        !audio_extn_compr_cap_format_supported(format))  ret = -EINVAL;
+        !audio_extn_compr_cap_format_supported(format))
+            ret = -EINVAL;
 
     switch (channel_count) {
     case 1:
@@ -3442,6 +3585,103 @@
     return ret;
 }
 
+static bool stream_get_parameter_channels(struct str_parms *query,
+                                          struct str_parms *reply,
+                                          audio_channel_mask_t *supported_channel_masks) {
+    int ret = -1;
+    char value[512];
+    bool first = true;
+    size_t i, j;
+
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
+        ret = 0;
+        value[0] = '\0';
+        i = 0;
+        while (supported_channel_masks[i] != 0) {
+            for (j = 0; j < ARRAY_SIZE(channels_name_to_enum_table); j++) {
+                if (channels_name_to_enum_table[j].value == supported_channel_masks[i]) {
+                    if (!first)
+                        strcat(value, "|");
+
+                    strcat(value, channels_name_to_enum_table[j].name);
+                    first = false;
+                    break;
+                }
+            }
+            i++;
+        }
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
+    }
+    return ret == 0;
+}
+
+static bool stream_get_parameter_formats(struct str_parms *query,
+                                         struct str_parms *reply,
+                                         audio_format_t *supported_formats) {
+    int ret = -1;
+    char value[256];
+    size_t i, j;
+    bool first = true;
+
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+        ret = 0;
+        value[0] = '\0';
+        i = 0;
+        while (supported_formats[i] != 0) {
+            for (j = 0; j < ARRAY_SIZE(formats_name_to_enum_table); j++) {
+                if (formats_name_to_enum_table[j].value == supported_formats[i]) {
+                    if (!first) {
+                        strcat(value, "|");
+                    }
+                    strcat(value, formats_name_to_enum_table[j].name);
+                    first = false;
+                    break;
+                }
+            }
+            i++;
+        }
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
+    }
+    return ret == 0;
+}
+
+static bool stream_get_parameter_rates(struct str_parms *query,
+                                       struct str_parms *reply,
+                                       uint32_t *supported_sample_rates) {
+
+    int i;
+    char value[256];
+    int ret = -1;
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
+        ret = 0;
+        value[0] = '\0';
+        i=0;
+        int cursor = 0;
+        while (supported_sample_rates[i]) {
+            int avail = sizeof(value) - cursor;
+            ret = snprintf(value + cursor, avail, "%s%d",
+                           cursor > 0 ? "|" : "",
+                           supported_sample_rates[i]);
+            if (ret < 0 || ret >= avail) {
+                // if cursor is at the last element of the array
+                //    overwrite with \0 is duplicate work as
+                //    snprintf already put a \0 in place.
+                // else
+                //    we had space to write the '|' at value[cursor]
+                //    (which will be overwritten) or no space to fill
+                //    the first element (=> cursor == 0)
+                value[cursor] = '\0';
+                break;
+            }
+            cursor += ret;
+            ++i;
+        }
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
+                          value);
+    }
+    return ret >= 0;
+}
+
 static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
 {
     struct stream_out *out = (struct stream_out *)stream;
@@ -3464,18 +3704,18 @@
         return NULL;
     }
 
-    ALOGV("%s: enter: keys - %s", __func__, keys);
+    ALOGV("%s: %s enter: keys - %s", __func__, use_case_table[out->usecase], keys);
     ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
     if (ret >= 0) {
         value[0] = '\0';
         i = 0;
         while (out->supported_channel_masks[i] != 0) {
-            for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
-                if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
+            for (j = 0; j < ARRAY_SIZE(channels_name_to_enum_table); j++) {
+                if (channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
                     if (!first) {
                         strlcat(value, "|", sizeof(value));
                     }
-                    strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value));
+                    strlcat(value, channels_name_to_enum_table[j].name, sizeof(value));
                     first = false;
                     break;
                 }
@@ -3513,12 +3753,12 @@
         i = 0;
         first = true;
         while (out->supported_formats[i] != 0) {
-            for (j = 0; j < ARRAY_SIZE(out_formats_name_to_enum_table); j++) {
-                if (out_formats_name_to_enum_table[j].value == out->supported_formats[i]) {
+            for (j = 0; j < ARRAY_SIZE(formats_name_to_enum_table); j++) {
+                if (formats_name_to_enum_table[j].value == out->supported_formats[i]) {
                     if (!first) {
                         strlcat(value, "|", sizeof(value));
                     }
-                    strlcat(value, out_formats_name_to_enum_table[j].name, sizeof(value));
+                    strlcat(value, formats_name_to_enum_table[j].name, sizeof(value));
                     first = false;
                     break;
                 }
@@ -3537,12 +3777,12 @@
         i = 0;
         first = true;
         while (out->supported_sample_rates[i] != 0) {
-            for (j = 0; j < ARRAY_SIZE(out_hdmi_sample_rates_name_to_enum_table); j++) {
-                if (out_hdmi_sample_rates_name_to_enum_table[j].value == out->supported_sample_rates[i]) {
+            for (j = 0; j < ARRAY_SIZE(out_sample_rates_name_to_enum_table); j++) {
+                if (out_sample_rates_name_to_enum_table[j].value == out->supported_sample_rates[i]) {
                     if (!first) {
                         strlcat(value, "|", sizeof(value));
                     }
-                    strlcat(value, out_hdmi_sample_rates_name_to_enum_table[j].name, sizeof(value));
+                    strlcat(value, out_sample_rates_name_to_enum_table[j].name, sizeof(value));
                     first = false;
                     break;
                 }
@@ -4705,10 +4945,16 @@
         return NULL;
     }
 
