diff --git a/configs/kona/mixer_paths.xml b/configs/kona/mixer_paths.xml
index a71a38a..3e40870 100644
--- a/configs/kona/mixer_paths.xml
+++ b/configs/kona/mixer_paths.xml
@@ -352,10 +352,10 @@
     <ctl name="HPHR Volume" value="20" />
     <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
 
-    <ctl name="TX_DEC0 Volume" value="84" />
-    <ctl name="TX_DEC1 Volume" value="84" />
-    <ctl name="TX_DEC2 Volume" value="84" />
-    <ctl name="TX_DEC3 Volume" value="84" />
+    <ctl name="TX_DEC0 Volume" value="102" />
+    <ctl name="TX_DEC1 Volume" value="102" />
+    <ctl name="TX_DEC2 Volume" value="102" />
+    <ctl name="TX_DEC3 Volume" value="102" />
     <ctl name="TX_DEC4 Volume" value="84" />
     <ctl name="TX_DEC5 Volume" value="84" />
     <ctl name="TX_DEC6 Volume" value="84" />
@@ -2016,7 +2016,7 @@
     </path>
 
     <path name="handset-mic">
-        <path name="dmic1" />
+        <path name="dmic2" />
     </path>
 
     <path name="headphones">
@@ -2056,6 +2056,7 @@
 
     <path name="headset-mic">
         <path name="amic2" />
+        <ctl name="TX_DEC0 Volume" value="84" />
     </path>
 
     <path name="headset-mic-liquid">
@@ -2196,17 +2197,17 @@
     <path name="handset-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="DMIC0" />
+        <ctl name="TX DMIC MUX0" value="DMIC1" />
         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="TX DMIC MUX1" value="DMIC2" />
+        <ctl name="TX DMIC MUX1" value="DMIC3" />
     </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="DMIC0" />
+        <ctl name="TX DMIC MUX0" value="DMIC4" />
         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="TX DMIC MUX1" value="DMIC4" />
+        <ctl name="TX DMIC MUX1" value="DMIC1" />
     </path>
 
     <path name="dmic-endfire">
@@ -2273,9 +2274,9 @@
     <path name="speaker-dmic-broadside">
         <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="DMIC0" />
+        <ctl name="TX DMIC MUX0" value="DMIC4" />
         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="TX DMIC MUX1" value="DMIC4" />
+        <ctl name="TX DMIC MUX1" value="DMIC1" />
     </path>
 
     <path name="dmic-broadside">
@@ -2301,9 +2302,9 @@
     <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="DMIC0" />
+        <ctl name="TX DMIC MUX0" value="DMIC4" />
         <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="TX DMIC MUX1" value="DMIC2" />
+        <ctl name="TX DMIC MUX1" value="DMIC1" />
         <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
         <ctl name="TX DMIC MUX2" value="DMIC3" />
     </path>
@@ -2316,13 +2317,13 @@
     <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="DMIC0" />
+        <ctl name="TX DMIC MUX0" value="DMIC4" />
         <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="DMIC2" />
+        <ctl name="TX DMIC MUX2" value="DMIC3" />
         <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
-        <ctl name="TX DMIC MUX3" value="DMIC4" />
+        <ctl name="TX DMIC MUX3" value="DMIC0" />
     </path>
 
     <path name="speaker-qmic-liquid">
@@ -2379,6 +2380,7 @@
     <path name="voice-tty-full-headset-mic">
         <path name="amic2" />
         <ctl name="ADC2 Volume" value="0" />
+        <ctl name="TX_DEC0 Volume" value="84" />
     </path>
 
     <path name="voice-tty-hco-headset-mic">
diff --git a/configs/kona/mixer_paths_qrd.xml b/configs/kona/mixer_paths_qrd.xml
index 35a1576..70f047d 100644
--- a/configs/kona/mixer_paths_qrd.xml
+++ b/configs/kona/mixer_paths_qrd.xml
@@ -2276,7 +2276,7 @@
     </path>
 
     <path name="speaker-mic">
-        <path name="dmic2" />
+        <path name="dmic3" />
     </path>
 
     <path name="speaker-protected">
@@ -2478,7 +2478,11 @@
     </path>
 
     <path name="speaker-dmic-endfire">
-        <path name="handset-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>
 
     <path name="dmic-endfire">
@@ -2577,6 +2581,36 @@
         <path name="three-mic" />
     </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" />
+    </path>
+
+    <path name="voice-speaker-tmic">
+        <path name="speaker-tmic" />
+    </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 DEC4" value="1" />
+         <ctl name="TX DMIC MUX2" value="DMIC3" />
+    </path>
+
+    <path name="voice-speaker-qmic">
+         <path name="speaker-qmic" />
+    </path>
+
     <path name="unprocessed-quad-mic">
         <path name="quad-mic" />
     </path>
diff --git a/configs/kona/sound_trigger_mixer_paths.xml b/configs/kona/sound_trigger_mixer_paths.xml
index 2b407bb..8cb6622 100644
--- a/configs/kona/sound_trigger_mixer_paths.xml
+++ b/configs/kona/sound_trigger_mixer_paths.xml
@@ -213,7 +213,7 @@
     <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="DMIC0" />
+        <ctl name="VA DMIC MUX0" value="DMIC4" />
     </path>
 
     <path name="listen-ape-handset-mic-preproc">
@@ -226,8 +226,8 @@
         <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="DMIC0" />
-        <ctl name="VA DMIC MUX1" value="DMIC1" />
+        <ctl name="VA DMIC MUX0" value="DMIC4" />
+        <ctl name="VA DMIC MUX1" value="DMIC0" />
     </path>
 
     <path name="listen-ape-handset-tmic">
@@ -238,9 +238,9 @@
         <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="DMIC0" />
-        <ctl name="VA DMIC MUX1" value="DMIC1" />
-        <ctl name="VA DMIC MUX2" value="DMIC4" />
+        <ctl name="VA DMIC MUX0" value="DMIC4" />
+        <ctl name="VA DMIC MUX1" value="DMIC0" />
+        <ctl name="VA DMIC MUX2" value="DMIC1" />
     </path>
 
     <path name="listen-ape-handset-qmic">
@@ -253,10 +253,10 @@
         <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="DMIC0" />
-        <ctl name="VA DMIC MUX1" value="DMIC1" />
-        <ctl name="VA DMIC MUX2" value="DMIC2" />
-        <ctl name="VA DMIC MUX3" value="DMIC4" />
+        <ctl name="VA DMIC MUX0" value="DMIC4" />
+        <ctl name="VA DMIC MUX1" value="DMIC0" />
+        <ctl name="VA DMIC MUX2" value="DMIC1" />
+        <ctl name="VA DMIC MUX3" value="DMIC2" />
     </path>
 
     <path name="listen-ape-headset-mic">
diff --git a/configs/msmnile/msmnile.mk b/configs/msmnile/msmnile.mk
index 004383a..efb3c46 100644
--- a/configs/msmnile/msmnile.mk
+++ b/configs/msmnile/msmnile.mk
@@ -12,7 +12,6 @@
 TARGET_USES_AOSP_FOR_AUDIO := false
 
 ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
-USE_CUSTOM_AUDIO_POLICY := 1
 AUDIO_FEATURE_QSSI_COMPLIANCE := false
 AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
 AUDIO_FEATURE_ENABLED_COMPRESS_INPUT := true
@@ -81,6 +80,81 @@
 AUDIO_FEATURE_ENABLED_BATTERY_LISTENER := true
 ##AUDIO_FEATURE_FLAGS
 
+AUDIO_HARDWARE := audio.a2dp.default
+AUDIO_HARDWARE += audio.usb.default
+AUDIO_HARDWARE += audio.r_submix.default
+AUDIO_HARDWARE += audio.primary.msmnile
+
+#HAL Wrapper
+AUDIO_WRAPPER := libqahw
+AUDIO_WRAPPER += libqahwwrapper
+
+#HAL Test app
+AUDIO_HAL_TEST_APPS := hal_play_test
+AUDIO_HAL_TEST_APPS += hal_rec_test
+
+PRODUCT_PACKAGES += $(AUDIO_HARDWARE)
+PRODUCT_PACKAGES += $(AUDIO_WRAPPER)
+PRODUCT_PACKAGES += $(AUDIO_HAL_TEST_APPS)
+
+ifeq ($(AUDIO_FEATURE_ENABLED_DLKM),true)
+BOARD_VENDOR_KERNEL_MODULES := \
+    $(KERNEL_MODULES_OUT)/audio_apr.ko \
+    $(KERNEL_MODULES_OUT)/audio_wglink.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_wcd.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_wcd934x.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd9360.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd_spi.ko \
+    $(KERNEL_MODULES_OUT)/audio_native.ko \
+    $(KERNEL_MODULES_OUT)/audio_machine_msmnile.ko
+endif
+
+# install lkdtm only for userdebug and eng build variants
+ifneq (,$(filter userdebug eng, $(TARGET_BUILD_VARIANT)))
+    ifeq (,$(findstring perf_defconfig, $(KERNEL_DEFCONFIG)))
+        BOARD_VENDOR_KERNEL_MODULES += $(KERNEL_MODULES_OUT)/lkdtm.ko
+    endif
+endif
+
+#Audio DLKM
+AUDIO_DLKM := audio_apr.ko
+AUDIO_DLKM += audio_wglink.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_wcd.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_wcd9360.ko
+AUDIO_DLKM += audio_wcd_spi.ko
+AUDIO_DLKM += audio_native.ko
+AUDIO_DLKM += audio_machine_msmnile.ko
+AUDIO_DLKM += audio_wcd934x.ko
+PRODUCT_PACKAGES += $(AUDIO_DLKM)
+
 ifneq ($(strip $(TARGET_USES_RRO)), true)
 #Audio Specific device overlays
 DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
@@ -137,6 +211,19 @@
 persist.vendor.audio.fluence.speaker=true\
 persist.vendor.audio.fluence.tmic.enabled=false
 
+#
+#snapdragon value add features
+#
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.ssr=false
+
+##fluencetype can be "fluence" or "fluencepro" or "none"
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.fluencetype=none\
+persist.audio.fluence.voicecall=true\
+persist.audio.fluence.voicerec=false\
+persist.audio.fluence.speaker=true
+
 #disable tunnel encoding
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.tunnel.encode=false
diff --git a/configs/msmsteppe/msmsteppe.mk b/configs/msmsteppe/msmsteppe.mk
index ce4a01c..4c404ea 100644
--- a/configs/msmsteppe/msmsteppe.mk
+++ b/configs/msmsteppe/msmsteppe.mk
@@ -5,7 +5,6 @@
 TARGET_USES_AOSP_FOR_AUDIO := false
 
 ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
-USE_CUSTOM_AUDIO_POLICY := 1
 AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
 AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := false
 AUDIO_FEATURE_ENABLED_DYNAMIC_ECNS := true
@@ -69,6 +68,93 @@
 AUDIO_FEATURE_ENABLED_BATTERY_LISTENER := true
 ##AUDIO_FEATURE_FLAGS
 
+AUDIO_HARDWARE := audio.a2dp.default
+AUDIO_HARDWARE += audio.usb.default
+AUDIO_HARDWARE += audio.r_submix.default
+AUDIO_HARDWARE += audio.primary.$(MSMSTEPPE)
+
+#HAL Wrapper
+AUDIO_WRAPPER := libqahw
+AUDIO_WRAPPER += libqahwwrapper
+
+#HAL Test app
+AUDIO_HAL_TEST_APPS := hal_play_test
+AUDIO_HAL_TEST_APPS += hal_rec_test
+
+PRODUCT_PACKAGES += $(AUDIO_HARDWARE)
+PRODUCT_PACKAGES += $(AUDIO_WRAPPER)
+PRODUCT_PACKAGES += $(AUDIO_HAL_TEST_APPS)
+
+ifeq ($(AUDIO_FEATURE_ENABLED_DLKM),true)
+BOARD_VENDOR_KERNEL_MODULES := \
+    $(KERNEL_MODULES_OUT)/audio_apr.ko \
+    $(KERNEL_MODULES_OUT)/audio_snd_event.ko \
+    $(KERNEL_MODULES_OUT)/audio_wglink.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_wcd.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_wcd934x.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_wcd_spi.ko \
+    $(KERNEL_MODULES_OUT)/audio_native.ko \
+    $(KERNEL_MODULES_OUT)/audio_machine_talos.ko
+endif
+
+BOARD_VENDOR_KERNEL_MODULES += $(shell ls $(KERNEL_MODULES_OUT)/*.ko)
+
+#Audio DLKM
+AUDIO_DLKM := audio_apr.ko
+AUDIO_DLKM += audio_snd_event.ko
+AUDIO_DLKM += audio_wglink.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_wcd.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_wcd_spi.ko
+AUDIO_DLKM += audio_native.ko
+AUDIO_DLKM += audio_machine_talos.ko
+AUDIO_DLKM += audio_wcd934x.ko
+AUDIO_DLKM += audio_pinctrl_lpi.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
+
+PRODUCT_PACKAGES += $(AUDIO_DLKM)
+
 #Audio Specific device overlays
 DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
 
@@ -123,6 +209,19 @@
 persist.vendor.audio.fluence.voicerec=false\
 persist.vendor.audio.fluence.speaker=true
 
+#
+#snapdragon value add features
+#
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.ssr=false
+
+##fluencetype can be "fluence" or "fluencepro" or "none"
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.fluencetype=none\
+persist.audio.fluence.voicecall=true\
+persist.audio.fluence.voicerec=false\
+persist.audio.fluence.speaker=true
+
 ##speaker protection v3 switch and ADSP AFE API version
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.vendor.audio.spv3.enable=true\
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index 74795c9..dc27ddc 100644
--- a/configs/sdm710/sdm710.mk
+++ b/configs/sdm710/sdm710.mk
@@ -4,7 +4,6 @@
 BOARD_USES_ALSA_AUDIO := true
 
 ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
-USE_CUSTOM_AUDIO_POLICY := 1
 AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
 AUDIO_FEATURE_ENABLED_COMPRESS_INPUT := true
 AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := false
@@ -78,6 +77,59 @@
 BOARD_SUPPORTS_FFV_EC_THREAD_RT_PRIORITY := false
 ##AUDIO_FEATURE_FLAGS
 
+AUDIO_HARDWARE := audio.a2dp.default
+AUDIO_HARDWARE += audio.usb.default
+AUDIO_HARDWARE += audio.r_submix.default
+AUDIO_HARDWARE += audio.primary.sdm710
+
+#HAL Wrapper
+AUDIO_WRAPPER := libqahw
+AUDIO_WRAPPER += libqahwwrapper
+
+#HAL Test app
+AUDIO_HAL_TEST_APPS := hal_play_test
+AUDIO_HAL_TEST_APPS += hal_rec_test
+
+PRODUCT_PACKAGES += $(AUDIO_HARDWARE)
+PRODUCT_PACKAGES += $(AUDIO_WRAPPER)
+PRODUCT_PACKAGES += $(AUDIO_HAL_TEST_APPS)
+
+ifeq ($(AUDIO_FEATURE_ENABLED_DLKM),true)
+BOARD_VENDOR_KERNEL_MODULES := \
+    $(KERNEL_MODULES_OUT)/qca_cld3_wlan.ko \
+    $(KERNEL_MODULES_OUT)/audio_apr.ko \
+    $(KERNEL_MODULES_OUT)/audio_wglink.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_wcd.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_cpe_lsm.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_wcd_spi.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd_cpe.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd9335.ko \
+    $(KERNEL_MODULES_OUT)/audio_wcd934x.ko \
+    $(KERNEL_MODULES_OUT)/audio_digital_cdc.ko \
+    $(KERNEL_MODULES_OUT)/audio_analog_cdc.ko \
+    $(KERNEL_MODULES_OUT)/audio_msm_sdw.ko \
+    $(KERNEL_MODULES_OUT)/audio_native.ko \
+    $(KERNEL_MODULES_OUT)/audio_machine_sdm710.ko
+endif
+
+BOARD_VENDOR_KERNEL_MODULES += \
+     $(KERNEL_MODULES_OUT)/llcc_perfmon.ko
+
 ifneq ($(strip $(TARGET_USES_RRO)), true)
 #Audio Specific device overlays
 DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
@@ -161,6 +213,19 @@
 persist.vendor.audio.fluence.audiorec=false\
 persist.vendor.audio.fluence.tmic.enabled=false
 
+#
+#snapdragon value add features
+#
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.ssr=false
+
+##fluencetype can be "fluence" or "fluencepro" or "none"
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.fluencetype=none\
+persist.audio.fluence.voicecall=true\
+persist.audio.fluence.voicerec=false\
+persist.audio.fluence.speaker=true
+
 # Mutlirec Apptype
 PRODUCT_PROPERTY_OVERRIDES += \
     vendor.audio.apptype.multirec.enabled=false \
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 98691f3..5710aaf 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -4,7 +4,6 @@
 BOARD_USES_ALSA_AUDIO := true
 
 ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
-USE_CUSTOM_AUDIO_POLICY := 1
 AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
 AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := false
 AUDIO_FEATURE_ENABLED_DYNAMIC_ECNS := false
@@ -66,6 +65,37 @@
 AUDIO_FEATURE_ENABLED_SVA_MULTI_STAGE := true
 ##AUDIO_FEATURE_FLAGS
 
+AUDIO_HARDWARE := audio.a2dp.default
+AUDIO_HARDWARE += audio.usb.default
+AUDIO_HARDWARE += audio.r_submix.default
+AUDIO_HARDWARE += audio.primary.sdm845
+
+#HAL Wrapper
+AUDIO_WRAPPER := libqahw
+AUDIO_WRAPPER += libqahwwrapper
+
+#HAL Test app
+AUDIO_HAL_TEST_APPS := hal_play_test
+AUDIO_HAL_TEST_APPS += hal_rec_test
+
+PRODUCT_PACKAGES += $(AUDIO_HARDWARE)
+PRODUCT_PACKAGES += $(AUDIO_WRAPPER)
+PRODUCT_PACKAGES += $(AUDIO_HAL_TEST_APPS)
+
+BOARD_VENDOR_KERNEL_MODULES := \
+    $(KERNEL_MODULES_OUT)/qca_cld3_wlan.ko \
+    $(KERNEL_MODULES_OUT)/wcd-core.ko \
+    $(KERNEL_MODULES_OUT)/pinctrl-wcd.ko \
+    $(KERNEL_MODULES_OUT)/swr-wcd-ctrl.ko \
+    $(KERNEL_MODULES_OUT)/snd-soc-wcd9xxx.ko \
+    $(KERNEL_MODULES_OUT)/wcd-dsp-glink.ko \
+    $(KERNEL_MODULES_OUT)/snd-soc-wcd934x.ko \
+    $(KERNEL_MODULES_OUT)/snd-soc-wcd-mbhc.ko \
+    $(KERNEL_MODULES_OUT)/snd-soc-wsa881x.ko \
+    $(KERNEL_MODULES_OUT)/snd-soc-sdm845.ko \
+    $(KERNEL_MODULES_OUT)/snd-soc-wcd-spi.ko \
+    $(KERNEL_MODULES_OUT)/llcc_perfmon.ko
+
 ifneq ($(strip $(TARGET_USES_RRO)), true)
 #Audio Specific device overlays
 DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
@@ -130,6 +160,19 @@
 persist.vendor.audio.fluence.voicerec=false\
 persist.vendor.audio.fluence.speaker=true
 
