diff --git a/configs/atoll/atoll.mk b/configs/atoll/atoll.mk
index 93d4cc9..9edaee7 100644
--- a/configs/atoll/atoll.mk
+++ b/configs/atoll/atoll.mk
@@ -105,6 +105,70 @@
 DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
 endif
 
+ifeq ($(AUDIO_FEATURE_ENABLED_DLKM),true)
+BOARD_VENDOR_KERNEL_MODULES := \
+    $(KERNEL_MODULES_OUT)/audio_apr.ko \
+    $(KERNEL_MODULES_OUT)/audio_q6_pdr.ko \
+    $(KERNEL_MODULES_OUT)/audio_q6_notifier.ko \
+    $(KERNEL_MODULES_OUT)/audio_adsp_loader.ko \
+    $(KERNEL_MODULES_OUT)/audio_q6.ko \
+    $(KERNEL_MODULES_OUT)/audio_usf.ko \
+    $(KERNEL_MODULES_OUT)/audio_pinctrl_lpi.ko \
+    $(KERNEL_MODULES_OUT)/audio_swr.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd_core.ko \
+    $(KERNEL_MODULES_OUT)/audio_swr_ctrl.ko \
+    $(KERNEL_MODULES_OUT)/audio_wsa881x.ko \
+    $(KERNEL_MODULES_OUT)/audio_platform.ko \
+    $(KERNEL_MODULES_OUT)/audio_hdmi.ko \
+    $(KERNEL_MODULES_OUT)/audio_stub.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd9xxx.ko \
+    $(KERNEL_MODULES_OUT)/audio_mbhc.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd938x.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd938x_slave.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd937x.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd937x_slave.ko \
+    $(KERNEL_MODULES_OUT)/audio_bolero_cdc.ko \
+    $(KERNEL_MODULES_OUT)/audio_wsa_macro.ko \
+    $(KERNEL_MODULES_OUT)/audio_va_macro.ko \
+    $(KERNEL_MODULES_OUT)/audio_rx_macro.ko \
+    $(KERNEL_MODULES_OUT)/audio_tx_macro.ko \
+    $(KERNEL_MODULES_OUT)/audio_native.ko \
+    $(KERNEL_MODULES_OUT)/audio_machine_atoll.ko \
+    $(KERNEL_MODULES_OUT)/audio_snd_event.ko
+endif
+
+#Audio DLKM
+AUDIO_DLKM := audio_apr.ko
+AUDIO_DLKM += audio_q6_pdr.ko
+AUDIO_DLKM += audio_q6_notifier.ko
+AUDIO_DLKM += audio_adsp_loader.ko
+AUDIO_DLKM += audio_q6.ko
+AUDIO_DLKM += audio_usf.ko
+AUDIO_DLKM += audio_pinctrl_lpi.ko
+AUDIO_DLKM += audio_swr.ko
+AUDIO_DLKM += audio_wcd_core.ko
+AUDIO_DLKM += audio_swr_ctrl.ko
+AUDIO_DLKM += audio_wsa881x.ko
+AUDIO_DLKM += audio_platform.ko
+AUDIO_DLKM += audio_hdmi.ko
+AUDIO_DLKM += audio_stub.ko
+AUDIO_DLKM += audio_wcd9xxx.ko
+AUDIO_DLKM += audio_mbhc.ko
+AUDIO_DLKM += audio_native.ko
+AUDIO_DLKM += audio_wcd938x.ko
+AUDIO_DLKM += audio_wcd938x_slave.ko
+AUDIO_DLKM += audio_wcd937x.ko
+AUDIO_DLKM += audio_wcd937x_slave.ko
+AUDIO_DLKM += audio_bolero_cdc.ko
+AUDIO_DLKM += audio_wsa_macro.ko
+AUDIO_DLKM += audio_va_macro.ko
+AUDIO_DLKM += audio_rx_macro.ko
+AUDIO_DLKM += audio_tx_macro.ko
+AUDIO_DLKM += audio_machine_atoll.ko
+AUDIO_DLKM += audio_snd_event.ko
+
+PRODUCT_PACKAGES += $(AUDIO_DLKM)
+
 PRODUCT_COPY_FILES += \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/audio_io_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_io_policy.conf \
     vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/audio_effects.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.conf \
diff --git a/configs/atoll/audio_platform_info_qrd.xml b/configs/atoll/audio_platform_info_qrd.xml
index 635f321..427647d 100644
--- a/configs/atoll/audio_platform_info_qrd.xml
+++ b/configs/atoll/audio_platform_info_qrd.xml
@@ -113,10 +113,13 @@
         <device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
+        <device name="SND_DEVICE_OUT_VOICE_HEADSET" backend="headset" interface="RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_ANC_HEADSET" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
+        <device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET" backend="headset" interface="RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
+        <device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET" backend="headset" interface="RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_HANDSET" backend="handset" interface="RX_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_SPEAKER" interface="WSA_CDC_DMA_RX_0"/>
         <device name="SND_DEVICE_OUT_SPEAKER_EXTERNAL_1" interface="WSA_CDC_DMA_RX_0"/>
diff --git a/configs/atoll/mixer_paths_qrd.xml b/configs/atoll/mixer_paths_qrd.xml
index c868451..b1823ef 100644
--- a/configs/atoll/mixer_paths_qrd.xml
+++ b/configs/atoll/mixer_paths_qrd.xml
@@ -146,6 +146,8 @@
     <!-- HFP end -->
     <!-- echo reference -->
     <ctl name="AUDIO_REF_EC_UL1 MUX" value="None" />
+    <ctl name="AUDIO_REF_EC_UL10 MUX" value="None" />
+    <ctl name="EC Reference Channels" value="Zero"/>
     <!-- usb headset -->
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="0" />
@@ -307,8 +309,12 @@
     <!-- TX Controls -->
     <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
     <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="0" />
+    <ctl name="ADC1_MIXER Switch" value="0" />
+    <ctl name="ADC1 MUX" value="ZERO" />
     <ctl name="ADC2_MIXER Switch" value="0" />
     <ctl name="ADC2 MUX" value="ZERO" />
+    <ctl name="ADC3_MIXER Switch" value="0" />
+    <ctl name="ADC3 MUX" value="ZERO" />
     <ctl name="TX_AIF1_CAP Mixer DEC0" value="0" />
     <ctl name="TX_AIF1_CAP Mixer DEC1" value="0" />
     <ctl name="TX_AIF1_CAP Mixer DEC2" value="0" />
@@ -434,30 +440,56 @@
 
     <path name="echo-reference">
         <ctl name="AUDIO_REF_EC_UL1 MUX" value="WSA_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="One"/>
+    </path>
+
+    <path name="echo-reference handset">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="RX_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="One"/>
     </path>
 
     <path name="echo-reference headphones">
         <ctl name="AUDIO_REF_EC_UL1 MUX" value="RX_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="Two"/>
+    </path>
+
+    <path name="echo-reference headset">
+        <path name="echo-reference headphones" />
     </path>
 
     <path name="echo-reference display-port">
         <ctl name="AUDIO_REF_EC_UL1 MUX" value="DISPLAY_PORT" />
+        <ctl name="EC Reference Channels" value="Two"/>
     </path>
 
     <path name="echo-reference headphones-44.1">
         <ctl name="AUDIO_REF_EC_UL1 MUX" value="RX_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="Two"/>
     </path>
 
     <path name="echo-reference-voip">
         <ctl name="AUDIO_REF_EC_UL10 MUX" value="WSA_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="One"/>
     </path>
 
     <path name="echo-reference-voip handset">
         <ctl name="AUDIO_REF_EC_UL10 MUX" value="RX_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="One"/>
     </path>
 
     <path name="echo-reference-voip headphones">
         <ctl name="AUDIO_REF_EC_UL10 MUX" value="RX_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="Two"/>
+    </path>
+
+    <path name="echo-reference-voip display-port">
+        <ctl name="AUDIO_REF_EC_UL10 MUX" value="DISPLAY_PORT" />
+        <ctl name="EC Reference Channels" value="Two"/>
+    </path>
+
+    <path name="echo-reference-voip headphones-44.1">
+        <ctl name="AUDIO_REF_EC_UL10 MUX" value="RX_CDC_DMA_RX_0" />
+        <ctl name="EC Reference Channels" value="Two"/>
     </path>
 
     <path name="deep-buffer-playback">
@@ -529,6 +561,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="1" />
     </path>
 
+    <path name="deep-buffer-playback headset">
+        <ctl name="deep-buffer-playback headphones" />
+    </path>
+
     <path name="deep-buffer-playback speaker-and-headphones">
         <path name="deep-buffer-playback headphones" />
         <path name="deep-buffer-playback" />
@@ -638,6 +674,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia5" value="1" />
     </path>
 
+    <path name="low-latency-playback headset">
+        <path name="low-latency-playback headphones" />
+    </path>
+
     <path name="low-latency-playback speaker-and-headphones">
         <path name="low-latency-playback headphones" />
         <path name="low-latency-playback" />
@@ -698,6 +738,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia8" value="1" />
     </path>
 
+    <path name="audio-ull-playback headset">
+        <path name="audio-ull-playback headphones" />
+    </path>
+
     <path name="audio-ull-playback speaker-and-headphones">
         <path name="audio-ull-playback" />
         <path name="audio-ull-playback headphones" />
@@ -858,6 +902,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
     </path>
 
+    <path name="compress-offload-playback headset">
+        <ctl name="compress-offload-playback headphones" />
+    </path>
+
     <path name="compress-offload-playback headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
     </path>
@@ -963,6 +1011,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
     </path>
 
+    <path name="compress-offload-playback2 headset">
+        <ctl name="compress-offload-playback2 headphones" />
+    </path>
+
     <path name="compress-offload-playback2 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
     </path>
@@ -1068,6 +1120,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="compress-offload-playback3 headset">
+        <path name="compress-offload-playback3 headphones" />
+    </path>
+
     <path name="compress-offload-playback3 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -1173,6 +1229,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
     </path>
 
+    <path name="compress-offload-playback4 headset">
+        <path name="compress-offload-playback4 headphones" />
+    </path>
+
     <path name="compress-offload-playback4 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
     </path>
@@ -1278,6 +1338,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
     </path>
 
+    <path name="compress-offload-playback5 headset">
+        <path name="compress-offload-playback5 headphones" />
+    </path>
+
     <path name="compress-offload-playback5 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
     </path>
@@ -1383,6 +1447,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
     </path>
 
+    <path name="compress-offload-playback6 headset">
+        <path name="compress-offload-playback6 headphones" />
+    </path>
+
     <path name="compress-offload-playback6 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
     </path>
@@ -1488,6 +1556,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
     </path>
 
+    <path name="compress-offload-playback7 headset">
+        <path name="compress-offload-playback7 headphones" />
+    </path>
+
     <path name="compress-offload-playback7 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
     </path>
@@ -1593,6 +1665,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
     </path>
 
+    <path name="compress-offload-playback8 headset">
+        <path name="compress-offload-playback8 headphones" />
+    </path>
+
     <path name="compress-offload-playback8 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
     </path>
@@ -1698,6 +1774,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="compress-offload-playback9 headset">
+        <path name="compress-offload-playback9 headphones" />
+    </path>
+
     <path name="compress-offload-playback9 headphones-44.1">
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
     </path>
@@ -1746,6 +1826,10 @@
         <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="1" />
     </path>
 
+    <path name="audio-record headset-mic">
+        <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="1" />
+    </path>
+
     <path name="audio-record usb-headset-mic">
         <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="1" />
     </path>
@@ -1772,6 +1856,10 @@
     </path>
 
     <path name="audio-record-compress headset-mic">
+        <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_3" value="1" />
+    </path>
+
+    <path name="audio-record-compress headset-mic">
         <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_4" value="1" />
     </path>
 
@@ -1841,6 +1929,10 @@
         <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
     </path>
 
+    <path name="incall-rec-uplink headste-mic">
+        <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+    </path>
+
     <path name="incall-rec-uplink bt-sco">
         <path name="incall-rec-uplink" />
     </path>
@@ -1865,6 +1957,10 @@
         <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="1" />
     </path>
 
+    <path name="incall-rec-uplink-compress headset-mic">
+        <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>
@@ -1889,6 +1985,10 @@
         <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
     </path>
 
+    <path name="incall-rec-downlink headset-mic">
+        <ctl name="MultiMedia1 Mixer VOC_REC_DL"  value="1" />
+    </path>
+
     <path name="incall-rec-downlink bt-sco">
         <path name="incall-rec-downlink" />
     </path>
@@ -1913,6 +2013,10 @@
         <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="1" />
     </path>
 
+    <path name="incall-rec-downlink-compress headset-mic">
+        <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>
@@ -1938,6 +2042,10 @@
         <path name="incall-rec-downlink" />
     </path>
 
+    <path name="incall-rec-uplink-and-downlink headset-mic">
+        <path name="incall-rec-uplink-and-downlink" />
+    </path>
+
     <path name="incall-rec-uplink-and-downlink bt-sco">
         <path name="incall-rec-uplink-and-downlink" />
     </path>
@@ -1963,6 +2071,10 @@
         <path name="incall-rec-downlink-compress" />
     </path>
 
+    <path name="incall-rec-uplink-and-downlink-compress headset-mic">
+         <path name="incall-rec-uplink-and-downlink-compress" />
+    </path>
+
     <path name="incall-rec-uplink-and-downlink-compress bt-sco">
         <path name="incall-rec-uplink-and-downlink-compress" />
     </path>
@@ -2017,6 +2129,11 @@
         <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
     </path>
 
+    <path name="compress-voip-call headset">
+        <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_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" />
@@ -2061,6 +2178,11 @@
         <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="1" />
     </path>
 
+    <path name="voicemmode1-call headset">
+        <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_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" />
@@ -2115,6 +2237,11 @@
         <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="1" />
     </path>
 
+    <path name="voicemmode2-call headset">
+        <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_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" />
@@ -2167,6 +2294,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="audio-playback-voip headset">
+        <path name="audio-playback-voip headphones" />
+    </path>
+
     <path name="audio-playback-voip bt-sco">
         <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -2215,6 +2346,10 @@
         <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_3" value="1" />
     </path>
 
+    <path name="audio-record-voip headset-mic">
+        <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_3" value="1" />
+    </path>
+
     <path name="audio-record-voip usb-headset-mic">
         <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="1" />
     </path>
@@ -2426,6 +2561,10 @@
         <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="mmap-playback headset">
+        <path name="mmap-playback headphones" />
+    </path>
+
     <path name="mmap-playback speaker-and-headphones">
         <path name="mmap-playback" />
         <path name="mmap-playback headphones" />
@@ -2503,6 +2642,10 @@
       <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_3" value="1" />
     </path>
 
+    <path name="mmap-record headset-mic">
+      <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_3" value="1" />
+    </path>
+
     <path name="mmap-record bt-sco">
       <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="1" />
     </path>
@@ -2616,6 +2759,10 @@
         <path name="incall_music_uplink" />
     </path>
 
+    <path name="incall_music_uplink headset">
+        <path name="incall_music_uplink" />
+    </path>
+
     <path name="incall_music_uplink speaker-and-headphones">
         <path name="incall_music_uplink" />
     </path>
@@ -2679,13 +2826,22 @@
 
     <path name="amic3">
         <ctl name="TX DEC0 MUX" value="SWR_MIC" />
-        <ctl name="TX SMIC MUX0" value="ADC2" />
+        <ctl name="TX SMIC MUX0" value="ADC1" />
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
         <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
         <ctl name="ADC2_MIXER Switch" value="1" />
         <ctl name="ADC2 MUX" value="INP3" />
     </path>
 
+    <path name="amic4">
+        <ctl name="TX DEC0 MUX" value="SWR_MIC" />
+        <ctl name="TX SMIC MUX0" value="ADC2" />
+        <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+        <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+        <ctl name="ADC3_MIXER Switch" value="1" />
+        <ctl name="ADC3 MUX" value="INP4" />
+    </path>
+
     <path name="dmic1">
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
         <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
@@ -2710,7 +2866,7 @@
         <ctl name="TX DMIC MUX0" value="DMIC3" />
     </path>
 	
-	<path name="dmic5">
+    <path name="dmic5">
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
         <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
         <ctl name="TX DMIC MUX0" value="DMIC4" />
@@ -2724,24 +2880,24 @@
 
