diff --git a/configs/msmcobalt/audio_output_policy.conf b/configs/msmcobalt/audio_output_policy.conf
index 67d79bf..1bbaad2 100644
--- a/configs/msmcobalt/audio_output_policy.conf
+++ b/configs/msmcobalt/audio_output_policy.conf
@@ -48,8 +48,8 @@
   }
   compress_passthrough_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING|AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
-    formats AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_E_AC3_JOC|AUDIO_FORMAT_DTS|AUDIO_FORMAT_DTS_HD
-    sampling_rates 32000|44100|48000|88200|96000|176400|192000
+    formats AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_E_AC3_JOC|AUDIO_FORMAT_DTS|AUDIO_FORMAT_DTS_HD|AUDIO_FORMAT_DSD
+    sampling_rates 32000|44100|48000|88200|96000|176400|192000|352800
     bit_width 16
     app_type 69941
   }
diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index 696a5d0..f5547dc 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -72,9 +72,12 @@
     <backend_names>
         <device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
+        <device name="SND_DEVICE_OUT_ANC_HEADSET" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
+        <device name="SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+        <device name="SND_DEVICE_OUT_VOICE_ANC_HEADSET" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
diff --git a/configs/msmcobalt/audio_policy.conf b/configs/msmcobalt/audio_policy.conf
index a3b0c55..166b9b6 100644
--- a/configs/msmcobalt/audio_policy.conf
+++ b/configs/msmcobalt/audio_policy.conf
@@ -71,6 +71,13 @@
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
       }
+      dsd_compress_passthrough {
+        sampling_rates 2822400|5644800
+        channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO
+        formats AUDIO_FORMAT_DSD
+        devices AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE
+        flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
+      }
       incall_music {
         sampling_rates 8000|16000|48000
         channel_masks AUDIO_CHANNEL_OUT_MONO
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index 4336aa2..b7da238 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -137,6 +137,12 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
                              channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
                 </mixPort>
+                <mixPort name="dsd_compress_passthrough" role="source"
+                         flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+                    <profile name="" format="AUDIO_FORMAT_DSD"
+                             samplingRates="2822400,5644800"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+                </mixPort>
                 <mixPort name="voice_tx" role="source">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
@@ -277,11 +283,11 @@
                 <route type="mix" sink="Speaker"
                        sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
                 <route type="mix" sink="Wired Headset"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
                 <route type="mix" sink="Wired Headphones"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
                 <route type="mix" sink="Line"
-                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
                 <route type="mix" sink="HDMI"
                        sources="primary output,raw,deep_buffer,multichannel,direct_pcm,compressed_offload,compress_passthrough"/>
                 <route type="mix" sink="Proxy"
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 3c6f642..eb5a150 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -1876,8 +1876,8 @@
         <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
         <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
         <ctl name="SLIM_6_RX Channels" value="Two" />
-        <ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
-        <ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
+        <ctl name= "RX INT1_2 MUX" value="RX2" />
+        <ctl name= "RX INT2_2 MUX" value="RX3" />
         <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
     </path>
@@ -1907,6 +1907,14 @@
         <ctl name= "RX INT2 SPLINE MIX HPHR Native Switch" value="1" />
     </path>
 
+    <path name="hph-highquality-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+    </path>
+
+    <path name="hph-lowpower-mode">
+        <ctl name="RX HPH Mode" value="CLS_H_LP" />
+    </path>
+
     <path name="line">
         <path name="headphones" />
     </path>
diff --git a/configs/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index 3a188f9..50e4df4 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -92,36 +92,46 @@
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia5" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="0" />
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="0" />
@@ -147,8 +157,13 @@
     <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
     <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
     <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
+    <ctl name="SLIM_2_RX Format" value="UNPACKED" />
+    <ctl name="SLIM_2_RX SampleRate" value="KHZ_48" />
+    <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
     <ctl name="SLIM_0_RX Channels" value="One" />
     <ctl name="SLIM_5_RX Channels" value="One" />
+    <ctl name="SLIM_6_RX Channels" value="One" />
+    <ctl name="SLIM_2_RX Channels" value="One" />
     <ctl name="SLIM_0_TX Channels" value="One" />
     <ctl name="SLIM_1_TX Channels" value="One" />
     <ctl name="AIF1_CAP Mixer SLIM TX7" value="0" />
@@ -286,10 +301,28 @@
     <!-- Codec controls -->
     <ctl name="SLIM RX0 MUX" value="ZERO" />
     <ctl name="SLIM RX1 MUX" value="ZERO" />
+    <ctl name="SLIM RX2 MUX" value="ZERO" />
+    <ctl name="SLIM RX3 MUX" value="ZERO" />
+    <ctl name="SLIM RX4 MUX" value="ZERO" />
+    <ctl name="SLIM RX5 MUX" value="ZERO" />
+    <ctl name="SLIM RX6 MUX" value="ZERO" />
+    <ctl name="SLIM RX7 MUX" value="ZERO" />
     <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
     <ctl name="CDC_IF RX1 MUX" value="SLIM RX1" />
+    <ctl name="CDC_IF RX2 MUX" value="SLIM RX2" />
+    <ctl name="CDC_IF RX3 MUX" value="SLIM RX3" />
+    <ctl name="CDC_IF RX4 MUX" value="SLIM RX4" />
+    <ctl name="CDC_IF RX5 MUX" value="SLIM RX5" />
+    <ctl name="CDC_IF RX6 MUX" value="SLIM RX6" />
+    <ctl name="CDC_IF RX7 MUX" value="SLIM RX7" />
+    <ctl name="RX INT1_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT2_1 MIX1 INP0" value="ZERO" />
+    <ctl name="RX INT1_2 MUX" value="ZERO" />
+    <ctl name="RX INT2_2 MUX" value="ZERO" />
     <ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
     <ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
+    <ctl name="COMP1 Switch" value="1" />
+    <ctl name="COMP2 Switch" value="1" />
     <ctl name="COMP7 Switch" value="0" />
     <ctl name="COMP8 Switch" value="0" />
     <ctl name="SpkrLeft COMP Switch" value="0" />
@@ -300,6 +333,16 @@
     <ctl name="SpkrRight VISENSE Switch" value="0" />
     <ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
     <ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
+
+    <ctl name="RX INT1_1 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT2_1 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT1_2 NATIVE MUX" value="OFF" />
+    <ctl name="RX INT2_2 NATIVE MUX" value="OFF" />
+
+    <ctl name="ASRC0 MUX" value="ZERO" />
+    <ctl name="RX INT1 SEC MIX HPHL Switch" value="0" />
+    <ctl name="ASRC1 MUX" value="ZERO" />
+    <ctl name="RX INT2 SEC MIX HPHR Switch" value="0" />
     <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="ZERO" />
     <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="ZERO" />
     <ctl name="VI_FEED_TX Channels" value="Two" />
@@ -307,6 +350,12 @@
     <ctl name="AIF4_VI Mixer SPKR_VI_2" value="0" />
     <ctl name="SLIM_4_TX Format" value="UNPACKED" />
 