+#
+#snapdragon value add features
+#
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.ssr=false
+
+##fluencetype can be "fluence" or "fluencepro" or "none"
+PRODUCT_PROPERTY_OVERRIDES += \
+ro.qc.sdk.audio.fluencetype=none\
+persist.audio.fluence.voicecall=true\
+persist.audio.fluence.voicerec=false\
+persist.audio.fluence.speaker=true
+
 #disable tunnel encoding
 PRODUCT_PROPERTY_OVERRIDES += \
 vendor.audio.tunnel.encode=false
diff --git a/hal/Android.mk b/hal/Android.mk
index 1377660..2abed28 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -340,6 +340,12 @@
     LOCAL_STATIC_LIBRARIES += libprofile_rt
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUDIO_ZOOM)), true)
+    LOCAL_CFLAGS += -DAUDIOZOOM_QDSP_ENABLED
+    LOCAL_SRC_FILES += audio_extn/audiozoom.c
+endif
+
+
 #ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUTO_HAL)),true)
 #    LOCAL_CFLAGS += -DAUDIO_EXTN_AUTO_HAL_ENABLED
 #    LOCAL_SRC_FILES += audio_extn/auto_hal.c
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
index 400d7d0..19c839c 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -62,6 +62,7 @@
 #define MEDIA_FMT_LDAC                                     0x00013224
 #define MEDIA_FMT_MP3                                      0x00010BE9
 #define MEDIA_FMT_APTX_ADAPTIVE                            0x00013204
+#define MEDIA_FMT_APTX_AD_SPEECH                           0x00013208
 #define MEDIA_FMT_AAC_AOT_LC                               2
 #define MEDIA_FMT_AAC_AOT_SBR                              5
 #define MEDIA_FMT_AAC_AOT_PS                               29
@@ -85,6 +86,7 @@
 #define MIXER_SAMPLE_RATE_DEFAULT  "BT SampleRate"
 #define MIXER_AFE_IN_CHANNELS      "AFE Input Channels"
 #define MIXER_ABR_TX_FEEDBACK_PATH "A2DP_SLIM7_UL_HL Switch"
+#define MIXER_ABR_RX_FEEDBACK_PATH "SCO_SLIM7_DL_HL Switch"
 #define MIXER_SET_FEEDBACK_CHANNEL "BT set feedback channel"
 #define MIXER_SINK_SAMPLE_RATE     "BT_TX SampleRate"
 #define MIXER_AFE_SINK_CHANNELS    "AFE Output Channels"
@@ -125,6 +127,9 @@
 // Slimbus Tx sample rate for ABR feedback channel
 #define ABR_TX_SAMPLE_RATE             "KHZ_8"
 
+// Slimbus Tx sample rate for APTX AD SPEECH
+#define SPEECH_TX_SAMPLE_RATE             "KHZ_96"
+
 // Purpose ID for Inter Module Communication (IMC) in AFE
 #define IMC_PURPOSE_ID_BT_INFO         0x000132E2
 
@@ -134,8 +139,13 @@
 // Instance identifier for A2DP
 #define MAX_INSTANCE_ID                (UINT32_MAX / 2)
 
+// Instance identifier for SWB
+#define APTX_AD_SPEECH_INSTANCE_ID                 37
+
+#define SAMPLING_RATE_96K               96000
 #define SAMPLING_RATE_48K               48000
 #define SAMPLING_RATE_441K              44100
+#define SAMPLING_RATE_32K               32000
 #define CH_STEREO                       2
 #define CH_MONO                         1
 #define SOURCE 0
@@ -172,6 +182,7 @@
     CODEC_TYPE_LDAC = AUDIO_FORMAT_LDAC, // 0x23000000UL
     CODEC_TYPE_CELT = 603979776u, // 0x24000000UL
     CODEC_TYPE_APTX_AD = 620756992u, // 0x25000000UL
+    CODEC_TYPE_APTX_AD_SPEECH = 637534208u, //0x26000000UL
     CODEC_TYPE_PCM = AUDIO_FORMAT_PCM_16_BIT, // 0x1u
 }codec_t;
 
@@ -237,6 +248,11 @@
 } imc_status_t;
 
 typedef enum {
+    SWAP_DISABLE,
+    SWAP_ENABLE,
+} swap_status_t;
+
+typedef enum {
     MTU_SIZE,
     PEAK_BIT_RATE,
 } frame_control_type_t;
@@ -265,6 +281,8 @@
     bool abr_started;
     /* ABR Tx path pcm handle */
     struct pcm *abr_tx_handle;
+    /* ABR Rx path pcm handle */
+    struct pcm *abr_rx_handle;
     /* ABR Inter Module Communication (IMC) instance ID */
     uint32_t imc_instance;
 };
@@ -319,6 +337,7 @@
     uint32_t dec_channels;
     bool a2dp_sink_started;
     int  a2dp_sink_total_active_session_requests;
+    bool swb_configured;
 };
 
 struct a2dp_data a2dp;
@@ -400,6 +419,18 @@
     struct imc_dec_enc_info imc_info;
 } __attribute__ ((packed));
 
+struct aptx_ad_speech_mode_cfg_t
+{
+    uint32_t mode;
+    uint32_t swapping;
+} __attribute__ ((packed));
+
+/* Structure for SWB voice dec config */
+struct aptx_ad_speech_dec_cfg_t {
+    struct abr_dec_cfg_t abr_cfg;
+    struct aptx_ad_speech_mode_cfg_t speech_mode;
+} __attribute__ ((packed));
+
 /* START of DSP configurable structures
  * These values should match with DSP interface defintion
  */
@@ -546,6 +577,15 @@
     struct abr_enc_cfg_t abr_cfg;
 } __attribute__ ((packed));
 
+/* APTX AD SPEECH structure */
+struct aptx_ad_speech_enc_cfg_t
+{
+    struct custom_enc_cfg_t  custom_cfg;
+    /* Information to set up IMC between decoder and encoder */
+    struct imc_dec_enc_info imc_info;
+    struct aptx_ad_speech_mode_cfg_t speech_mode;
+} __attribute__ ((packed));
+
 struct ldac_specific_enc_cfg_t
 {
     uint32_t      bit_rate;
@@ -715,7 +755,9 @@
 static int stop_abr()
 {
     struct mixer_ctl *ctl_abr_tx_path = NULL;
+    struct mixer_ctl *ctl_abr_rx_path = NULL;
     struct mixer_ctl *ctl_set_bt_feedback_channel = NULL;
+    int ret = 0;
 
     /* This function can be used if !abr_started for clean up */
     ALOGV("%s: enter", __func__);
@@ -725,6 +767,10 @@
         pcm_close(a2dp.abr_config.abr_tx_handle);
         a2dp.abr_config.abr_tx_handle = NULL;
     }
+    if (a2dp.abr_config.abr_rx_handle != NULL) {
+        pcm_close(a2dp.abr_config.abr_rx_handle);
+        a2dp.abr_config.abr_rx_handle = NULL;
+    }
     a2dp.abr_config.abr_started = false;
     a2dp.abr_config.imc_instance = 0;
 
@@ -733,11 +779,10 @@
                                         MIXER_SET_FEEDBACK_CHANNEL);
     if (!ctl_set_bt_feedback_channel) {
         ALOGE("%s: ERROR Set usecase mixer control not identifed", __func__);
-        return -ENOSYS;
-    }
-    if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 0) != 0) {
+        ret = -ENOSYS;
+    } else if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 0) != 0) {
         ALOGE("%s: Failed to set BT usecase", __func__);
-        return -ENOSYS;
+        ret = -ENOSYS;
     }
 
     // Reset ABR Tx feedback path
@@ -746,19 +791,31 @@
                                         MIXER_ABR_TX_FEEDBACK_PATH);
     if (!ctl_abr_tx_path) {
         ALOGE("%s: ERROR ABR Tx feedback path mixer control not identifed", __func__);
-        return -ENOSYS;
-    }
-    if (mixer_ctl_set_value(ctl_abr_tx_path, 0, 0) != 0) {
+        ret = -ENOSYS;
+    } else if (mixer_ctl_set_value(ctl_abr_tx_path, 0, 0) != 0) {
         ALOGE("%s: Failed to set ABR Tx feedback path", __func__);
-        return -ENOSYS;
+        ret = -ENOSYS;
     }
 
-   return 0;
+    // Reset ABR Rx feedback path
+    ALOGV("%s: Disable ABR Rx feedback path", __func__);
+    ctl_abr_rx_path = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                        MIXER_ABR_RX_FEEDBACK_PATH);
+    if (!ctl_abr_rx_path) {
+        ALOGE("%s: ERROR ABR Rx feedback path mixer control not identifed", __func__);
+        ret = -ENOSYS;
+    } else if (mixer_ctl_set_value(ctl_abr_rx_path, 0, 0) != 0) {
+        ALOGE("%s: Failed to set ABR Rx feedback path", __func__);
+        ret = -ENOSYS;
+    }
+
+   return ret;
 }
 
 static int start_abr()
 {
     struct mixer_ctl *ctl_abr_tx_path = NULL;
+    struct mixer_ctl *ctl_abr_rx_path = NULL;
     struct mixer_ctl *ctl_set_bt_feedback_channel = NULL;
     int abr_device_id;
     int ret = 0;
@@ -792,11 +849,11 @@
                                         MIXER_SET_FEEDBACK_CHANNEL);
     if (!ctl_set_bt_feedback_channel) {
         ALOGE("%s: ERROR Set usecase mixer control not identifed", __func__);
-        return -ENOSYS;
+        goto fail;
     }
     if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 1) != 0) {
         ALOGE("%s: Failed to set BT usecase", __func__);
-        return -ENOSYS;
+        goto fail;
     }
 
     // Open hostless front end and prepare ABR Tx path
@@ -806,19 +863,60 @@
         a2dp.abr_config.abr_tx_handle = pcm_open(a2dp.adev->snd_card,
                                                  abr_device_id, PCM_IN,
                                                  &pcm_config_abr);
-        if (a2dp.abr_config.abr_tx_handle == NULL ||
-            !pcm_is_ready(a2dp.abr_config.abr_tx_handle))
+        if (a2dp.abr_config.abr_tx_handle == NULL) {
+            ALOGE("%s: Can't open abr tx device", __func__);
             goto fail;
+        }
+        if (!(pcm_is_ready(a2dp.abr_config.abr_tx_handle) &&
+              !pcm_start(a2dp.abr_config.abr_tx_handle))) {
+            ALOGE("%s: tx: %s", __func__, pcm_get_error(a2dp.abr_config.abr_tx_handle));
+            goto fail;
+        }
     }
-    ret = pcm_start(a2dp.abr_config.abr_tx_handle);
-    if (ret < 0)
-        goto fail;
+
+    // Enable Slimbus 7 Rx feedback path for HD Voice use case
+    if (a2dp.bt_encoder_format == CODEC_TYPE_APTX_AD_SPEECH) {
+        ALOGV("%s: Enable ABR Rx feedback path", __func__);
+        ctl_abr_rx_path = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                            MIXER_ABR_RX_FEEDBACK_PATH);
+        if (!ctl_abr_rx_path) {
+            ALOGE("%s: ERROR ABR Rx feedback path mixer control not identifed", __func__);
+            goto fail;
+        }
+        if (mixer_ctl_set_value(ctl_abr_rx_path, 0, 1) != 0) {
+            ALOGE("%s: Failed to set ABR Rx feedback path", __func__);
+            goto fail;
+        }
+
+        if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 1) != 0) {
+            ALOGE("%s: Failed to set BT usecase", __func__);
+            goto fail;
+        }
+
+        // Open hostless front end and prepare ABR Rx path
+        abr_device_id = fp_platform_get_pcm_device_id(USECASE_AUDIO_A2DP_ABR_FEEDBACK,
+                                                   PCM_PLAYBACK);
+        if (!a2dp.abr_config.abr_rx_handle) {
+            a2dp.abr_config.abr_rx_handle = pcm_open(a2dp.adev->snd_card,
+                                                     abr_device_id, PCM_OUT,
+                                                     &pcm_config_abr);
+            if (a2dp.abr_config.abr_rx_handle == NULL) {
+                ALOGE("%s: Can't open abr rx device", __func__);
+                goto fail;
+            }
+            if (!(pcm_is_ready(a2dp.abr_config.abr_rx_handle) &&
+                  !pcm_start(a2dp.abr_config.abr_rx_handle))) {
+                ALOGE("%s: rx: %s", __func__, pcm_get_error(a2dp.abr_config.abr_rx_handle));
+                goto fail;
+            }
+        }
+    }
+
     a2dp.abr_config.abr_started = true;
 