     <path name="speaker">
         <ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
-        <ctl name="WSA RX1 MUX" value="AIF1_PB" />
-        <ctl name="WSA_RX1 INP0" value="RX1" />
-        <ctl name="WSA_COMP2 Switch" value="1" />
-        <ctl name="SpkrRight COMP Switch" value="1" />
-        <ctl name="SpkrRight BOOST Switch" value="1" />
-        <ctl name="SpkrRight VISENSE Switch" value="1" />
-        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+        <ctl name="WSA RX0 MUX" value="AIF1_PB" />
+        <ctl name="WSA_RX0 INP0" value="RX0" />
+        <ctl name="WSA_COMP1 Switch" value="1" />
+        <ctl name="SpkrLeft COMP Switch" value="1" />
+        <ctl name="SpkrLeft BOOST Switch" value="1" />
+        <ctl name="SpkrLeft VISENSE Switch" value="1" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
     </path>
 
     <path name="speaker-mono">
         <ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
-        <ctl name="WSA RX1 MUX" value="AIF1_PB" />
-        <ctl name="WSA_RX1 INP0" value="RX1" />
-        <ctl name="WSA_COMP2 Switch" value="1" />
-        <ctl name="SpkrRight COMP Switch" value="1" />
-        <ctl name="SpkrRight BOOST Switch" value="1" />
-        <ctl name="SpkrRight VISENSE Switch" value="1" />
-        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+        <ctl name="WSA RX0 MUX" value="AIF1_PB" />
+        <ctl name="WSA_RX0 INP0" value="RX0" />
+        <ctl name="WSA_COMP1 Switch" value="1" />
+        <ctl name="SpkrLeft COMP Switch" value="1" />
+        <ctl name="SpkrLeft BOOST Switch" value="1" />
+        <ctl name="SpkrLeft VISENSE Switch" value="1" />
+        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
     </path>
 
     <path name="speaker-mono-2">
@@ -2780,11 +2936,11 @@
     </path>
 
     <path name="speaker-mic">
-        <path name="dmic3" />
+        <path name="amic3" />
     </path>
 
     <path name="speaker-protected">
-        <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
+        <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
         <ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
         <path name="speaker" />
         <ctl name="VI_FEED_TX Channels" value="One" />
@@ -2792,7 +2948,7 @@
     </path>
 
     <path name="voice-speaker-protected">
-        <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
+        <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
         <ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
         <path name="speaker-mono" />
         <ctl name="VI_FEED_TX Channels" value="One" />
@@ -2800,11 +2956,7 @@
     </path>
 
     <path name="voice-speaker-2-protected">
-        <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
-        <ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
-        <path name="speaker-mono-2" />
-        <ctl name="VI_FEED_TX Channels" value="One" />
-        <ctl name="WSA_RX_0_VI_FB_RCH_MUX" value="WSA_CDC_DMA_TX_0" />
+        <path name="voice-speaker-protected" />
     </path>
 
     <path name="voice-speaker-stereo-protected">
@@ -2831,7 +2983,7 @@
     </path>
 
     <path name="handset-mic">
-        <path name="dmic2" />
+        <path name="amic1" />
     </path>
 
     <path name="headphones">
@@ -2903,6 +3055,10 @@
         <path name="headphones" />
     </path>
 
+    <path name="voice-headset">
+        <path name="headphones" />
+    </path>
+
     <path name="voice-line">
         <path name="voice-headphones" />
     </path>
@@ -2996,18 +3152,19 @@
     <!-- Dual MIC devices -->
     <path name="handset-dmic-endfire">
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Two" />
+        <ctl name="TX DEC0 MUX" value="SWR_MIC" />
+        <ctl name="TX SMIC MUX0" value="ADC0" />
         <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
-        <ctl name="TX DMIC MUX0" value="DMIC1" />
+        <ctl name="ADC1_MIXER Switch" value="1" />
+        <ctl name="TX DEC1 MUX" value="SWR_MIC" />
+        <ctl name="TX SMIC MUX1" value="ADC1" />
         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="TX DMIC MUX0" value="DMIC3" />
+        <ctl name="ADC2_MIXER Switch" value="1" />
+        <ctl name="ADC2 MUX" value="INP3" />
     </path>
 
     <path name="speaker-dmic-endfire">
-        <ctl name="TX_CDC_DMA_TX_3 Channels" value="Two" />
-        <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
-        <ctl name="TX DMIC MUX0" value="DMIC2" />
-        <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="TX DMIC MUX0" value="DMIC1" />
+        <path name="handset-dmic-endfire" />
     </path>
 
     <path name="dmic-endfire">
@@ -3087,7 +3244,7 @@
     </path>
 
     <path name="voice-tty-vco-handset-mic">
-        <path name="dmic3" />
+        <path name="amic3" />
     </path>
 
     <path name="unprocessed-handset-mic">
@@ -3106,14 +3263,40 @@
         <path name="three-mic" />
     </path>
 
+    <path name="three-mic">
+         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
+         <ctl name="TX DEC0 MUX" value="SWR_MIC" />
+         <ctl name="TX SMIC MUX0" value="ADC0" />
+         <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+         <ctl name="ADC1_MIXER Switch" value="1" />
+         <ctl name="TX DEC1 MUX" value="SWR_MIC" />
+         <ctl name="TX SMIC MUX1" value="ADC1" />
+         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
+         <ctl name="ADC2_MIXER Switch" value="1" />
+         <ctl name="ADC2 MUX" value="INP3" />
+         <ctl name="TX DEC2 MUX" value="SWR_MIC" />
+         <ctl name="TX SMIC MUX2" value="ADC2" />
+         <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
+         <ctl name="ADC3_MIXER Switch" value="1" />
+         <ctl name="ADC3 MUX" value="INP4" />
+    </path>
+
     <path name="speaker-tmic">
-        <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
-        <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
-        <ctl name="TX DMIC MUX0" value="DMIC2" />
-        <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="TX DMIC MUX1" value="DMIC1" />
-        <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
-        <ctl name="TX DMIC MUX2" value="DMIC3" />
+         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
+         <ctl name="TX DEC0 MUX" value="SWR_MIC" />
+         <ctl name="TX SMIC MUX0" value="ADC0" />
+         <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+         <ctl name="ADC1_MIXER Switch" value="1" />
+         <ctl name="TX DEC1 MUX" value="SWR_MIC" />
+         <ctl name="TX SMIC MUX1" value="ADC1" />
+         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
+         <ctl name="ADC2_MIXER Switch" value="1" />
+         <ctl name="ADC2 MUX" value="INP3" />
+         <ctl name="TX DEC2 MUX" value="SWR_MIC" />
+         <ctl name="TX SMIC MUX2" value="ADC2" />
+         <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
+         <ctl name="ADC3_MIXER Switch" value="1" />
+         <ctl name="ADC3 MUX" value="INP4" />
     </path>
 
     <path name="voice-speaker-tmic">
@@ -3121,15 +3304,6 @@
     </path>
 
     <path name="speaker-qmic">
-         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Four" />
-         <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
-         <ctl name="TX DMIC MUX0" value="DMIC2" />
-         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-         <ctl name="TX DMIC MUX1" value="DMIC1" />
-         <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
-         <ctl name="TX DMIC MUX2" value="DMIC3" />
-         <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
-         <ctl name="TX DMIC MUX3" value="DMIC4" />
     </path>
 
     <path name="voice-speaker-qmic">
diff --git a/configs/atoll/sound_trigger_mixer_paths_qrd.xml b/configs/atoll/sound_trigger_mixer_paths_qrd.xml
index 8e6513c..487ea11 100644
--- a/configs/atoll/sound_trigger_mixer_paths_qrd.xml
+++ b/configs/atoll/sound_trigger_mixer_paths_qrd.xml
@@ -81,6 +81,12 @@
     <ctl name="EC Reference Channels" value="Zero"/>
     <ctl name="EC Reference Bit Format" value="0"/>
     <ctl name="EC Reference SampleRate" value="0"/>
+    <ctl name="ADC1_MIXER Switch" value="0" />
+    <ctl name="ADC2_MIXER Switch" value="0" />
+    <ctl name="ADC3_MIXER Switch" value="0" />
+    <ctl name="ADC2 MUX" value="ZERO" />
+    <ctl name="ADC3 MUX" value="ZERO" />
+    <ctl name="ADC4 MUX" value="ZERO" />
 
     <path name="listen-voice-wakeup-1">
         <ctl name="LSM1 Mixer VA_CDC_DMA_TX_0" value="1" />
@@ -212,8 +218,9 @@
 
     <path name="listen-ape-handset-mic">
         <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
-        <ctl name="VA DEC0 MUX" value="MSM_DMIC" />
-        <ctl name="VA DMIC MUX0" value="DMIC1" />
+        <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+        <ctl name="VA SMIC MUX0" value="ADC0" />
+        <ctl name="ADC1_MIXER Switch" value="1" />
     </path>
 
     <path name="listen-ape-handset-mic-preproc">
@@ -222,41 +229,36 @@
 
     <path name="listen-ape-handset-dmic">
         <ctl name="VA_CDC_DMA_TX_0 Channels" value="Two" />
+        <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+        <ctl name="VA SMIC MUX0" value="ADC0" />
         <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
+        <ctl name="ADC1_MIXER Switch" value="1" />
+        <ctl name="VA DEC1 MUX" value="SWR_MIC" />
+        <ctl name="VA SMIC MUX1" value="ADC2" />
         <ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="VA DEC0 MUX" value="MSM_DMIC" />
-        <ctl name="VA DEC1 MUX" value="MSM_DMIC" />
-        <ctl name="VA DMIC MUX0" value="DMIC1" />
-        <ctl name="VA DMIC MUX1" value="DMIC2" />
+        <ctl name="ADC2_MIXER Switch" value="1" />
+        <ctl name="ADC2 MUX" value="INP3" />
     </path>
 
     <path name="listen-ape-handset-tmic">
         <ctl name="VA_CDC_DMA_TX_0 Channels" value="Three" />
+        <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+        <ctl name="VA SMIC MUX0" value="ADC0" />
         <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
+        <ctl name="ADC1_MIXER Switch" value="1" />
+        <ctl name="VA DEC1 MUX" value="SWR_MIC" />
+        <ctl name="VA SMIC MUX1" value="ADC1" />
         <ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
+        <ctl name="ADC2_MIXER Switch" value="1" />
+        <ctl name="ADC2 MUX" value="INP3" />
+        <ctl name="VA DEC2 MUX" value="SWR_MIC" />
+        <ctl name="VA SMIC MUX2" value="ADC2" />
         <ctl name="VA_AIF1_CAP Mixer DEC2" value="1" />
-        <ctl name="VA DEC0 MUX" value="MSM_DMIC" />
-        <ctl name="VA DEC1 MUX" value="MSM_DMIC" />
-        <ctl name="VA DEC2 MUX" value="MSM_DMIC" />
-        <ctl name="VA DMIC MUX0" value="DMIC1" />
-        <ctl name="VA DMIC MUX1" value="DMIC2" />
-        <ctl name="VA DMIC MUX2" value="DMIC4" />
+        <ctl name="ADC3_MIXER Switch" value="1" />
+        <ctl name="ADC3 MUX" value="INP4" />
     </path>
 
     <path name="listen-ape-handset-qmic">
-        <ctl name="VA_CDC_DMA_TX_0 Channels" value="Four" />
-        <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
-        <ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="VA_AIF1_CAP Mixer DEC2" value="1" />
-        <ctl name="VA_AIF1_CAP Mixer DEC3" value="1" />
-        <ctl name="VA DEC0 MUX" value="MSM_DMIC" />
-        <ctl name="VA DEC1 MUX" value="MSM_DMIC" />
-        <ctl name="VA DEC2 MUX" value="MSM_DMIC" />
-        <ctl name="VA DEC3 MUX" value="MSM_DMIC" />
-        <ctl name="VA DMIC MUX0" value="DMIC1" />
-        <ctl name="VA DMIC MUX1" value="DMIC2" />
-        <ctl name="VA DMIC MUX2" value="DMIC3" />
-        <ctl name="VA DMIC MUX3" value="DMIC4" />
     </path>
 
     <path name="listen-ape-headset-mic">
@@ -289,4 +291,8 @@
         <ctl name="EC Reference SampleRate" value="48000"/>
     </path>
 
+    <path name="echo-reference line">
+        <path name="echo-reference headset" />
+    </path>
+
 </mixer>
diff --git a/configs/bengal/audio_policy_configuration.xml b/configs/bengal/audio_policy_configuration.xml
index a33356b..950b2b4 100644
--- a/configs/bengal/audio_policy_configuration.xml
+++ b/configs/bengal/audio_policy_configuration.xml
@@ -173,11 +173,6 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
                 </mixPort>
-                <mixPort name="surround_sound" role="sink">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
-                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3,AUDIO_CHANNEL_INDEX_MASK_4,AUDIO_CHANNEL_IN_5POINT1,AUDIO_CHANNEL_INDEX_MASK_6"/>
-                </mixPort>
                 <mixPort name="usb_surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
@@ -358,8 +353,6 @@
                        sources="Telephony Rx"/>
                 <route type="mix" sink="primary input"
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
-                <route type="mix" sink="surround_sound"
-                       sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="usb_surround_sound"
                        sources="USB Device In,USB Headset In"/>
                 <route type="mix" sink="record_24"
diff --git a/configs/kona/kona.mk b/configs/kona/kona.mk
index cfd5a23..0b33ce4 100644
--- a/configs/kona/kona.mk
+++ b/configs/kona/kona.mk
@@ -401,7 +401,7 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.feature.a2dp_offload.enable=true \
 vendor.audio.feature.afe_proxy.enable=true \
-vendor.audio.feature.anc_headset.enable=true \
+vendor.audio.feature.anc_headset.enable=false \
 vendor.audio.feature.battery_listener.enable=true \
 vendor.audio.feature.compr_cap.enable=false \
 vendor.audio.feature.compress_in.enable=true \
diff --git a/configs/kona/mixer_paths.xml b/configs/kona/mixer_paths.xml
index 4163190..afed79a 100644
--- a/configs/kona/mixer_paths.xml
+++ b/configs/kona/mixer_paths.xml
@@ -1734,6 +1734,10 @@
         <path name="audio-record-compress bt-sco" />
     </path>
 
+    <path name="audio-record-compress headset-mic">
+        <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="audio-record-compress usb-headset-mic">
         <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
@@ -1780,6 +1784,10 @@
         <path name="low-latency-record bt-sco" />
     </path>
 
+    <path name="low-latency-record headset-mic">
+        <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="low-latency-record usb-headset-mic">
         <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
@@ -3236,6 +3244,10 @@
         <ctl name="MultiMedia2 Mixer TX_CDC_DMA_TX_3" value="1" />
     </path>
 
+    <path name="hifi-record headset-mic">
+        <ctl name="MultiMedia2 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="hifi-record usb-headset-mic">
         <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
     </path>
diff --git a/configs/kona/mixer_paths_cdp.xml b/configs/kona/mixer_paths_cdp.xml
index 705de15..eeb6496 100644
--- a/configs/kona/mixer_paths_cdp.xml
+++ b/configs/kona/mixer_paths_cdp.xml
@@ -1609,6 +1609,10 @@
         <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="audio-record-compress headset-mic">
+        <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="audio-record-compress2">
         <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="1" />
     </path>
@@ -1655,6 +1659,10 @@
         <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="low-latency-record headset-mic">
+        <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="low-latency-record capture-fm">
       <ctl name="MultiMedia8 Mixer TERT_MI2S_TX" value="1" />
     </path>
@@ -3061,6 +3069,10 @@
         <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="hifi-record headset-mic">
+        <ctl name="MultiMedia2 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="incall_music_uplink">
         <ctl name="Incall_Music Audio Mixer MultiMedia9" value="1" />
     </path>
diff --git a/configs/kona/mixer_paths_qrd.xml b/configs/kona/mixer_paths_qrd.xml
index be056b2..f76d7bd 100644
--- a/configs/kona/mixer_paths_qrd.xml
+++ b/configs/kona/mixer_paths_qrd.xml
@@ -1609,6 +1609,10 @@
         <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="audio-record-compress headset-mic">
+        <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="audio-record-compress2">
         <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="1" />
     </path>
@@ -1655,6 +1659,10 @@
         <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="low-latency-record headset-mic">
+        <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="low-latency-record capture-fm">
       <ctl name="MultiMedia8 Mixer TERT_MI2S_TX" value="1" />
     </path>
@@ -2395,6 +2403,10 @@
         <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
+    <path name="hifi-record headset-mic">
+        <ctl name="MultiMedia2 Mixer TX_CDC_DMA_TX_4" value="1" />
+    </path>
+
     <path name="incall_music_uplink">
         <ctl name="Incall_Music Audio Mixer MultiMedia9" value="1" />
     </path>
@@ -3067,6 +3079,16 @@
         <path name="three-mic" />
     </path>
 