+    <ctl name="DSD_L IF MUX" value="ZERO" />
+    <ctl name="DSD_R IF MUX" value="ZERO" />
+    <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="0" />
+    <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="0" />
+    <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+    <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
     <ctl name="AIF1_CAP Mixer SLIM TX0" value="0" />
     <ctl name="AIF1_CAP Mixer SLIM TX2" value="0" />
     <ctl name="CDC_IF TX0 MUX" value="ZERO" />
@@ -350,6 +399,7 @@
     </path>
 
     <path name="echo-reference headphones">
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
     </path>
 
     <path name="echo-reference headphones-44.1">
@@ -553,6 +603,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="1" />
     </path>
 
+    <path name="compress-offload-playback headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
     <path name="compress-offload-playback speaker-and-headphones">
         <path name="compress-offload-playback headphones" />
         <path name="compress-offload-playback" />
@@ -601,6 +655,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="1" />
     </path>
 
+    <path name="compress-offload-playback2 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
     <path name="compress-offload-playback2 speaker-and-headphones">
         <path name="compress-offload-playback2 headphones" />
         <path name="compress-offload-playback2" />
@@ -649,6 +707,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="compress-offload-playback3 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
     <path name="compress-offload-playback3 speaker-and-headphones">
         <path name="compress-offload-playback3 headphones" />
         <path name="compress-offload-playback3" />
@@ -697,6 +759,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="1" />
     </path>
 
+    <path name="compress-offload-playback4 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
     <path name="compress-offload-playback4 speaker-and-headphones">
         <path name="compress-offload-playback4 headphones" />
         <path name="compress-offload-playback4" />
@@ -745,6 +811,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="1" />
     </path>
 
+    <path name="compress-offload-playback5 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
     <path name="compress-offload-playback5 speaker-and-headphones">
         <path name="compress-offload-playback5 headphones" />
         <path name="compress-offload-playback5" />
@@ -793,6 +863,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="1" />
     </path>
 
+    <path name="compress-offload-playback6 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
     <path name="compress-offload-playback6 speaker-and-headphones">
         <path name="compress-offload-playback6 headphones" />
         <path name="compress-offload-playback6" />
@@ -841,6 +915,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="1" />
     </path>
 
+    <path name="compress-offload-playback7 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
     <path name="compress-offload-playback7 speaker-and-headphones">
         <path name="compress-offload-playback7 headphones" />
         <path name="compress-offload-playback7" />
@@ -889,6 +967,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="1" />
     </path>
 
+    <path name="compress-offload-playback8 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
     <path name="compress-offload-playback8 speaker-and-headphones">
         <path name="compress-offload-playback8 headphones" />
         <path name="compress-offload-playback8" />
@@ -937,6 +1019,10 @@
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="compress-offload-playback9 headphones-dsd">
+        <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
     <path name="compress-offload-playback9 speaker-and-headphones">
         <path name="compress-offload-playback9 headphones" />
         <path name="compress-offload-playback9" />
@@ -1446,9 +1532,41 @@
     </path>
 
     <path name="headphones">
+        <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+        <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+        <ctl name="SLIM_6_RX Channels" value="Two" />
+        <ctl name="RX INT1_2 MUX" value="RX2" />
+        <ctl name="RX INT2_2 MUX" value="RX3" />
     </path>
 
     <path name="headphones-44.1">
+        <ctl name="SLIM RX4 MUX" value="AIF3_PB" />
+        <ctl name="SLIM RX5 MUX" value="AIF3_PB" />
+        <ctl name="SLIM_5_RX Channels" value="Two" />
+        <ctl name="RX INT1_1 MIX1 INP0" value="RX4" />
+        <ctl name="RX INT2_1 MIX1 INP0" value="RX5" />
+        <ctl name="RX INT1_1 NATIVE MUX" value="ON" />
+        <ctl name="RX INT2_1 NATIVE MUX" value="ON" />
+    </path>
+
+    <path name="asrc-mode">
+        <ctl name="RX INT1_2 NATIVE MUX" value="ON" />
+        <ctl name="RX INT2_2 NATIVE MUX" value="ON" />
+        <ctl name="ASRC0 MUX" value="ASRC_IN_HPHL" />
+        <ctl name="RX INT1 SEC MIX HPHL Switch" value="1" />
+        <ctl name="ASRC1 MUX" value="ASRC_IN_HPHR" />
+        <ctl name="RX INT2 SEC MIX HPHR Switch" value="1" />
+    </path>
+
+    <path name="headphones-dsd">
+        <ctl name="SLIM RX6 MUX" value="AIF2_PB" />
+        <ctl name="SLIM RX7 MUX" value="AIF2_PB" />
+        <ctl name="SLIM_2_RX Channels" value="Two" />
+        <ctl name="DSD_L IF MUX" value="RX6" />
+        <ctl name="DSD_R IF MUX" value="RX7" />
+        <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="1" />
+        <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="1" />
+        <ctl name="SLIM_2_RX Format" value="DSD_DOP" />
     </path>
 