     return ret;
 
 fail:
-    ALOGE("%s: %s", __func__, pcm_get_error(a2dp.abr_config.abr_tx_handle));
     stop_abr();
     return -ENOSYS;
 }
@@ -965,6 +1063,7 @@
     a2dp.abr_config.abr_started = false;
     a2dp.abr_config.imc_instance = 0;
     a2dp.abr_config.abr_tx_handle = NULL;
+    a2dp.abr_config.abr_rx_handle = NULL;
     a2dp.bt_state_source = A2DP_STATE_DISCONNECTED;
 
     return 0;
@@ -1079,9 +1178,12 @@
 
         if (direction == SOURCE) {
             /* Set Tx backend sample rate */
-            if (a2dp.abr_config.is_abr_enabled)
-            rate_str = ABR_TX_SAMPLE_RATE;
-
+            if (a2dp.abr_config.is_abr_enabled) {
+                if (a2dp.bt_encoder_format == CODEC_TYPE_APTX_AD_SPEECH)
+                    rate_str = SPEECH_TX_SAMPLE_RATE;
+                else
+                    rate_str = ABR_TX_SAMPLE_RATE;
+            }
             ALOGD("%s: set backend tx sample rate = %s", __func__, rate_str);
             ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
                                             MIXER_SOURCE_SAMPLE_RATE_TX);
@@ -1282,17 +1384,18 @@
             ALOGE("%s: Failed to reset backend sample rate = %s", __func__, rate_str);
             return -ENOSYS;
         }
-
-        ctl_sample_rate_tx = mixer_get_ctl_by_name(a2dp.adev->mixer,
-                                        MIXER_SOURCE_SAMPLE_RATE_TX);
-        if (!ctl_sample_rate_tx) {
+        if (a2dp.abr_config.is_abr_enabled) {
+            ctl_sample_rate_tx = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                            MIXER_SOURCE_SAMPLE_RATE_TX);
+            if (!ctl_sample_rate_tx) {
                 ALOGE("%s: ERROR Tx backend sample rate mixer control not identifed", __func__);
                 return -ENOSYS;
-        }
+            }
 
-        if (mixer_ctl_set_enum_by_string(ctl_sample_rate_tx, rate_str) != 0) {
-            ALOGE("%s: Failed to reset Tx backend sample rate = %s", __func__, rate_str);
-            return -ENOSYS;
+            if (mixer_ctl_set_enum_by_string(ctl_sample_rate_tx, rate_str) != 0) {
+                ALOGE("%s: Failed to reset Tx backend sample rate = %s", __func__, rate_str);
+                return -ENOSYS;
+            }
         }
     } else {
 
@@ -1707,7 +1810,7 @@
 bool configure_aptx_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
 {
     struct mixer_ctl *ctl_enc_data = NULL;
-    int mixer_size;
+    int mixer_size = 0;
     bool is_configured = false;
     int ret = 0;
     int sample_rate_backup;
@@ -2209,7 +2312,7 @@
         return -ENOSYS;
     }
 
-    if (a2dp.a2dp_source_suspended == true) {
+    if (a2dp.a2dp_source_suspended || a2dp.swb_configured) {
         //session will be restarted after suspend completion
         ALOGD("a2dp start requested during suspend state");
         return -ENOSYS;
@@ -2434,6 +2537,16 @@
     }
 }
 
+static void reset_codec_config()
+{
+    reset_a2dp_enc_config_params();
+    reset_a2dp_source_dec_config_params();
+    a2dp_reset_backend_cfg(SOURCE);
+    if (a2dp.abr_config.is_abr_enabled && a2dp.abr_config.abr_started)
+        stop_abr();
+    a2dp.abr_config.is_abr_enabled = false;
+}
+
 int a2dp_stop_playback()
 {
     int ret =0;
@@ -2456,14 +2569,9 @@
             ALOGE("stop stream to BT IPC lib failed");
         else
             ALOGV("stop steam to BT IPC lib successful");
-        reset_a2dp_enc_config_params();
-        reset_a2dp_source_dec_config_params();
-        a2dp_reset_backend_cfg(SOURCE);
-        if (a2dp.abr_config.is_abr_enabled && a2dp.abr_config.abr_started)
-            stop_abr();
-        a2dp.abr_config.is_abr_enabled = false;
+        if (!a2dp.a2dp_source_suspended && !a2dp.swb_configured)
+            reset_codec_config();
         a2dp.a2dp_source_started = false;
-        a2dp_reset_backend_cfg(SOURCE);
     }
     if (!a2dp.a2dp_source_total_active_session_requests)
        a2dp.a2dp_source_started = false;
@@ -2578,8 +2686,8 @@
                         pthread_mutex_lock(&a2dp.adev->lock);
                     }
                 }
-                reset_a2dp_enc_config_params();
-                reset_a2dp_source_dec_config_params();
+                if (!a2dp.swb_configured)
+                    reset_codec_config();
                 if (a2dp.audio_source_suspend)
                    a2dp.audio_source_suspend();
             } else if (a2dp.a2dp_source_suspended == true) {
@@ -2699,8 +2807,11 @@
   a2dp.abr_config.abr_started = false;
   a2dp.abr_config.imc_instance = 0;
   a2dp.abr_config.abr_tx_handle = NULL;
+  a2dp.abr_config.abr_rx_handle = NULL;
   a2dp.is_tws_mono_mode_on = false;
   a2dp_source_init();
+  a2dp.swb_configured = false;
+
   // init function pointers
   fp_platform_get_pcm_device_id =
               init_config.fp_platform_get_pcm_device_id;
@@ -2802,3 +2913,112 @@
 
     return 0;
 }
+
+
+bool configure_aptx_ad_speech_enc_fmt() {
+    struct mixer_ctl *ctl_enc_data = NULL;
+    int mixer_size = 0;
+    int ret = 0;
+    struct aptx_ad_speech_enc_cfg_t aptx_dsp_cfg;
+
+    ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identifed");
+        return false;
+    }
+
+    /* Initialize dsp configuration params */
+    memset(&aptx_dsp_cfg, 0x0, sizeof(struct aptx_ad_speech_enc_cfg_t));
+    aptx_dsp_cfg.custom_cfg.enc_format = MEDIA_FMT_APTX_AD_SPEECH;
+    aptx_dsp_cfg.custom_cfg.sample_rate = SAMPLING_RATE_32K;
+    aptx_dsp_cfg.custom_cfg.num_channels = CH_MONO;
+    aptx_dsp_cfg.custom_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+    aptx_dsp_cfg.imc_info.direction = IMC_RECEIVE;
+    aptx_dsp_cfg.imc_info.enable = IMC_ENABLE;
+    aptx_dsp_cfg.imc_info.purpose = IMC_PURPOSE_ID_BT_INFO;
+    aptx_dsp_cfg.imc_info.comm_instance = APTX_AD_SPEECH_INSTANCE_ID;
+    aptx_dsp_cfg.speech_mode.mode = a2dp.adev->swb_speech_mode;
+    aptx_dsp_cfg.speech_mode.swapping = SWAP_ENABLE;
+
+    /* Configure AFE DSP configuration */
+    mixer_size = sizeof(struct aptx_ad_speech_enc_cfg_t);
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+                  mixer_size);
+    if (ret != 0) {
+        ALOGE("%s: Failed to set SWB encoder config", __func__);
+        return false;
+    }
+
+    /* Configure AFE Input Bit Format as PCM_16 */
+    ret = a2dp_set_bit_format(DEFAULT_ENCODER_BIT_FORMAT);
+    if (ret != 0) {
+        ALOGE("%s: Failed to set SWB bit format", __func__);
+        return false;
+    }
+
+    return true;
+}
+
+bool configure_aptx_ad_speech_dec_fmt()
+{
+    struct mixer_ctl *ctl_dec_data = NULL;
+    struct aptx_ad_speech_dec_cfg_t dec_cfg;
+    int ret = 0;
+
+    ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_SOURCE_DEC_CONFIG_BLOCK);
+    if (!ctl_dec_data) {
+        ALOGE("%s: ERROR codec config data mixer control not identifed", __func__);
+        return false;
+    }
+    memset(&dec_cfg, 0x0, sizeof(dec_cfg));
+    dec_cfg.abr_cfg.dec_format = MEDIA_FMT_APTX_AD_SPEECH;
+    dec_cfg.abr_cfg.imc_info.direction = IMC_TRANSMIT;
+    dec_cfg.abr_cfg.imc_info.enable = IMC_ENABLE;
+    dec_cfg.abr_cfg.imc_info.purpose = IMC_PURPOSE_ID_BT_INFO;
+    dec_cfg.abr_cfg.imc_info.comm_instance = APTX_AD_SPEECH_INSTANCE_ID;
+    dec_cfg.speech_mode.mode = a2dp.adev->swb_speech_mode;
+    dec_cfg.speech_mode.swapping = SWAP_ENABLE;
+
+    ret = mixer_ctl_set_array(ctl_dec_data, &dec_cfg,
+                              sizeof(dec_cfg));
+    if (ret != 0) {
+        ALOGE("%s: Failed to set decoder config", __func__);
+        return false;
+    }
+      return true;
+}
+
+int sco_start_configuration()
+{
+    ALOGD("sco_start_configuration start");
+
+    if (!a2dp.swb_configured) {
+        a2dp.bt_encoder_format = CODEC_TYPE_APTX_AD_SPEECH;
+        /* Configure AFE codec*/
+        if (configure_aptx_ad_speech_enc_fmt() &&
+            configure_aptx_ad_speech_dec_fmt()) {
+            ALOGD("%s: SCO enc/dec configured successfully", __func__);
+        } else {
+            ALOGE("%s: failed to send SCO configuration", __func__);
+            return -ETIMEDOUT;
+        }
+        /* Configure backend*/
+        a2dp.enc_sampling_rate = SAMPLING_RATE_96K;
+        a2dp.enc_channels = CH_MONO;
+        a2dp.abr_config.is_abr_enabled = true;
+        a2dp_set_backend_cfg(SOURCE);
+        /* Start abr*/
+        start_abr();
+        a2dp.swb_configured = true;
+    }
+    return 0;
+}
+
+void sco_reset_configuration()
+{
+    ALOGD("sco_reset_configuration start");
+
+    reset_codec_config();
+    a2dp.bt_encoder_format = CODEC_TYPE_INVALID;
+    a2dp.swb_configured = false;
+}
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 6afcc65..24ed4c5 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -3792,6 +3792,13 @@
 typedef int (*a2dp_stop_capture_t)();
 static a2dp_stop_capture_t a2dp_stop_capture;
 