+    <path name="three-mic">
+        <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
+        <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
+        <ctl name="TX DMIC MUX3" value="DMIC1" />
+        <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
+        <ctl name="TX DMIC MUX1" value="DMIC3" />
+        <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
+        <ctl name="TX DMIC MUX2" value="DMIC2" />
+    </path>
+
     <path name="speaker-tmic">
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
         <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
diff --git a/configs/kona/sound_trigger_platform_info.xml b/configs/kona/sound_trigger_platform_info.xml
index 2defe5b..17a937c 100644
--- a/configs/kona/sound_trigger_platform_info.xml
+++ b/configs/kona/sound_trigger_platform_info.xml
@@ -51,6 +51,7 @@
         <param dedicated_headset_path="false" />
         <param platform_lpi_enable="true" />
         <param enable_debug_dumps="false" />
+        <param support_non_lpi_without_ec="false" />
     </common_config>
     <acdb_ids>
         <param DEVICE_HANDSET_MIC_APE="100" />
@@ -121,8 +122,14 @@
             <!-- is FFECNS -->
             <param fluence_type="FLUENCE_TMIC" />
             <param execution_mode="ADSP" />
-            <!-- lpi_mode: "NON_LPI", "LPI" -->
-            <param lpi_mode="NON_LPI" />
+            <!-- lpi_mode: "NON_LPI_BARGE_IN", "NON_LPI", "LPI" -->
+            <!-- NON_LPI_BARGE_IN: Default non-LPI mode type. lsm_usecase -->
+            <!-- must be present with this mode type to handle barge-in. -->
+            <!-- NON_LPI: If another lsm_usecase is present with this mode -->
+            <!-- type, it will be used for non-LPI non-barge-in usecases. -->
+            <!-- If not present, NON_LPI_BARGE_IN mode type will be used. -->
+            <!-- LPI: This mode type will be used for LPI usecases. -->
+            <param lpi_mode="NON_LPI_BARGE_IN" />
             <param app_type="2" /> <!-- app type used in ACDB -->
             <param in_channels="5"/> <!-- Module input channels -->
             <param load_sound_model_ids="0x00012C1C, 0x0, 0x00012C14" />
@@ -144,7 +151,13 @@
             <!-- is FFECNS -->
             <param fluence_type="FLUENCE_TMIC" />
             <param execution_mode="ADSP" />
-            <!-- lpi_mode: "NON_LPI", "LPI" -->
+            <!-- lpi_mode: "NON_LPI_BARGE_IN", "NON_LPI", "LPI" -->
+            <!-- NON_LPI_BARGE_IN: Default non-LPI mode type. lsm_usecase -->
+            <!-- must be present with this mode type to handle barge-in. -->
+            <!-- NON_LPI: If another lsm_usecase is present with this mode -->
+            <!-- type, it will be used for non-LPI non-barge-in usecases. -->
+            <!-- If not present, NON_LPI_BARGE_IN mode type will be used. -->
+            <!-- LPI: This mode type will be used for LPI usecases. -->
             <param lpi_mode="LPI" />
             <param app_type="2" /> <!-- app type used in ACDB -->
             <param in_channels="3"/> <!-- Module input channels -->
diff --git a/configs/lito/mixer_paths_qrd.xml b/configs/lito/mixer_paths_qrd.xml
index b6bbddc..7870f82 100644
--- a/configs/lito/mixer_paths_qrd.xml
+++ b/configs/lito/mixer_paths_qrd.xml
@@ -3291,6 +3291,16 @@
         <path name="three-mic" />
     </path>
 
+    <path name="three-mic">
+        <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
+        <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
+        <ctl name="TX DMIC MUX3" value="DMIC1" />
+        <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
+        <ctl name="TX DMIC MUX1" value="DMIC3" />
+        <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
+        <ctl name="TX DMIC MUX2" value="DMIC2" />
+    </path>
+
     <path name="speaker-tmic">
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
         <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
diff --git a/configs/msmnile_au/audio_platform_info.xml b/configs/msmnile_au/audio_platform_info.xml
index 55f0d66..4fdf786 100644
--- a/configs/msmnile_au/audio_platform_info.xml
+++ b/configs/msmnile_au/audio_platform_info.xml
@@ -35,8 +35,8 @@
         <device name="SND_DEVICE_OUT_BUS_NAV" acdb_id="14"/>
         <device name="SND_DEVICE_OUT_BUS_PHN" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_BUS_RSE" acdb_id="60"/>
-        <device name="SND_DEVICE_OUT_BT_SCO" acdb_id="94"/>
-        <device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="94"/>
+        <device name="SND_DEVICE_OUT_BT_SCO" acdb_id="21"/>
+        <device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="39"/>
         <device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="60"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" acdb_id="94"/>
@@ -49,8 +49,8 @@
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" acdb_id="95"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC" acdb_id="95"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" acdb_id="95"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" acdb_id="20"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" acdb_id="38"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" acdb_id="80"/>
         <device name="SND_DEVICE_IN_SPEAKER_DMIC" acdb_id="80"/>
         <device name="SND_DEVICE_IN_QUAD_MIC" acdb_id="47"/>
@@ -111,10 +111,14 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_MMAP" type="out" id="28" />
         <usecase name="USECASE_AUDIO_RECORD_MMAP" type="in" id="28" />
         <usecase name="USECASE_AUDIO_RECORD" type="in" id="0" />
-        <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="36" />
-        <usecase name="USECASE_AUDIO_HFP_SCO" type="out" id="36" />
-        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="36" />
-        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="out" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO" type="out" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="out" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_DOWNLINK" type="in" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_DOWNLINK" type="out" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB_DOWNLINK" type="in" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB_DOWNLINK" type="out" id="36" />
     </pcm_ids>
 
     <config_params>
@@ -131,6 +135,7 @@
         <param key="true_32_bit" value="true"/>
         <!-- In the below value string, the value indicates sidetone gain in dB -->
         <param key="usb_sidetone_gain" value="35"/>
+        <param key="hfp_pcm_dev_id" value="36"/>
     </config_params>
 
     <gain_db_to_level_mapping>
@@ -145,8 +150,8 @@
         <device name="SND_DEVICE_OUT_HANDSET" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_SPEAKER" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_HEADPHONES" interface="TERT_TDM_RX_0"/>
-        <device name="SND_DEVICE_OUT_BT_SCO" interface="TERT_TDM_RX_2"/>
-        <device name="SND_DEVICE_OUT_BT_SCO_WB" interface="TERT_TDM_RX_2"/>
+        <device name="SND_DEVICE_OUT_BT_SCO" interface="AUX_PCM_RX"/>
+        <device name="SND_DEVICE_OUT_BT_SCO_WB" interface="AUX_PCM_RX"/>
         <device name="SND_DEVICE_OUT_BT_A2DP" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="TERT_TDM_RX_2"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="TERT_TDM_RX_2"/>
@@ -163,8 +168,8 @@
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" interface="TERT_TDM_TX_0"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="TERT_TDM_TX_0"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="TERT_TDM_TX_0"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="AUX_PCM_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="AUX_PCM_TX"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_SPEAKER_DMIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_QUAD_MIC" interface="TERT_TDM_TX_0"/>
diff --git a/configs/msmnile_au/mixer_paths_adp.xml b/configs/msmnile_au/mixer_paths_adp.xml
index 02dd367..566f1e9 100644
--- a/configs/msmnile_au/mixer_paths_adp.xml
+++ b/configs/msmnile_au/mixer_paths_adp.xml
@@ -1601,8 +1601,6 @@
     </path>
 
     <path name="hfp-sco">
-        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
-        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
         <ctl name="AUX_PCM_RX Audio Mixer MultiMedia6" value="1" />
         <ctl name="TERT_TDM_TX_0 Channels" value="One" />
         <ctl name="MultiMedia6 Mixer TERT_TDM_TX_0" value="1" />
@@ -1630,20 +1628,38 @@
         <path name="hfp-sco headphones" />
     </path>
 
+    <path name="hfp-sco bt-sco">
+        <path name="hfp-sco" />
+    </path>
+
     <path name="hfp-sco-wb">
-        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
         <ctl name="PRIM_AUX_PCM_RX SampleRate" value="KHZ_16" />
         <path name="hfp-sco" />
     </path>
 
+    <path name="hfp-sco-wb bt-sco-wb">
+        <path name="hfp-sco-wb" />
+    </path>
+
     <path name="hfp-sco-wb headphones">
         <path name="hfp-sco headphones" />
     </path>
 
+
     <path name="hfp-sco-wb headset">
         <path name="hfp-sco headphones" />
     </path>
 
+    <path name="hfp-sco-downlink">
+        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
+        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
+    </path>
+
+    <path name="hfp-sco-wb-downlink">
+        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
+        <path name="hfp-sco-downlink" />
+    </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" />
@@ -2105,6 +2121,10 @@
         <path name="speaker-mono" />
     </path>
 
+    <path name="voice-speaker-hfp">
+        <path name="speaker-mono" />
+    </path>
+
     <path name="voice-speaker-2">
         <path name="speaker-mono-2" />
     </path>
@@ -2121,6 +2141,10 @@
         <path name="speaker-mic-qrd" />
     </path>
 
+    <path name="voice-speaker-mic-hfp">
+        <path name="speaker-mic" />
+    </path>
+
     <path name="voice-headphones">
         <path name="headphones" />
     </path>
diff --git a/configs/msmnile_au/mixer_paths_custom.xml b/configs/msmnile_au/mixer_paths_custom.xml
index 8ededba..2136c79 100644
--- a/configs/msmnile_au/mixer_paths_custom.xml
+++ b/configs/msmnile_au/mixer_paths_custom.xml
@@ -1589,8 +1589,6 @@
     </path>
 
     <path name="hfp-sco">
-        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
-        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
         <ctl name="AUX_PCM_RX Audio Mixer MultiMedia6" value="1" />
         <ctl name="TERT_TDM_TX_0 Channels" value="One" />
         <ctl name="MultiMedia6 Mixer TERT_TDM_TX_0" value="1" />
@@ -1618,20 +1616,38 @@
         <path name="hfp-sco headphones" />
     </path>
 
+    <path name="hfp-sco bt-sco">
+        <path name="hfp-sco" />
+    </path>
+
     <path name="hfp-sco-wb">
-        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
         <ctl name="PRIM_AUX_PCM_RX SampleRate" value="KHZ_16" />
         <path name="hfp-sco" />
     </path>
 
+    <path name="hfp-sco-wb bt-sco-wb">
+        <path name="hfp-sco-wb" />
+    </path>
+
     <path name="hfp-sco-wb headphones">
         <path name="hfp-sco headphones" />
     </path>
 
+
     <path name="hfp-sco-wb headset">
         <path name="hfp-sco headphones" />
     </path>
 
+    <path name="hfp-sco-downlink">
+        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
+        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
+    </path>
+
+    <path name="hfp-sco-wb-downlink">
+        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
+        <path name="hfp-sco-downlink" />
+    </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" />
@@ -2093,6 +2109,10 @@
         <path name="speaker-mono" />
     </path>
 
+    <path name="voice-speaker-hfp">
+        <path name="speaker-mono" />
+    </path>
+
     <path name="voice-speaker-2">
         <path name="speaker-mono-2" />
     </path>
@@ -2109,6 +2129,10 @@
         <path name="speaker-mic-qrd" />
     </path>
 
+    <path name="voice-speaker-mic-hfp">
+        <path name="speaker-mic" />
+    </path>
+
     <path name="voice-headphones">
         <path name="headphones" />
     </path>
diff --git a/configs/msmsteppe_au/audio_platform_info.xml b/configs/msmsteppe_au/audio_platform_info.xml
index 55f0d66..4fdf786 100644
--- a/configs/msmsteppe_au/audio_platform_info.xml
+++ b/configs/msmsteppe_au/audio_platform_info.xml
@@ -35,8 +35,8 @@
         <device name="SND_DEVICE_OUT_BUS_NAV" acdb_id="14"/>
         <device name="SND_DEVICE_OUT_BUS_PHN" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_BUS_RSE" acdb_id="60"/>
-        <device name="SND_DEVICE_OUT_BT_SCO" acdb_id="94"/>
-        <device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="94"/>
+        <device name="SND_DEVICE_OUT_BT_SCO" acdb_id="21"/>
+        <device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="39"/>
         <device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="60"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" acdb_id="94"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" acdb_id="94"/>
@@ -49,8 +49,8 @@
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" acdb_id="95"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC" acdb_id="95"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" acdb_id="95"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" acdb_id="20"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" acdb_id="38"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" acdb_id="80"/>
         <device name="SND_DEVICE_IN_SPEAKER_DMIC" acdb_id="80"/>
         <device name="SND_DEVICE_IN_QUAD_MIC" acdb_id="47"/>
@@ -111,10 +111,14 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_MMAP" type="out" id="28" />
         <usecase name="USECASE_AUDIO_RECORD_MMAP" type="in" id="28" />
         <usecase name="USECASE_AUDIO_RECORD" type="in" id="0" />
-        <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="36" />
-        <usecase name="USECASE_AUDIO_HFP_SCO" type="out" id="36" />
-        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="36" />
-        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="out" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO" type="out" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="out" id="29" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_DOWNLINK" type="in" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_DOWNLINK" type="out" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB_DOWNLINK" type="in" id="36" />
+        <usecase name="USECASE_AUDIO_HFP_SCO_WB_DOWNLINK" type="out" id="36" />
     </pcm_ids>
 
     <config_params>
@@ -131,6 +135,7 @@
         <param key="true_32_bit" value="true"/>
         <!-- In the below value string, the value indicates sidetone gain in dB -->
         <param key="usb_sidetone_gain" value="35"/>
+        <param key="hfp_pcm_dev_id" value="36"/>
     </config_params>
 
     <gain_db_to_level_mapping>
@@ -145,8 +150,8 @@
         <device name="SND_DEVICE_OUT_HANDSET" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_SPEAKER" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_HEADPHONES" interface="TERT_TDM_RX_0"/>
-        <device name="SND_DEVICE_OUT_BT_SCO" interface="TERT_TDM_RX_2"/>
-        <device name="SND_DEVICE_OUT_BT_SCO_WB" interface="TERT_TDM_RX_2"/>
+        <device name="SND_DEVICE_OUT_BT_SCO" interface="AUX_PCM_RX"/>
+        <device name="SND_DEVICE_OUT_BT_SCO_WB" interface="AUX_PCM_RX"/>
         <device name="SND_DEVICE_OUT_BT_A2DP" interface="TERT_TDM_RX_0"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="TERT_TDM_RX_2"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="TERT_TDM_RX_2"/>
@@ -163,8 +168,8 @@
         <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" interface="TERT_TDM_TX_0"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="TERT_TDM_TX_0"/>
-        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="TERT_TDM_TX_0"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC" interface="AUX_PCM_TX"/>
+        <device name="SND_DEVICE_IN_BT_SCO_MIC_WB" interface="AUX_PCM_TX"/>
         <device name="SND_DEVICE_IN_HANDSET_DMIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_SPEAKER_DMIC" interface="TERT_TDM_TX_0"/>
         <device name="SND_DEVICE_IN_QUAD_MIC" interface="TERT_TDM_TX_0"/>
diff --git a/configs/msmsteppe_au/mixer_paths_adp.xml b/configs/msmsteppe_au/mixer_paths_adp.xml
index 6cebd13..7622dd8 100644
--- a/configs/msmsteppe_au/mixer_paths_adp.xml
+++ b/configs/msmsteppe_au/mixer_paths_adp.xml
@@ -1372,8 +1372,6 @@
     </path>
 
     <path name="hfp-sco">
-        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
-        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
         <ctl name="AUX_PCM_RX Audio Mixer MultiMedia6" value="1" />
         <ctl name="TERT_TDM_TX_0 Channels" value="One" />
         <ctl name="MultiMedia6 Mixer TERT_TDM_TX_0" value="1" />
@@ -1397,16 +1395,33 @@
     <path name="hfp-sco headphones">
     </path>
 
+    <path name="hfp-sco bt-sco">
+        <path name="hfp-sco" />
+    </path>
+
     <path name="hfp-sco-wb">
-        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
         <ctl name="PRIM_AUX_PCM_RX SampleRate" value="KHZ_16" />
         <path name="hfp-sco" />
     </path>
 
+    <path name="hfp-sco-wb bt-sco-wb">
+        <path name="hfp-sco-wb" />
+    </path>
+
     <path name="hfp-sco-wb headphones">
         <path name="hfp-sco headphones" />
     </path>
 
+    <path name="hfp-sco-downlink">
+        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
+        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
+    </path>
+
+    <path name="hfp-sco-wb-downlink">
+        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
+        <path name="hfp-sco-downlink" />
+    </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" />
@@ -1831,6 +1846,10 @@
         <path name="speaker-mono" />
     </path>
 
+    <path name="voice-speaker-hfp">
+        <path name="speaker-mono" />
+    </path>
+
     <path name="voice-speaker-2">
         <path name="speaker-mono-2" />
     </path>
@@ -1847,6 +1866,10 @@
         <path name="speaker-mic-qrd" />
     </path>
 