-    ALOGV("%s: enter: keys - %s", __func__, keys);
+    ALOGV("%s: enter: keys - %s %s ", __func__, use_case_table[in->usecase], keys);
 
     voice_extn_in_get_parameters(in, query, reply);
 
+    stream_get_parameter_channels(query, reply,
+                                  &in->supported_channel_masks[0]);
+    stream_get_parameter_formats(query, reply,
+                                 &in->supported_formats[0]);
+    stream_get_parameter_rates(query, reply,
+                               &in->supported_sample_rates[0]);
     str = str_parms_to_str(reply);
     str_parms_destroy(query);
     str_parms_destroy(reply);
@@ -5043,6 +5289,10 @@
     audio_format_t format;
     struct adsp_hdlr_stream_cfg hdlr_stream_cfg;
     bool is_direct_passthough = false;
+    bool is_hdmi = devices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+    bool is_usb_dev = audio_is_usb_out_device(devices) &&
+                      (devices != AUDIO_DEVICE_OUT_USB_ACCESSORY);
+    bool direct_dev = is_hdmi || is_usb_dev;
 
     *stream_out = NULL;
 
@@ -5087,18 +5337,34 @@
         (property_get_bool("audio.matrix.limiter.enable", false)))
         platform_set_device_params(out, DEVICE_PARAM_LIMITER_ID, 1);
 
-    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
-        (flags & AUDIO_OUTPUT_FLAG_DIRECT)) {
-        pthread_mutex_lock(&adev->lock);
-        ALOGV("AUDIO_DEVICE_OUT_AUX_DIGITAL and DIRECT|OFFLOAD, check hdmi caps");
-        ret = read_hdmi_sink_caps(out);
-        pthread_mutex_unlock(&adev->lock);
-        if (ret != 0) {
+    if (audio_is_linear_pcm(out->format) &&
+        out->flags == AUDIO_OUTPUT_FLAG_NONE && direct_dev) {
+       pthread_mutex_lock(&adev->lock);
+       if (is_hdmi) {
+           ALOGV("AUDIO_DEVICE_OUT_AUX_DIGITAL and DIRECT|OFFLOAD, check hdmi caps");
+           ret = read_hdmi_sink_caps(out);
+       } else if (is_usb_dev) {
+            ret = read_usb_sup_params_and_compare(true /*is_playback*/,
+                                                  &config->format,
+                                                  &out->supported_formats[0],
+                                                  MAX_SUPPORTED_FORMATS,
+                                                  &config->channel_mask,
+                                                  &out->supported_channel_masks[0],
+                                                  MAX_SUPPORTED_CHANNEL_MASKS,
+                                                  &config->sample_rate,
+                                                  &out->supported_sample_rates[0],
+                                                  MAX_SUPPORTED_SAMPLE_RATES);
+            ALOGV("plugged dev USB ret %d", ret);
+       } else {
+           ret = -1;
+       }
+       pthread_mutex_unlock(&adev->lock);
+       if (ret != 0) {
             if (ret == -ENOSYS) {
                 /* ignore and go with default */
                 ret = 0;
             } else {
-                ALOGE("error reading hdmi sink caps");
+                ALOGE("error reading direct dev sink caps");
                 goto error_open;
             }
         }
@@ -5128,6 +5394,19 @@
             goto error_open;
         }
 #endif
+    } else if (audio_is_linear_pcm(out->format) &&
+        out->flags == AUDIO_OUTPUT_FLAG_NONE && is_usb_dev) {
+        out->channel_mask = config->channel_mask;
+        out->sample_rate = config->sample_rate;
+        out->format = config->format;
+        out->usecase = USECASE_AUDIO_PLAYBACK_HIFI;
+        // does this change?
+        out->config = is_hdmi ? pcm_config_hdmi_multi : pcm_config_hifi;
+        out->config.rate = config->sample_rate;
+        out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
+        out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels *
+                                                         audio_bytes_per_sample(config->format));
+        out->config.format = pcm_format_from_audio_format(out->format);
     } else if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ||
                (out->flags == AUDIO_OUTPUT_FLAG_DIRECT)) {
         pthread_mutex_lock(&adev->lock);
@@ -5617,7 +5896,7 @@
     struct audio_device *adev = out->dev;
     int ret = 0;
 
