diff --git a/configs/msm8937/audio_output_policy.conf b/configs/msm8937/audio_output_policy.conf
index 8f70451..3d6b978 100644
--- a/configs/msm8937/audio_output_policy.conf
+++ b/configs/msm8937/audio_output_policy.conf
@@ -32,23 +32,30 @@
     bit_width 16
     app_type 69936
   }
-  direct_pcm {
+  direct_pcm_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
-    formats AUDIO_FORMAT_PCM_16_BIT
+    formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
     sampling_rates 44100|48000|96000|192000
     bit_width 16
     app_type 69936
   }
+  direct_pcm_24 {
+    flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
+    sampling_rates 44100|48000|96000|192000
+    bit_width 24
+    app_type 69940
+  }
   compress_offload_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+    formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
     sampling_rates 44100|48000|96000|192000
     bit_width 16
     app_type 69936
   }
   compress_offload_24 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
+    formats AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
     sampling_rates 44100|48000|96000|192000
     bit_width 24
     app_type 69940
diff --git a/configs/msm8937/audio_policy.conf b/configs/msm8937/audio_policy.conf
index f69cfa5..3e59ba2 100644
--- a/configs/msm8937/audio_policy.conf
+++ b/configs/msm8937/audio_policy.conf
@@ -57,14 +57,14 @@
       direct_pcm {
         sampling_rates 8000|11025|16000|22050|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_PCM_16_BIT
+        formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_AUX_DIGITAL
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
       }
       compress_offload {
         sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
       }
diff --git a/configs/msm8953/audio_output_policy.conf b/configs/msm8953/audio_output_policy.conf
index 8f70451..3d6b978 100644
--- a/configs/msm8953/audio_output_policy.conf
+++ b/configs/msm8953/audio_output_policy.conf
@@ -32,23 +32,30 @@
     bit_width 16
     app_type 69936
   }
-  direct_pcm {
+  direct_pcm_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
-    formats AUDIO_FORMAT_PCM_16_BIT
+    formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
     sampling_rates 44100|48000|96000|192000
     bit_width 16
     app_type 69936
   }
+  direct_pcm_24 {
+    flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
+    sampling_rates 44100|48000|96000|192000
+    bit_width 24
+    app_type 69940
+  }
   compress_offload_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+    formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
     sampling_rates 44100|48000|96000|192000
     bit_width 16
     app_type 69936
   }
   compress_offload_24 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
+    formats AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
     sampling_rates 44100|48000|96000|192000
     bit_width 24
     app_type 69940
diff --git a/configs/msm8953/audio_policy.conf b/configs/msm8953/audio_policy.conf
index f69cfa5..3e59ba2 100644
--- a/configs/msm8953/audio_policy.conf
+++ b/configs/msm8953/audio_policy.conf
@@ -57,14 +57,14 @@
       direct_pcm {
         sampling_rates 8000|11025|16000|22050|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_PCM_16_BIT
+        formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_AUX_DIGITAL
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
       }
       compress_offload {
         sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
       }
diff --git a/configs/msm8996/audio_output_policy.conf b/configs/msm8996/audio_output_policy.conf
index 2f01bc9..93cd0c2 100644
--- a/configs/msm8996/audio_output_policy.conf
+++ b/configs/msm8996/audio_output_policy.conf
@@ -32,13 +32,20 @@
     bit_width 16
     app_type 69936
   }
-  direct_pcm {
+  direct_pcm_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
-    formats AUDIO_FORMAT_PCM_16_BIT
+    formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
     sampling_rates 44100|48000|96000|192000
     bit_width 16
     app_type 69936
   }
+  direct_pcm_24 {
+    flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
+    sampling_rates 44100|48000|96000|192000
+    bit_width 24
+    app_type 69940
+  }
   compress_passthrough {
     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
@@ -48,14 +55,14 @@
   }
   compress_offload_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+    formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
     sampling_rates 44100|48000|96000|192000
     bit_width 16
     app_type 69936
   }
   compress_offload_24 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
-    formats AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO
+    formats AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO
     sampling_rates 44100|48000|96000|192000
     bit_width 24
     app_type 69940
diff --git a/configs/msm8996/audio_policy.conf b/configs/msm8996/audio_policy.conf
index 453aca4..8a3bd30 100644
--- a/configs/msm8996/audio_policy.conf
+++ b/configs/msm8996/audio_policy.conf
@@ -58,16 +58,16 @@
         flags AUDIO_OUTPUT_FLAG_DIRECT
       }
       direct_pcm {
-        sampling_rates 44100|48000|64000|88200|96000|176400|192000
+        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_PCM_16_BIT
+        formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
       }
       compress_offload {
         sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
       }
@@ -99,18 +99,18 @@
         formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_AMR_NB|AUDIO_FORMAT_AMR_WB|AUDIO_FORMAT_QCELP|AUDIO_FORMAT_EVRC|AUDIO_FORMAT_EVRCB|AUDIO_FORMAT_EVRCWB|AUDIO_FORMAT_EVRCNW
         devices AUDIO_DEVICE_IN_WIRED_HEADSET|AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET|AUDIO_DEVICE_IN_FM_TUNER|AUDIO_DEVICE_IN_VOICE_CALL
       }
-      float {
-        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|96000|192000
-        channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK
-        formats AUDIO_FORMAT_PCM_24_BIT_PACKED
-        devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET
-      }
       surround_sound {
         sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000
         channel_masks AUDIO_CHANNEL_IN_5POINT1|AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK
         formats AUDIO_FORMAT_PCM_16_BIT
         devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC
       }
+      record_24 {
+        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|96000|192000
+        channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK
+        formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT|AUDIO_FORMAT_PCM_FLOAT
+        devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET
+      }
       voice_rx {
         sampling_rates 8000|16000|48000
         channel_masks AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_MONO
diff --git a/configs/msm8996/audio_policy_configuration.xml b/configs/msm8996/audio_policy_configuration.xml
index d0c4cdd..3f30e32 100644
--- a/configs/msm8996/audio_policy_configuration.xml
+++ b/configs/msm8996/audio_policy_configuration.xml
@@ -152,6 +152,17 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_IN_5POINT1"/>
                 </mixPort>
+                <mixPort name="record_24" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+                    <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+                </mixPort>
                 <mixPort name="voice_rx" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
@@ -268,6 +279,8 @@
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
+                <route type="mix" sink="record_24"
+                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
                 <route type="mix" sink="voice_rx"
diff --git a/configs/msm8996/msm8996.mk b/configs/msm8996/msm8996.mk
index 23b9213..2b70a81 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -189,3 +189,9 @@
 #enable hw aac encoder by default
 PRODUCT_PROPERTY_OVERRIDES += \
 qcom.hw.aac.encoder=true
+
+#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/audio_output_policy.conf b/configs/msmcobalt/audio_output_policy.conf
index cbfd43e..46fa191 100644
--- a/configs/msmcobalt/audio_output_policy.conf
+++ b/configs/msmcobalt/audio_output_policy.conf
@@ -32,13 +32,20 @@
     bit_width 16
     app_type 69936
   }
-  direct_pcm {
+  direct_pcm_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
-    formats AUDIO_FORMAT_PCM_16_BIT
+    formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
     sampling_rates 44100|48000|96000|192000
     bit_width 16
     app_type 69936
   }
+  direct_pcm_24 {
+    flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
+    sampling_rates 44100|48000|96000|192000
+    bit_width 24
+    app_type 69940
+  }
   compress_offload_16 {
     flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
     formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index acc2e4e..6bf14e2 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -67,5 +67,15 @@
         <!-- followed by perf lock options                             -->
         <param key="perf_lock_opts" value="4, 0x101, 0x704, 0x20F, 0x1E01"/>
     </config_params>
+    <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_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_VOICE_HEADPHONES" 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"/>
+    </backend_names>
 </audio_platform_info>
 