+    <path name="voice-speaker-mic-hfp">
+        <path name="speaker-mic" />
+    </path>
+
     <path name="voice-headphones">
         <path name="headphones" />
     </path>
diff --git a/configs/msmsteppe_au/mixer_paths_custom.xml b/configs/msmsteppe_au/mixer_paths_custom.xml
index 8620565..dd0da54 100644
--- a/configs/msmsteppe_au/mixer_paths_custom.xml
+++ b/configs/msmsteppe_au/mixer_paths_custom.xml
@@ -1361,8 +1361,6 @@
     </path>
 
     <path name="hfp-sco">
-        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
-        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
         <ctl name="AUX_PCM_RX Audio Mixer MultiMedia6" value="1" />
         <ctl name="TERT_TDM_TX_0 Channels" value="One" />
         <ctl name="MultiMedia6 Mixer TERT_TDM_TX_0" value="1" />
@@ -1386,16 +1384,33 @@
     <path name="hfp-sco headphones">
     </path>
 
+    <path name="hfp-sco bt-sco">
+        <path name="hfp-sco" />
+    </path>
+
     <path name="hfp-sco-wb">
-        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
         <ctl name="PRIM_AUX_PCM_RX SampleRate" value="KHZ_16" />
         <path name="hfp-sco" />
     </path>
 
+    <path name="hfp-sco-wb bt-sco-wb">
+        <path name="hfp-sco-wb" />
+    </path>
+
     <path name="hfp-sco-wb headphones">
         <path name="hfp-sco headphones" />
     </path>
 
+    <path name="hfp-sco-downlink">
+        <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="1" />
+        <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="1" />
+    </path>
+
+    <path name="hfp-sco-wb-downlink">
+        <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
+        <path name="hfp-sco-downlink" />
+    </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" />
@@ -1820,6 +1835,10 @@
         <path name="speaker-mono" />
     </path>
 
+    <path name="voice-speaker-hfp">
+        <path name="speaker-mono" />
+    </path>
+
     <path name="voice-speaker-2">
         <path name="speaker-mono-2" />
     </path>
@@ -1836,6 +1855,10 @@
         <path name="speaker-mic-qrd" />
     </path>
 
+    <path name="voice-speaker-mic-hfp">
+        <path name="speaker-mic" />
+    </path>
+
     <path name="voice-headphones">
         <path name="headphones" />
     </path>
diff --git a/configs/sdm710/audio_platform_info.xml b/configs/sdm710/audio_platform_info.xml
index 553e02f..0932e7c 100644
--- a/configs/sdm710/audio_platform_info.xml
+++ b/configs/sdm710/audio_platform_info.xml
@@ -88,7 +88,7 @@
         <usecase name="USECASE_AUDIO_PLAYBACK_FM" type="out" id="5"/>
         <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="37"/>
+        <usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="42"/>
         <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" />
diff --git a/configs/trinket/audio_policy_configuration.xml b/configs/trinket/audio_policy_configuration.xml
index 6d92ad5..6c36e0c 100644
--- a/configs/trinket/audio_policy_configuration.xml
+++ b/configs/trinket/audio_policy_configuration.xml
@@ -178,11 +178,6 @@
                              samplingRates="8000,16000,48000"
                              channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </mixPort>
-                <mixPort name="surround_sound" role="sink" maxActiveCount="2">
-                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
-                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3,AUDIO_CHANNEL_INDEX_MASK_4,AUDIO_CHANNEL_IN_5POINT1,AUDIO_CHANNEL_INDEX_MASK_6"/>
-                </mixPort>
                 <mixPort name="usb_surround_sound" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
@@ -362,8 +357,6 @@
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
                 <route type="mix" sink="voip_tx"
                        sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
-                <route type="mix" sink="surround_sound"
-                       sources="Built-In Mic,Built-In Back Mic"/>
                 <route type="mix" sink="usb_surround_sound"
                        sources="USB Device In,USB Headset In"/>
                 <route type="mix" sink="record_24"
diff --git a/hal/audio_extn/Android.mk b/hal/audio_extn/Android.mk
index 56d9179..ccc1b2e 100644
--- a/hal/audio_extn/Android.mk
+++ b/hal/audio_extn/Android.mk
@@ -143,7 +143,8 @@
     libtinycompress \
     libaudioroute \
     libdl \
-    libexpat
+    libexpat \
+    libprocessgroup
 
 LOCAL_C_INCLUDES := \
     vendor/qcom/opensource/audio-hal/primary-hal/hal \
@@ -168,7 +169,7 @@
 
 LOCAL_HEADER_LIBRARIES += libhardware_headers
 LOCAL_HEADER_LIBRARIES += libsystem_headers
-#include $(BUILD_SHARED_LIBRARY)
+include $(BUILD_SHARED_LIBRARY)
 
 #--------------------------------------------
 #          Build HDMI_EDID LIB
@@ -475,6 +476,10 @@
   MULTIPLE_HW_VARIANTS_ENABLED := true
 endif
 
+ifeq ($(TARGET_BOARD_AUTO),true)
+  LOCAL_CFLAGS += -DPLATFORM_AUTO
+endif
+
 LOCAL_SRC_FILES:= \
         hfp.c
 
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 8512a4e..11ee448 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -68,7 +68,7 @@
 #define MAX_NUM_CHANNELS 8
 #define Q14_GAIN_UNITY 0x4000
 
-static int is_running_vendor_enhanced_fwk = 0;
+static int  vendor_enhanced_info = 0;
 static bool is_compress_meta_data_enabled = false;
 
 struct snd_card_split cur_snd_card_split = {
@@ -1125,8 +1125,10 @@
 
 bool audio_extn_get_anc_enabled(void)
 {
-    ALOGD("%s: anc_enabled:%d", __func__, aextnmod.anc_enabled);
-    return (aextnmod.anc_enabled ? true: false);
+    ALOGD("%s: anc_enabled:%d", __func__,
+        (aextnmod.anc_enabled && audio_extn_anc_headset_feature_enabled));
+    return (aextnmod.anc_enabled &&
+        audio_extn_anc_headset_feature_enabled);
 }
 
 bool audio_extn_should_use_handset_anc(int in_channels)
@@ -1772,10 +1774,10 @@
 
 //START: SPEAKER_PROTECTION ==========================================================
 #ifdef __LP64__
-#define SPKR_PROT_LIB_PATH  "/vendor/lib64/libspkrprot.so"
+#define SPKR_PROT_LIB_PATH         "/vendor/lib64/libspkrprot.so"
 #define CIRRUS_SPKR_PROT_LIB_PATH  "/vendor/lib64/libcirrusspkrprot.so"
 #else
-#define SPKR_PROT_LIB_PATH  "/vendor/lib/libspkrprot.so"
+#define SPKR_PROT_LIB_PATH         "/vendor/lib/libspkrprot.so"
 #define CIRRUS_SPKR_PROT_LIB_PATH  "/vendor/lib/libcirrusspkrprot.so"
 #endif
 
@@ -1815,22 +1817,22 @@
 
 void spkr_prot_feature_init(bool is_feature_enabled)
 {
-    ALOGD("%s: Called with feature %s, is_running_with_enhanced_fwk %d", __func__,
-            is_feature_enabled?"Enabled":"NOT Enabled", is_running_vendor_enhanced_fwk);
+    ALOGD("%s: Called with feature %s, vendor_enhanced_info 0x%x", __func__,
+            is_feature_enabled ? "Enabled" : "NOT Enabled", vendor_enhanced_info);
     if (is_feature_enabled) {
-        //dlopen lib
-        if (is_running_vendor_enhanced_fwk)
-            spkr_prot_lib_handle = dlopen(SPKR_PROT_LIB_PATH, RTLD_NOW);
-        else
+        // dlopen lib
+        if ((vendor_enhanced_info & 0x3) == 0x0) // Pure AOSP
             spkr_prot_lib_handle = dlopen(CIRRUS_SPKR_PROT_LIB_PATH, RTLD_NOW);
+        else
+            spkr_prot_lib_handle = dlopen(SPKR_PROT_LIB_PATH, RTLD_NOW);
 
         if (spkr_prot_lib_handle == NULL) {
             ALOGE("%s: dlopen failed", __func__);
             goto feature_disabled;
         }
-        //map each function
-        //if mandatoy functions are not found, disble feature
 
+        // map each function
+        // if mandatoy functions are not found, disble feature
         // Mandatory functions
         if (((spkr_prot_init =
              (spkr_prot_init_t)dlsym(spkr_prot_lib_handle, "spkr_prot_init")) == NULL) ||
@@ -1851,7 +1853,6 @@
         }
 
         // optional functions, can be NULL
-
         spkr_prot_set_parameters = NULL;
         fbsp_set_parameters = NULL;
         fbsp_get_parameters = NULL;
@@ -4509,6 +4510,10 @@
         init_config.fp_disable_audio_route = disable_audio_route;
         init_config.fp_disable_snd_device = disable_snd_device;
         init_config.fp_voice_get_mic_mute = voice_get_mic_mute;
+        init_config.fp_audio_extn_auto_hal_start_hfp_downlink =
+                                        audio_extn_auto_hal_start_hfp_downlink;
+        init_config.fp_audio_extn_auto_hal_stop_hfp_downlink =
+                                        audio_extn_auto_hal_stop_hfp_downlink;
 
         hfp_init(init_config);
         ALOGD("%s:: ---- Feature HFP is Enabled ----", __func__);
@@ -5461,10 +5466,6 @@
 typedef bool (*auto_hal_is_bus_device_usecase_t)(audio_usecase_t);
 static auto_hal_is_bus_device_usecase_t auto_hal_is_bus_device_usecase;
 
-typedef snd_device_t (*auto_hal_get_snd_device_for_car_audio_stream_t)(
-                                struct stream_out*);
-static auto_hal_get_snd_device_for_car_audio_stream_t auto_hal_get_snd_device_for_car_audio_stream;
-
 typedef int (*auto_hal_get_audio_port_t)(struct audio_hw_device*,
                                 struct audio_port*);
 static auto_hal_get_audio_port_t auto_hal_get_audio_port;
@@ -5477,6 +5478,22 @@
                                 struct str_parms*);
 static auto_hal_set_parameters_t auto_hal_set_parameters;
 
+typedef int (*auto_hal_start_hfp_downlink_t)(struct audio_device*,
+                                struct audio_usecase*);
+static auto_hal_start_hfp_downlink_t auto_hal_start_hfp_downlink;
+
+typedef int (*auto_hal_stop_hfp_downlink_t)(struct audio_device*,
+                                struct audio_usecase*);
+static auto_hal_stop_hfp_downlink_t auto_hal_stop_hfp_downlink;
+
+typedef snd_device_t (*auto_hal_get_input_snd_device_t)(struct audio_device*,
+                                audio_usecase_t);
+static auto_hal_get_input_snd_device_t auto_hal_get_input_snd_device;
+
+typedef snd_device_t (*auto_hal_get_output_snd_device_t)(struct audio_device*,
+                                audio_usecase_t);
+static auto_hal_get_output_snd_device_t auto_hal_get_output_snd_device;
+
 int auto_hal_feature_init(bool is_feature_enabled)
 {
     ALOGD("%s: Called with feature %s", __func__,
@@ -5509,9 +5526,6 @@
             !(auto_hal_is_bus_device_usecase =
                  (auto_hal_is_bus_device_usecase_t)dlsym(
                             auto_hal_lib_handle, "auto_hal_is_bus_device_usecase")) ||
-            !(auto_hal_get_snd_device_for_car_audio_stream =
-                 (auto_hal_get_snd_device_for_car_audio_stream_t)dlsym(
-                            auto_hal_lib_handle, "auto_hal_get_snd_device_for_car_audio_stream")) ||
             !(auto_hal_get_audio_port =
                  (auto_hal_get_audio_port_t)dlsym(
                             auto_hal_lib_handle, "auto_hal_get_audio_port")) ||
@@ -5520,7 +5534,19 @@
                             auto_hal_lib_handle, "auto_hal_set_audio_port_config")) ||
             !(auto_hal_set_parameters =
                  (auto_hal_set_parameters_t)dlsym(
-                            auto_hal_lib_handle, "auto_hal_set_parameters"))) {
+                            auto_hal_lib_handle, "auto_hal_set_parameters")) ||
+            !(auto_hal_start_hfp_downlink =
+                 (auto_hal_start_hfp_downlink_t)dlsym(
+                            auto_hal_lib_handle, "auto_hal_start_hfp_downlink")) ||
+            !(auto_hal_stop_hfp_downlink =
+                 (auto_hal_stop_hfp_downlink_t)dlsym(
+                            auto_hal_lib_handle, "auto_hal_stop_hfp_downlink")) ||
+            !(auto_hal_get_input_snd_device =
+                 (auto_hal_get_input_snd_device_t)dlsym(
+                            auto_hal_lib_handle, "auto_hal_get_input_snd_device")) ||
+            !(auto_hal_get_output_snd_device =
+                 (auto_hal_get_output_snd_device_t)dlsym(
+                            auto_hal_lib_handle, "auto_hal_get_output_snd_device"))) {
             ALOGE("%s: dlsym failed", __func__);
             goto feature_disabled;
         }
@@ -5541,10 +5567,13 @@
     auto_hal_get_car_audio_stream_from_address = NULL;
     auto_hal_open_output_stream = NULL;
     auto_hal_is_bus_device_usecase = NULL;
-    auto_hal_get_snd_device_for_car_audio_stream = NULL;
     auto_hal_get_audio_port = NULL;
     auto_hal_set_audio_port_config = NULL;
     auto_hal_set_parameters = NULL;
+    auto_hal_start_hfp_downlink = NULL;
+    auto_hal_stop_hfp_downlink = NULL;
+    auto_hal_get_input_snd_device = NULL;
+    auto_hal_get_output_snd_device = NULL;
 
     ALOGW(":: %s: ---- Feature AUTO_HAL is disabled ----", __func__);
     return -ENOSYS;
@@ -5561,6 +5590,11 @@
         auto_hal_init_config.fp_get_usecase_from_list = get_usecase_from_list;
         auto_hal_init_config.fp_get_output_period_size = get_output_period_size;
         auto_hal_init_config.fp_audio_extn_ext_hw_plugin_set_audio_gain = audio_extn_ext_hw_plugin_set_audio_gain;
+        auto_hal_init_config.fp_select_devices = select_devices;
+        auto_hal_init_config.fp_disable_audio_route = disable_audio_route;
+        auto_hal_init_config.fp_disable_snd_device = disable_snd_device;
+        auto_hal_init_config.fp_adev_get_active_input = adev_get_active_input;
+        auto_hal_init_config.fp_platform_set_echo_reference = platform_set_echo_reference;
         return auto_hal_init(adev, auto_hal_init_config);
     }
     else
@@ -5599,25 +5633,19 @@
 int audio_extn_auto_hal_get_car_audio_stream_from_address(const char *address)
 {
     return ((auto_hal_get_car_audio_stream_from_address) ?
-                            auto_hal_get_car_audio_stream_from_address(address): 0);
+                            auto_hal_get_car_audio_stream_from_address(address): -ENOSYS);
 }
 
 int audio_extn_auto_hal_open_output_stream(struct stream_out *out)
 {
     return ((auto_hal_open_output_stream) ?
-                            auto_hal_open_output_stream(out): 0);
+                            auto_hal_open_output_stream(out): -ENOSYS);
 }
 
 bool audio_extn_auto_hal_is_bus_device_usecase(audio_usecase_t uc_id)
 {
     return ((auto_hal_is_bus_device_usecase) ?
-                            auto_hal_is_bus_device_usecase(uc_id): 0);
-}
-
-snd_device_t audio_extn_auto_hal_get_snd_device_for_car_audio_stream(struct stream_out *out)
-{
-    return ((auto_hal_get_snd_device_for_car_audio_stream) ?
-                            auto_hal_get_snd_device_for_car_audio_stream(out): 0);
+                            auto_hal_is_bus_device_usecase(uc_id): false);
 }
 
 int audio_extn_auto_hal_get_audio_port(struct audio_hw_device *dev,
@@ -5640,11 +5668,39 @@
     if (auto_hal_set_parameters)
         auto_hal_set_parameters(adev, parms);
 }
+
+int audio_extn_auto_hal_start_hfp_downlink(struct audio_device *adev,
+                                struct audio_usecase *uc_info)
+{
+    return ((auto_hal_start_hfp_downlink) ?
+                            auto_hal_start_hfp_downlink(adev, uc_info): 0);
+}
+
+int audio_extn_auto_hal_stop_hfp_downlink(struct audio_device *adev,
+                                struct audio_usecase *uc_info)
+{
+    return ((auto_hal_stop_hfp_downlink) ?
+                            auto_hal_stop_hfp_downlink(adev, uc_info): 0);
+}
+
+snd_device_t audio_extn_auto_hal_get_input_snd_device(struct audio_device *adev,
+                                audio_usecase_t uc_id)
+{
+    return ((auto_hal_get_input_snd_device) ?
+                            auto_hal_get_input_snd_device(adev, uc_id): SND_DEVICE_NONE);
+}
+
+snd_device_t audio_extn_auto_hal_get_output_snd_device(struct audio_device *adev,
+                                audio_usecase_t uc_id)
+{
+    return ((auto_hal_get_output_snd_device) ?
+                            auto_hal_get_output_snd_device(adev, uc_id): SND_DEVICE_NONE);
+}
 // END: AUTO_HAL ===================================================================
 
 void audio_extn_feature_init()
 {
-    is_running_vendor_enhanced_fwk = audio_extn_utils_is_vendor_enhanced_fwk();
+    vendor_enhanced_info = audio_extn_utils_get_vendor_enhanced_info();
 