-    ALOGD("%s: enter:stream_handle(%p)",__func__, out);
+    ALOGD("%s: enter:stream_handle(%s)",__func__, use_case_table[out->usecase]);
 
     // must deregister from sndmonitor first to prevent races
     // between the callback and close_stream
@@ -6016,6 +6295,42 @@
             false /* is_low_latency: since we don't know, be conservative */);
 }
 
+static bool adev_input_allow_hifi_record(struct audio_device *adev,
+                                         audio_devices_t devices,
+                                         audio_input_flags_t flags,
+                                         audio_source_t source) {
+    const bool allowed = true;
+
+    if (!audio_is_usb_in_device(devices))
+        return !allowed;
+
+    switch (flags) {
+        case AUDIO_INPUT_FLAG_NONE:
+        case AUDIO_INPUT_FLAG_FAST: // just fast, not fast|raw || fast|mmap
+            break;
+        default:
+            return !allowed;
+    }
+
+    switch (source) {
+        case AUDIO_SOURCE_DEFAULT:
+        case AUDIO_SOURCE_MIC:
+        case AUDIO_SOURCE_UNPROCESSED:
+            break;
+        default:
+            return !allowed;
+    }
+
+    switch (adev->mode) {
+        case 0:
+            break;
+        default:
+            return !allowed;
+    }
+
+    return allowed;
+}
+
 static int adev_open_input_stream(struct audio_hw_device *dev,
                                   audio_io_handle_t handle,
                                   audio_devices_t devices,
@@ -6031,11 +6346,26 @@
     int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
     bool is_low_latency = false;
     bool channel_mask_updated = false;
+    bool is_usb_dev = audio_is_usb_in_device(devices);
+    bool may_use_hifi_record = adev_input_allow_hifi_record(adev,
+                                                            devices,
+                                                            flags,
+                                                            source);
 
     *stream_in = NULL;
-    if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) {
-        ALOGE("%s: invalid input parameters", __func__);
-        return -EINVAL;
+
+    if (!(is_usb_dev && may_use_hifi_record)) {
+        if (config->sample_rate == 0)
+            config->sample_rate = 48000;
+        if (config->channel_mask == AUDIO_CHANNEL_NONE)
+            config->channel_mask = AUDIO_CHANNEL_IN_MONO;
+        if (config->format == AUDIO_FORMAT_DEFAULT)
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
+
+        channel_count = audio_channel_count_from_in_mask(config->channel_mask);
+
+        if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
+            return -EINVAL;
     }
 
     in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
@@ -6071,47 +6401,31 @@
     in->source = source;
     in->dev = adev;
     in->standby = 1;
-    in->channel_mask = config->channel_mask;
     in->capture_handle = handle;
     in->flags = flags;
 