diff --git a/configs/msmcobalt/audio_policy.conf b/configs/msmcobalt/audio_policy.conf
index b32bce6..a9f4644 100644
--- a/configs/msmcobalt/audio_policy.conf
+++ b/configs/msmcobalt/audio_policy.conf
@@ -51,16 +51,16 @@
         flags AUDIO_OUTPUT_FLAG_DIRECT
       }
       direct_pcm {
-        sampling_rates 44100|48000|64000|88200|96000|176400|192000
+        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_PCM_16_BIT
+        formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
       }
       compress_offload {
         sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
         channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
-        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+        formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
       }
@@ -98,6 +98,12 @@
         formats AUDIO_FORMAT_PCM_16_BIT
         devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC
       }
+      record_24 {
+        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|96000|192000
+        channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK
+        formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
+        devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET
+      }
       voice_rx {
         sampling_rates 8000|16000|48000
         channel_masks AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_MONO
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index d0c4cdd..3f30e32 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -152,6 +152,17 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_IN_5POINT1"/>
                 </mixPort>
+                <mixPort name="record_24" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+                    <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+                    <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+                </mixPort>
                 <mixPort name="voice_rx" role="sink">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
@@ -268,6 +279,8 @@
                        sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
                 <route type="mix" sink="surround_sound"
                        sources="Built-In Mic,Built-In Back Mic"/>
+                <route type="mix" sink="record_24"
+                       sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
                 <route type="mix" sink="Telephony Tx"
                        sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
                 <route type="mix" sink="voice_rx"
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 68f91cb..93d671d 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -94,6 +94,18 @@
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="0" />
+    <ctl name="SLIMBUS_6_RX Port Mixer SLIM_0_TX" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia5" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia10" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia11" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia12" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia13" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia14" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia15" value="0" />
+    <ctl name="AUX_PCM_RX Audio Mixer MultiMedia16" value="0" />
     <ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia2" value="0" />
     <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="0" />
@@ -127,27 +139,40 @@
     <ctl name="HDMI Mixer MultiMedia15" value="0" />
     <ctl name="HDMI Mixer MultiMedia16" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia2" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_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_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_6_RX Audio Mixer MultiMedia7" 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_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_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_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_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_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_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_6_RX Audio Mixer MultiMedia16" value="0" />
     <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
     <ctl name="IIR0 INP0 MUX" value="ZERO" />
     <ctl name="IIR0 INP1 MUX" value="ZERO" />
@@ -286,8 +311,10 @@
     <ctl name="SLIM_0_TX Channels" value="One" />
     <ctl name="SLIM_1_TX Channels" value="One" />
     <ctl name="SLIM RX0 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="EAR PA Gain" value="G_6_DB" />
     <ctl name="SpkrLeft COMP Switch" value="0" />
     <ctl name="SpkrRight COMP Switch" value="0" />
@@ -308,6 +335,7 @@
     <ctl name="AIF1_CAP Mixer SLIM TX1" value="0"/>
     <ctl name="AIF1_CAP Mixer SLIM TX0" value="0"/>
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+    <ctl name="SLIMBUS_6_RX Port Mixer AUX_PCM_UL_TX" value="0" />
     <ctl name="HDMI Mixer MultiMedia4" value="0" />
     <!-- HFP start -->
     <ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
@@ -338,6 +366,7 @@
 
     <!-- Voice -->
     <ctl name="SLIM_0_RX_Voice Mixer CSVoice" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer CSVoice" value="0" />
     <ctl name="Voice_Tx Mixer SLIM_0_TX_Voice" value="0" />
     <!-- Voice HDMI -->
     <ctl name="HDMI_RX_Voice Mixer CSVoice" value="0" />
@@ -352,6 +381,7 @@
 
     <!-- Voice2 -->
     <ctl name="SLIM_0_RX_Voice Mixer Voice2" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer Voice2" value="0" />
     <ctl name="Voice2_Tx Mixer SLIM_0_TX_Voice2" value="0" />
     <!-- Voice2 HDMI -->
     <ctl name="HDMI_RX_Voice Mixer Voice2" value="0" />
@@ -365,6 +395,7 @@
 
     <!-- VoLTE -->
     <ctl name="SLIM_0_RX_Voice Mixer VoLTE" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer VoLTE" value="0" />
     <ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="0" />
     <!-- VoLTE HDMI -->
     <ctl name="HDMI_RX_Voice Mixer VoLTE" value="0" />
@@ -378,6 +409,7 @@
 
     <!-- Multimode Voice1 -->
     <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode1" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="0" />
     <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="0" />
     <!-- Multimode Voice1 HDMI -->
     <ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="0" />
@@ -391,6 +423,7 @@
 
     <!-- Multimode Voice2 -->
     <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode2" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="0" />
     <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="0" />
     <!-- Multimode Voice2 HDMI -->
     <ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="0" />
@@ -427,6 +460,7 @@
 
     <!-- compress-voip-call start -->
     <ctl name="SLIM_0_RX_Voice Mixer Voip" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="0" />
     <ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
     <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
@@ -436,6 +470,7 @@
 
     <!-- QCHAT start -->
     <ctl name="SLIM_0_RX_Voice Mixer QCHAT" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer QCHAT" value="0" />
     <ctl name="QCHAT_Tx Mixer SLIM_0_TX_QCHAT" value="0" />
     <ctl name="SLIM_7_RX_Voice Mixer QCHAT" value="0" />
     <ctl name="QCHAT_Tx Mixer SLIM_7_TX_QCHAT" value="0" />
@@ -443,6 +478,7 @@
 
     <!-- VoWLAN start -->
     <ctl name="SLIM_0_RX_Voice Mixer VoWLAN" value="0" />
+    <ctl name="SLIM_6_RX_Voice Mixer VoWLAN" value="0" />
     <ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="0" />
     <ctl name="HDMI_RX_Voice Mixer VoWLAN" value="0" />
     <ctl name="SLIM_7_RX_Voice Mixer VoWLAN" value="0" />
@@ -533,6 +569,7 @@
 
     <!-- ADSP testfwk -->
     <ctl name="SLIMBUS_DL_HL Switch" value="0" />
+    <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
     <!-- ADSP testfwk end-->
 
     <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
@@ -566,7 +603,7 @@
     </path>
 
     <path name="echo-reference headphones">
-        <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
+        <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
     </path>
 
     <path name="echo-reference headphones-44.1">
@@ -613,6 +650,11 @@
     </path>
 
     <path name="deep-buffer-playback headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-headphones">
+        <path name="deep-buffer-playback headphones" />
         <path name="deep-buffer-playback" />
     </path>
 
@@ -656,6 +698,11 @@
     </path>
 
     <path name="low-latency-playback headphones">
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-headphones">
+        <path name="low-latency-playback headphones" />
         <path name="low-latency-playback" />
     </path>
 
@@ -668,7 +715,7 @@
     </path>
 
     <path name="audio-ull-playback headphones">
-      <path name="audio-ull-playback" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-and-headphones">
@@ -745,13 +792,18 @@
     </path>
 
     <path name="compress-offload-playback headphones">
-        <path name="compress-offload-playback" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="1" />
     </path>
 
     <path name="compress-offload-playback headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="1" />
     </path>
 