+typedef int (*sco_start_configuration_t)();
+static sco_start_configuration_t sco_start_configuration;
+
+typedef void (*sco_reset_configuration_t)();
+static sco_reset_configuration_t sco_reset_configuration;
+
+
 int a2dp_offload_feature_init(bool is_feature_enabled)
 {
     ALOGD("%s: Called with feature %s", __func__,
@@ -3843,6 +3850,15 @@
             ALOGE("%s: dlsym failed", __func__);
             goto feature_disabled;
         }
+        // initialize APIs for SWB extension
+        if (!(sco_start_configuration =
+                 (sco_start_configuration_t)dlsym(a2dp_lib_handle, "sco_start_configuration")) ||
+             !(sco_reset_configuration =
+                 (sco_reset_configuration_t)dlsym(a2dp_lib_handle, "sco_reset_configuration"))) {
+            ALOGE("%s: dlsym failed for swb APIs", __func__);
+            sco_start_configuration = NULL;
+            sco_reset_configuration = NULL;
+        }
         ALOGD("%s:: ---- Feature A2DP_OFFLOAD is Enabled ----", __func__);
         return 0;
     }
@@ -3966,6 +3982,16 @@
     return (a2dp_stop_capture ? a2dp_stop_capture() : 0);
 }
 
+int audio_extn_sco_start_configuration()
+{
+    return (sco_start_configuration? sco_start_configuration() : 0);
+}
+
+void audio_extn_sco_reset_configuration()
+{
+    return (sco_reset_configuration? sco_reset_configuration() : 0);
+}
+
 // END: A2DP_OFFLOAD =====================================================================
 
 // START: HFP ======================================================================
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index adcb459..e1e3ca0 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -313,6 +313,9 @@
 bool audio_extn_a2dp_source_is_suspended();
 int audio_extn_a2dp_start_capture();
 int audio_extn_a2dp_stop_capture();
+int audio_extn_sco_start_configuration();
+void audio_extn_sco_reset_configuration();
+
 
 // --- Function pointers from audio_extn needed by A2DP_OFFLOAD
 typedef int (*fp_check_a2dp_restore_t)(struct audio_device *,
diff --git a/hal/audio_extn/audiozoom.c b/hal/audio_extn/audiozoom.c
new file mode 100644
index 0000000..73e4862
--- /dev/null
+++ b/hal/audio_extn/audiozoom.c
@@ -0,0 +1,257 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_audiozoom"
+/*#define LOG_NDEBUG 0*/
+
+#include <errno.h>
+#include <log/log.h>
+#include <stdlib.h>
+#include <expat.h>
+#include <audio_hw.h>
+#include <system/audio.h>
+#include <platform_api.h>
+#include "audio_extn.h"
+
+#include "audiozoom.h"
+
+#include <resolv.h>
+
+#define AUDIOZOOM_PRESET_FILE "/vendor/etc/audiozoom.xml"
+
+typedef struct qdsp_audiozoom_cfg {
+    uint32_t             topo_id;
+    uint32_t             module_id;
+    uint32_t             instance_id;
+    uint32_t             zoom_param_id;
+    uint32_t             wide_param_id;
+    uint32_t             dir_param_id;
+    uint32_t             app_type;
+} qdsp_audiozoom_cfg_t;
+
+static qdsp_audiozoom_cfg_t qdsp_audiozoom;
+
+static void start_tag(void *userdata __unused, const XML_Char *tag_name,
+                      const XML_Char **attr)
+{
+    uint32_t index = 0;
+
+    if (!attr) {
+        ALOGE("%s: NULL platform/tag_name/attr", __func__);
+        return;
+    }
+
+    if (strcmp(tag_name, "topo") == 0) {
+        if (strcmp(attr[0], "id") == 0) {
+            if (attr[1])
+                qdsp_audiozoom.topo_id = atoi(attr[1]);
+        }
+    } else if (strcmp(tag_name, "module") == 0) {
+        if (strcmp(attr[0], "id") == 0) {
+            if (attr[1])
+                qdsp_audiozoom.module_id = atoi(attr[1]);
+        }
+    } else if (strcmp(tag_name, "param") == 0) {
+        while (attr[index] != NULL) {
+            if (strcmp(attr[index], "zoom_id") == 0) {
+                index++;
+                if (attr[index])
+                    qdsp_audiozoom.zoom_param_id = atoi(attr[index]);
+                else
+                    break;
+            } else if (strcmp(attr[index], "wide_id") == 0) {
+                index++;
+                if (attr[index])
+                    qdsp_audiozoom.wide_param_id = atoi(attr[index]);
+                else
+                    break;
+            } else if (strcmp(attr[index], "dir_id") == 0) {
+                index++;
+                if (attr[index])
+                    qdsp_audiozoom.dir_param_id = atoi(attr[index]);
+                else
+                    break;
+            }
+            index++;
+        }
+    } else if (strcmp(tag_name, "app_type") == 0) {
+        if (strcmp(attr[0], "id") == 0) {
+            if (attr[1])
+                qdsp_audiozoom.app_type = atoi(attr[1]);
+        }
+    } else if (strcmp(tag_name, "instance") == 0) {
+        if (strcmp(attr[0], "id") == 0) {
+            if (attr[1])
+                qdsp_audiozoom.instance_id = atoi(attr[1]);
+        }
+    } else {
+        ALOGE("%s: %s is not a supported tag", __func__, tag_name);
+    }
+
+    return;
+}
+
+static void end_tag(void *userdata __unused, const XML_Char *tag_name)
+{
+    if (strcmp(tag_name, "topo") == 0) {
+    } else if (strcmp(tag_name, "module") == 0) {
+    } else if (strcmp(tag_name, "param") == 0) {
+    } else if (strcmp(tag_name, "app_type") == 0) {
+    } else if (strcmp(tag_name, "instance") == 0) {
+    } else {
+        ALOGE("%s: %s is not a supported tag", __func__, tag_name);
+    }
+}
+
+static int audio_extn_audiozoom_parse_info(const char *filename)
+{
+    XML_Parser      parser;
+    FILE            *file;
+    int             ret = 0;
+    int             bytes_read;
+    void            *buf;
+    static const uint32_t kBufSize = 1024;
+
+    file = fopen(filename, "r");
+    if (!file) {
+        ALOGE("%s: Failed to open %s", __func__, filename);
+        ret = -ENODEV;
+        goto done;
+    }
+
+    parser = XML_ParserCreate(NULL);
+    if (!parser) {
+        ALOGE("%s: Failed to create XML parser!", __func__);
+        ret = -ENODEV;
+        goto err_close_file;
+    }
+
+    XML_SetElementHandler(parser, start_tag, end_tag);
+
+    while (1) {
+        buf = XML_GetBuffer(parser, kBufSize);
+        if (buf == NULL) {
+            ALOGE("%s: XML_GetBuffer failed", __func__);
+            ret = -ENOMEM;
+            goto err_free_parser;
+        }
+
+        bytes_read = fread(buf, 1, kBufSize, file);
+        if (bytes_read < 0) {
+            ALOGE("%s: fread failed, bytes read = %d", __func__, bytes_read);
+             ret = bytes_read;
+            goto err_free_parser;
+        }
+
+        if (XML_ParseBuffer(parser, bytes_read,
+                            bytes_read == 0) == XML_STATUS_ERROR) {
+            ALOGE("%s: XML_ParseBuffer failed, for %s",
+                __func__, filename);
+            ret = -EINVAL;
+            goto err_free_parser;
+        }
+
+        if (bytes_read == 0)
+            break;
+    }
+
+err_free_parser:
+    XML_ParserFree(parser);
+err_close_file:
+    fclose(file);
+done:
+    return ret;
+}
+
+int audio_extn_audiozoom_set_microphone_direction(
+    struct stream_in *in, audio_microphone_direction_t dir)
+{
+    (void)in;
+    (void)dir;
+    return 0;
+}
+
+static int audio_extn_audiozoom_set_microphone_field_dimension_zoom(
+    struct stream_in *in, float zoom)
+{
+    struct audio_device *adev = in->dev;
+    struct str_parms *parms = str_parms_create();
+    /* The encoding process in b64_ntop represents 24-bit groups of input bits
+       as output strings of 4 encoded characters. */
+    char data[((sizeof(zoom) + 2) / 3) * 4 + 1] = {0};
+    int32_t ret;
+
+    if (zoom > 1.0 || zoom < 0)
+        return -EINVAL;
+
+    if (qdsp_audiozoom.topo_id == 0 || qdsp_audiozoom.module_id == 0 ||
+        qdsp_audiozoom.zoom_param_id == 0)
+        return -ENOSYS;
+
+    str_parms_add_int(parms, "cal_devid", in->device);
+    str_parms_add_int(parms, "cal_apptype", in->app_type_cfg.app_type);
+    str_parms_add_int(parms, "cal_topoid", qdsp_audiozoom.topo_id);
+    str_parms_add_int(parms, "cal_moduleid", qdsp_audiozoom.module_id);
+    str_parms_add_int(parms, "cal_instanceid", qdsp_audiozoom.instance_id);
+    str_parms_add_int(parms, "cal_paramid", qdsp_audiozoom.zoom_param_id);
+
+    ret = b64_ntop((uint8_t*)&zoom, sizeof(zoom), data, sizeof(data));
+    if (ret > 0) {
+        str_parms_add_str(parms, "cal_data", data);
+
+        platform_set_parameters(adev->platform, parms);
+    } else {
+        ALOGE("%s: failed to convert data to string, ret %d", __func__, ret);
+    }
+
+    str_parms_destroy(parms);
+
+    return 0;
+}
+
+static int audio_extn_audiozoom_set_microphone_field_dimension_wide_angle(
+    struct stream_in *in, float zoom)
+{
+    (void)in;
+    (void)zoom;
+    return 0;
+}
+
+int audio_extn_audiozoom_set_microphone_field_dimension(
+    struct stream_in *in, float zoom)
+{
+    if (zoom > 1.0 || zoom < -1.0)
+        return -EINVAL;
+
+    if (zoom >= 0 && zoom <= 1.0)
+        return audio_extn_audiozoom_set_microphone_field_dimension_zoom(in, zoom);
+
+    if (zoom >= -1.0 && zoom <= 0)
+        return audio_extn_audiozoom_set_microphone_field_dimension_wide_angle(in, zoom);
+
+    return 0;
+}
+
+int audio_extn_audiozoom_init()
+{
+    audio_extn_audiozoom_parse_info(AUDIOZOOM_PRESET_FILE);
+
+    ALOGV("%s: topo_id=%d, module_id=%d, instance_id=%d, zoom__id=%d, dir_id=%d, app_type=%d",
+        __func__, qdsp_audiozoom.topo_id, qdsp_audiozoom.module_id, qdsp_audiozoom.instance_id,
+        qdsp_audiozoom.zoom_param_id, qdsp_audiozoom.dir_param_id,qdsp_audiozoom.app_type);
+
+    return 0;
+}
diff --git a/hal/audio_extn/audiozoom.h b/hal/audio_extn/audiozoom.h
new file mode 100644
index 0000000..2c0ad71
--- /dev/null
+++ b/hal/audio_extn/audiozoom.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AUDIOZOOM_H_
+#define AUDIOZOOM_H_
+
+#ifndef AUDIOZOOM_QDSP_ENABLED
+#define audio_extn_audiozoom_init()                                          (0)
+#define audio_extn_audiozoom_set_microphone_direction(stream, dir)           (-ENOSYS)
+#define audio_extn_audiozoom_set_microphone_field_dimension(stream, zoom)    (-ENOSYS)
+#else
+int audio_extn_audiozoom_init();
+int audio_extn_audiozoom_set_microphone_direction(struct stream_in *stream,
+                                           audio_microphone_direction_t dir);
+int audio_extn_audiozoom_set_microphone_field_dimension(struct stream_in *stream, float zoom);
+#endif
+
+#endif /* AUDIOZOOM_H_ */
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 254ecb1..2294b0c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -75,6 +75,7 @@
 #include "voice_extn.h"
 #include "ip_hdlr_intf.h"
 #include "audio_feature_manager.h"
+#include "audio_extn/audiozoom.h"
 