     // register feature init functions here
     // each feature needs a vendor property
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 91204a7..db769da 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -682,6 +682,10 @@
                                                 struct audio_usecase *);
 typedef int (*fp_disable_snd_device_t)(struct audio_device *, snd_device_t);
 typedef bool (*fp_voice_get_mic_mute_t)(struct audio_device *);
+typedef int (*fp_audio_extn_auto_hal_start_hfp_downlink_t)(struct audio_device *,
+                                                        struct audio_usecase *);
+typedef int (*fp_audio_extn_auto_hal_stop_hfp_downlink_t)(struct audio_device *,
+                                                        struct audio_usecase *);
 
 typedef struct hfp_init_config {
     fp_platform_set_mic_mute_t                   fp_platform_set_mic_mute;
@@ -694,6 +698,8 @@
     fp_disable_audio_route_t                     fp_disable_audio_route;
     fp_disable_snd_device_t                      fp_disable_snd_device;
     fp_voice_get_mic_mute_t                      fp_voice_get_mic_mute;
+    fp_audio_extn_auto_hal_start_hfp_downlink_t  fp_audio_extn_auto_hal_start_hfp_downlink;
+    fp_audio_extn_auto_hal_stop_hfp_downlink_t   fp_audio_extn_auto_hal_stop_hfp_downlink;
 } hfp_init_config_t;
 
 
@@ -814,7 +820,8 @@
 int audio_extn_utils_get_sample_rate_from_string(const char *);
 int audio_extn_utils_get_channels_from_string(const char *);
 void audio_extn_utils_release_snd_device(snd_device_t snd_device);
-int audio_extn_utils_is_vendor_enhanced_fwk();
+bool audio_extn_utils_is_vendor_enhanced_fwk();
+int audio_extn_utils_get_vendor_enhanced_info();
 int audio_extn_utils_get_app_sample_rate_for_device(struct audio_device *adev,
                                     struct audio_usecase *usecase, int snd_device);
 
@@ -1309,18 +1316,26 @@
 int audio_extn_auto_hal_get_car_audio_stream_from_address(const char *address);
 int audio_extn_auto_hal_open_output_stream(struct stream_out *out);
 bool audio_extn_auto_hal_is_bus_device_usecase(audio_usecase_t uc_id);
-snd_device_t audio_extn_auto_hal_get_snd_device_for_car_audio_stream(struct stream_out *out);
 int audio_extn_auto_hal_get_audio_port(struct audio_hw_device *dev,
                                 struct audio_port *config);
 int audio_extn_auto_hal_set_audio_port_config(struct audio_hw_device *dev,
                                 const struct audio_port_config *config);
 void audio_extn_auto_hal_set_parameters(struct audio_device *adev,
                                 struct str_parms *parms);
+int audio_extn_auto_hal_start_hfp_downlink(struct audio_device *adev,
+                                struct audio_usecase *uc_info);
+int audio_extn_auto_hal_stop_hfp_downlink(struct audio_device *adev,
+                                struct audio_usecase *uc_info);
+snd_device_t audio_extn_auto_hal_get_input_snd_device(struct audio_device *adev,
+                                audio_usecase_t uc_id);
+snd_device_t audio_extn_auto_hal_get_output_snd_device(struct audio_device *adev,
+                                audio_usecase_t uc_id);
 
 typedef streams_input_ctxt_t* (*fp_in_get_stream_t)(struct audio_device*, audio_io_handle_t);
 typedef streams_output_ctxt_t* (*fp_out_get_stream_t)(struct audio_device*, audio_io_handle_t);
 typedef size_t (*fp_get_output_period_size_t)(uint32_t, audio_format_t, int, int);
 typedef int (*fp_audio_extn_ext_hw_plugin_set_audio_gain_t)(void*, struct audio_usecase*, uint32_t);
+typedef struct stream_in* (*fp_adev_get_active_input_t)(const struct audio_device*);
 
 typedef struct auto_hal_init_config {
     fp_in_get_stream_t                           fp_in_get_stream;
@@ -1330,6 +1345,11 @@
     fp_get_usecase_from_list_t                   fp_get_usecase_from_list;
     fp_get_output_period_size_t                  fp_get_output_period_size;
     fp_audio_extn_ext_hw_plugin_set_audio_gain_t fp_audio_extn_ext_hw_plugin_set_audio_gain;
+    fp_select_devices_t                          fp_select_devices;
+    fp_disable_audio_route_t                     fp_disable_audio_route;
+    fp_disable_snd_device_t                      fp_disable_snd_device;
+    fp_adev_get_active_input_t                   fp_adev_get_active_input;
+    fp_platform_set_echo_reference_t             fp_platform_set_echo_reference;
 } auto_hal_init_config_t;
 // END: AUTO_HAL FEATURE ==================================================
 
diff --git a/hal/audio_extn/auto_hal.c b/hal/audio_extn/auto_hal.c
old mode 100644
new mode 100755
index c18b55f..84f578f
--- a/hal/audio_extn/auto_hal.c
+++ b/hal/audio_extn/auto_hal.c
@@ -54,12 +54,19 @@
 static fp_get_usecase_from_list_t                   fp_get_usecase_from_list;
 static fp_get_output_period_size_t                  fp_get_output_period_size;
 static fp_audio_extn_ext_hw_plugin_set_audio_gain_t fp_audio_extn_ext_hw_plugin_set_audio_gain;
+static fp_select_devices_t                          fp_select_devices;
+static fp_disable_audio_route_t                     fp_disable_audio_route;
+static fp_disable_snd_device_t                      fp_disable_snd_device;
+static fp_adev_get_active_input_t                   fp_adev_get_active_input;
+static fp_platform_set_echo_reference_t             fp_platform_set_echo_reference;
 
 /* Auto hal module struct */
 static struct auto_hal_module *auto_hal = NULL;
 
 int auto_hal_release_audio_patch(struct audio_hw_device *dev,
                                 audio_patch_handle_t handle);
+int auto_hal_stop_hfp_downlink(struct audio_device *adev,
+                               struct audio_usecase *uc_info);
 
 static struct audio_patch_record *get_patch_from_list(struct audio_device *adev,
                                                     audio_patch_handle_t patch_id)
@@ -151,6 +158,11 @@
                                                 sources->ext.device.address);
         } else {
             address = (char *)calloc(1, 1);
+            if (address == NULL) {
+                ALOGE("%s: failed to get address",__func__);
+                ret = -EFAULT;
+                goto error;
+            }
         }
         parms = str_parms_create_str(address);
         if (!parms) {
@@ -185,6 +197,11 @@
                                                 sinks->ext.device.address);
         } else {
             address = (char *)calloc(1, 1);
+            if (address == NULL) {
+                ALOGE("%s: failed to get address",__func__);
+                ret = -EFAULT;
+                goto error;
+            }
         }
         parms = str_parms_create_str(address);
         if (!parms) {
@@ -677,6 +694,236 @@
     ALOGV("%s: exit", __func__);
 }
 
+int auto_hal_start_hfp_downlink(struct audio_device *adev,
+                                struct audio_usecase *uc_info)
+{
+    int32_t ret = 0;
+    struct audio_usecase *uc_downlink_info;
+
+    ALOGD("%s: enter", __func__);
+
+    uc_downlink_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+
+    if (!uc_downlink_info)
+        return -ENOMEM;
+
+    uc_downlink_info->type = PCM_HFP_CALL;
+    uc_downlink_info->stream.out = adev->primary_output;
+    uc_downlink_info->devices = adev->primary_output->devices;
+    uc_downlink_info->in_snd_device = SND_DEVICE_NONE;
+    uc_downlink_info->out_snd_device = SND_DEVICE_NONE;
+
+    switch (uc_info->id) {
+    case USECASE_AUDIO_HFP_SCO:
+        uc_downlink_info->id = USECASE_AUDIO_HFP_SCO_DOWNLINK;
+        break;
+    case USECASE_AUDIO_HFP_SCO_WB:
+        uc_downlink_info->id = USECASE_AUDIO_HFP_SCO_WB_DOWNLINK;
+        break;
+    default:
+        ALOGE("%s: Invalid usecase %d", __func__, uc_info->id);
+        free(uc_downlink_info);
+        return -EINVAL;
+    }
+
+    list_add_tail(&adev->usecase_list, &uc_downlink_info->list);
+
+    ret = fp_select_devices(adev, uc_downlink_info->id);
+    if (ret) {
+        ALOGE("%s: Select devices failed %d", __func__, ret);
+        goto exit;
+    }
+
+    ALOGD("%s: exit: status(%d)", __func__, ret);
+    return 0;
+
+exit:
+    auto_hal_stop_hfp_downlink(adev, uc_info);
+    ALOGE("%s: Problem in start hfp downlink: status(%d)", __func__, ret);
+    return ret;
+}
+
+int auto_hal_stop_hfp_downlink(struct audio_device *adev,
+                               struct audio_usecase *uc_info)
+{
+    int32_t ret = 0;
+    struct audio_usecase *uc_downlink_info;
+    audio_usecase_t ucid;
+
+    ALOGD("%s: enter", __func__);
+
+    switch (uc_info->id) {
+    case USECASE_AUDIO_HFP_SCO:
+        ucid = USECASE_AUDIO_HFP_SCO_DOWNLINK;
+        break;
+    case USECASE_AUDIO_HFP_SCO_WB:
+        ucid = USECASE_AUDIO_HFP_SCO_WB_DOWNLINK;
+        break;
+    default:
+        ALOGE("%s: Invalid usecase %d", __func__, uc_info->id);
+        return -EINVAL;
+    }
+
+    uc_downlink_info = fp_get_usecase_from_list(adev, ucid);
+    if (uc_downlink_info == NULL) {
+        ALOGE("%s: Could not find the usecase (%d) in the list",
+              __func__, ucid);
+        return -EINVAL;
+    }
+
+    /* Get and set stream specific mixer controls */
+    fp_disable_audio_route(adev, uc_downlink_info);
+
+    /* Disable the rx and tx devices */
+    fp_disable_snd_device(adev, uc_downlink_info->out_snd_device);
+    fp_disable_snd_device(adev, uc_downlink_info->in_snd_device);
+
+    list_remove(&uc_downlink_info->list);
+    free(uc_downlink_info);
+
+    ALOGD("%s: exit: status(%d)", __func__, ret);
+    return ret;
+}
+
+snd_device_t auto_hal_get_input_snd_device(struct audio_device *adev,
+                                audio_usecase_t uc_id)
+{
+    snd_device_t snd_device = SND_DEVICE_NONE;
+    audio_devices_t out_device = AUDIO_DEVICE_NONE;
+    struct audio_usecase *usecase = NULL;
+    struct stream_in *in = fp_adev_get_active_input(adev);
+    audio_devices_t in_device = ((in == NULL) ?
+                                    AUDIO_DEVICE_NONE : in->device)
+                                & ~AUDIO_DEVICE_BIT_IN;
+
+    if (uc_id == USECASE_INVALID) {
+        ALOGE("%s: Invalid usecase (%d)", __func__, uc_id);
+        return -EINVAL;
+    }
+
+    usecase = fp_get_usecase_from_list(adev, uc_id);
+    if (usecase == NULL) {
+        ALOGE("%s: Could not find the usecase (%d)", __func__, uc_id);
+        return -EINVAL;
+    }
+
+    if (usecase->stream.out == NULL) {
+        ALOGE("%s: stream.out is NULL", __func__);
+        return -EINVAL;
+    }
+
+    out_device = usecase->stream.out->devices;
+    if (out_device == AUDIO_DEVICE_NONE ||
+        out_device & AUDIO_DEVICE_BIT_IN) {
+        ALOGE("%s: Invalid output devices (%#x)", __func__, out_device);
+        return -EINVAL;
+    }
+
+    ALOGV("%s: output device(%#x), input device(%#x), usecase(%d)",
+        __func__, out_device, in_device, uc_id);
+
+    if (out_device & AUDIO_DEVICE_OUT_BUS) {
+        /* usecase->id is token as judgement for HFP calls */
+        switch (usecase->id) {
+        case USECASE_AUDIO_HFP_SCO:
+        case USECASE_AUDIO_HFP_SCO_WB:
+            snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP;
+            if (adev->enable_hfp)
+                fp_platform_set_echo_reference(adev, true, out_device);
+            break;
+        case USECASE_AUDIO_HFP_SCO_DOWNLINK:
+            snd_device = SND_DEVICE_IN_BT_SCO_MIC;
+            break;
+        case USECASE_AUDIO_HFP_SCO_WB_DOWNLINK:
+            snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB;
+            break;
+        case USECASE_VOICE_CALL:
+            snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC;
+            break;
+        default:
+            ALOGE("%s: Usecase (%d) not supported", __func__, uc_id);
+            return -EINVAL;
+        }
+    } else {
+        ALOGE("%s: Output devices (%#x) not supported", __func__, out_device);
+        return -EINVAL;
+    }
+
+    return snd_device;
+}
+
+snd_device_t auto_hal_get_output_snd_device(struct audio_device *adev,
+                                audio_usecase_t uc_id)
+{
+    snd_device_t snd_device = SND_DEVICE_NONE;
+    audio_devices_t devices = AUDIO_DEVICE_NONE;
+    struct audio_usecase *usecase = NULL;
+
+    if (uc_id == USECASE_INVALID) {
+        ALOGE("%s: Invalid usecase (%d)", __func__, uc_id);
+        return -EINVAL;
+    }
+
+    usecase = fp_get_usecase_from_list(adev, uc_id);
+    if (usecase == NULL) {
+        ALOGE("%s: Could not find the usecase (%d)", __func__, uc_id);
+        return -EINVAL;
+    }
+
+    if (usecase->stream.out == NULL) {
+        ALOGE("%s: stream.out is NULL", __func__);
+        return -EINVAL;
+    }
+
+    devices = usecase->stream.out->devices;
+    if (devices == AUDIO_DEVICE_NONE ||
+        devices & AUDIO_DEVICE_BIT_IN) {
+        ALOGE("%s: Invalid output devices (%#x)", __func__, devices);
+        return -EINVAL;
+    }
+
+    ALOGV("%s: output devices(%#x), usecase(%d)", __func__, devices, uc_id);
+
+    if (devices & AUDIO_DEVICE_OUT_BUS) {
+        /* usecase->id is token as judgement for HFP calls */
+        switch (usecase->id) {
+        case USECASE_AUDIO_HFP_SCO:
+            snd_device = SND_DEVICE_OUT_BT_SCO;
+            break;
+        case USECASE_AUDIO_HFP_SCO_WB:
+            snd_device = SND_DEVICE_OUT_BT_SCO_WB;
+            break;
+        case USECASE_AUDIO_HFP_SCO_DOWNLINK:
+        case USECASE_AUDIO_HFP_SCO_WB_DOWNLINK:
+            snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_HFP;
+            break;
+        case USECASE_VOICE_CALL:
+            snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+            break;
+        case USECASE_AUDIO_PLAYBACK_MEDIA:
+            snd_device = SND_DEVICE_OUT_BUS_MEDIA;
+            break;
+        case USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION:
+            snd_device = SND_DEVICE_OUT_BUS_SYS;
+            break;
+        case USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE:
+            snd_device = SND_DEVICE_OUT_BUS_NAV;
+            break;
+        case USECASE_AUDIO_PLAYBACK_PHONE:
+            snd_device = SND_DEVICE_OUT_BUS_PHN;
+            break;
+        default:
+            ALOGE("%s: Usecase (%d) not supported", __func__, uc_id);
+            return -EINVAL;
+        }
+    } else {
+        ALOGE("%s: Output devices (%#x) not supported", __func__, devices);
+        return -EINVAL;
+    }
+
+    return snd_device;
+}
+
 int auto_hal_init(struct audio_device *adev, auto_hal_init_config_t init_config)
 {
     int ret = 0;
@@ -704,6 +951,11 @@
     fp_get_usecase_from_list = init_config.fp_get_usecase_from_list;
     fp_get_output_period_size = init_config.fp_get_output_period_size;
     fp_audio_extn_ext_hw_plugin_set_audio_gain = init_config.fp_audio_extn_ext_hw_plugin_set_audio_gain;
+    fp_select_devices = init_config.fp_select_devices;
+    fp_disable_audio_route = init_config.fp_disable_audio_route;
+    fp_disable_snd_device = init_config.fp_disable_snd_device;
+    fp_adev_get_active_input = init_config.fp_adev_get_active_input;
+    fp_platform_set_echo_reference = init_config.fp_platform_set_echo_reference;
 