-    in->usecase = USECASE_AUDIO_RECORD;
-    if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
-        (flags & AUDIO_INPUT_FLAG_FAST) != 0) {
-        is_low_latency = true;
-#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->format = config->format;
-    if ((config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE) &&
-               ((in->flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0)) {
-        in->realtime = 0;
-        in->usecase = USECASE_AUDIO_RECORD_MMAP;
-        in->config = pcm_config_mmap_capture;
-        in->stream.start = in_start;
-        in->stream.stop = in_stop;
-        in->stream.create_mmap_buffer = in_create_mmap_buffer;
-        in->stream.get_mmap_position = in_get_mmap_position;
-        in->af_period_multiplier = 1;
-        ALOGV("%s: USECASE_AUDIO_RECORD_MMAP", __func__);
-    } else if (in->realtime) {
-        in->config = pcm_config_audio_capture_rt;
-        in->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;
-
     /* restrict 24 bit capture for unprocessed source only
      * for other sources if 24 bit requested reject 24 and set 16 bit capture only
      */
-    if (config->format == AUDIO_FORMAT_DEFAULT) {
+    if (is_usb_dev && may_use_hifi_record) {
+        /* HiFi record selects an appropriate format, channel, rate combo
+           depending on sink capabilities*/
+        ret = read_usb_sup_params_and_compare(false /*is_playback*/,
+                                              &config->format,
+                                              &in->supported_formats[0],
+                                              MAX_SUPPORTED_FORMATS,
+                                              &config->channel_mask,
+                                              &in->supported_channel_masks[0],
+                                              MAX_SUPPORTED_CHANNEL_MASKS,
+                                              &config->sample_rate,
+                                              &in->supported_sample_rates[0],
+                                              MAX_SUPPORTED_SAMPLE_RATES);
+        if (ret != 0) {
+            ret = -EINVAL;
+            goto err_open;
+        }
+        channel_count = audio_channel_count_from_in_mask(config->channel_mask);
+    } else if (config->format == AUDIO_FORMAT_DEFAULT) {
         config->format = AUDIO_FORMAT_PCM_16_BIT;
     } else if ((config->format == AUDIO_FORMAT_PCM_FLOAT) ||
                (config->format == AUDIO_FORMAT_PCM_32_BIT) ||
@@ -6149,6 +6463,42 @@
         }
     }
 
+    in->channel_mask = config->channel_mask;
+    in->format = config->format;
+
+    in->usecase = USECASE_AUDIO_RECORD;
+    if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
+            (flags & AUDIO_INPUT_FLAG_FAST) != 0) {
+        is_low_latency = true;
+#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);
+    }
+
+    if ((config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE) &&
+               ((in->flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0)) {
+        in->realtime = 0;
+        in->usecase = USECASE_AUDIO_RECORD_MMAP;
+        in->config = pcm_config_mmap_capture;
+        in->stream.start = in_start;
+        in->stream.stop = in_stop;
+        in->stream.create_mmap_buffer = in_create_mmap_buffer;
+        in->stream.get_mmap_position = in_get_mmap_position;
+        in->af_period_multiplier = 1;
+        ALOGV("%s: USECASE_AUDIO_RECORD_MMAP", __func__);
+    } else if (in->realtime) {
+        in->config = pcm_config_audio_capture_rt;
+        in->sample_rate = in->config.rate;
+        in->af_period_multiplier = af_period_multiplier;
+    } else {
+        in->config = pcm_config_audio_capture;
+        in->config.rate = config->sample_rate;
+        in->sample_rate = config->sample_rate;
+        in->af_period_multiplier = 1;
+    }
+    in->bit_width = 16;
+
     /* Update config params with the requested sample rate and channels */
     if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) &&
           (adev->mode != AUDIO_MODE_IN_CALL)) {
@@ -6156,7 +6506,20 @@
         goto err_open;
     }
 
-    if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) ||
+    if (is_usb_dev && may_use_hifi_record) {
+        in->usecase = USECASE_AUDIO_RECORD_HIFI;
+        in->config = pcm_config_audio_capture;
+        frame_size = audio_stream_in_frame_size(&in->stream);
+        buffer_size = get_input_buffer_size(config->sample_rate,
+                                            config->format,
+                                            channel_count,
+                                            false /*is_low_latency*/);
+        in->config.period_size = buffer_size / frame_size;
+        in->config.rate = config->sample_rate;
+        in->af_period_multiplier = 1;
+        in->config.format = pcm_format_from_audio_format(config->format);
+        in->config.channels = channel_count;
+    } else if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) ||
              (in->device == AUDIO_DEVICE_IN_PROXY)) {
         if (config->sample_rate == 0)
             config->sample_rate = AFE_PROXY_SAMPLING_RATE;
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index fda0672..487121b 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -119,6 +119,7 @@
     USECASE_AUDIO_PLAYBACK_OFFLOAD9,
     USECASE_AUDIO_PLAYBACK_ULL,
     USECASE_AUDIO_PLAYBACK_MMAP,
+    USECASE_AUDIO_PLAYBACK_HIFI,
 
     /* FM usecase */
     USECASE_AUDIO_PLAYBACK_FM,
@@ -135,6 +136,7 @@
     USECASE_AUDIO_RECORD_COMPRESS4,
     USECASE_AUDIO_RECORD_LOW_LATENCY,
     USECASE_AUDIO_RECORD_FM_VIRTUAL,
+    USECASE_AUDIO_RECORD_HIFI,
 