     <path name="true-native-mode">
diff --git a/configs/msmcobalt/msmcobalt.mk b/configs/msmcobalt/msmcobalt.mk
index aaf9db5..2c721b6 100644
--- a/configs/msmcobalt/msmcobalt.mk
+++ b/configs/msmcobalt/msmcobalt.mk
@@ -4,7 +4,7 @@
 BOARD_USES_ALSA_AUDIO := true
 USE_CUSTOM_AUDIO_POLICY := 1
 USE_XML_AUDIO_POLICY_CONF := 1
-BOARD_SUPPORTS_SOUND_TRIGGER := true
+BOARD_SUPPORTS_SOUND_TRIGGER_HAL := true
 AUDIO_USE_LL_AS_PRIMARY_OUTPUT := true
 
 AUDIO_FEATURE_ENABLED_VBAT_MONITOR := true
@@ -195,3 +195,9 @@
 #split a2dp DSP supported encoder list
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.bt.a2dp_offload_cap=sbc-aptx
+
+#enable software decoders for ALAC and APE
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.alac.decoder=true
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.ape.decoder=true
diff --git a/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
new file mode 100755
index 0000000..3c75b8e
--- /dev/null
+++ b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
@@ -0,0 +1,115 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2014-2016, The Linux Foundation. All rights reserved.       -->
+<!---                                                                           -->
+<!--- Redistribution and use in source and binary forms, with or without        -->
+<!--- modification, are permitted provided that the following conditions are    -->
+<!--- met:                                                                      -->
+<!---     * Redistributions of source code must retain the above copyright      -->
+<!---       notice, this list of conditions and the following disclaimer.       -->
+<!---     * Redistributions in binary form must reproduce the above             -->
+<!---       copyright notice, this list of conditions and the following         -->
+<!---       disclaimer in the documentation and/or other materials provided     -->
+<!---       with the distribution.                                              -->
+<!---     * Neither the name of The Linux Foundation nor the names of its       -->
+<!---       contributors may be used to endorse or promote products derived     -->
+<!---       from this software without specific prior written permission.       -->
+<!---                                                                           -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED              -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF      -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT    -->
+<!--- ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS    -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR    -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF      -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR           -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,     -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE      -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN    -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                             -->
+
+<mixer>
+    <!-- These are the initial mixer settings -->
+    <ctl name="LSM1 MUX" value="None" />
+    <ctl name="LSM2 MUX" value="None" />
+    <ctl name="LSM3 MUX" value="None" />
+    <ctl name="LSM4 MUX" value="None" />
+    <ctl name="LSM5 MUX" value="None" />
+    <ctl name="LSM6 MUX" value="None" />
+    <ctl name="LSM7 MUX" value="None" />
+    <ctl name="LSM8 MUX" value="None" />
+    <ctl name="SLIMBUS_5_TX LSM Function" value="None" />
+    <ctl name="MADONOFF Switch" value="0" />
+    <ctl name="MAD Input" value="DMIC1" />
+    <ctl name="MAD_BROADCAST Switch" value="0" />
+    <ctl name="TX13 INP MUX" value="CDC_DEC_5" />
+    <ctl name="AIF4_MAD Mixer SLIM TX12" value="0" />
+    <ctl name="AIF4_MAD Mixer SLIM TX13" value="0" />
+    <ctl name="CPE AFE MAD Enable" value="0"/>
+    <ctl name="CLK MODE" value="EXTERNAL" />
+    <ctl name="EC BUF MUX INP" value="ZERO" />
+    <ctl name="ADC MUX1" value="DMIC" />
+    <ctl name="DMIC MUX1" value="ZERO" />
+
+    <path name="listen-voice-wakeup-1">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+    </path>
+
+    <path name="listen-voice-wakeup-2">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-3">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-4">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-5">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-6">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-7">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+    </path>
+    <path name="listen-voice-wakeup-8">
+        <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+        <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+    </path>
+
+    <path name="listen-cpe-handset-mic">
+        <ctl name="MAD Input" "DMIC0" />
+        <ctl name="MAD_SEL MUX" "SPE" />
+        <ctl name="MAD_INP MUX" "MAD" />
+        <ctl name="MAD_CPE1 Switch" 1 />
+    </path>
+
+    <path name="listen-cpe-handset-mic-ecpp">
+        <ctl name="CLK MODE" value="INTERNAL" />
+        <ctl name="EC BUF MUX INP" value="DEC1" />
+        <ctl name="ADC MUX1" value="DMIC" />
+        <ctl name="DMIC MUX1" value="DMIC0" />
+    </path>
+
+    <!-- path name used for low bandwidth FTRT codec interface -->
+    <path name="listen-cpe-handset-mic low-speed-intf">
+        <ctl name="MADONOFF Switch" value="1" />
+        <ctl name="AIF4_MAD Mixer SLIM TX12" value="1" />
+        <ctl name="MAD Input" value="DMIC0" />
+        <ctl name="CPE AFE MAD Enable" value="1"/>
+    </path>
+
+    <path name="listen-ape-handset-mic">
+        <ctl name="MAD_BROADCAST Switch" value="1" />
+        <ctl name="TX13 INP MUX" value="MAD_BRDCST" />
+        <ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
+        <ctl name="MAD Input" value="DMIC0" />
+    </path>
+
+</mixer>
diff --git a/configs/msmcobalt/sound_trigger_platform_info.xml b/configs/msmcobalt/sound_trigger_platform_info.xml
index b92ea48..1f90bd5 100644
--- a/configs/msmcobalt/sound_trigger_platform_info.xml
+++ b/configs/msmcobalt/sound_trigger_platform_info.xml
@@ -29,8 +29,7 @@
     <param version="0x0101" /> <!-- this must be the first param -->
 