     return ret;
 }
diff --git a/hal/audio_extn/ext_hw_plugin.c b/hal/audio_extn/ext_hw_plugin.c
index 7460caa..6e784cb 100644
--- a/hal/audio_extn/ext_hw_plugin.c
+++ b/hal/audio_extn/ext_hw_plugin.c
@@ -182,6 +182,8 @@
         break;
     case USECASE_AUDIO_HFP_SCO:
     case USECASE_AUDIO_HFP_SCO_WB:
+    case USECASE_AUDIO_HFP_SCO_DOWNLINK:
+    case USECASE_AUDIO_HFP_SCO_WB_DOWNLINK:
         *plugin_usecase = AUDIO_HAL_PLUGIN_USECASE_HFP_VOICE_CALL;
         break;
     case USECASE_VOICE_CALL:
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 1088a0c..9b60083 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -125,6 +125,8 @@
 static fp_disable_audio_route_t                     fp_disable_audio_route;
 static fp_disable_snd_device_t                      fp_disable_snd_device;
 static fp_voice_get_mic_mute_t                      fp_voice_get_mic_mute;
+static fp_audio_extn_auto_hal_start_hfp_downlink_t  fp_audio_extn_auto_hal_start_hfp_downlink;
+static fp_audio_extn_auto_hal_stop_hfp_downlink_t   fp_audio_extn_auto_hal_stop_hfp_downlink;
 
 static int32_t hfp_set_volume(struct audio_device *adev, float value)
 {
@@ -332,15 +334,6 @@
     ALOGD("%s: HFP PCM devices (rx: %d tx: %d pcm dev id: %d) usecase(%d)",
               __func__, pcm_dev_rx_id, pcm_dev_tx_id, hfpmod.hfp_pcm_dev_id, uc_info->id);
 
-    hfpmod.hfp_sco_rx = pcm_open(adev->snd_card,
-                                  pcm_dev_asm_rx_id,
-                                  PCM_OUT, &pcm_config_hfp);
-    if (hfpmod.hfp_sco_rx && !pcm_is_ready(hfpmod.hfp_sco_rx)) {
-        ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_sco_rx));
-        ret = -EIO;
-        goto exit;
-    }
-
     hfpmod.hfp_pcm_rx = pcm_open(adev->snd_card,
                                  pcm_dev_rx_id,
                                  PCM_OUT, &pcm_config_hfp);
@@ -350,15 +343,6 @@
         goto exit;
     }
 
-    hfpmod.hfp_sco_tx = pcm_open(adev->snd_card,
-                                  pcm_dev_asm_tx_id,
-                                  PCM_IN, &pcm_config_hfp);
-    if (hfpmod.hfp_sco_tx && !pcm_is_ready(hfpmod.hfp_sco_tx)) {
-        ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_sco_tx));
-        ret = -EIO;
-        goto exit;
-    }
-
     hfpmod.hfp_pcm_tx = pcm_open(adev->snd_card,
                                  pcm_dev_tx_id,
                                  PCM_IN, &pcm_config_hfp);
@@ -368,17 +352,6 @@
         goto exit;
     }
 
-    if (pcm_start(hfpmod.hfp_sco_rx) < 0) {
-        ALOGE("%s: pcm start for hfp sco rx failed", __func__);
-        ret = -EINVAL;
-        goto exit;
-    }
-    if (pcm_start(hfpmod.hfp_sco_tx) < 0) {
-        ALOGE("%s: pcm start for hfp sco tx failed", __func__);
-        ret = -EINVAL;
-        goto exit;
-    }
-
     if (pcm_start(hfpmod.hfp_pcm_rx) < 0) {
         ALOGE("%s: pcm start for hfp pcm rx failed", __func__);
         ret = -EINVAL;
@@ -390,6 +363,38 @@
         goto exit;
     }
 
+    if (fp_audio_extn_auto_hal_start_hfp_downlink(adev, uc_info))
+        ALOGE("%s: start hfp downlink failed", __func__);
+
+    hfpmod.hfp_sco_rx = pcm_open(adev->snd_card,
+                                  pcm_dev_asm_rx_id,
+                                  PCM_OUT, &pcm_config_hfp);
+    if (hfpmod.hfp_sco_rx && !pcm_is_ready(hfpmod.hfp_sco_rx)) {
+        ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_sco_rx));
+        ret = -EIO;
+        goto exit;
+    }
+
+    hfpmod.hfp_sco_tx = pcm_open(adev->snd_card,
+                                  pcm_dev_asm_tx_id,
+                                  PCM_IN, &pcm_config_hfp);
+    if (hfpmod.hfp_sco_tx && !pcm_is_ready(hfpmod.hfp_sco_tx)) {
+        ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_sco_tx));
+        ret = -EIO;
+        goto exit;
+    }
+
+    if (pcm_start(hfpmod.hfp_sco_rx) < 0) {
+        ALOGE("%s: pcm start for hfp sco rx failed", __func__);
+        ret = -EINVAL;
+        goto exit;
+    }
+    if (pcm_start(hfpmod.hfp_sco_tx) < 0) {
+        ALOGE("%s: pcm start for hfp sco tx failed", __func__);
+        ret = -EINVAL;
+        goto exit;
+    }
+
     hfpmod.is_hfp_running = true;
     hfp_set_volume(adev, hfpmod.hfp_volume);
 
@@ -455,6 +460,9 @@
     fp_disable_snd_device(adev, uc_info->out_snd_device);
     fp_disable_snd_device(adev, uc_info->in_snd_device);
 
+    if (fp_audio_extn_auto_hal_stop_hfp_downlink(adev, uc_info))
+        ALOGE("%s: stop hfp downlink failed", __func__);
+
     /* Set the unmute Tx mixer control */
     if (fp_voice_get_mic_mute(adev)) {
         fp_platform_set_mic_mute(adev->platform, false);
@@ -483,6 +491,10 @@
     fp_disable_audio_route = init_config.fp_disable_audio_route;
     fp_disable_snd_device = init_config.fp_disable_snd_device;
     fp_voice_get_mic_mute = init_config.fp_voice_get_mic_mute;
+    fp_audio_extn_auto_hal_start_hfp_downlink =
+                                init_config.fp_audio_extn_auto_hal_start_hfp_downlink;
+    fp_audio_extn_auto_hal_stop_hfp_downlink =
+                                init_config.fp_audio_extn_auto_hal_stop_hfp_downlink;
 }
 
 bool hfp_is_active(struct audio_device *adev)
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index a29b6e5..5bdc881 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -117,9 +117,15 @@
 #define VNDK_FWK_LIB_PATH "/vendor/lib/libqti_vndfwk_detect.so"
 #endif
 
-static void *vndk_fwk_lib_handle = NULL;
-typedef int (*vndk_fwk_isVendorEnhancedFwk_t)();
-static vndk_fwk_isVendorEnhancedFwk_t vndk_fwk_isVendorEnhancedFwk;
+typedef struct vndkfwk_s {
+    void *lib_handle;
+    int (*isVendorEnhancedFwk)(void);
+    int (*getVendorEnhancedInfo)(void);
+    const char *lib_name;
+} vndkfwk_t;
+
+static vndkfwk_t mVndkFwk = {
+    NULL, NULL, NULL, VNDK_FWK_LIB_PATH};
 
 typedef struct {
     const char *id_string;
@@ -832,6 +838,25 @@
     return adev->vr_audio_mode_enabled;
 }
 
+static void audio_extn_btsco_get_sample_rate(int snd_device, int *sample_rate)
+{
+    switch (snd_device) {
+    case SND_DEVICE_OUT_BT_SCO:
+    case SND_DEVICE_IN_BT_SCO_MIC:
+    case SND_DEVICE_IN_BT_SCO_MIC_NREC:
+        *sample_rate = 8000;
+        break;
+    case SND_DEVICE_OUT_BT_SCO_WB:
+    case SND_DEVICE_IN_BT_SCO_MIC_WB:
+    case SND_DEVICE_IN_BT_SCO_MIC_WB_NREC:
+        *sample_rate = 16000;
+        break;
+    default:
+        ALOGD("%s:Not a BT SCO device, need not update sampling rate\n", __func__);
+        break;
+    }
+}
+
 void audio_extn_utils_update_stream_app_type_cfg_for_usecase(
                                     struct audio_device *adev,
                                     struct audio_usecase *usecase)
@@ -880,31 +905,46 @@
                                                 &usecase->stream.inout->out_app_type_cfg);
         ALOGV("%s Selected apptype: %d", __func__, usecase->stream.inout->out_app_type_cfg.app_type);
         break;
+    case PCM_HFP_CALL:
+        switch (usecase->id) {
+        case USECASE_AUDIO_HFP_SCO:
+        case USECASE_AUDIO_HFP_SCO_WB:
+            audio_extn_btsco_get_sample_rate(usecase->out_snd_device,
+                                             &usecase->out_app_type_cfg.sample_rate);
+            usecase->in_app_type_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+            break;
+        case USECASE_AUDIO_HFP_SCO_DOWNLINK:
+        case USECASE_AUDIO_HFP_SCO_WB_DOWNLINK:
+            audio_extn_btsco_get_sample_rate(usecase->in_snd_device,
+                                             &usecase->in_app_type_cfg.sample_rate);
+            usecase->out_app_type_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+            break;
+        default:
+            ALOGE("%s: usecase id (%d) not supported, use default sample rate",
+                __func__, usecase->id);
+            usecase->in_app_type_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+            usecase->out_app_type_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+            break;
+        }
+        /* update out_app_type_cfg */
+        usecase->out_app_type_cfg.bit_width =
+                                platform_get_snd_device_bit_width(usecase->out_snd_device);
+        usecase->out_app_type_cfg.app_type =
+                                platform_get_default_app_type_v2(adev->platform, PCM_PLAYBACK);
+        /* update in_app_type_cfg */
+        usecase->in_app_type_cfg.bit_width =
+                                platform_get_snd_device_bit_width(usecase->in_snd_device);
+        usecase->in_app_type_cfg.app_type =
+                                platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE);
+        ALOGV("%s Selected apptype: playback %d capture %d",
+            __func__, usecase->out_app_type_cfg.app_type, usecase->in_app_type_cfg.app_type);
+        break;
     default:
         ALOGE("%s: app type cfg not supported for usecase type (%d)",
             __func__, usecase->type);
     }
 }
 
-void audio_extn_btsco_get_sample_rate(int snd_device, int *sample_rate)
-{
-    switch (snd_device) {
-    case SND_DEVICE_OUT_BT_SCO:
-    case SND_DEVICE_IN_BT_SCO_MIC:
-    case SND_DEVICE_IN_BT_SCO_MIC_NREC:
-        *sample_rate = 8000;
-        break;
-    case SND_DEVICE_OUT_BT_SCO_WB:
-    case SND_DEVICE_IN_BT_SCO_MIC_WB:
-    case SND_DEVICE_IN_BT_SCO_MIC_WB_NREC:
-        *sample_rate = 16000;
-        break;
-    default:
-        ALOGD("%s:Not a BT SCO device, need not update sampling rate\n", __func__);
-        break;
-    }
-}
-
 static int set_stream_app_type_mixer_ctrl(struct audio_device *adev,
                                           int pcm_device_id, int app_type,
                                           int acdb_dev_id, int sample_rate,
@@ -955,21 +995,27 @@
     int pcm_device_id, acdb_dev_id = 0, snd_device = usecase->out_snd_device;
     int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
     int app_type = 0, rc = 0;
+    bool is_bus_dev_usecase = false;
 
     ALOGV("%s", __func__);
 
     if (usecase->type != PCM_HFP_CALL) {
-        ALOGV("%s: not a playback or HFP path, no need to cfg app type", __func__);
+        ALOGV("%s: not a HFP path, no need to cfg app type", __func__);
         rc = 0;
         goto exit_send_app_type_cfg;
     }
     if ((usecase->id != USECASE_AUDIO_HFP_SCO) &&
-        (usecase->id != USECASE_AUDIO_HFP_SCO_WB)) {
-        ALOGV("%s: a playback path where app type cfg is not required", __func__);
+        (usecase->id != USECASE_AUDIO_HFP_SCO_WB) &&
+        (usecase->id != USECASE_AUDIO_HFP_SCO_DOWNLINK) &&
+        (usecase->id != USECASE_AUDIO_HFP_SCO_WB_DOWNLINK)) {
+        ALOGV("%s: a usecase where app type cfg is not required", __func__);
         rc = 0;
         goto exit_send_app_type_cfg;
     }
 
+    if (usecase->devices & AUDIO_DEVICE_OUT_BUS)
+        is_bus_dev_usecase = true;
+
     snd_device = usecase->out_snd_device;
     pcm_device_id = platform_get_pcm_device_id(usecase->id, PCM_PLAYBACK);
 
@@ -983,24 +1029,47 @@
     if (usecase->type == PCM_HFP_CALL) {
 
         /* config HFP session:1 playback path */
-        app_type = platform_get_default_app_type_v2(adev->platform, PCM_PLAYBACK);
-        sample_rate= CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        if (is_bus_dev_usecase) {
+            app_type = usecase->out_app_type_cfg.app_type;
+            sample_rate= usecase->out_app_type_cfg.sample_rate;
+        } else {
+            snd_device = SND_DEVICE_NONE; // use legacy behavior
+            app_type = platform_get_default_app_type_v2(adev->platform, PCM_PLAYBACK);
+            sample_rate= CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        }
         rc = set_stream_app_type_mixer_ctrl(adev, pcm_device_id, app_type,
                                             acdb_dev_id, sample_rate,
                                             PCM_PLAYBACK,
-                                            SND_DEVICE_NONE); // use legacy behavior
+                                            snd_device);
         if (rc < 0)
             goto exit_send_app_type_cfg;
 
         /* config HFP session:1 capture path */
-        app_type = platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE);
+        if (is_bus_dev_usecase) {
+            snd_device = usecase->in_snd_device;
+            pcm_device_id = platform_get_pcm_device_id(usecase->id, PCM_CAPTURE);
+            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__);
+                rc = -EINVAL;
+                goto exit_send_app_type_cfg;
+            }
+            app_type = usecase->in_app_type_cfg.app_type;
+            sample_rate= usecase->in_app_type_cfg.sample_rate;
+        } else {
+            snd_device = SND_DEVICE_NONE; // use legacy behavior
+            app_type = platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE);
+        }
         rc = set_stream_app_type_mixer_ctrl(adev, pcm_device_id, app_type,
                                             acdb_dev_id, sample_rate,
                                             PCM_CAPTURE,
-                                            SND_DEVICE_NONE);
+                                            snd_device);
         if (rc < 0)
             goto exit_send_app_type_cfg;
 
+        if (is_bus_dev_usecase)
+            goto exit_send_app_type_cfg;
+
         /* config HFP session:2 capture path */
         pcm_device_id = HFP_ASM_RX_TX;
         snd_device = usecase->in_snd_device;
@@ -2915,15 +2984,15 @@
             LISTEN_EVENT_SND_DEVICE_FREE);
 }
 
-int audio_extn_utils_get_license_params
-(
-const struct audio_device *adev,
-struct audio_license_params *license_params
-)
+int audio_extn_utils_get_license_params(
+        const struct audio_device *adev,
+        struct audio_license_params *license_params)
 {
     if(!license_params)
         return -EINVAL;
-    return platform_get_license_by_product(adev->platform, (const char*)license_params->product, &license_params->key, license_params->license);
+
+    return platform_get_license_by_product(adev->platform,
+            (const char*)license_params->product, &license_params->key, license_params->license);
 }
 
 int audio_extn_utils_send_app_type_gain(struct audio_device *adev,
@@ -2948,32 +3017,78 @@
                                sizeof(gain_cfg)/sizeof(gain_cfg[0]));
 }
 
-int audio_extn_utils_is_vendor_enhanced_fwk()
+static void vndk_fwk_init()
 {
-    static int is_running_with_enhanced_fwk = -EINVAL;
+    if (mVndkFwk.lib_handle != NULL)
+        return;
 