     USECASE_AUDIO_PLAYBACK_VOIP,
     USECASE_AUDIO_RECORD_VOIP,
@@ -361,6 +363,11 @@
     struct audio_device *dev;
     card_status_t card_status;
     int capture_started;
+
+    /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
+    audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
+    audio_format_t supported_formats[MAX_SUPPORTED_FORMATS + 1];
+    uint32_t supported_sample_rates[MAX_SUPPORTED_SAMPLE_RATES + 1];
 };
 
 typedef enum {
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 6dfa0eb..18e355d 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -311,6 +311,8 @@
                                            LOWLATENCY_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_MULTI_CH] = {MULTIMEDIA2_PCM_DEVICE,
                                         MULTIMEDIA2_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_HIFI] = {MULTIMEDIA2_PCM_DEVICE,
+                                     MULTIMEDIA2_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_OFFLOAD] =
                      {PLAYBACK_OFFLOAD_DEVICE, PLAYBACK_OFFLOAD_DEVICE},
     /* Below entries are initialized with invalid values
@@ -335,6 +337,8 @@
                                           LOWLATENCY_PCM_DEVICE},
     [USECASE_AUDIO_RECORD_FM_VIRTUAL] = {MULTIMEDIA2_PCM_DEVICE,
                                   MULTIMEDIA2_PCM_DEVICE},
+    [USECASE_AUDIO_RECORD_HIFI] = {MULTIMEDIA2_PCM_DEVICE,
+                                   MULTIMEDIA2_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_FM] = {FM_PLAYBACK_PCM_DEVICE, FM_CAPTURE_PCM_DEVICE},
     [USECASE_AUDIO_HFP_SCO] = {HFP_PCM_RX, HFP_SCO_RX},
     [USECASE_AUDIO_HFP_SCO_WB] = {HFP_PCM_RX, HFP_SCO_RX},
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 13247ff..77a614a 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -299,6 +299,10 @@
 #define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
 #define LOW_LATENCY_CAPTURE_USE_CASE 1
 
+#define HIFI_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define HIFI_BUFFER_OUTPUT_PERIOD_DURATION 40 /* 40 millisecs */
+#define HIFI_BUFFER_OUTPUT_PERIOD_COUNT 2
+
 #define HDMI_MULTI_PERIOD_SIZE  336
 #define HDMI_MULTI_PERIOD_COUNT 8
 #define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 61ef223..639e645 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -281,6 +281,8 @@
                                             MULTIMEDIA3_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_MULTI_CH] = {MULTIMEDIA2_PCM_DEVICE,
                                          MULTIMEDIA2_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_HIFI] = {MULTIMEDIA2_PCM_DEVICE,
+                                     MULTIMEDIA2_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_OFFLOAD] =
                      {PLAYBACK_OFFLOAD_DEVICE, PLAYBACK_OFFLOAD_DEVICE},
     [USECASE_AUDIO_PLAYBACK_OFFLOAD2] =
@@ -310,6 +312,8 @@
                                           LOWLATENCY_PCM_DEVICE},
     [USECASE_AUDIO_RECORD_FM_VIRTUAL] = {MULTIMEDIA2_PCM_DEVICE,
                                   MULTIMEDIA2_PCM_DEVICE},
+    [USECASE_AUDIO_RECORD_HIFI] = {MULTIMEDIA2_PCM_DEVICE,
+                                   MULTIMEDIA2_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_FM] = {FM_PLAYBACK_PCM_DEVICE, FM_CAPTURE_PCM_DEVICE},
     [USECASE_AUDIO_HFP_SCO] = {HFP_PCM_RX, HFP_SCO_RX},
     [USECASE_AUDIO_HFP_SCO_WB] = {HFP_PCM_RX, HFP_SCO_RX},
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index ba1a195..11a4b91 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -302,6 +302,9 @@
 #define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
 #define HDMI_MULTI_PERIOD_BYTES (HDMI_MULTI_PERIOD_SIZE * HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)
 
+#define HIFI_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define HIFI_BUFFER_OUTPUT_PERIOD_DURATION 40 /* 40 milisecs */
+#define HIFI_BUFFER_OUTPUT_PERIOD_COUNT 2
 
 /* Used in calculating fragment size for pcm offload */
 #define PCM_OFFLOAD_BUFFER_DURATION 80 /* 80 millisecs */