     <common_config>
-        <param execution_type="CPE" /> <!-- value: "CPE" "APE" -->
-        <param max_cpe_sessions="1" />
+        <param max_cpe_sessions="2" />
         <param max_ape_sessions="8" />
         <param enable_failure_detection="false" />
     </common_config>
@@ -41,11 +40,12 @@
         <param DEVICE_HANDSET_CPE_ECPP_ACDB_ID="128" />
     </acdb_ids>
 
-    <!-- Multiple sound_model_config tags can be listed, each with unique    -->
-    <!-- vendor_uuid. The below tag represents QTI SVA engine sound model    -->
-    <!-- configuration. ISV must use their own unique vendor_uuid.           -->
+    <!-- Multiple sound_model_config tags can be listed, each with unique   -->
+    <!-- vendor_uuid. The below tag represents QTI SVA engine sound model   -->
+    <!-- configuration. ISV must use their own unique vendor_uuid.          -->
     <sound_model_config>
         <param vendor_uuid="68ab2d40-e860-11e3-95ef-0002a5d5c51b" />
+        <param execution_type="WDSP" /> <!-- value: "WDSP" "ADSP" "DYNAMIC" -->
         <param app_type="2" /> <!-- app type used in ACDB -->
         <param library="libsmwrapper.so" />
         <param max_cpe_phrases="6" />
@@ -54,7 +54,18 @@
         <param max_ape_users="10" />
         <param sample_rate="16000" />
 
-        <!-- Module and param ids with which the algorithm is integrated in firmware -->
+        <gcs_uid>
+            <param uid="0x1" />
+            <param did="0x4" />
+            <param load_sound_model_ids="0x00012C0D, 0x0, 0x00012C14" />
+            <param confidence_levels_ids="0x00012C0D, 0x0, 0x00012C28" />
+            <param operation_mode_ids="0x00012C0D, 0x0, 0x00012C28" />
+            <param detection_event_ids="0x00012C0D, 0x0, 0x00012C29" />
+            <param capture_event_ids="0x00020013, 0x0,0x00020015" />
+        </gcs_uid>
+
+        <!-- Module and param ids with which the algorithm is integrated
+            in non-graphite firmware (note these must come after gcs params) -->
         <param load_sound_model_ids="0x00012C0D, 0x00012C14" />
         <param unload_sound_model_ids="0x00012C0D, 0x00012C15" />
         <param confidence_levels_ids="0x00012C0D, 0x00012C07" />
@@ -62,7 +73,8 @@
 
         <!-- format: "ADPCM_packet" or "PCM_packet" !-->
         <!-- transfer_mode: "FTRT" or "RT" -->
-        <!--  kw_duration is in milli seconds. It is valid only for FTRT transfer mode -->
+        <!--  kw_duration is in milli seconds. It is valid only for FTRT
+            transfer mode -->
         <param capture_keyword="PCM_packet, RT, 2000" />
         <param client_capture_read_delay="2000" />
     </sound_model_config>
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index d186a5f..12a8082 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -401,6 +401,10 @@
 
 #endif
 
+#ifndef AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
+#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH  0x10000
+#endif
+
 #ifndef HDMI_PASSTHROUGH_ENABLED
 #define audio_extn_passthru_update_stream_configuration(adev, out)            (0)
 #define audio_extn_passthru_is_convert_supported(adev, out)                   (0)
@@ -419,8 +423,6 @@
 #define audio_extn_passthru_set_parameters(a, p) (-ENOSYS)
 #define audio_extn_passthru_init(a) do {} while(0)
 #define audio_extn_passthru_should_standby(o) (1)
-
-#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH  0x1000
 #else
 bool audio_extn_passthru_is_convert_supported(struct audio_device *adev,
                                                  struct stream_out *out);
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index e3f1b6c..26c43b4 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -90,9 +90,7 @@
 #ifdef INCALL_MUSIC_ENABLED
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_INCALL_MUSIC),
 #endif
-#ifdef HDMI_PASSTHROUGH_ENABLED
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH),
-#endif
 };
 
 const struct string_to_enum s_format_name_to_enum_table[] = {
@@ -133,6 +131,7 @@
     STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_LC),
     STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V1),
     STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V2),
+    STRING_TO_ENUM(AUDIO_FORMAT_DSD),
 #endif
 };
 
@@ -515,6 +514,21 @@
                                __func__, sample_rate);
         }
     }
+
+    /* Set sampling rate to 176.4 for DSD64
+     * and 352.8Khz for DSD128.
+     * Set Bit Width to 16. output will be 16 bit
+     * post DoP in ASM.
+     */
+    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH) &&
+        (format == AUDIO_FORMAT_DSD)) {
+        bit_width = 16;
+        if (sample_rate == INPUT_SAMPLING_RATE_DSD64)
+            sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+        else if (sample_rate == INPUT_SAMPLING_RATE_DSD128)
+            sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+    }
+
     ALOGV("%s: flags: %x, format: %x sample_rate %d",
            __func__, flags, format, sample_rate);
     list_for_each(node_i, streams_output_cfg_list) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index af399a1..b617407 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -82,6 +82,7 @@
 /* ToDo: Check and update a proper value in msec */
 #define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
 #define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define DSD_VOLUME_MIN_DB (-110)
 