-    if (is_running_with_enhanced_fwk == -EINVAL) {
-        vndk_fwk_lib_handle = dlopen(VNDK_FWK_LIB_PATH, RTLD_NOW);
-        if (vndk_fwk_lib_handle != NULL) {
-            vndk_fwk_isVendorEnhancedFwk = (vndk_fwk_isVendorEnhancedFwk_t)
-                        dlsym(vndk_fwk_lib_handle, "isRunningWithVendorEnhancedFramework");
-            if (vndk_fwk_isVendorEnhancedFwk == NULL) {
-                ALOGW("%s: dlsym failed, defaulting to enhanced_fwk configuration",
-                       __func__);
-                is_running_with_enhanced_fwk = 1;
-            } else {
-                is_running_with_enhanced_fwk = vndk_fwk_isVendorEnhancedFwk();
-            }
-            dlclose(vndk_fwk_lib_handle);
-            vndk_fwk_lib_handle = NULL;
-        } else {
-            ALOGW("%s: VNDK_FWK_LIB not found, setting stock configuration", __func__);
-            is_running_with_enhanced_fwk = 0;
-        }
-        ALOGV("%s: vndk_fwk_isVendorEnhancedFwk=%d", __func__, is_running_with_enhanced_fwk);
+    mVndkFwk.lib_handle = dlopen(VNDK_FWK_LIB_PATH, RTLD_NOW);
+    if (mVndkFwk.lib_handle == NULL) {
+        ALOGW("%s: failed to dlopen VNDK_FWK_LIB %s", __func__,  strerror(errno));
+        return;
     }
 
-    return is_running_with_enhanced_fwk;
+    *(void **)(&mVndkFwk.isVendorEnhancedFwk) =
+        dlsym(mVndkFwk.lib_handle, "isRunningWithVendorEnhancedFramework");
+    if (mVndkFwk.isVendorEnhancedFwk == NULL) {
+        ALOGW("%s: dlsym failed %s", __func__, strerror(errno));
+        if (mVndkFwk.lib_handle) {
+            dlclose(mVndkFwk.lib_handle);
+            mVndkFwk.lib_handle = NULL;
+        }
+        return;
+    }
+
+
+    *(void **)(&mVndkFwk.getVendorEnhancedInfo) =
+        dlsym(mVndkFwk.lib_handle, "getVendorEnhancedInfo");
+    if (mVndkFwk.getVendorEnhancedInfo == NULL) {
+        ALOGW("%s: dlsym failed %s", __func__, strerror(errno));
+        if (mVndkFwk.lib_handle) {
+            dlclose(mVndkFwk.lib_handle);
+            mVndkFwk.lib_handle = NULL;
+        }
+    }
+
+    return;
+}
+
+bool audio_extn_utils_is_vendor_enhanced_fwk()
+{
+    static int is_vendor_enhanced_fwk = -EINVAL;
+    if (is_vendor_enhanced_fwk != -EINVAL)
+        return (bool)is_vendor_enhanced_fwk;
+
+    vndk_fwk_init();
+
+    if (mVndkFwk.isVendorEnhancedFwk != NULL) {
+        is_vendor_enhanced_fwk = mVndkFwk.isVendorEnhancedFwk();
+        ALOGW("%s: is_vendor_enhanced_fwk %d", __func__, is_vendor_enhanced_fwk);
+    } else {
+        is_vendor_enhanced_fwk = 0;
+        ALOGW("%s: default to non enhanced_fwk config", __func__);
+    }
+
+    return (bool)is_vendor_enhanced_fwk;
+}
+
+int audio_extn_utils_get_vendor_enhanced_info()
+{
+    static int vendor_enhanced_info = -EINVAL;
+    if (vendor_enhanced_info != -EINVAL)
+        return vendor_enhanced_info;
+
+    vndk_fwk_init();
+
+    if (mVndkFwk.getVendorEnhancedInfo != NULL) {
+        vendor_enhanced_info = mVndkFwk.getVendorEnhancedInfo();
+        ALOGW("%s: vendor_enhanced_info 0x%x", __func__, vendor_enhanced_info);
+    } else {
+        vendor_enhanced_info = 0x0;
+        ALOGW("%s: default to vendor_enhanced_info 0x0", __func__);
+    }
+
+    return vendor_enhanced_info;
 }
 
 int audio_extn_utils_get_perf_mode_flag(void)
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index bc2f159..0580353 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -357,8 +357,10 @@
 
     [USECASE_AUDIO_HFP_SCO] = "hfp-sco",
     [USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb",
-    [USECASE_VOICE_CALL] = "voice-call",
+    [USECASE_AUDIO_HFP_SCO_DOWNLINK] = "hfp-sco-downlink",
+    [USECASE_AUDIO_HFP_SCO_WB_DOWNLINK] = "hfp-sco-wb-downlink",
 
+    [USECASE_VOICE_CALL] = "voice-call",
     [USECASE_VOICE2_CALL] = "voice2-call",
     [USECASE_VOLTE_CALL] = "volte-call",
     [USECASE_QCHAT_CALL] = "qchat-call",
@@ -511,6 +513,8 @@
 static void in_snd_mon_cb(void * stream, struct str_parms * parms);
 static void out_snd_mon_cb(void * stream, struct str_parms * parms);
 
+static int configure_btsco_sample_rate(snd_device_t snd_device);
+
 #ifdef AUDIO_FEATURE_ENABLED_GCOV
 extern void  __gcov_flush();
 static void enable_gcov()
@@ -1303,6 +1307,7 @@
             goto err;
         }
 
+        configure_btsco_sample_rate(snd_device);
         /* 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,
@@ -2310,6 +2315,50 @@
     return bt_soc_status;
 }
 
+static int configure_btsco_sample_rate(snd_device_t snd_device)
+{
+    struct mixer_ctl *ctl = NULL;
+    struct mixer_ctl *ctl_sr_rx = NULL, *ctl_sr_tx = NULL, *ctl_sr = NULL;
+    char *rate_str = NULL;
+    bool is_rx_dev = true;
+
+    if (is_btsco_device(snd_device, snd_device)) {
+        ctl_sr_tx = mixer_get_ctl_by_name(adev->mixer, "BT SampleRate TX");
+        ctl_sr_rx = mixer_get_ctl_by_name(adev->mixer, "BT SampleRate RX");
+        if (!ctl_sr_tx || !ctl_sr_rx) {
+            ctl_sr = mixer_get_ctl_by_name(adev->mixer, "BT SampleRate");
+            if (!ctl_sr)
+                return -ENOSYS;
+        }
+
+        switch (snd_device) {
+        case SND_DEVICE_OUT_BT_SCO:
+            rate_str = "KHZ_8";
+            break;
+        case SND_DEVICE_IN_BT_SCO_MIC_NREC:
+        case SND_DEVICE_IN_BT_SCO_MIC:
+            rate_str = "KHZ_8";
+            is_rx_dev = false;
+            break;
+        case SND_DEVICE_OUT_BT_SCO_WB:
+            rate_str = "KHZ_16";
+            break;
+        case SND_DEVICE_IN_BT_SCO_MIC_WB_NREC:
+        case SND_DEVICE_IN_BT_SCO_MIC_WB:
+            rate_str = "KHZ_16";
+            is_rx_dev = false;
+            break;
+        default:
+            return 0;
+        }
+
+        ctl = (ctl_sr == NULL) ? (is_rx_dev ? ctl_sr_rx : ctl_sr_tx) : ctl_sr;
+        if (mixer_ctl_set_enum_by_string(ctl, rate_str) != 0)
+            return -ENOSYS;
+    }
+    return 0;
+}
+
 int out_standby_l(struct audio_stream *stream);
 
 struct stream_in *adev_get_active_input(const struct audio_device *adev)
@@ -2430,11 +2479,18 @@
             ALOGE("%s: stream.out is NULL", __func__);
             return -EINVAL;
         }
-        out_snd_device = platform_get_output_snd_device(adev->platform,
-                                                        usecase->stream.out);
-        in_snd_device = platform_get_input_snd_device(adev->platform,
-                                                      NULL,
-                                                      usecase->stream.out->devices);
+        if (usecase->devices & AUDIO_DEVICE_OUT_BUS) {
+            out_snd_device = audio_extn_auto_hal_get_output_snd_device(adev,
+                                                                       uc_id);
+            in_snd_device = audio_extn_auto_hal_get_input_snd_device(adev,
+                                                                     uc_id);
+        } else {
+            out_snd_device = platform_get_output_snd_device(adev->platform,
+                                                            usecase->stream.out);
+            in_snd_device = platform_get_input_snd_device(adev->platform,
+                                                          NULL,
+                                                          usecase->stream.out->devices);
+        }
         usecase->devices = usecase->stream.out->devices;
     } else if (usecase->type == TRANSCODE_LOOPBACK_RX) {
         if (usecase->stream.inout == NULL) {
@@ -2515,8 +2571,11 @@
             if (out_snd_device == SND_DEVICE_NONE) {
                 struct stream_out *voip_out = adev->primary_output;
                 struct stream_in *voip_in = get_voice_communication_input(adev);
-                out_snd_device = platform_get_output_snd_device(adev->platform,
-                                                                usecase->stream.out);
+                if (usecase->devices & AUDIO_DEVICE_OUT_BUS)
+                    out_snd_device = audio_extn_auto_hal_get_output_snd_device(adev, uc_id);
+                else
+                    out_snd_device = platform_get_output_snd_device(adev->platform,
+                                                                    usecase->stream.out);
                 voip_usecase = get_usecase_from_list(adev, USECASE_AUDIO_PLAYBACK_VOIP);
 
                 if (voip_usecase)
@@ -2578,10 +2637,11 @@
             return 0;
     }
 
-    if ((is_btsco_device(out_snd_device,in_snd_device) && !adev->bt_sco_on) ||
-         (is_a2dp_device(out_snd_device) && !audio_extn_a2dp_source_is_ready())) {
-          ALOGD("SCO/A2DP is selected but they are not connected/ready hence dont route");
-          return 0;
+    if (!(usecase->devices & AUDIO_DEVICE_OUT_BUS) &&
+        ((is_btsco_device(out_snd_device,in_snd_device) && !adev->bt_sco_on) ||
+            (is_a2dp_device(out_snd_device) && !audio_extn_a2dp_source_is_ready()))) {
+        ALOGD("SCO/A2DP is selected but they are not connected/ready hence dont route");
+        return 0;
     }
 
     if (out_snd_device != SND_DEVICE_NONE &&
@@ -5872,25 +5932,24 @@
                 size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
 
                 uint64_t signed_frames = 0;
+                uint64_t frames_temp = 0;
 
-                if (avail > kernel_buffer_size)
-                    avail = kernel_buffer_size;
-
-                if (out->written >= (kernel_buffer_size - avail))
-                    signed_frames = out->written - kernel_buffer_size + avail;
+                frames_temp = (kernel_buffer_size > avail) ? (kernel_buffer_size - avail) : 0;
+                if (out->written >= frames_temp)
+                    signed_frames = out->written - frames_temp;
 
                 // This adjustment accounts for buffering after app processor.
                 // It is based on estimated DSP latency per use case, rather than exact.
-                if (signed_frames >= (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL))
-                    signed_frames -=
-                        (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+                frames_temp = platform_render_latency(out->usecase) * out->sample_rate / 1000000LL;
+                if (signed_frames >= frames_temp)
+                    signed_frames -= frames_temp;
 
                 // Adjustment accounts for A2dp encoder latency with non offload usecases
                 // Note: Encoder latency is returned in ms, while platform_render_latency in us.
                 if (AUDIO_DEVICE_OUT_ALL_A2DP & out->devices) {
-                    if (signed_frames >= (audio_extn_a2dp_get_encoder_latency() * out->sample_rate / 1000))
-                        signed_frames -=
-                            (audio_extn_a2dp_get_encoder_latency() * out->sample_rate / 1000);
+                    frames_temp = audio_extn_a2dp_get_encoder_latency() * out->sample_rate / 1000;
+                    if (signed_frames >= frames_temp)
+                        signed_frames -= frames_temp;
                 }
 
                 // It would be unusual for this value to be negative, but check just in case ...
@@ -8677,6 +8736,7 @@
         //XXX needed for voice_extn_compress_voip_open_input_stream
         in->config.rate = config->sample_rate;
         if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||
+             in->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
              voice_extn_compress_voip_is_active(in->dev)) &&
             (voice_extn_compress_voip_is_format_supported(in->format)) &&
             valid_rate && valid_ch) {
@@ -9062,8 +9122,9 @@
             }
         }
     }
+    if (audio_extn_ssr_get_stream() != in)
+        in->config.channels = channel_count;
 
-    in->config.channels = channel_count;
     in->sample_rate  = in->config.rate;
 
     audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index b1e61ad..5e9627b 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -159,6 +159,8 @@
     /* HFP Use case*/
     USECASE_AUDIO_HFP_SCO,
     USECASE_AUDIO_HFP_SCO_WB,
+    USECASE_AUDIO_HFP_SCO_DOWNLINK,
+    USECASE_AUDIO_HFP_SCO_WB_DOWNLINK,
 
     /* Capture usecases */
     USECASE_AUDIO_RECORD,
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 3094183..73369ea 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -6115,7 +6115,8 @@
             if (rate_str == NULL) {
                 switch (sample_rate) {
                 case 32000:
-                    if (passthrough_enabled || (backend_idx == HDMI_TX_BACKEND )) {
+                    if (passthrough_enabled || (backend_idx == HDMI_TX_BACKEND) ||
+                            (backend_idx == DISP_PORT_RX_BACKEND)) {
                         rate_str = "KHZ_32";
                         break;
                     }
@@ -7569,37 +7570,47 @@
     struct platform_data *my_data = (struct platform_data *)platform;
     struct audio_device *adev = my_data->adev;
     edid_audio_info *info = NULL;
-    int channel_count = 2;
-    int i, ret;
+    int ret;
     char default_channelMap[MAX_CHANNELS_SUPPORTED] = {0};
     struct audio_device_config_param *adev_device_cfg_ptr = adev->device_cfg_params;
+    int channel_alloc = 0;
+    int max_supported_channels = 0;
 
     ret = platform_get_edid_info(platform);
     info = (edid_audio_info *)my_data->edid_info;
     adev_device_cfg_ptr += HDMI_RX_BACKEND;
     if(ret == 0 && info != NULL) {
-        if (channels > 2) {
-
+        if ((channels > 2) && (channels <= MAX_HDMI_CHANNEL_CNT)) {
             ALOGV("%s:able to get HDMI sink capabilities multi channel playback",
                    __func__);
-            for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
-                if (info->audio_blocks_array[i].format_id == LPCM &&
-                      info->audio_blocks_array[i].channels > channel_count &&
-                      info->audio_blocks_array[i].channels <= MAX_HDMI_CHANNEL_CNT) {
-                    channel_count = info->audio_blocks_array[i].channels;
-                }
+            max_supported_channels = platform_edid_get_max_channels(my_data);
+            if (channels > max_supported_channels)
+                channels = max_supported_channels;
+            // refer to HDMI spec CEA-861-E: Table 28 Audio InfoFrame Data Byte 4
+            switch (channels) {
+            case 3:
+                channel_alloc = 0x02; break;
+            case 4:
+                channel_alloc = 0x06; break;
+            case 5:
+                channel_alloc = 0x0A; break;
+            case 6:
+                channel_alloc = 0x0B; break;
+            case 7:
+                channel_alloc = 0x12; break;
+            case 8:
+                channel_alloc = 0x13; break;
+            default:
+                ALOGE("%s: invalid channel %d", __func__, channels);
+                return -EINVAL;
             }
-            ALOGV("%s:channel_count:%d", __func__, channel_count);
-            /*
-             * Channel map is set for supported hdmi max channel count even
-             * though the input channel count set on adm is less than or equal to
-             * max supported channel count
-             */
+            ALOGVV("%s:channels:%d", __func__, channels);
+
             if (adev_device_cfg_ptr->use_client_dev_cfg) {
                 platform_set_channel_map(platform, adev_device_cfg_ptr->dev_cfg_params.channels,
                                    (char *)adev_device_cfg_ptr->dev_cfg_params.channel_map, -1, -1);
             } else {
-                platform_set_channel_map(platform, channel_count, info->channel_map, -1, -1);
+                platform_set_channel_map(platform, channels, info->channel_map, -1, -1);
             }
 
             if (adev_device_cfg_ptr->use_client_dev_cfg) {
@@ -7608,8 +7619,8 @@
                 platform_set_channel_allocation(platform,
                        adev_device_cfg_ptr->dev_cfg_params.channel_allocation);
             } else {
-                platform_set_channel_allocation(platform, info->channel_allocation);
-           }
+                platform_set_channel_allocation(platform, channel_alloc);
+            }
         } else {
             if (adev_device_cfg_ptr->use_client_dev_cfg) {
                 default_channelMap[0] = adev_device_cfg_ptr->dev_cfg_params.channel_map[0];
@@ -7619,7 +7630,7 @@
                 default_channelMap[1] = PCM_CHANNEL_FR;
             }
             platform_set_channel_map(platform, 2, default_channelMap, -1, -1);
-            platform_set_channel_allocation(platform,0);
+            platform_set_channel_allocation(platform, 0);
         }
     }
 