 #include "sound/compress_params.h"
 #include "sound/asound.h"
@@ -506,6 +507,11 @@
 }
 #endif
 
+static int in_set_microphone_direction(const struct audio_stream_in *stream,
+                                           audio_microphone_direction_t dir);
+static int in_set_microphone_field_dimension(const struct audio_stream_in *stream, float zoom);
+
+
 static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
                                int flags __unused)
 {
@@ -1183,6 +1189,14 @@
             goto err;
         }
 
+        if (((SND_DEVICE_OUT_BT_SCO_SWB == snd_device) ||
+             (SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC == snd_device) ||
+             (SND_DEVICE_IN_BT_SCO_MIC_SWB == snd_device)) &&
+            (audio_extn_sco_start_configuration() < 0)) {
+            ALOGE(" fail to configure sco control path ");
+            goto err;
+        }
+
         /* 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,
@@ -1275,7 +1289,14 @@
             audio_extn_a2dp_stop_playback();
         else if (snd_device == SND_DEVICE_IN_BT_A2DP)
             audio_extn_a2dp_stop_capture();
-        else if ((snd_device == SND_DEVICE_OUT_HDMI) ||
+        else if ((snd_device == SND_DEVICE_OUT_BT_SCO_SWB) ||
+                 (snd_device == SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC) ||
+                 (snd_device == SND_DEVICE_IN_BT_SCO_MIC_SWB)) {
+            if ((adev->snd_dev_ref_cnt[SND_DEVICE_OUT_BT_SCO_SWB] == 0) &&
+                (adev->snd_dev_ref_cnt[SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] == 0) &&
+                (adev->snd_dev_ref_cnt[SND_DEVICE_IN_BT_SCO_MIC_SWB] == 0))
+                audio_extn_sco_reset_configuration();
+       } else if ((snd_device == SND_DEVICE_OUT_HDMI) ||
                 (snd_device == SND_DEVICE_OUT_DISPLAY_PORT))
             adev->is_channel_status_set = false;
         else if ((snd_device == SND_DEVICE_OUT_HEADPHONES) &&
@@ -2761,6 +2782,8 @@
     }
 
     check_and_enable_effect(adev);
+    audio_extn_audiozoom_set_microphone_direction(in, in->zoom);
+    audio_extn_audiozoom_set_microphone_field_dimension(in, in->direction);
 
 done_open:
     audio_extn_perf_lock_release(&adev->perf_lock_handle);
@@ -4373,6 +4396,37 @@
     return ret;
 }
 
+static int in_set_microphone_direction(const struct audio_stream_in *stream,
+                                           audio_microphone_direction_t dir) {
+    struct stream_in *in = (struct stream_in *)stream;
+
+    ALOGVV("%s: standby %d source %d dir %d", __func__, in->standby, in->source, dir);
+
+    in->direction = dir;
+
+    if (in->standby)
+        return 0;
+
+    return audio_extn_audiozoom_set_microphone_direction(in, dir);
+}
+
+static int in_set_microphone_field_dimension(const struct audio_stream_in *stream, float zoom) {
+    struct stream_in *in = (struct stream_in *)stream;
+
+    ALOGVV("%s: standby %d source %d zoom %f", __func__, in->standby, in->source, zoom);
+
+    if (zoom > 1.0 || zoom < -1.0)
+        return -EINVAL;
+
+    in->zoom = zoom;
+
+    if (in->standby)
+        return 0;
+
+    return audio_extn_audiozoom_set_microphone_field_dimension(in, zoom);
+}
+
+
 static bool stream_get_parameter_channels(struct str_parms *query,
                                           struct str_parms *reply,
                                           audio_channel_mask_t *supported_channel_masks) {
@@ -6602,6 +6656,53 @@
 
     return ret;
 }
+
+static void in_update_sink_metadata(struct audio_stream_in *stream,
+                                    const struct sink_metadata *sink_metadata) {
+
+    if (stream == NULL
+            || sink_metadata == NULL
+            || sink_metadata->tracks == NULL) {
+        return;
+    }
+
+    int error = 0;
+    struct stream_in *in = (struct stream_in *)stream;
+    struct audio_device *adev = in->dev;
+    audio_devices_t device = AUDIO_DEVICE_NONE;
+
+    if (sink_metadata->track_count != 0)
+        device = sink_metadata->tracks->dest_device;
+
+    lock_input_stream(in);
+    pthread_mutex_lock(&adev->lock);
+    ALOGV("%s: in->usecase: %d, device: %x", __func__, in->usecase, device);
+
+    if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY
+            && device != AUDIO_DEVICE_NONE
+            && adev->voice_tx_output != NULL) {
+        /* Use the rx device from afe-proxy record to route voice call because
+           there is no routing if tx device is on primary hal and rx device
+           is on other hal during voice call. */
+        adev->voice_tx_output->devices = device;
+
+        if (!voice_is_call_state_active(adev)) {
+            if (adev->mode == AUDIO_MODE_IN_CALL) {
+                adev->current_call_output = adev->voice_tx_output;
+                error = voice_start_call(adev);
+                if (error != 0)
+                    ALOGE("%s: start voice call failed %d", __func__, error);
+            }
+        } else {
+            adev->current_call_output = adev->voice_tx_output;
+            voice_update_devices_for_all_voice_usecases(adev);
+        }
+    }
+
+    pthread_mutex_unlock(&adev->lock);
+    pthread_mutex_unlock(&in->lock);
+}
+
 int adev_open_output_stream(struct audio_hw_device *dev,
                             audio_io_handle_t handle,
                             audio_devices_t devices,
@@ -7574,6 +7675,12 @@
             adev->bt_wb_speech_enabled = false;
     }
 
+    ret = str_parms_get_str(parms, "bt_swb", value, sizeof(value));
+    if (ret >= 0) {
+        val = atoi(value);
+        adev->swb_speech_mode = val;
+    }
+
     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
     if (ret >= 0) {
         val = atoi(value);
@@ -8021,6 +8128,9 @@
     in->stream.get_input_frames_lost = in_get_input_frames_lost;
     in->stream.get_capture_position = in_get_capture_position;
     in->stream.get_active_microphones = in_get_active_microphones;
+    in->stream.set_microphone_direction = in_set_microphone_direction;
+    in->stream.set_microphone_field_dimension = in_set_microphone_field_dimension;
+    in->stream.update_sink_metadata = in_update_sink_metadata;
 
     in->device = devices;
     in->source = source;
@@ -8030,6 +8140,8 @@
     in->flags = flags;
     in->bit_width = 16;
     in->af_period_multiplier = 1;
+    in->direction = MIC_DIRECTION_UNSPECIFIED;
+    in->zoom = 0;
 
     ALOGV("%s: source = %d, config->channel_mask = %d", __func__, source, config->channel_mask);
     if (source == AUDIO_SOURCE_VOICE_UPLINK ||
@@ -9014,6 +9126,7 @@
 
     adev->enable_voicerx = false;
     adev->bt_wb_speech_enabled = false;
+    adev->swb_speech_mode = SPEECH_MODE_INVALID;
     //initialize this to false for now,
     //this will be set to true through set param
     adev->vr_audio_mode_enabled = false;
@@ -9090,6 +9203,7 @@
     adev->is_charging = audio_extn_battery_properties_is_charging();
     audio_extn_sound_trigger_init(adev); /* dependent on snd_mon_init() */
     audio_extn_sound_trigger_update_battery_status(adev->is_charging);
+    audio_extn_audiozoom_init();
     pthread_mutex_unlock(&adev->lock);
     /* Allocate memory for Device config params */
     adev->device_cfg_params = (struct audio_device_config_param*)
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 69c13e9..1a04cff 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -85,6 +85,9 @@
 #define ACDB_DEV_TYPE_OUT 1
 #define ACDB_DEV_TYPE_IN 2
 