 #define PROXY_OPEN_RETRY_COUNT           100
 #define PROXY_OPEN_WAIT_TIME             20
@@ -501,6 +502,7 @@
         format == AUDIO_FORMAT_FLAC ||
         format == AUDIO_FORMAT_ALAC ||
         format == AUDIO_FORMAT_APE ||
+        format == AUDIO_FORMAT_DSD ||
         format == AUDIO_FORMAT_VORBIS ||
         format == AUDIO_FORMAT_WMA ||
         format == AUDIO_FORMAT_WMA_PRO)
@@ -541,6 +543,9 @@
     case AUDIO_FORMAT_APE:
         id = SND_AUDIOCODEC_APE;
         break;
+    case AUDIO_FORMAT_DSD:
+        id = SND_AUDIOCODEC_DSD;
+        break;
     case AUDIO_FORMAT_VORBIS:
         id = SND_AUDIOCODEC_VORBIS;
         break;
@@ -616,6 +621,36 @@
     return 0;
 }
 
+/*
+ * Enable ASRC mode if native or DSD stream is active.
+ */
+static void audio_check_and_set_asrc_mode(struct audio_device *adev, snd_device_t snd_device)
+{
+    if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
+       !adev->asrc_mode_enabled) {
+        struct listnode *node = NULL;
+        struct audio_usecase *uc = NULL;
+        struct stream_out *curr_out = NULL;
+
+        list_for_each(node, &adev->usecase_list) {
+            uc = node_to_item(node, struct audio_usecase, list);
+            curr_out = (struct stream_out*) uc->stream.out;
+
+            if (curr_out && PCM_PLAYBACK == uc->type) {
+                if((platform_get_backend_index(uc->out_snd_device) == HEADPHONE_44_1_BACKEND) ||
+                      (platform_get_backend_index(uc->out_snd_device) == DSD_NATIVE_BACKEND)) {
+                    ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware",
+                          __func__);
+                    audio_route_apply_and_update_path(adev->audio_route,
+                                                  "asrc-mode");
+                    adev->asrc_mode_enabled = true;
+                    break;
+                }
+            }
+        }
+    }
+}
+
 int pcm_ioctl(struct pcm *pcm, int request, ...)
 {
     va_list ap;
@@ -767,7 +802,8 @@
             audio_route_apply_and_update_path(adev->audio_route,
                                               "true-native-mode");
             adev->native_playback_enabled = true;
-        }
+        } else
+            audio_check_and_set_asrc_mode(adev, snd_device);
     }
     return 0;
 }
@@ -824,6 +860,11 @@
             audio_route_reset_and_update_path(adev->audio_route,
                                               "true-native-mode");
             adev->native_playback_enabled = false;
+        } else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
+                 adev->asrc_mode_enabled) {
+            ALOGD("%s: %d: disabling asrc mode in hardware", __func__, __LINE__);
+            audio_route_reset_and_update_path(adev->audio_route, "asrc-mode");
+            adev->asrc_mode_enabled = false;
         }
 
         audio_extn_dev_arbi_release(snd_device);
@@ -895,7 +936,9 @@
             ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
              (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
              (force_restart_session)) &&
-            platform_check_backends_match(snd_device, usecase->out_snd_device)) {
+            (platform_check_backends_match(snd_device, usecase->out_snd_device)||
+             (platform_check_codec_asrc_support(adev->platform) && !adev->asrc_mode_enabled &&
+              platform_check_if_backend_has_to_be_disabled(snd_device,usecase->out_snd_device)))) {
                 ALOGD("%s:becf: check_usecases (%s) is active on (%s) - disabling ..",
                     __func__, use_case_table[usecase->id],
                       platform_get_snd_device_name(usecase->out_snd_device));
@@ -1166,6 +1209,28 @@
     return active;
 }
 
+/*
+ * if native DSD playback active
+ */
+bool audio_is_dsd_native_stream_active(struct audio_device *adev)
+{
+    bool active = false;
+    struct listnode *node = NULL;
+    struct audio_usecase *uc = NULL;
+    struct stream_out *curr_out = NULL;
+
+    list_for_each(node, &adev->usecase_list) {
+        uc = node_to_item(node, struct audio_usecase, list);
+        curr_out = (struct stream_out*) uc->stream.out;
+
+        if (curr_out && PCM_PLAYBACK == uc->type &&
+               (DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) {
+            active = true;
+            ALOGV("%s:DSD playback is active", __func__);
+        }
+    }
+    return active;
+}
 
 static bool force_device_switch(struct audio_usecase *usecase)
 {
@@ -2537,6 +2602,14 @@
     return latency;
 }
 
+static float AmpToDb(float amplification)
+{
+    if (amplification == 0) {
+        return DSD_VOLUME_MIN_DB;
+    }
+    return 20 * log10(amplification);
+}
+
 static int out_set_volume(struct audio_stream_out *stream, float left,
                           float right)
 {
@@ -2555,6 +2628,20 @@
              * Mute is 0 and unmute 1
              */
             audio_extn_passthru_set_volume(out, (left == 0.0f));
+        } else if (out->format == AUDIO_FORMAT_DSD){
+            char mixer_ctl_name[128] =  "DSD Volume";
+            struct audio_device *adev = out->dev;
+            struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+
+            if (!ctl) {
+                ALOGE("%s: Could not get ctl for mixer cmd - %s",
+                      __func__, mixer_ctl_name);
+                return -EINVAL;
+            }
+            volume[0] = (int)(AmpToDb(left));
+            volume[1] = (int)(AmpToDb(right));
+            mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
+            return 0;
         } else {
             char mixer_ctl_name[128];
             struct audio_device *adev = out->dev;
@@ -3666,12 +3753,24 @@
                 __func__, config->offload_info.version,
                 config->offload_info.bit_rate);
 