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 26fae75..8c000ba 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -531,6 +531,10 @@
     } else if (!strncmp(snd_card_name, "atoll-wcd937x-snd-card",
                  sizeof("atoll-wcd937x-snd-card"))) {
         strlcpy(hw_info->name, "atoll", sizeof(hw_info->name));
+    } else if (!strncmp(snd_card_name, "atoll-qrd-snd-card",
+                 sizeof("atoll-qrd-snd-card"))) {
+        strlcpy(hw_info->name, "atoll", sizeof(hw_info->name));
+        hw_info->is_stereo_spkr = false;
     } else if (!strncmp(snd_card_name, "kona-qrd-snd-card",
                  sizeof("kona-qrd-snd-card"))) {
         strlcpy(hw_info->name, "kona", sizeof(hw_info->name));
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index fb65f88..1cab0a2 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -417,6 +417,8 @@
     [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},
+    [USECASE_AUDIO_HFP_SCO_DOWNLINK] = {HFP_ASM_RX_TX, HFP_ASM_RX_TX},
+    [USECASE_AUDIO_HFP_SCO_WB_DOWNLINK] = {HFP_ASM_RX_TX, HFP_ASM_RX_TX},
     [USECASE_VOICE_CALL] = {VOICE_CALL_PCM_DEVICE, VOICE_CALL_PCM_DEVICE},
     [USECASE_AUDIO_PLAYBACK_MMAP] = {MMAP_PLAYBACK_PCM_DEVICE,
             MMAP_PLAYBACK_PCM_DEVICE},
@@ -1276,6 +1278,8 @@
     {TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
     {TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO)},
     {TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO_WB)},
+    {TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO_DOWNLINK)},
+    {TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO_WB_DOWNLINK)},
     {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_FM)},
     {TO_NAME_INDEX(USECASE_AUDIO_RECORD_FM_VIRTUAL)},
     {TO_NAME_INDEX(USECASE_AUDIO_SPKR_CALIB_RX)},
@@ -1690,7 +1694,9 @@
          !strncmp(snd_card_name, "atoll-wcd937x-snd-card",
                    sizeof("atoll-wcd937x-snd-card")) ||
          !strncmp(snd_card_name, "atoll-idp-snd-card",
-                   sizeof("atoll-idp-snd-card"))) {
+                   sizeof("atoll-idp-snd-card")) ||
+         !strncmp(snd_card_name, "atoll-qrd-snd-card",
+                   sizeof("atoll-qrd-snd-card"))) {
          ALOGI("%s: snd_card_name: %s",__func__,snd_card_name);
          my_data->is_internal_codec = true;
          my_data->is_slimbus_interface = false;
@@ -3245,6 +3251,9 @@
     else if (!strncmp(snd_card_name, "lito-qrd-snd-card",
                sizeof("lito-qrd-snd-card")))
         platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
+    else if (!strncmp(snd_card_name, "atoll-qrd-snd-card",
+               sizeof("atoll-qrd-snd-card")))
+        platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
     else if (!strncmp(snd_card_name, "qcs405-wsa-snd-card",
                sizeof("qcs405-wsa-snd-card")))
         platform_info_init(PLATFORM_INFO_XML_PATH_WSA, my_data, PLATFORM);
@@ -4981,10 +4990,14 @@
     snd_device_t incall_rec_device;
     int sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
     struct audio_backend_cfg backend_cfg = {0};
+    bool is_bus_dev_usecase = false;
 
     if (voice_is_in_call(my_data->adev))
         is_incall_rec_usecase = voice_is_in_call_rec_stream(usecase->stream.in);
 
+    if (usecase->devices & AUDIO_DEVICE_OUT_BUS)
+        is_bus_dev_usecase = true;
+
     if (usecase->type == PCM_PLAYBACK)
         snd_device = usecase->out_snd_device;
     else if ((usecase->type == PCM_CAPTURE) && is_incall_rec_usecase)
@@ -5013,6 +5026,11 @@
             new_snd_device[0] = snd_device;
         }
     }
+    if ((usecase->type == PCM_HFP_CALL) && is_bus_dev_usecase) {
+        num_devices = 2;
+        new_snd_device[0] = usecase->in_snd_device;
+        new_snd_device[1] = usecase->out_snd_device;
+    }
 
     for (i = 0; i < num_devices; i++) {
         if (!is_incall_rec_usecase) {
@@ -5032,6 +5050,17 @@
         if ((usecase->type == PCM_CAPTURE) && (app_type == DEFAULT_APP_TYPE_RX_PATH)) {
             ALOGD("Resetting app type for Tx path to default");
             app_type  = DEFAULT_APP_TYPE_TX_PATH;
+        } else if ((usecase->type == PCM_HFP_CALL) && is_bus_dev_usecase) {
+            if (new_snd_device[i] >= SND_DEVICE_OUT_BEGIN &&
+                new_snd_device[i] < SND_DEVICE_OUT_END) {
+                app_type  = usecase->out_app_type_cfg.app_type;
+                sample_rate = usecase->out_app_type_cfg.sample_rate;
+            } else {
+                app_type  = usecase->in_app_type_cfg.app_type;
+                sample_rate = usecase->in_app_type_cfg.sample_rate;
+            }
+            ALOGD("%s: Updating to app type (%d) and sample rate (%d)",
+                  __func__, app_type, sample_rate);
         }
 
         /* Override backend cfg sample rate in calibration for vi feedback usecase */
@@ -6073,8 +6102,7 @@
                 snd_device = SND_DEVICE_OUT_BT_SCO_WB;
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
-        } else if ((devices & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE)) ||
-                   (devices & AUDIO_DEVICE_OUT_BUS)) {
+        } else if (devices & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
             if (my_data->is_vbat_speaker || my_data->is_bcl_speaker) {
                 if (hw_info_is_stereo_spkr(my_data->hw_info)) {
                     if (my_data->mono_speaker == SPKR_1)
@@ -6250,7 +6278,7 @@
         snd_device = SND_DEVICE_OUT_AFE_PROXY;
         audio_extn_set_afe_proxy_channel_mixer(adev, channel_count, snd_device);
     } else if (devices & AUDIO_DEVICE_OUT_BUS) {
-        snd_device = audio_extn_auto_hal_get_snd_device_for_car_audio_stream(out);
+        snd_device = audio_extn_auto_hal_get_output_snd_device(adev, out->usecase);
     } else {
         ALOGE("%s: Unknown device(s) %#x", __func__, devices);
     }
@@ -6629,8 +6657,7 @@
         } else if (out_device & AUDIO_DEVICE_OUT_SPEAKER ||
                    out_device & AUDIO_DEVICE_OUT_SPEAKER_SAFE ||
                    out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
-                   out_device & AUDIO_DEVICE_OUT_LINE ||
-                   out_device & AUDIO_DEVICE_OUT_BUS) {
+                   out_device & AUDIO_DEVICE_OUT_LINE) {
             if (my_data->fluence_type != FLUENCE_NONE &&
                 (my_data->fluence_in_voice_call ||
                  my_data->fluence_in_hfp_call) &&
@@ -8474,9 +8501,10 @@
             if (rate_str == NULL) {
                 switch (sample_rate) {
                 case 32000:
-                    if (passthrough_enabled || (backend_idx == SPDIF_TX_BACKEND ) ||
-                        (backend_idx == HDMI_TX_BACKEND ) ||
-                        (backend_idx == HDMI_ARC_TX_BACKEND )) {
+                    if (passthrough_enabled || (backend_idx == SPDIF_TX_BACKEND) ||
+                        (backend_idx == HDMI_TX_BACKEND) ||
+                        (backend_idx == HDMI_ARC_TX_BACKEND) ||
+                        (backend_idx == DISP_PORT_RX_BACKEND)) {
                         rate_str = "KHZ_32";
                         break;
                     }
@@ -10305,11 +10333,12 @@
     struct platform_data *my_data = (struct platform_data *)platform;
     struct audio_device *adev = my_data->adev;
     edid_audio_info *info = NULL;
-    int channel_count = 2;
-    int i, ret;
+    int ret;
     char default_channelMap[MAX_CHANNELS_SUPPORTED] = {0};
     struct audio_device_config_param *adev_device_cfg_ptr = adev->device_cfg_params;
     int be_idx = -1;
+    int channel_alloc = 0;
+    int max_supported_channels = 0;
 
     if ((backend_idx != HDMI_RX_BACKEND) &&
             (backend_idx != DISP_PORT_RX_BACKEND) &&
@@ -10332,28 +10361,37 @@
     info = (edid_audio_info *)my_data->ext_disp[controller][stream].edid_info;
     adev_device_cfg_ptr += backend_idx;
     if(ret == 0 && info != NULL) {
-        if (channels > 2) {
-
+        if ((channels > 2) && (channels <= MAX_HDMI_CHANNEL_CNT)) {
             ALOGV("%s:able to get HDMI/DP sink capabilities multi channel playback",
                    __func__);
-            for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
-                if (info->audio_blocks_array[i].format_id == LPCM &&
-                      info->audio_blocks_array[i].channels > channel_count &&
-                      info->audio_blocks_array[i].channels <= MAX_HDMI_CHANNEL_CNT) {
-                    channel_count = info->audio_blocks_array[i].channels;
-                }
+            max_supported_channels = platform_edid_get_max_channels(my_data);
+            if (channels > max_supported_channels)
+                channels = max_supported_channels;
+            // refer to HDMI spec CEA-861-E: Table 28 Audio InfoFrame Data Byte 4
+            switch (channels) {
+            case 3:
+                channel_alloc = 0x02; break;
+            case 4:
+                channel_alloc = 0x06; break;
+            case 5:
+                channel_alloc = 0x0A; break;
+            case 6:
+                channel_alloc = 0x0B; break;
+            case 7:
+                channel_alloc = 0x12; break;
+            case 8:
+                channel_alloc = 0x13; break;
+            default:
+                ALOGE("%s: invalid channel %d", __func__, channels);
+                return -EINVAL;
             }
-            ALOGVV("%s:channel_count:%d", __func__, channel_count);
-            /*
-             * Channel map is set for supported hdmi max channel count even
-             * though the input channel count set on adm is less than or equal to
-             * max supported channel count
-             */
+            ALOGVV("%s:channels:%d", __func__, channels);
+
             if (adev_device_cfg_ptr->use_client_dev_cfg) {
                 platform_set_channel_map(platform, adev_device_cfg_ptr->dev_cfg_params.channels,
                                      (char *)adev_device_cfg_ptr->dev_cfg_params.channel_map, -1, be_idx);
             } else {
-                platform_set_channel_map(platform, channel_count, info->channel_map, -1, be_idx);
+                platform_set_channel_map(platform, channels, info->channel_map, -1, be_idx);
             }
 
             if (adev_device_cfg_ptr->use_client_dev_cfg) {
@@ -10363,9 +10401,9 @@
                        adev_device_cfg_ptr->dev_cfg_params.channel_allocation,
                        controller, stream);
             } else {
-                platform_set_channel_allocation_v2(platform, info->channel_allocation,
+                platform_set_channel_allocation_v2(platform, channel_alloc,
                                                 controller, stream);
-           }
+            }
         } else {
             if (adev_device_cfg_ptr->use_client_dev_cfg) {
                 default_channelMap[0] = adev_device_cfg_ptr->dev_cfg_params.channel_map[0];
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 2a570ce..7f54e7f 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -451,7 +451,7 @@
 #define FM_CAPTURE_PCM_DEVICE  6
 
 #ifdef PLATFORM_AUTO
-#define HFP_PCM_RX 36
+#define HFP_PCM_RX 29
 #else
 #define HFP_PCM_RX 5
 #endif
@@ -638,8 +638,8 @@
 #define HFP_SCO_RX 17
 #define HFP_ASM_RX_TX 18
 #elif PLATFORM_AUTO
-#define HFP_SCO_RX 36
-#define HFP_ASM_RX_TX 29
+#define HFP_SCO_RX 29
+#define HFP_ASM_RX_TX 36
 #elif defined (PLATFORM_MSMFALCON) || defined (PLATFORM_MSM8937)
 #define HFP_SCO_RX 17
 #define HFP_ASM_RX_TX 18
diff --git a/hal/platform_info.c b/hal/platform_info.c
index d73792c..1969f52 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -135,6 +135,7 @@
 };
 
 static struct platform_info my_data;
+static pthread_mutex_t parser_lock = PTHREAD_MUTEX_INITIALIZER;
 
 
 struct audio_string_to_enum {
@@ -1447,6 +1448,7 @@
     void            *buf;
     char            platform_info_file_name[MIXER_PATH_MAX_LENGTH]= {0};
 
+    pthread_mutex_lock(&parser_lock);
     if (filename == NULL)
         strlcpy(platform_info_file_name, PLATFORM_INFO_XML_PATH,
                 MIXER_PATH_MAX_LENGTH);
@@ -1511,5 +1513,6 @@
 err_close_file:
     fclose(file);
 done:
+    pthread_mutex_unlock(&parser_lock);
     return ret;
 }
diff --git a/qahw/Android.mk b/qahw/Android.mk
index a64b913..84645b8 100644
--- a/qahw/Android.mk
+++ b/qahw/Android.mk
@@ -11,6 +11,10 @@
 LOCAL_MODULE_TAGS := optional
 LOCAL_C_INCLUDES   := $(libqahw-inc)
 
+LOCAL_HEADER_LIBRARIES := libutils_headers \
+    libsystem_headers \
+    libhardware_headers
+
 LOCAL_SRC_FILES := \
     src/qahw.c \
     src/qahw_effect.c
@@ -27,15 +31,10 @@
 LOCAL_COPY_HEADERS      := inc/qahw.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_api.h
 
-LOCAL_PRELINK_MODULE    := false
+LOCAL_PROPRIETARY_MODULE := true
 LOCAL_VENDOR_MODULE     := true
 
 include $(BUILD_SHARED_LIBRARY)
 
-include $(CLEAR_VARS)
-LOCAL_COPY_HEADERS_TO   := mm-audio/qahw_api/inc
-LOCAL_COPY_HEADERS      := inc/qahw_defs.h
-
-include $(BUILD_COPY_HEADERS)
 endif
 endif
diff --git a/qahw/src/qahw.c b/qahw/src/qahw.c
index 545152c..ceec657 100644
--- a/qahw/src/qahw.c
+++ b/qahw/src/qahw.c
@@ -35,7 +35,7 @@
 #include <utils/Log.h>
 #include <stdlib.h>
 #include <cutils/list.h>
-
+#include <pthread.h>
 #include <hardware/audio.h>
 #include <hardware/sound_trigger.h>
 #include "qahw.h"
diff --git a/qahw/src/qahw_effect.c b/qahw/src/qahw_effect.c
index 9bee2a1..ff92f73 100644
--- a/qahw/src/qahw_effect.c
+++ b/qahw/src/qahw_effect.c
@@ -37,7 +37,7 @@
 #include <hardware/audio.h>
 #include <hardware/audio_effect.h>
 #include <stdlib.h>
-
+#include <pthread.h>
 #include "qahw.h"
 
 // The current effect API version.
diff --git a/qahw_api/Android.mk b/qahw_api/Android.mk
index 1cad75d..a873a7d 100644
--- a/qahw_api/Android.mk
+++ b/qahw_api/Android.mk
@@ -25,12 +25,14 @@
     libcutils \
     libhardware \
     libdl \
+    libutils \
     libqahwwrapper
 
 LOCAL_CFLAGS += -Wall -Werror
 
 LOCAL_COPY_HEADERS_TO   := mm-audio/qahw_api/inc
-LOCAL_COPY_HEADERS      := inc/qahw_api.h
+LOCAL_COPY_HEADERS      := inc/qahw_defs.h
+LOCAL_COPY_HEADERS      += inc/qahw_api.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_audiosphere.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_bassboost.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_environmentalreverb.h
@@ -39,7 +41,6 @@
 LOCAL_COPY_HEADERS      += inc/qahw_effect_virtualizer.h
 LOCAL_COPY_HEADERS      += inc/qahw_effect_visualizer.h
 
-LOCAL_PRELINK_MODULE    := false
 LOCAL_VENDOR_MODULE     := true
 
 include $(BUILD_SHARED_LIBRARY)