+/* SCO SWB codec mode */
+#define SPEECH_MODE_INVALID  0xFFFF
+
 /* support positional and index masks to 8ch */
 #define MAX_SUPPORTED_CHANNEL_MASKS (2 * FCC_8)
 #define MAX_SUPPORTED_FORMATS 15
@@ -409,6 +412,8 @@
     struct audio_device *dev;
     card_status_t card_status;
     int capture_started;
+    float zoom;
+    audio_microphone_direction_t direction;
 
     /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
     audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
@@ -525,6 +530,7 @@
     unsigned int cur_hdmi_bit_width;
     unsigned int cur_wfd_channels;
     bool bt_wb_speech_enabled;
+    unsigned int swb_speech_mode;
     bool allow_afe_proxy_usage;
     bool is_charging; // from battery listener
     bool mic_break_enabled;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 45fadf5..7b15f53 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -6159,7 +6159,6 @@
         ALOGV("%s: Format doesnt have to be set", __func__);
     }
 
-    format = format & AUDIO_FORMAT_MAIN_MASK;
     /* Set data format only if there is a change from PCM to compressed
        and vice versa */
     if (set_mi2s_tx_data_format && (format ^ my_data->current_backend_cfg[backend_idx].format)) {
@@ -6169,7 +6168,7 @@
                   __func__, ext_disp_format);
             return -EINVAL;
         }
-        if (format == AUDIO_FORMAT_PCM) {
+        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
             ALOGE("%s:MI2S data format LPCM", __func__);
             mixer_ctl_set_enum_by_string(ctl, "LPCM");
         } else {
@@ -6724,18 +6723,25 @@
               __func__, bit_width, sample_rate, channels);
     }
 
-    ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
-          "sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
+    ALOGI("%s:txbecf: afe: current backend bit_width %d sample_rate %d channels %d, format %x",
+                            __func__,
+                            my_data->current_backend_cfg[backend_idx].bit_width,
+                            my_data->current_backend_cfg[backend_idx].sample_rate,
+                            my_data->current_backend_cfg[backend_idx].channels,
+                            my_data->current_backend_cfg[backend_idx].format);
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
+    // Note that below if block would be entered even if main format is same
+    // but subformat is different for e.g. current backend is configured for 16 bit PCM
+    // as compared to 24 bit PCM backend requested
     if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
         (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
         (channels != my_data->current_backend_cfg[backend_idx].channels) ||
-        ((format & AUDIO_FORMAT_MAIN_MASK) != my_data->current_backend_cfg[backend_idx].format)) {
+        (format != my_data->current_backend_cfg[backend_idx].format)) {
         backend_cfg->bit_width = bit_width;
         backend_cfg->sample_rate= sample_rate;
         backend_cfg->channels = channels;
-        backend_cfg->format = format & AUDIO_FORMAT_MAIN_MASK;
+        backend_cfg->format = format;
         backend_change = true;
         ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
               "new sample rate: %d new channel: %d new format: %d",
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 01df5bf..1c46ed7 100755
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -684,6 +684,10 @@
                  sizeof("sm6150-qrd-snd-card"))) {
         hw_info->is_stereo_spkr = false;
         strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
+    } else if (!strncmp(snd_card_name, "sm6150-wcd9375qrd-snd-card",
+                 sizeof("sm6150-wcd9375qrd-snd-card"))) {
+        strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
+        hw_info->is_stereo_spkr = false;
     } else if (!strncmp(snd_card_name, "sm6150-tavil-snd-card",
                  sizeof("sm6150-tavil-snd-card"))) {
         strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index a428f9a..79b78ca 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -498,6 +498,7 @@
     [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = "speaker-and-display-port",
     [SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
     [SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+    [SND_DEVICE_OUT_BT_SCO_SWB] = "bt-sco-headset-swb",
     [SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
     [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
     [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] = "speaker-safe-and-bt-a2dp",
@@ -544,9 +545,12 @@
     [SND_DEVICE_OUT_SPEAKER_AND_BT_SCO] = "speaker-and-bt-sco",
     [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO] = "speaker-safe-and-bt-sco",
     [SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB] = "speaker-and-bt-sco-wb",
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB] = "speaker-and-bt-sco-swb",
     [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] = "speaker-safe-and-bt-sco-wb",
+    [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] = "speaker-safe-and-bt-sco-swb",
     [SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO] = "wsa-speaker-and-bt-sco",
     [SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB] = "wsa-speaker-and-bt-sco-wb",
+    [SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB] = "wsa-speaker-and-bt-sco-wb",
 