+        /*Check if DSD audio format is supported in codec
+         *and there is no active native DSD use case
+         */
+
+        if ((config->format == AUDIO_FORMAT_DSD) &&
+               (!platform_check_codec_dsd_support(adev->platform) ||
+               audio_is_dsd_native_stream_active(adev))) {
+            ret = -EINVAL;
+            goto error_open;
+        }
+
         /* Disable gapless if any of the following is true
          * passthrough playback
          * AV playback
          * Direct PCM playback
          */
         if (audio_extn_passthru_is_passthrough_stream(out) ||
+            (config->format == AUDIO_FORMAT_DSD) ||
             config->offload_info.has_video ||
             out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
             check_and_set_gapless_mode(adev, false);
@@ -3681,6 +3780,10 @@
         if (audio_extn_passthru_is_passthrough_stream(out)) {
             out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
         }
+        if (config->format == AUDIO_FORMAT_DSD) {
+            out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
+            out->compr_config.codec->compr_passthr = PASSTHROUGH_DSD;
+        }
     } else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
         ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
         if (ret != 0) {
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index ee28157..197807c 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -389,6 +389,7 @@
     int perf_lock_opts[MAX_PERF_LOCK_OPTS];
     int perf_lock_opts_size;
     bool native_playback_enabled;
+    bool asrc_mode_enabled;
 };
 
 int select_devices(struct audio_device *adev,
@@ -410,6 +411,8 @@
 
 bool audio_is_true_native_stream_active(struct audio_device *adev);
 
+bool audio_is_dsd_native_stream_active(struct audio_device *adev);
+
 int pcm_ioctl(struct pcm *pcm, int request, ...);
 
 int get_snd_card_state(struct audio_device *adev);
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index e0400c6..9c6cc6f 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -2448,7 +2448,7 @@
     return ret;
 }
 
-static int platform_get_backend_index(snd_device_t snd_device)
+int platform_get_backend_index(snd_device_t snd_device)
 {
     int32_t port = DEFAULT_CODEC_BACKEND;
 
@@ -4343,8 +4343,8 @@
             struct audio_usecase *uc;
             uc = node_to_item(node, struct audio_usecase, list);
             struct stream_out *out = (struct stream_out*) uc->stream.out;
-            unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
             if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+                unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
 
                 ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
                       "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
@@ -5408,3 +5408,19 @@
     }
     return 0;
 }
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+    return false;
+}
+
+bool platform_check_codec_asrc_support(void *platform __unused)
+{
+    return false;
+}
+
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device __unused,
+                                                  snd_device_t cuurent_snd_device __unused)
+{
+    return false;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index dcd351a..6c89d0a 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -198,9 +198,12 @@
     SND_DEVICE_MAX = SND_DEVICE_IN_END,
 
 };
-
+#define INPUT_SAMPLING_RATE_DSD64       2822400
+#define INPUT_SAMPLING_RATE_DSD128      5644800
 #define DEFAULT_OUTPUT_SAMPLING_RATE 48000
 #define OUTPUT_SAMPLING_RATE_44100      44100
+#define OUTPUT_SAMPLING_RATE_DSD64       176400
+#define OUTPUT_SAMPLING_RATE_DSD128      352800
 #define MAX_PORT                        6
 #define ALL_CODEC_BACKEND_PORT          0
 #define HEADPHONE_44_1_BACKEND_PORT     5
@@ -208,6 +211,8 @@
 enum {
     DEFAULT_CODEC_BACKEND,
     SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+    DSD_NATIVE_BACKEND,
+    SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
     HEADPHONE_44_1_BACKEND,
     SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
     HEADPHONE_BACKEND,
@@ -356,7 +361,8 @@
 enum {
     LEGACY_PCM = 0,
     PASSTHROUGH,
-    PASSTHROUGH_CONVERT
+    PASSTHROUGH_CONVERT,
+    PASSTHROUGH_DSD
 };
 /*
  * ID for setting mute and lateny on the device side
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 2b6a1d7..e5d42bd 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1257,3 +1257,24 @@
     }
     return 0;
 }
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+    return false;
+}
+
+int platform_get_backend_index(snd_device_t snd_device __unused);
+{
+    return 0;
+}
+
+bool platform_check_codec_asrc_support(void *platform __unused)
+{
+    return false;
+}
+
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device __unused,
+                                                  snd_device_t cuurent_snd_device __unused)
+{
+    return false;
+}
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index e42af8c..07060b6 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -112,6 +112,12 @@
 #define SOUND_CARD 0
 
 #define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+#define INPUT_SAMPLING_RATE_DSD64       2822400
+#define INPUT_SAMPLING_RATE_DSD128      5644800
+#define OUTPUT_SAMPLING_RATE_DSD64       176400
+#define OUTPUT_SAMPLING_RATE_DSD128      352800
+#define DSD_NATIVE_BACKEND 1
+#define PASSTHROUGH_DSD 3
 
 #define ALL_SESSION_VSID                0xFFFFFFFF
 #define DEFAULT_MUTE_RAMP_DURATION_MS   20
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 8daa715..7d6f02b 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -247,6 +247,8 @@
     int metainfo_key;
     int source_mic_type;
     int max_mic_count;
+    bool is_dsd_supported;
+    bool is_asrc_supported;
 };
 
 static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -334,6 +336,7 @@
     [SND_DEVICE_OUT_SPEAKER_VBAT] = "speaker-vbat",
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
     [SND_DEVICE_OUT_HEADPHONES] = "headphones",
+    [SND_DEVICE_OUT_HEADPHONES_DSD] = "headphones-dsd",
     [SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
     [SND_DEVICE_OUT_LINE] = "line",
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
@@ -451,6 +454,7 @@
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
     [SND_DEVICE_OUT_LINE] = 10,
     [SND_DEVICE_OUT_HEADPHONES] = 10,
+    [SND_DEVICE_OUT_HEADPHONES_DSD] = 10,
     [SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
     [SND_DEVICE_OUT_SPEAKER_AND_LINE] = 10,
@@ -568,6 +572,7 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_DSD)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_LINE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
@@ -1102,11 +1107,13 @@
     backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
     backend_tag_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
     backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
+    backend_tag_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("headphones-dsd");
     backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
     backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
     backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
     backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
 