+    <path name="compress-offload-playback speaker-and-headphones">
+        <path name="compress-offload-playback headphones" />
+        <path name="compress-offload-playback" />
+    </path>
+
     <path name="compress-offload-playback2">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="1" />
     </path>
@@ -788,13 +840,18 @@
     </path>
 
     <path name="compress-offload-playback2 headphones">
-        <path name="compress-offload-playback2" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="1" />
     </path>
 
     <path name="compress-offload-playback2 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="1" />
     </path>
 
+    <path name="compress-offload-playback2 speaker-and-headphones">
+        <path name="compress-offload-playback2 headphones" />
+        <path name="compress-offload-playback2" />
+    </path>
+
     <path name="compress-offload-playback3">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
     </path>
@@ -831,13 +888,18 @@
     </path>
 
     <path name="compress-offload-playback3 headphones">
-        <path name="compress-offload-playback3" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="1" />
     </path>
 
     <path name="compress-offload-playback3 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="1" />
     </path>
 
+    <path name="compress-offload-playback3 speaker-and-headphones">
+        <path name="compress-offload-playback3 headphones" />
+        <path name="compress-offload-playback3" />
+    </path>
+
     <path name="compress-offload-playback4">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="1" />
     </path>
@@ -874,13 +936,18 @@
     </path>
 
     <path name="compress-offload-playback4 headphones">
-        <path name="compress-offload-playback4" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="1" />
     </path>
 
     <path name="compress-offload-playback4 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="1" />
     </path>
 
+    <path name="compress-offload-playback4 speaker-and-headphones">
+        <path name="compress-offload-playback4 headphones" />
+        <path name="compress-offload-playback4" />
+    </path>
+
     <path name="compress-offload-playback5">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="1" />
     </path>
@@ -917,13 +984,18 @@
     </path>
 
     <path name="compress-offload-playback5 headphones">
-        <path name="compress-offload-playback5" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="1" />
     </path>
 
     <path name="compress-offload-playback5 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="1" />
     </path>
 
+    <path name="compress-offload-playback5 speaker-and-headphones">
+        <path name="compress-offload-playback5 headphones" />
+        <path name="compress-offload-playback5" />
+    </path>
+
     <path name="compress-offload-playback6">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="1" />
     </path>
@@ -960,13 +1032,18 @@
     </path>
 
     <path name="compress-offload-playback6 headphones">
-        <path name="compress-offload-playback6" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="1" />
     </path>
 
     <path name="compress-offload-playback6 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="1" />
     </path>
 
+    <path name="compress-offload-playback6 speaker-and-headphones">
+        <path name="compress-offload-playback6 headphones" />
+        <path name="compress-offload-playback6" />
+    </path>
+
     <path name="compress-offload-playback7">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="1" />
     </path>
@@ -1003,13 +1080,18 @@
     </path>
 
     <path name="compress-offload-playback7 headphones">
-        <path name="compress-offload-playback7" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="1" />
     </path>
 
     <path name="compress-offload-playback7 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="1" />
     </path>
 
+    <path name="compress-offload-playback7 speaker-and-headphones">
+        <path name="compress-offload-playback7 headphones" />
+        <path name="compress-offload-playback7" />
+    </path>
+
     <path name="compress-offload-playback8">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="1" />
     </path>
@@ -1046,13 +1128,18 @@
     </path>
 
     <path name="compress-offload-playback8 headphones">
-        <path name="compress-offload-playback8" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="1" />
     </path>
 
     <path name="compress-offload-playback8 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="1" />
     </path>
 
+    <path name="compress-offload-playback8 speaker-and-headphones">
+        <path name="compress-offload-playback8 headphones" />
+        <path name="compress-offload-playback8" />
+    </path>
+
     <path name="compress-offload-playback9">
         <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
     </path>
@@ -1089,13 +1176,18 @@
     </path>
 
     <path name="compress-offload-playback9 headphones">
-        <path name="compress-offload-playback9" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="1" />
     </path>
 
     <path name="compress-offload-playback9 headphones-44.1">
         <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="1" />
     </path>
 
+    <path name="compress-offload-playback9 speaker-and-headphones">
+        <path name="compress-offload-playback9 headphones" />
+        <path name="compress-offload-playback9" />
+    </path>
+
     <path name="audio-record">
         <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="1" />
     </path>
@@ -1189,6 +1281,11 @@
         <ctl name="Voice_Tx Mixer AFE_PCM_TX_Voice" value="1" />
     </path>
 
+    <path name="voice-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer CSVoice" value="1" />
+        <ctl name="Voice_Tx Mixer SLIM_0_TX_Voice" value="1" />
+    </path>
+
     <path name="voice2-call">
         <ctl name="SLIM_0_RX_Voice Mixer Voice2" value="1" />
         <ctl name="Voice2_Tx Mixer SLIM_0_TX_Voice2" value="1" />
@@ -1229,6 +1326,11 @@
         <path name="voice2-call"/>
     </path>
 
+    <path name="voice2-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer Voice2" value="1" />
+        <ctl name="Voice2_Tx Mixer SLIM_0_TX_Voice2" value="1" />
+    </path>
+
     <path name="play-fm">
         <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
         <ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="1" />
@@ -1236,7 +1338,9 @@
     </path>
 
     <path name="play-fm headphones">
-        <path name="play-fm" />
+       <ctl name="Tert MI2S LOOPBACK Volume" value="1" />
+       <ctl name="SLIMBUS_6_RX Port Mixer TERT_MI2S_TX" value="1" />
+       <ctl name="SLIMBUS6_DL_HL Switch" value="1" />
     </path>
 
     <path name="incall-rec-uplink">
@@ -1369,16 +1473,34 @@
         <ctl name="SLIMBUS_DL_HL Switch" value="1" />
    </path>
 
+    <path name="hfp-sco headphones">
+        <ctl name="HFP_PRI_AUX_UL_HL Switch" value="1" />
+        <ctl name="SLIMBUS_6_RX Port Mixer AUX_PCM_UL_TX" value="1" />
+        <ctl name="AUX_PCM_RX Audio Mixer MultiMedia6" value="1" />
+        <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="1" />
+        <ctl name="SLIMBUS6_DL_HL Switch" value="1" />
+    </path>
+
    <path name="hfp-sco-wb">
         <ctl name="BT_SCO SampleRate" value="16000" />
         <path name="hfp-sco" />
    </path>
 
+    <path name="hfp-sco-wb headphones">
+        <ctl name="AUX PCM SampleRate" value="16000" />
+        <path name="hfp-sco headphones" />
+    </path>
+
     <path name="volte-call">
         <ctl name="SLIM_0_RX_Voice Mixer VoLTE" value="1" />
         <ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="1" />
     </path>
 
+    <path name="volte-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer VoLTE" value="1" />
+        <ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="1" />
+    </path>
+
     <path name="volte-call hdmi">
         <ctl name="HDMI_RX_Voice Mixer VoLTE" value="1" />
         <ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="1" />
@@ -1414,6 +1536,12 @@
         <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="1" />
     </path>
 
+    <path name="compress-voip-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer Voip" value="1" />
+        <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="1" />
+    </path>
+
+
     <path name="compress-voip-call bt-sco">
         <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
         <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="1" />
@@ -1444,6 +1572,11 @@
         <ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="1" />
     </path>
 
+    <path name="vowlan-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer VoWLAN" value="1" />
+        <ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="1" />
+    </path>
+
     <path name="vowlan-call hdmi">
         <ctl name="HDMI_RX_Voice Mixer VoWLAN" value="1" />
         <ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="1" />
@@ -1479,6 +1612,11 @@
         <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
     </path>
 