     /* Capture sound devices */
     [SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -593,6 +597,8 @@
     [SND_DEVICE_IN_BT_SCO_MIC_NREC] = "bt-sco-mic",
     [SND_DEVICE_IN_BT_SCO_MIC_WB] = "bt-sco-mic-wb",
     [SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = "bt-sco-mic-wb",
+    [SND_DEVICE_IN_BT_SCO_MIC_SWB] = "bt-sco-mic-swb",
+    [SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = "bt-sco-mic-swb",
     [SND_DEVICE_IN_BT_A2DP] = "bt-a2dp-cap",
     [SND_DEVICE_IN_CAMCORDER_MIC] = "camcorder-mic",
     [SND_DEVICE_IN_VOICE_DMIC] = "voice-dmic-ef",
@@ -751,7 +757,9 @@
     [SND_DEVICE_OUT_BT_SCO] = 22,
     [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO] = 14,
     [SND_DEVICE_OUT_BT_SCO_WB] = 39,
+    [SND_DEVICE_OUT_BT_SCO_SWB] = 39,
     [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] = 14,
+    [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] = 14,
     [SND_DEVICE_OUT_BT_A2DP] = 20,
     [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
     [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] = 14,
@@ -834,6 +842,8 @@
     [SND_DEVICE_IN_BT_SCO_MIC_NREC] = 122,
     [SND_DEVICE_IN_BT_SCO_MIC_WB] = 38,
     [SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = 123,
+    [SND_DEVICE_IN_BT_SCO_MIC_SWB] = 38,
+    [SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = 123,
     [SND_DEVICE_IN_BT_A2DP] = 21,
     [SND_DEVICE_IN_CAMCORDER_MIC] = 4,
     [SND_DEVICE_IN_VOICE_DMIC] = 41,
@@ -954,7 +964,9 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_SWB)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP)},
@@ -965,8 +977,10 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_USB)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_USB)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TX)},
@@ -1044,6 +1058,8 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_NREC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_WB)},
     {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_WB_NREC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_SWB)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_BT_A2DP)},
     {TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_DMIC)},
@@ -1515,7 +1531,9 @@
          !strncmp(snd_card_name, "sdm439-sku1-snd-card",
                    sizeof("sdm439-sku1-snd-card")) ||
          !strncmp(snd_card_name, "sdm439-snd-card-mtp",
-                   sizeof("sdm439-snd-card-mtp"))) {
+                   sizeof("sdm439-snd-card-mtp")) ||
+         !strncmp(snd_card_name, "sm6150-wcd9375qrd-snd-card",
+                   sizeof("sm6150-wcd9375qrd-snd-card"))) {
          ALOGI("%s: snd_card_name: %s", __func__, snd_card_name);
          my_data->is_internal_codec = true;
          my_data->is_slimbus_interface = false;
@@ -1894,11 +1912,14 @@
     backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_WB] = strdup("bt-sco-wb");
     backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_NREC] = strdup("bt-sco");
     backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = strdup("bt-sco-wb");
+    backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_SWB] = strdup("bt-sco-swb");
+    backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = strdup("bt-sco-swb");
     backend_tag_table[SND_DEVICE_IN_SPDIF] = strdup("spdif-in");
     backend_tag_table[SND_DEVICE_IN_HDMI_MIC] = strdup("hdmi-in");
     backend_tag_table[SND_DEVICE_IN_HDMI_ARC] = strdup("hdmi-arc-in");
     backend_tag_table[SND_DEVICE_OUT_BT_SCO] = strdup("bt-sco");
     backend_tag_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
+    backend_tag_table[SND_DEVICE_OUT_BT_SCO_SWB] = strdup("bt-sco-swb");
     backend_tag_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
     backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
     backend_tag_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("display-port");
@@ -1920,6 +1941,8 @@
         strdup("speaker-safe-and-bt-sco");
     backend_tag_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] =
         strdup("speaker-safe-and-bt-sco-wb");
+    backend_tag_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] =
+        strdup("speaker-safe-and-bt-sco-swb");
     backend_tag_table[SND_DEVICE_IN_VOICE_TTY_FULL_USB_MIC] = strdup("usb-headset-mic");
     backend_tag_table[SND_DEVICE_IN_VOICE_TTY_HCO_USB_MIC] = strdup("usb-headset-mic");
     backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
@@ -1987,6 +2010,7 @@
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("SLIMBUS_0_RX-and-DISPLAY_PORT");
     hw_interface_table[SND_DEVICE_OUT_BT_SCO] = strdup("SLIMBUS_7_RX");
     hw_interface_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("SLIMBUS_7_RX");
+    hw_interface_table[SND_DEVICE_OUT_BT_SCO_SWB] = strdup("SLIMBUS_7_RX");
     hw_interface_table[SND_DEVICE_OUT_BT_A2DP] = strdup("SLIMBUS_7_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("SLIMBUS_0_RX-and-SLIMBUS_7_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] =
@@ -2020,8 +2044,10 @@
     hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_HFP] = strdup("SLIMBUS_0_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO] = strdup("SLIMBUS_0_RX-and-SEC_AUX_PCM_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB] = strdup("SLIMBUS_0_RX-and-SEC_AUX_PCM_RX");
+    hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB] = strdup("SLIMBUS_0_RX-and-SEC_AUX_PCM_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO] = strdup("QUAT_TDM_RX_0-and-SLIMBUS_7_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] = strdup("QUAT_TDM_RX_0-and-SLIMBUS_7_RX");
+    hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] = strdup("QUAT_TDM_RX_0-and-SLIMBUS_7_RX");
     hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = strdup("SLIMBUS_0_RX");
     hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_STEREO_PROTECTED] = strdup("SLIMBUS_0_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = strdup("SLIMBUS_0_RX");
@@ -2076,6 +2102,8 @@
     hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_NREC] = strdup("SLIMBUS_7_TX");
     hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_WB] = strdup("SLIMBUS_7_TX");
     hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = strdup("SLIMBUS_7_TX");
+    hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_SWB] = strdup("SLIMBUS_7_TX");
+    hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = strdup("SLIMBUS_7_TX");
     hw_interface_table[SND_DEVICE_IN_BT_A2DP] = strdup("SLIMBUS_7_TX");
     hw_interface_table[SND_DEVICE_IN_CAMCORDER_MIC] = strdup("SLIMBUS_0_TX");
     hw_interface_table[SND_DEVICE_IN_VOICE_DMIC] = strdup("SLIMBUS_0_TX");
@@ -2987,6 +3015,9 @@
     else if (!strncmp(snd_card_name, "sm6150-qrd-snd-card",
                sizeof("sm6150-qrd-snd-card")))
         platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
+    else if (!strncmp(snd_card_name, "sm6150-wcd9375qrd-snd-card",
+               sizeof("sm6150-wcd9375qrd-snd-card")))
+        platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
     else if (!strncmp(snd_card_name, "kona-qrd-snd-card",
                sizeof("kona-qrd-snd-card")))
         platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
@@ -4940,6 +4971,19 @@
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_SAFE;
         new_snd_devices[1] = SND_DEVICE_OUT_BT_SCO_WB;
         ret = 0;
+    } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB &&
+               !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_BT_SCO_SWB)) {
+        *num_devices = 2;
+        new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+        new_snd_devices[1] = SND_DEVICE_OUT_BT_SCO_SWB;
+        ret = 0;
+    } else if (snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB &&
+               !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER_SAFE,
+                                              SND_DEVICE_OUT_BT_SCO_SWB)) {
+        *num_devices = 2;
+        new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_SAFE;
+        new_snd_devices[1] = SND_DEVICE_OUT_BT_SCO_SWB;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_USB_HEADSET &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER_SAFE, SND_DEVICE_OUT_USB_HEADSET)) {
         *num_devices = 2;
@@ -5131,19 +5175,29 @@
             snd_device = SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP;
         } else if ((devices & AUDIO_DEVICE_OUT_ALL_SCO) &&
                    ((devices & ~AUDIO_DEVICE_OUT_ALL_SCO) == AUDIO_DEVICE_OUT_SPEAKER)) {
-            if (my_data->is_wsa_speaker)
-                snd_device = adev->bt_wb_speech_enabled ?
-                        SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB :
-                        SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO;
-            else
-                snd_device = adev->bt_wb_speech_enabled ?
-                        SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB :
-                        SND_DEVICE_OUT_SPEAKER_AND_BT_SCO;
+            if (my_data->is_wsa_speaker) {
+                if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+                    snd_device = SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB;
+                else
+                    snd_device = adev->bt_wb_speech_enabled ?
+                            SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB :
+                            SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO;
+            } else {
+                if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+                    snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB;
+                else
+                    snd_device = adev->bt_wb_speech_enabled ?
+                            SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB :
+                            SND_DEVICE_OUT_SPEAKER_AND_BT_SCO;
+            }
         } else if ((devices & AUDIO_DEVICE_OUT_ALL_SCO) &&
                          ((devices & ~AUDIO_DEVICE_OUT_ALL_SCO) == AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
-            snd_device = adev->bt_wb_speech_enabled ?
-                    SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB :
-                    SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO;
+            if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+                snd_device = SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB;
+            else
+                snd_device = adev->bt_wb_speech_enabled ?
+                        SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB :
+                        SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO;
         } else if ((devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
                                AUDIO_DEVICE_OUT_SPEAKER_SAFE)) ||
                 (devices == (AUDIO_DEVICE_OUT_USB_HEADSET |
@@ -5225,7 +5279,9 @@
                                  SND_DEVICE_OUT_VOICE_USB_HEADPHONES;
             }
         } else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
-            if (adev->bt_wb_speech_enabled)
+            if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+                snd_device = SND_DEVICE_OUT_BT_SCO_SWB;
+            else if (adev->bt_wb_speech_enabled)
                 snd_device = SND_DEVICE_OUT_BT_SCO_WB;
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
@@ -5351,7 +5407,9 @@
         else
             snd_device = SND_DEVICE_OUT_SPEAKER;
     } else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
-        if (adev->bt_wb_speech_enabled)
+        if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+                snd_device = SND_DEVICE_OUT_BT_SCO_SWB;
+        else if (adev->bt_wb_speech_enabled)
             snd_device = SND_DEVICE_OUT_BT_SCO_WB;
         else
             snd_device = SND_DEVICE_OUT_BT_SCO;
@@ -5740,7 +5798,12 @@
             if (audio_extn_hfp_is_active(adev))
                 platform_set_echo_reference(adev, true, out_device);
         } else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
-            if (adev->bt_wb_speech_enabled) {
+            if (adev->swb_speech_mode != SPEECH_MODE_INVALID) {
+                if (adev->bluetooth_nrec)
+                    snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC;
+                else
+                    snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB;
+            } else if (adev->bt_wb_speech_enabled) {
                 if (adev->bluetooth_nrec)
                     snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB_NREC;
                 else
@@ -6050,7 +6113,12 @@
         } else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
             snd_device = SND_DEVICE_IN_HEADSET_MIC;
         } else if (in_device & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
-            if (adev->bt_wb_speech_enabled) {
+            if (adev->swb_speech_mode != SPEECH_MODE_INVALID) {
+                if (adev->bluetooth_nrec)
+                    snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC;
+                else
+                    snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB;
+            } else if (adev->bt_wb_speech_enabled) {
                 if (adev->bluetooth_nrec)
                     snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB_NREC;
                 else
@@ -6113,7 +6181,12 @@
             snd_device = my_data->fluence_sb_enabled ? SND_DEVICE_IN_HANDSET_MIC_SB
                              : SND_DEVICE_IN_HANDSET_MIC;
         } else if (out_device & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET) {
-            if (adev->bt_wb_speech_enabled) {
+            if (adev->swb_speech_mode != SPEECH_MODE_INVALID) {
+                if (adev->bluetooth_nrec)
+                    snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC;
+                else
+                    snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB;
+            } else if (adev->bt_wb_speech_enabled) {
                 if (adev->bluetooth_nrec)
                     snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB_NREC;
                 else
@@ -7389,7 +7462,6 @@
     int ret = -EINVAL;
     int backend_idx = platform_get_backend_index(snd_device);
     struct platform_data *my_data = (struct platform_data *)adev->platform;
-    backend_idx = platform_get_backend_index(snd_device);
     unsigned int bit_width = backend_cfg.bit_width;
     unsigned int sample_rate = backend_cfg.sample_rate;
     unsigned int channels = backend_cfg.channels;
@@ -7611,7 +7683,6 @@
         ALOGV("%s: Format doesnt have to be set", __func__);
     }
 
-    format = format & AUDIO_FORMAT_MAIN_MASK;
     /* Set data format only if there is a change from PCM to compressed
        and vice versa */
     if (set_mi2s_tx_data_format && (format ^ my_data->current_backend_cfg[backend_idx].format)) {
@@ -7621,7 +7692,7 @@
                   __func__, ext_disp_format);
             return -EINVAL;
         }
-        if (format == AUDIO_FORMAT_PCM) {
+        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
             ALOGE("%s:Set %s to LPCM", __func__, ext_disp_format);
             mixer_ctl_set_enum_by_string(ctl, "LPCM");
         } else {
@@ -7790,6 +7861,7 @@
     if (snd_device == SND_DEVICE_OUT_BT_A2DP ||
         snd_device == SND_DEVICE_OUT_BT_SCO ||
         snd_device == SND_DEVICE_OUT_BT_SCO_WB ||
+        snd_device == SND_DEVICE_OUT_BT_SCO_SWB ||
         snd_device == SND_DEVICE_IN_BT_A2DP ||
         snd_device == SND_DEVICE_OUT_AFE_PROXY) {
         backend_change = false;
@@ -8193,18 +8265,25 @@
               __func__, bit_width, sample_rate, channels);
     }
 
-    ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
-          "sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
+    ALOGI("%s:txbecf: afe: current backend bit_width %d sample_rate %d channels %d, format %x",
+                            __func__,
+                            my_data->current_backend_cfg[backend_idx].bit_width,
+                            my_data->current_backend_cfg[backend_idx].sample_rate,
+                            my_data->current_backend_cfg[backend_idx].channels,
+                            my_data->current_backend_cfg[backend_idx].format);
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
+    // Note that below if block would be entered even if main format is same
+    // but subformat is different for e.g. current backend is configured for 16 bit PCM
+    // as compared to 24 bit PCM backend requested
     if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
         (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
         (channels != my_data->current_backend_cfg[backend_idx].channels) ||
-        ((format & AUDIO_FORMAT_MAIN_MASK) != my_data->current_backend_cfg[backend_idx].format)) {
+        (format != my_data->current_backend_cfg[backend_idx].format)) {
         backend_cfg->bit_width = bit_width;
         backend_cfg->sample_rate= sample_rate;
         backend_cfg->channels = channels;
-        backend_cfg->format = format & AUDIO_FORMAT_MAIN_MASK;
+        backend_cfg->format = format;
         backend_change = true;
         ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
               "new sample rate: %d new channel: %d new format: %d",
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 2cb1741..ec6843b 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -110,6 +110,7 @@
     SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
     SND_DEVICE_OUT_BT_SCO,
     SND_DEVICE_OUT_BT_SCO_WB,
+    SND_DEVICE_OUT_BT_SCO_SWB,
     SND_DEVICE_OUT_BT_A2DP,
     SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
     SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP,
@@ -118,8 +119,11 @@
     SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO,
     SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB,
     SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB,
+    SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB,
+    SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB,
     SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO,
     SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB,
+    SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -219,6 +223,8 @@
     SND_DEVICE_IN_BT_SCO_MIC_NREC,
     SND_DEVICE_IN_BT_SCO_MIC_WB,
     SND_DEVICE_IN_BT_SCO_MIC_WB_NREC,
+    SND_DEVICE_IN_BT_SCO_MIC_SWB,
+    SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC,
     SND_DEVICE_IN_BT_A2DP,
     SND_DEVICE_IN_CAMCORDER_MIC,
     SND_DEVICE_IN_VOICE_DMIC,