+    hw_interface_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("SLIMBUS_2_RX");
     hw_interface_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("SLIMBUS_5_RX");
     hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
@@ -1715,6 +1722,11 @@
     my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
         strdup("SLIM_0_RX SampleRate");
 
+    my_data->current_backend_cfg[DSD_NATIVE_BACKEND].bitwidth_mixer_ctl =
+        strdup("SLIM_2_RX Format");
+    my_data->current_backend_cfg[DSD_NATIVE_BACKEND].samplerate_mixer_ctl =
+        strdup("SLIM_2_RX SampleRate");
+
     my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].bitwidth_mixer_ctl =
         strdup("SLIM_5_RX Format");
     my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
@@ -1745,6 +1757,13 @@
         }
     }
 
+    if(strstr(snd_card_name, "tavil")) {
+        ALOGD("%s:DSD playback is supported", __func__);
+        my_data->is_dsd_supported = true;
+        my_data->is_asrc_supported = true;
+        platform_set_native_support(NATIVE_AUDIO_MODE_MULTIPLE_44_1);
+    }
+
     my_data->current_backend_cfg[HEADPHONE_BACKEND].bitwidth_mixer_ctl =
         strdup("SLIM_6_RX Format");
     my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
@@ -1907,6 +1926,32 @@
     return result;
 }
 
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device,
+                                                  snd_device_t cuurent_snd_device)
+{
+    bool result = false;
+
+    ALOGV("%s: current snd device = %s, new snd device = %s", __func__,
+                platform_get_snd_device_name(cuurent_snd_device),
+                platform_get_snd_device_name(new_snd_device));
+
+    if ((new_snd_device < SND_DEVICE_MIN) || (new_snd_device >= SND_DEVICE_OUT_END) ||
+            (cuurent_snd_device < SND_DEVICE_MIN) || (cuurent_snd_device >= SND_DEVICE_OUT_END)) {
+        ALOGE("%s: Invalid snd_device",__func__);
+        return false;
+    }
+
+    if (cuurent_snd_device == SND_DEVICE_OUT_HEADPHONES &&
+            (new_snd_device == SND_DEVICE_OUT_HEADPHONES_44_1 ||
+             new_snd_device == SND_DEVICE_OUT_HEADPHONES_DSD)) {
+        result = true;
+    }
+
+    ALOGV("%s: Need to disable current backend %s, %d",
+          __func__, platform_get_snd_device_name(cuurent_snd_device), result);
+    return result;
+}
+
 int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type)
 {
     int device_id;
@@ -2092,7 +2137,8 @@
 
 int platform_set_native_support(int na_mode)
 {
-    if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode) {
+    if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode
+        || NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode) {
         na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled = true;
         na_props.na_mode = na_mode;
         ALOGD("%s:napb: native audio playback enabled in (%s) mode v2.0", __func__,
@@ -2107,6 +2153,18 @@
     return 0;
 }
 
+bool platform_check_codec_dsd_support(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    return my_data->is_dsd_supported;
+}
+
+bool platform_check_codec_asrc_support(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    return my_data->is_asrc_supported;
+}
+
 int platform_get_native_support()
 {
     int ret = NATIVE_AUDIO_MODE_INVALID;
@@ -2159,6 +2217,8 @@
             mode = NATIVE_AUDIO_MODE_SRC;
         else if (value && !strncmp(value, "true", sizeof("true")))
             mode = NATIVE_AUDIO_MODE_TRUE_44_1;
+        else if (value && !strncmp(value, "multiple", sizeof("multiple")))
+            mode = NATIVE_AUDIO_MODE_MULTIPLE_44_1;
         else {
             mode = NATIVE_AUDIO_MODE_INVALID;
             ALOGE("%s:napb:native_audio_mode in platform info xml,invalid mode string",
@@ -2238,7 +2298,7 @@
     return ret;
 }
 
-static int platform_get_backend_index(snd_device_t snd_device)
+int platform_get_backend_index(snd_device_t snd_device)
 {
     int32_t port = DEFAULT_CODEC_BACKEND;
 
@@ -2247,6 +2307,9 @@
                 if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
                             sizeof("headphones-44.1")) == 0)
                         port = HEADPHONE_44_1_BACKEND;
+                else if (strncmp(backend_tag_table[snd_device], "headphones-dsd",
+                            sizeof("headphones-dsd")) == 0)
+                        port = DSD_NATIVE_BACKEND;
                 else if (strncmp(backend_tag_table[snd_device], "headphones",
                             sizeof("headphones")) == 0)
                         port = HEADPHONE_BACKEND;
@@ -2764,6 +2827,12 @@
         } else if (NATIVE_AUDIO_MODE_SRC == na_mode &&
                    OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
                 snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+        } else if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode &&
+                   (sample_rate % OUTPUT_SAMPLING_RATE_44100 == 0) &&
+                   (out->format != AUDIO_FORMAT_DSD)) {
+                snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+        } else if (out->format == AUDIO_FORMAT_DSD) {
+                snd_device = SND_DEVICE_OUT_HEADPHONES_DSD;
         } else
             snd_device = SND_DEVICE_OUT_HEADPHONES;
     } else if (devices & AUDIO_DEVICE_OUT_LINE) {
@@ -4070,14 +4139,6 @@
               my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
     }
 