+    <path name="voicemmode1-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="1" />
+        <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+    </path>
+
     <path name="voicemmode1-call hdmi">
         <ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="1" />
         <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
@@ -1514,6 +1652,11 @@
         <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
     </path>
 
+    <path name="voicemmode2-call headphones">
+        <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="1" />
+        <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+    </path>
+
     <path name="voicemmode2-call hdmi">
         <ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="1" />
         <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
@@ -1851,22 +1994,22 @@
     </path>
 
     <path name="headphones">
-        <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM_0_RX Channels" value="Two" />
-        <ctl name="RX INT1_2 MUX" value="RX0" />
-        <ctl name="RX INT2_2 MUX" value="RX1" />
+        <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 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
     </path>
 
     <path name="headphones-44.1">
-        <ctl name="SLIM RX3 MUX" value="AIF3_PB" />
         <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="SLIM_5_RX SampleRate" value="KHZ_44P1" />
-        <ctl name="RX INT1_1 MIX1 INP0" value="RX3" />
-        <ctl name="RX INT2_1 MIX1 INP1" value="RX4" />
+        <ctl name="RX INT1_1 MIX1 INP0" value="RX4" />
+        <ctl name="RX INT2_1 MIX1 INP1" value="RX5" />
         <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="SPL SRC0 MUX" value="SRC_IN_HPHL" />
@@ -1879,8 +2022,8 @@
     <path name="true-native-mode">
         <ctl name="RX INT1_2 MUX" value="ZERO" />
         <ctl name="RX INT2_2 MUX" value="ZERO" />
-        <ctl name= "RX INT1_1 MIX1 INP0" value="RX0" />
-        <ctl name= "RX INT2_1 MIX1 INP0" value="RX1" />
+        <ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
+        <ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
         <ctl name= "RX INT1 SPLINE MIX HPHL Native Switch" value="1" />
         <ctl name= "RX INT2 SPLINE MIX HPHR Native Switch" value="1" />
     </path>
@@ -1939,16 +2082,7 @@
 
     <path name="speaker-and-headphones">
         <path name="headphones" />
-        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
-        <ctl name="SpkrLeft COMP Switch" value="1" />
-        <ctl name="SpkrRight COMP Switch" value="1" />
-        <ctl name="SpkrLeft BOOST Switch" value="1" />
-        <ctl name="SpkrRight BOOST Switch" value="1" />
-        <ctl name="SpkrLeft VISENSE Switch" value="1" />
-        <ctl name="SpkrRight VISENSE Switch" value="1" />
-        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
-        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+        <path name="speaker" />
     </path>
 
     <path name="speaker-and-line">
@@ -1957,16 +2091,7 @@
 
     <path name="speaker-and-headphones-liquid">
         <path name="headphones" />
-        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
-        <ctl name="SpkrLeft COMP Switch" value="1" />
-        <ctl name="SpkrRight COMP Switch" value="1" />
-        <ctl name="SpkrLeft BOOST Switch" value="1" />
-        <ctl name="SpkrRight BOOST Switch" value="1" />
-        <ctl name="SpkrLeft VISENSE Switch" value="1" />
-        <ctl name="SpkrRight VISENSE Switch" value="1" />
-        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
-        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+        <path name="speaker" />
     </path>
 
     <path name="speaker-and-line-liquid">
@@ -1992,11 +2117,11 @@
         <ctl name="ANC1 FB MUX" value="ANC_IN_HPHR" />
         <ctl name="ADC3 Volume" value="8" />
         <ctl name="ADC4 Volume" value="8" />
-        <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM_0_RX Channels" value="Two" />
-        <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT2_1 MIX1 INP0" value="RX1" />
+        <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 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="RX1 Digital Volume" value="81" />
@@ -2009,16 +2134,7 @@
 
     <path name="speaker-and-anc-headphones">
         <path name="anc-headphones" />
-        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
-        <ctl name="SpkrLeft COMP Switch" value="1" />
-        <ctl name="SpkrRight COMP Switch" value="1" />
-        <ctl name="SpkrLeft BOOST Switch" value="1" />
-        <ctl name="SpkrRight BOOST Switch" value="1" />
-        <ctl name="SpkrLeft VISENSE Switch" value="1" />
-        <ctl name="SpkrRight VISENSE Switch" value="1" />
-        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
-        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+        <path name="speaker" />
     </path>
 
     <path name="anc-fb-headphones">
@@ -2028,16 +2144,7 @@
 
     <path name="speaker-and-anc-fb-headphones">
         <path name="anc-fb-headphones" />
-        <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
-        <ctl name="SpkrLeft COMP Switch" value="1" />
-        <ctl name="SpkrRight COMP Switch" value="1" />
-        <ctl name="SpkrLeft BOOST Switch" value="1" />
-        <ctl name="SpkrRight BOOST Switch" value="1" />
-        <ctl name="SpkrLeft VISENSE Switch" value="1" />
-        <ctl name="SpkrRight VISENSE Switch" value="1" />
-        <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
-        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+        <path name="speaker" />
     </path>
 
     <path name="voice-anc-headphones">
@@ -2053,11 +2160,11 @@
         <ctl name="ANC1 FB MUX" value="ANC_IN_HPHR" />
         <ctl name="ADC3 Volume" value="8" />
         <ctl name="ADC4 Volume" value="8" />
-        <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM_0_RX Channels" value="Two" />
-        <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT2_1 MIX1 INP0" value="RX1" />
+        <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 HPH Mode" value="CLS_H_LP" />
         <ctl name="IIR0 Enable Band1" value="1" />
         <ctl name="IIR0 Enable Band2" value="1" />
@@ -2088,11 +2195,11 @@
         <ctl name="ANC1 FB MUX" value="ANC_IN_HPHR" />
         <ctl name="ADC3 Volume" value="8" />
         <ctl name="ADC4 Volume" value="8" />
-        <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM_0_RX Channels" value="Two" />
-        <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
-        <ctl name="RX INT2_1 MIX1 INP0" value="RX1" />
+        <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 HPH Mode" value="CLS_H_LP" />
         <ctl name="IIR0 Enable Band1" value="1" />
         <ctl name="IIR0 Enable Band2" value="1" />
@@ -2343,9 +2450,9 @@
     <!-- TTY devices -->
 
     <path name="tty-headphones">
-        <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
-        <ctl name="SLIM_0_RX Channels" value="One" />
-        <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
+        <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+        <ctl name="SLIM_6_RX Channels" value="One" />
+        <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
         <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
     </path>
 
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 91d7c0a..3ce83b1 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -40,17 +40,6 @@
 
 #include <cutils/str_parms.h>
 
-#ifndef PCM_OFFLOAD_ENABLED
-#define AUDIO_FORMAT_PCM_OFFLOAD 0x1A000000UL
-#define AUDIO_FORMAT_PCM_16_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_16_BIT)
-#define AUDIO_FORMAT_PCM_24_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_8_24_BIT)
-#define AUDIO_OFFLOAD_CODEC_FORMAT  "music_offload_codec_format"
-#define audio_is_offload_pcm(format) (0)
-#define OFFLOAD_USE_SMALL_BUFFER false
-#else
-#define OFFLOAD_USE_SMALL_BUFFER ((info->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD)
-#endif
-
 #ifndef AFE_PROXY_ENABLED
 #define AUDIO_DEVICE_OUT_PROXY 0x40000
 #endif
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index b3ba2b5..cd9ead7 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -97,6 +97,8 @@
 
 const struct string_to_enum s_format_name_to_enum_table[] = {
     STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
     STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
     STRING_TO_ENUM(AUDIO_FORMAT_MP3),
     STRING_TO_ENUM(AUDIO_FORMAT_AAC),
@@ -117,8 +119,6 @@
     STRING_TO_ENUM(AUDIO_FORMAT_QCELP),
     STRING_TO_ENUM(AUDIO_FORMAT_MP2),
     STRING_TO_ENUM(AUDIO_FORMAT_EVRCNW),
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT_OFFLOAD),
-    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_OFFLOAD),
     STRING_TO_ENUM(AUDIO_FORMAT_FLAC),
     STRING_TO_ENUM(AUDIO_FORMAT_ALAC),
     STRING_TO_ENUM(AUDIO_FORMAT_APE),
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 8486e18..ad565de 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -74,6 +74,8 @@
 #include "sound/asound.h"
 