-    /*
-     * Backend sample rate configuration follows:
-     * 16 bit playback - 48khz for streams at any valid sample rate
-     * 24 bit playback - 48khz for stream sample rate less than 48khz
-     * 24 bit playback - 96khz for sample rate range of 48khz to 96khz
-     * 24 bit playback - 192khz for sample rate range of 96khz to 192 khz
-     * Upper limit is inclusive in the sample rate range.
-     */
     if (sample_rate !=
        my_data->current_backend_cfg[backend_idx].sample_rate) {
             char *rate_str = NULL;
@@ -4096,14 +4157,24 @@
                 rate_str = "KHZ_44P1";
                 break;
             case 64000:
-            case 88200:
             case 96000:
                 rate_str = "KHZ_96";
                 break;
+            case 88200:
+                rate_str = "KHZ_88P2";
+                break;
             case 176400:
+                rate_str = "KHZ_176P4";
+                break;
             case 192000:
                 rate_str = "KHZ_192";
                 break;
+            case 352800:
+                rate_str = "KHZ_352P8";
+                break;
+            case 384000:
+                rate_str = "KHZ_384";
+                break;
             default:
                 rate_str = "KHZ_48";
                 break;
@@ -4180,6 +4251,17 @@
         }
     }
 
+    if (snd_device == SND_DEVICE_OUT_HEADPHONES || snd_device ==
+        SND_DEVICE_OUT_HEADPHONES_44_1) {
+        if (sample_rate > 48000 || (sample_rate == 48000 && bit_width >= 24)) {
+            ALOGV("%s: apply HPH HQ mode\n", __func__);
+            audio_route_apply_and_update_path(adev->audio_route, "hph-highquality-mode");
+        } else {
+            ALOGV("%s: apply HPH LP mode\n", __func__);
+            audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
+        }
+    }
+
     return ret;
 }
 
@@ -4310,8 +4392,8 @@
             struct audio_usecase *uc;
             uc = node_to_item(node, struct audio_usecase, list);
             struct stream_out *out = (struct stream_out*) uc->stream.out;
-            unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
             if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+                unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
 
                 ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
                       "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
@@ -4400,6 +4482,24 @@
             channels_updated = true;
     }
 
+    /*
+     * Map native sampling rates to upper limit range
+     * if multiple of native sampling rates are not supported.
+     */
+    if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 != na_mode) {
+        switch (sample_rate) {
+            case 88200:
+                sample_rate = 96000;
+                break;
+            case 176400:
+                sample_rate = 192000;
+                break;
+            case 352800:
+                sample_rate = 192000;
+                break;
+        }
+    }
+
     ALOGI("%s:becf: afe: Codec selected backend: %d updated bit width: %d and sample rate: %d",
           __func__, backend_idx , bit_width, sample_rate);
 
@@ -4440,6 +4540,17 @@
     /*this is populated by check_codec_backend_cfg hence set default value to false*/
     backend_cfg.passthrough_enabled = false;
 
+    /* Set Backend sampling rate to 176.4 for DSD64 and
+     * 352.8Khz for DSD128.
+     * Set Bit Width to 16
+     */
+    if ((backend_idx == DSD_NATIVE_BACKEND) && (backend_cfg.format == AUDIO_FORMAT_DSD)) {
+        backend_cfg.bit_width = 16;
+        if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD64)
+            backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+        else if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD128)
+            backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+    }
     ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
           ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
           backend_cfg.sample_rate, backend_cfg.channels, backend_idx, usecase->id,
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 48bfb2b..9394ef8 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -80,6 +80,7 @@
     SND_DEVICE_OUT_SPEAKER_VBAT,
     SND_DEVICE_OUT_LINE,
     SND_DEVICE_OUT_HEADPHONES,
+    SND_DEVICE_OUT_HEADPHONES_DSD,
     SND_DEVICE_OUT_HEADPHONES_44_1,
     SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
     SND_DEVICE_OUT_SPEAKER_AND_LINE,
@@ -192,13 +193,18 @@
     SND_DEVICE_MAX = SND_DEVICE_IN_END,
 
 };
-
+#define INPUT_SAMPLING_RATE_DSD64       2822400
+#define INPUT_SAMPLING_RATE_DSD128      5644800
 #define DEFAULT_OUTPUT_SAMPLING_RATE    48000
 #define OUTPUT_SAMPLING_RATE_44100      44100
+#define OUTPUT_SAMPLING_RATE_DSD64       176400
+#define OUTPUT_SAMPLING_RATE_DSD128      352800
 #define MAX_CODEC_TX_BACKENDS           1
 enum {
     DEFAULT_CODEC_BACKEND,
     SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+    DSD_NATIVE_BACKEND,
+    SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
     HEADPHONE_44_1_BACKEND,
     SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
     HEADPHONE_BACKEND,
@@ -447,7 +453,8 @@
 enum {
     LEGACY_PCM = 0,
     PASSTHROUGH,
-    PASSTHROUGH_CONVERT
+    PASSTHROUGH_CONVERT,
+    PASSTHROUGH_DSD
 };
 /*
  * ID for setting mute and lateny on the device side
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 0bb73f3..625f4eb 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -27,6 +27,7 @@
 enum {
     NATIVE_AUDIO_MODE_SRC = 1,
     NATIVE_AUDIO_MODE_TRUE_44_1,
+    NATIVE_AUDIO_MODE_MULTIPLE_44_1,
     NATIVE_AUDIO_MODE_INVALID
 };
 
@@ -151,4 +152,8 @@
                           bool enable,
                           char * str);
 bool platform_supports_true_32bit();
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device, snd_device_t cuurent_snd_device);
+bool platform_check_codec_dsd_support(void *platform);
+bool platform_check_codec_asrc_support(void *platform);
+int platform_get_backend_index(snd_device_t snd_device);
 #endif // AUDIO_PLATFORM_API_H