 #define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+/*DIRECT PCM has same buffer sizes as DEEP Buffer*/
+#define DIRECT_PCM_NUM_FRAGMENTS 2
 /* ToDo: Check and update a proper value in msec */
 #define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
 #define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
@@ -287,17 +289,17 @@
     return ret_val;
 }
 
-static int check_and_set_gapless_mode(struct audio_device *adev) {
-
-
-    char value[PROPERTY_VALUE_MAX] = {0};
+static int check_and_set_gapless_mode(struct audio_device *adev, bool enable_gapless)
+{
     bool gapless_enabled = false;
     const char *mixer_ctl_name = "Compress Gapless Playback";
     struct mixer_ctl *ctl;
 
     ALOGV("%s:", __func__);
-    property_get("audio.offload.gapless.enabled", value, NULL);
-    gapless_enabled = atoi(value) || !strncmp("true", value, 4);
+    gapless_enabled = property_get_bool("audio.offload.gapless.enabled", false);
+
+    /*Disable gapless if its AV playback*/
+    gapless_enabled = gapless_enabled && enable_gapless;
 
     ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
     if (!ctl) {
@@ -323,8 +325,8 @@
         format == AUDIO_FORMAT_AAC_ADTS_LC ||
         format == AUDIO_FORMAT_AAC_ADTS_HE_V1 ||
         format == AUDIO_FORMAT_AAC_ADTS_HE_V2 ||
-        format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
-        format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD ||
+        format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+        format == AUDIO_FORMAT_PCM_8_24_BIT ||
         format == AUDIO_FORMAT_PCM_16_BIT ||
         format == AUDIO_FORMAT_FLAC ||
         format == AUDIO_FORMAT_ALAC ||
@@ -351,7 +353,6 @@
     case AUDIO_FORMAT_AAC_ADTS:
         id = SND_AUDIOCODEC_AAC;
         break;
-    case AUDIO_FORMAT_PCM_OFFLOAD:
     case AUDIO_FORMAT_PCM:
         id = SND_AUDIOCODEC_PCM;
         break;
@@ -1036,9 +1037,9 @@
         if (voice_is_in_call(adev) && adev->mode != AUDIO_MODE_NORMAL) {
             vc_usecase = get_usecase_from_list(adev,
                                                get_usecase_id_from_usecase_type(adev, VOICE_CALL));
-            if (((vc_usecase) && ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
-                                 (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND))) ||
-                                (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL)) {
+            if ((vc_usecase) && (((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+                                 (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) ||
+                                (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
                 in_snd_device = vc_usecase->in_snd_device;
                 out_snd_device = vc_usecase->out_snd_device;
             }
@@ -1956,7 +1957,7 @@
            for the default max poll time (20s) in the event of an SSR.
            Reduce the poll time to observe and deal with SSR faster.
         */
-        if (out->use_small_bufs) {
+        if (!out->non_blocking) {
             compress_set_max_poll_wait(out->compr, 1000);
         }
 
@@ -1973,7 +1974,7 @@
             if (adev->visualizer_start_output != NULL)
                 adev->visualizer_start_output(out->handle, out->pcm_device_id);
             if (adev->offload_effects_start_output != NULL)
-                adev->offload_effects_start_output(out->handle, out->pcm_device_id);
+                adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer);
             audio_extn_check_and_set_dts_hpx_state(adev);
         }
     }
@@ -2061,6 +2062,37 @@
     return size;
 }
 
+static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
+{
+    uint64_t actual_frames_rendered = 0;
+    size_t kernel_buffer_size = out->compr_config.fragment_size * out->compr_config.fragments;
+
+    /* This adjustment accounts for buffering after app processor.
+     * It is based on estimated DSP latency per use case, rather than exact.
+     */
+    int64_t platform_latency =  platform_render_latency(out->usecase) *
+                                out->sample_rate / 1000000LL;
+
+    /* not querying actual state of buffering in kernel as it would involve an ioctl call
+     * which then needs protection, this causes delay in TS query for pcm_offload usecase
+     * hence only estimate.
+     */
+    int64_t signed_frames = out->written - kernel_buffer_size;
+
+    signed_frames = signed_frames / (audio_bytes_per_sample(out->format) * popcount(out->channel_mask)) - platform_latency;
+
+    if (signed_frames > 0)
+        actual_frames_rendered = signed_frames;
+
+    ALOGVV("%s signed frames %lld out_written %lld kernel_buffer_size %d"
+            "bytes/sample %zu channel count %d", __func__,(long long int)signed_frames,
+             (long long int)out->written, (int)kernel_buffer_size,
+             audio_bytes_per_sample(out->compr_config.codec->format),
+             popcount(out->channel_mask));
+
+    return actual_frames_rendered;
+}
+
 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
 {
     struct stream_out *out = (struct stream_out *)stream;
@@ -2553,6 +2585,9 @@
             out_standby(&out->stream.common);
             return ret;
         }
+        if ( ret == (ssize_t)bytes && !out->non_blocking)
+            out->written += bytes;
+
         if (!out->playback_started && ret >= 0) {
             compress_start(out->compr);
             audio_extn_dts_eagle_fade(adev, true, out);
@@ -2627,14 +2662,24 @@
     *dsp_frames = 0;
     if (is_offload_usecase(out->usecase)) {
         ssize_t ret = 0;
+
+        /* Below piece of code is not guarded against any lock beacuse audioFliner serializes
+         * this operation and adev_close_output_stream(where out gets reset).
+         */
+        if (!out->non_blocking && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+            *dsp_frames = get_actual_pcm_frames_rendered(out);
+             ALOGVV("dsp_frames %d sampleRate %d",(int)*dsp_frames,out->sample_rate);
+             return 0;
+        }
+
         lock_output_stream(out);
-        if (out->compr != NULL) {
+        if (out->compr != NULL && out->non_blocking) {
             ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
                     &out->sample_rate);
             if (ret < 0)
                 ret = -errno;
             ALOGVV("%s rendered frames %d sample_rate %d",
-                   __func__, *dsp_frames, out->sample_rate);
+                    __func__, *dsp_frames, out->sample_rate);
         }
         pthread_mutex_unlock(&out->lock);
         if (-ENETRESET == ret) {
@@ -2686,27 +2731,37 @@
     int ret = -1;
     unsigned long dsp_frames;
 
+    /* below piece of code is not guarded against any lock because audioFliner serializes
+     * this operation and adev_close_output_stream( where out gets reset).
+     */
+    if (is_offload_usecase(out->usecase) && !out->non_blocking &&
+        (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+        *frames = get_actual_pcm_frames_rendered(out);
+        /* this is the best we can do */
+        clock_gettime(CLOCK_MONOTONIC, timestamp);
+        ALOGVV("frames %lld playedat %lld",(long long int)*frames,
+             timestamp->tv_sec * 1000000LL + timestamp->tv_nsec / 1000);
+        return 0;
+    }
+
     lock_output_stream(out);
 
-    if (is_offload_usecase(out->usecase)) {
-        if (out->compr != NULL) {
-            ret = compress_get_tstamp(out->compr, &dsp_frames,
-                    &out->sample_rate);
-            ALOGVV("%s rendered frames %ld sample_rate %d",
-                   __func__, dsp_frames, out->sample_rate);
-            *frames = dsp_frames;
-            if (ret < 0)
-                ret = -errno;
-            if (-ENETRESET == ret) {
-                ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
-                set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
-                ret = -EINVAL;
-            } else
-                ret = 0;
-
-            /* this is the best we can do */
-            clock_gettime(CLOCK_MONOTONIC, timestamp);
-        }
+    if (is_offload_usecase(out->usecase) && out->compr != NULL && out->non_blocking) {
+        ret = compress_get_tstamp(out->compr, &dsp_frames,
+                 &out->sample_rate);
+        ALOGVV("%s rendered frames %ld sample_rate %d",
+               __func__, dsp_frames, out->sample_rate);
+        *frames = dsp_frames;
+        if (ret < 0)
+            ret = -errno;
+        if (-ENETRESET == ret) {
+            ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
+            set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+            ret = -EINVAL;
+        } else
+            ret = 0;
+         /* this is the best we can do */
+        clock_gettime(CLOCK_MONOTONIC, timestamp);
     } else {
         if (out->pcm) {
             unsigned int avail;
@@ -2837,6 +2892,7 @@
         ALOGD("copl(%p):calling compress flush", out);
         lock_output_stream(out);
         stop_compressed_output_l(out);
+        out->written = 0;
         pthread_mutex_unlock(&out->lock);
         ALOGD("copl(%p):out of compress flush", out);
         return 0;
@@ -3238,7 +3294,6 @@
     out->handle = handle;
     out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
     out->non_blocking = 0;
-    out->use_small_bufs = false;
 
     if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
         (flags & AUDIO_OUTPUT_FLAG_DIRECT)) {
@@ -3337,6 +3392,9 @@
         }
 
         if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+            out->stream.pause = out_pause;
+            out->stream.flush = out_flush;
+            out->stream.resume = out_resume;
             out->usecase = get_offload_usecase(adev, true);
             ALOGV("DIRECT_PCM usecase ... usecase selected %d ", out->usecase);
         } else {
@@ -3381,18 +3439,19 @@
             out->compr_config.codec->id =
                 get_snd_codec_id(config->offload_info.format);
 
-        if (((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD)||
-             ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM)) {
+        if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
             out->compr_config.fragment_size =
                platform_get_pcm_offload_buffer_size(&config->offload_info);
+            out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS;
         } else if (audio_extn_dolby_is_passthrough_stream(out)) {
             out->compr_config.fragment_size =
                audio_extn_dolby_get_passt_buffer_size(&config->offload_info);
+            out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
         } else {
             out->compr_config.fragment_size =
                platform_get_compress_offload_buffer_size(&config->offload_info);
+            out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
         }
-        out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
         out->compr_config.codec->sample_rate =
                     config->offload_info.sample_rate;
         out->compr_config.codec->bit_rate =
@@ -3408,16 +3467,12 @@
              out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
         if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS)
             out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
-        if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD)
-            out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
-        if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
-            out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
         if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT)
             out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
-
-        if (out->bit_width == 24) {
+        if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+            out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_3LE;
+        if (config->offload_info.format == AUDIO_FORMAT_PCM_8_24_BIT)
             out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
-        }
 
         if (config->offload_info.format == AUDIO_FORMAT_FLAC)
             out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH;
@@ -3425,14 +3480,6 @@
         if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
             out->non_blocking = 1;
 
-        if (platform_use_small_buffer(&config->offload_info)) {
-            //this flag is set from framework only if its for PCM formats
-            //no need to check for PCM format again
-            out->non_blocking = 0;
-            out->use_small_bufs = true;
-            ALOGI("Keep write blocking for small buff: non_blockling %d",
-                  out->non_blocking);
-        }
 
         out->send_new_metadata = 1;
         out->send_next_track_params = false;
@@ -3446,11 +3493,18 @@
         ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
                 __func__, config->offload_info.version,
                 config->offload_info.bit_rate);
-        //Decide if we need to use gapless mode by default
-        if (!audio_extn_dolby_is_passthrough_stream(out)) {
-            ALOGV("%s: don't enable gapless for passthrough", __func__);
-            check_and_set_gapless_mode(adev);
-        }
+
+        /* Disable gapless if any of the following is true
+         * passthrough playback
+         * AV playback
+         * Direct PCM playback
+         */
+        if (audio_extn_dolby_is_passthrough_stream(out) ||
+            config->offload_info.has_video ||
+            out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+            check_and_set_gapless_mode(adev, false);
+        } else
+            check_and_set_gapless_mode(adev, true);
 
         if (audio_extn_dolby_is_passthrough_stream(out)) {
             out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
@@ -4288,7 +4342,7 @@
             ALOGV("%s: DLOPEN successful for %s", __func__,
                   OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
             adev->offload_effects_start_output =
-                        (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
+                        (int (*)(audio_io_handle_t, int, struct mixer *))dlsym(adev->offload_effects_lib,
                                          "offload_effects_bundle_hal_start_output");
             adev->offload_effects_stop_output =
                         (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 31184d5..6c97840 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -209,7 +209,6 @@
     struct stream_app_type_cfg app_type_cfg;
 
     int non_blocking;
-    bool use_small_bufs;
     int playback_started;
     int offload_state;
     pthread_cond_t offload_cond;
@@ -343,7 +342,7 @@
     int (*visualizer_start_output)(audio_io_handle_t, int);
     int (*visualizer_stop_output)(audio_io_handle_t, int);
     void *offload_effects_lib;
-    int (*offload_effects_start_output)(audio_io_handle_t, int);
+    int (*offload_effects_start_output)(audio_io_handle_t, int, struct mixer *);
     int (*offload_effects_stop_output)(audio_io_handle_t, int);
 
     struct sound_card_status snd_card_status;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 857d9e1..46611fc 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -760,6 +760,7 @@
 
 
 #define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
 #define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
 
 static bool is_misc_usecase(audio_usecase_t usecase) {
@@ -3651,7 +3652,7 @@
     free(kv_pairs);
 }
 
-/* Delay in Us */
+/* Delay in Us, only to be used for PCM formats */
 int64_t platform_render_latency(audio_usecase_t usecase)
 {
     switch (usecase) {
@@ -3659,6 +3660,9 @@
             return DEEP_BUFFER_PLATFORM_DELAY;
         case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
             return LOW_LATENCY_PLATFORM_DELAY;
+        case USECASE_AUDIO_PLAYBACK_OFFLOAD:
+        case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
+            return PCM_OFFLOAD_PLATFORM_DELAY;
         default:
             return 0;
     }
@@ -3846,19 +3850,17 @@
 uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
 {
     uint32_t fragment_size = 0;
-    uint32_t bits_per_sample = 16;
+    uint32_t bytes_per_sample;
     uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
 
-    if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
-        bits_per_sample = 32;
-    }
+    bytes_per_sample = audio_bytes_per_sample(info->format);
 
     //duration is set to 40 ms worth of stereo data at 48Khz
     //with 16 bit per sample, modify this when the channel
     //configuration is different
     fragment_size = (pcm_offload_time
                      * info->sample_rate
-                     * (bits_per_sample >> 3)
+                     * bytes_per_sample
                      * popcount(info->channel_mask))/1000;
     if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
         fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
@@ -3867,23 +3869,18 @@
     // To have same PCM samples for all channels, the buffer size requires to
     // be multiple of (number of channels * bytes per sample)
     // For writes to succeed, the buffer must be written at address which is multiple of 32
-    fragment_size = ALIGN(fragment_size, ((bits_per_sample >> 3)* popcount(info->channel_mask) * 32));
+    fragment_size = ALIGN(fragment_size, (bytes_per_sample * popcount(info->channel_mask) * 32));
 
     ALOGI("PCM offload Fragment size to %d bytes", fragment_size);
     return fragment_size;
 }
 
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
-    return OFFLOAD_USE_SMALL_BUFFER;
-}
-
 /*
  * configures afe with bit width and Sample Rate
  */
 static int platform_set_codec_backend_cfg(struct audio_device* adev,
-                         snd_device_t snd_device,
-                         unsigned int bit_width, unsigned int sample_rate)
+                         snd_device_t snd_device, unsigned int bit_width,
+                         unsigned int sample_rate, audio_format_t format)
 {
     int ret = 0;
     int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3908,13 +3905,17 @@
         }
 
         if (bit_width == 24) {
+            if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+                mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+            else
                 mixer_ctl_set_enum_by_string(ctl, "S24_LE");
         } else {
             mixer_ctl_set_enum_by_string(ctl, "S16_LE");
         }
         my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
-        ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
-              my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+        ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+              my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl,
+              bit_width, format);
     }
 
     /*
@@ -4150,11 +4151,13 @@
     int new_snd_devices[SND_DEVICE_OUT_END];
     int i, num_devices = 1;
     bool ret = false;
+    audio_format_t format;
 
     backend_idx = platform_get_backend_index(snd_device);
 
     new_bit_width = usecase->stream.out->bit_width;
     new_sample_rate = usecase->stream.out->sample_rate;
+    format = usecase->stream.out->format;
 
     ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
           ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
@@ -4171,7 +4174,7 @@
         if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
                                              &new_bit_width, &new_sample_rate)) {
                 platform_set_codec_backend_cfg(adev, new_snd_devices[i],
-                                               new_bit_width, new_sample_rate);
+                                               new_bit_width, new_sample_rate, format);
                 ret = true;
         }
     }
@@ -4691,8 +4694,8 @@
         format = DTS_HD;
         break;
     case AUDIO_FORMAT_PCM_16_BIT:
-    case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
-    case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+    case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+    case AUDIO_FORMAT_PCM_8_24_BIT:
         ALOGV("%s:PCM", __func__);
         format = LPCM;
         break;
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 5a41b07..2c60f3b 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1119,11 +1119,6 @@
     return 0;
 }
 
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
-    return false;
-}
-
 int platform_get_edid_info(void *platform __unused)
 {
    return -ENOSYS;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 385d20b..c0e4088 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -667,6 +667,8 @@
     {TO_NAME_INDEX(USECASE_INCALL_REC_DOWNLINK)},
     {TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
     {TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO)},
+    {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_FM)},
+    {TO_NAME_INDEX(USECASE_AUDIO_RECORD_FM_VIRTUAL)},
 };
 
 #define NO_COLS 2
@@ -757,6 +759,7 @@
 
 
 #define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
 #define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
 
 bool platform_send_gain_dep_cal(void *platform, int level) {
@@ -1064,7 +1067,6 @@
     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] = strdup("headphones");
     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");
 
@@ -3677,7 +3679,7 @@
     free(kv_pairs);
 }
 
-/* Delay in Us */
+/* Delay in Us, only to be used for PCM formats */
 int64_t platform_render_latency(audio_usecase_t usecase)
 {
     switch (usecase) {
@@ -3685,6 +3687,9 @@
             return DEEP_BUFFER_PLATFORM_DELAY;
         case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
             return LOW_LATENCY_PLATFORM_DELAY;
+        case USECASE_AUDIO_PLAYBACK_OFFLOAD:
+        case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
+             return PCM_OFFLOAD_PLATFORM_DELAY;
         default:
             return 0;
     }
@@ -3782,44 +3787,38 @@
 uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
 {
     uint32_t fragment_size = 0;
-    uint32_t bits_per_sample = 16;
+    uint32_t bytes_per_sample;
     uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
 
-    if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
-        bits_per_sample = 32;
-    }
+    bytes_per_sample = audio_bytes_per_sample(info->format);
 
     //duration is set to 40 ms worth of stereo data at 48Khz
     //with 16 bit per sample, modify this when the channel
     //configuration is different
     fragment_size = (pcm_offload_time
                      * info->sample_rate
-                     * (bits_per_sample >> 3)
+                     * bytes_per_sample
                      * popcount(info->channel_mask))/1000;
     if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
         fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
     else if(fragment_size > MAX_PCM_OFFLOAD_FRAGMENT_SIZE)
         fragment_size = MAX_PCM_OFFLOAD_FRAGMENT_SIZE;
+
     // To have same PCM samples for all channels, the buffer size requires to
     // be multiple of (number of channels * bytes per sample)
     // For writes to succeed, the buffer must be written at address which is multiple of 32
-    fragment_size = ALIGN(fragment_size, ((bits_per_sample >> 3)* popcount(info->channel_mask) * 32));
+    fragment_size = ALIGN(fragment_size, ((bytes_per_sample) * popcount(info->channel_mask) * 32));
 
     ALOGI("PCM offload Fragment size to %d bytes", fragment_size);
     return fragment_size;
 }
 
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
-    return OFFLOAD_USE_SMALL_BUFFER;
-}
-
 /*
  * configures afe with bit width and Sample Rate
  */
 static int platform_set_codec_backend_cfg(struct audio_device* adev,
-                         snd_device_t snd_device,
-                         unsigned int bit_width, unsigned int sample_rate)
+                         snd_device_t snd_device, unsigned int bit_width,
+                         unsigned int sample_rate, audio_format_t format)
 {
     int ret = 0;
     int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3845,13 +3844,16 @@
         }
 
         if (bit_width == 24) {
-            mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+            if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+                 mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+            else
+                 mixer_ctl_set_enum_by_string(ctl, "S24_LE");
         } else {
             mixer_ctl_set_enum_by_string(ctl, "S16_LE");
         }
         my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
-        ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
-              my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+        ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+              my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
     }
 
     /*
@@ -4053,11 +4055,13 @@
     int i, num_devices = 1;
     bool ret = false;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
+    audio_format_t format;
 
     backend_idx = platform_get_backend_index(snd_device);
 
     new_bit_width = usecase->stream.out->bit_width;
     new_sample_rate = usecase->stream.out->sample_rate;
+    format = usecase->stream.out->format;
 
     ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
           ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
@@ -4073,7 +4077,7 @@
         if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
                                              &new_bit_width, &new_sample_rate)) {
                 platform_set_codec_backend_cfg(adev, new_snd_devices[i],
-                                               new_bit_width, new_sample_rate);
+                                               new_bit_width, new_sample_rate, format);
                 ret = true;
         }
     }
@@ -4591,8 +4595,8 @@
         format = DTS_HD;
         break;
     case AUDIO_FORMAT_PCM_16_BIT:
-    case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
-    case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+    case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+    case AUDIO_FORMAT_PCM_8_24_BIT:
         ALOGV("%s:PCM", __func__);
         format = LPCM;
         break;
diff --git a/hal/platform_api.h b/hal/platform_api.h
index df80a0c..cb177b6 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -109,7 +109,6 @@
 struct audio_offload_info_t;
 uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
 uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
-bool platform_use_small_buffer(audio_offload_info_t* info);
 uint32_t platform_get_compress_passthrough_buffer_size(audio_offload_info_t* info);
 
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 24d8f74..839f1eb 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -460,80 +460,43 @@
         return false;
     }
 
-    char propValue[PROPERTY_VALUE_MAX];
-    bool pcmOffload = false;
-#ifdef PCM_OFFLOAD_ENABLED
-    if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD) {
-        bool prop_enabled = false;
-        if ((AUDIO_FORMAT_PCM_16_BIT_OFFLOAD == offloadInfo.format) &&
-               property_get("audio.offload.pcm.16bit.enable", propValue, NULL)) {
-            prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
-        }
-
-#ifdef PCM_OFFLOAD_ENABLED_24
-        if ((AUDIO_FORMAT_PCM_24_BIT_OFFLOAD == offloadInfo.format) &&
-               property_get("audio.offload.pcm.24bit.enable", propValue, NULL)) {
-            prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
-        }
-#endif
-
-        if (prop_enabled) {
-            ALOGI("PCM offload property is enabled");
-            pcmOffload = true;
-        }
-
-        if (!pcmOffload) {
-            ALOGD("system property not enabled for PCM offload format[%x]",offloadInfo.format);
-            return false;
-        }
+    //check if it's multi-channel AAC (includes sub formats) and FLAC format
+    if ((popcount(offloadInfo.channel_mask) > 2) &&
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+        ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
+        ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
+        return false;
     }
-#endif
-    if (!pcmOffload) {
-
-        bool compressedOffloadDisabled = property_get_bool("audio.offload.compress.disable", false);
-        if (compressedOffloadDisabled) {
-            ALOGI("compressed offload disabled by audio.offload.compress.disable=%d", compressedOffloadDisabled);
-            return false;
-        }
-
-        //check if it's multi-channel AAC (includes sub formats) and FLAC format
-        if ((popcount(offloadInfo.channel_mask) > 2) &&
-           (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
-            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
-               ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
-               return false;
-        }
 
 #ifdef AUDIO_EXTN_FORMATS_ENABLED
-            //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
-        if ((popcount(offloadInfo.channel_mask) > 2) &&
-            (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
-            (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) ||
-            (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) ||
-            (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) ||
-            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
-                ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
-            return false;
-        }
+    //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
+    if ((popcount(offloadInfo.channel_mask) > 2) &&
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) ||
+        ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
+            ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
+        return false;
+    }
 #endif
-        //TODO: enable audio offloading with video when ready
-        const bool allowOffloadWithVideo =
-                property_get_bool("audio.offload.video", false /* default_value */);
-        if (offloadInfo.has_video && !allowOffloadWithVideo) {
-            ALOGV("isOffloadSupported: has_video == true, returning false");
-            return false;
-        }
+    //TODO: enable audio offloading with video when ready
+    const bool allowOffloadWithVideo =
+            property_get_bool("audio.offload.video", false /* default_value */);
+    if (offloadInfo.has_video && !allowOffloadWithVideo) {
+        ALOGV("isOffloadSupported: has_video == true, returning false");
+        return false;
+    }
 
-        const bool allowOffloadStreamingWithVideo = property_get_bool("av.streaming.offload.enable",
-                                                                   false /*default value*/);
-        if(offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
-            ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
-            return false;
-        }
-
+    const bool allowOffloadStreamingWithVideo = property_get_bool("av.streaming.offload.enable",
+                                                               false /*default value*/);
+    if (offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
+        ALOGW("offload disabled by av.streaming.offload.enable %d",allowOffloadStreamingWithVideo);
+        return false;
     }
 
     //If duration is less than minimum value defined in property, return false
+    char propValue[PROPERTY_VALUE_MAX];
     if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
         if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
             ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
@@ -545,18 +508,18 @@
         //do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
         if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
-            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS) ||
-#ifdef AUDIO_EXTN_FORMATS_ENABLED
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
-            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))
+            return false;
+
+#ifdef AUDIO_EXTN_FORMATS_ENABLED
+        if (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
             ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) ||
-            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS) ||
-#endif
-            pcmOffload)
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
             return false;
-
+#endif
     }
 
     // Do not allow offloading if one non offloadable effect is enabled. This prevents from
@@ -1334,6 +1297,19 @@
     return false;
 }
 
+bool static isDirectPCMEnabled(int bitWidth)
+{
+    bool directPCMEnabled = false;
+    if (bitWidth == 24 || bitWidth == 32)
+        directPCMEnabled =
+             property_get_bool("audio.offload.pcm.24bit.enable", false);
+    else
+        directPCMEnabled =
+             property_get_bool("audio.offload.pcm.16bit.enable", false);
+
+    return directPCMEnabled;
+}
+
 status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr,
                                               audio_io_handle_t *output,
                                               audio_session_t session,
@@ -1349,23 +1325,21 @@
     audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
 
     bool offloadDisabled = property_get_bool("audio.offload.disable", false);
-    bool pcmOffloadEnabled = false;
+    uint32_t bitWidth = (audio_bytes_per_sample(format) * 8);
 
     if (offloadDisabled) {
         ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
     }
 
-    //read track offload property only if the global offload switch is off.
-    if (!offloadDisabled) {
-         pcmOffloadEnabled = property_get_bool("audio.offload.track.enable", false);
-    }
+    if (!offloadDisabled && (offloadInfo == NULL) &&
+        isDirectPCMEnabled(bitWidth) &&
+        (flags == AUDIO_OUTPUT_FLAG_NONE)) {
 
-    if (offloadInfo == NULL && pcmOffloadEnabled) {
         tOffloadInfo.sample_rate  = samplingRate;
         tOffloadInfo.channel_mask = channelMask;
         tOffloadInfo.format = format;
         tOffloadInfo.stream_type = *stream;
-        tOffloadInfo.bit_width = 16;    //hard coded for PCM_16
+        tOffloadInfo.bit_width = bitWidth;
         if (attr != NULL) {
             ALOGV("found attribute .. setting usage %d ", attr->usage);
             tOffloadInfo.usage = attr->usage;
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index d39a8b7..464bc0d 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -209,7 +209,7 @@
  * Interface from audio HAL
  */
 __attribute__ ((visibility ("default")))
-int offload_effects_bundle_hal_start_output(audio_io_handle_t output, int pcm_id)
+int offload_effects_bundle_hal_start_output(audio_io_handle_t output, int pcm_id, struct mixer *mixer)
 {
     int ret = 0;
     struct listnode *node;
@@ -245,19 +245,19 @@
     /* populate the mixer control to send offload parameters */
     snprintf(mixer_string, sizeof(mixer_string),
              "%s %d", "Audio Effects Config", out_ctxt->pcm_device_id);
-    out_ctxt->mixer = mixer_open(MIXER_CARD);
-    if (!out_ctxt->mixer) {
-        ALOGE("Failed to open mixer");
+
+    if (!mixer) {
+        ALOGE("Invalid mixer");
         out_ctxt->ctl = NULL;
         out_ctxt->ref_ctl = NULL;
         ret = -EINVAL;
         free(out_ctxt);
         goto exit;
     } else {
+        out_ctxt->mixer = mixer;
         out_ctxt->ctl = mixer_get_ctl_by_name(out_ctxt->mixer, mixer_string);
         if (!out_ctxt->ctl) {
             ALOGE("mixer_get_ctl_by_name failed");
-            mixer_close(out_ctxt->mixer);
             out_ctxt->mixer = NULL;
             ret = -EINVAL;
             free(out_ctxt);
@@ -314,9 +314,6 @@
             fx_ctxt->ops.stop(fx_ctxt, out_ctxt);
     }
 
-    if (out_ctxt->mixer)
-        mixer_close(out_ctxt->mixer);
-
     list_remove(&out_ctxt->outputs_list_node);
 
 #ifdef DTS_EAGLE