diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index 512e8ee..696a5d0 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -55,6 +55,8 @@
         <usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="35"/>
         <usecase name="USECASE_AUDIO_PLAYBACK_AFE_PROXY" type="out" id="6"/>
         <usecase name="USECASE_AUDIO_RECORD_AFE_PROXY" type="in" id="7"/>
+        <usecase name="USECASE_AUDIO_RECORD_LOW_LATENCY" type="in" id="17" />
+        <usecase name="USECASE_AUDIO_PLAYBACK_ULL" type="out" id="17" />
     </pcm_ids>
     <config_params>
         <param key="spkr_1_tz_name" value="wsatz.13"/>
diff --git a/configs/msmcobalt/audio_policy.conf b/configs/msmcobalt/audio_policy.conf
index dd827fe..a3b0c55 100644
--- a/configs/msmcobalt/audio_policy.conf
+++ b/configs/msmcobalt/audio_policy.conf
@@ -26,21 +26,21 @@
         sampling_rates 44100|48000
         channel_masks AUDIO_CHANNEL_OUT_STEREO
         formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE
+        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY
       }
       raw {
         sampling_rates 48000
         channel_masks AUDIO_CHANNEL_OUT_STEREO
         formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE
+        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW
       }
       deep_buffer {
          sampling_rates 44100|48000
          channel_masks AUDIO_CHANNEL_OUT_STEREO
          formats AUDIO_FORMAT_PCM_16_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_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE
+         devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
          flags AUDIO_OUTPUT_FLAG_DEEP_BUFFER
       }
       compress_passthrough {
@@ -61,14 +61,14 @@
         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|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_USB_DEVICE
+        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_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_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_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|AUDIO_DEVICE_OUT_USB_DEVICE
+        devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
         flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
       }
       incall_music {
@@ -108,14 +108,6 @@
     }
   }
   a2dp {
-    outputs {
-      a2dp {
-        sampling_rates 44100
-        channel_masks AUDIO_CHANNEL_OUT_STEREO
-        formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_ALL_A2DP
-      }
-    }
     inputs {
       a2dp {
         sampling_rates 44100|48000
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index 235c157..4336aa2 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -228,6 +228,18 @@
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
+                <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
 
                 <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -288,12 +300,37 @@
                        sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
+                <route type="mix" sink="BT A2DP Out"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                <route type="mix" sink="BT A2DP Headphones"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+                <route type="mix" sink="BT A2DP Speaker"
+                       sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
             </routes>
 
         </module>
 
-        <!-- A2dp Audio HAL -->
-        <xi:include href="a2dp_audio_policy_configuration.xml"/>
+        <!-- A2DP Audio HAL -->
+        <module name="a2dp" halVersion="2.0">
+            <mixPorts>
+                <mixPort name="a2dp input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+            </mixPorts>
+
+            <devicePorts>
+                <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+                </devicePort>
+            </devicePorts>
+
+            <routes>
+                <route type="mix" sink="a2dp input"
+                       sources="BT A2DP In"/>
+            </routes>
+        </module>
 
         <!-- Usb Audio HAL -->
         <xi:include href="usb_audio_policy_configuration.xml"/>
diff --git a/configs/msmcobalt/graphite_ipc_platform_info.xml b/configs/msmcobalt/graphite_ipc_platform_info.xml
new file mode 100644
index 0000000..f6775be
--- /dev/null
+++ b/configs/msmcobalt/graphite_ipc_platform_info.xml
@@ -0,0 +1,47 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2016, The Linux Foundation. All rights reserved.       -->
+<!---                                                                           -->
+<!--- Redistribution and use in source and binary forms, with or without        -->
+<!--- modification, are permitted provided that the following conditions are    -->
+<!--- met:                                                                      -->
+<!---     * Redistributions of source code must retain the above copyright      -->
+<!---       notice, this list of conditions and the following disclaimer.       -->
+<!---     * Redistributions in binary form must reproduce the above             -->
+<!---       copyright notice, this list of conditions and the following         -->
+<!---       disclaimer in the documentation and/or other materials provided     -->
+<!---       with the distribution.                                              -->
+<!---     * Neither the name of The Linux Foundation nor the names of its       -->
+<!---       contributors may be used to endorse or promote products derived     -->
+<!---       from this software without specific prior written permission.       -->
+<!---                                                                           -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED              -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF      -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT    -->
+<!--- ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS    -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR    -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF      -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR           -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,     -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE      -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN    -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                             -->
+<graphite_ipc_platform_info>
+    <no_of_glink_channels value="4">
+    </no_of_glink_channels>
+    <!-- channel 1 configuration -->
+    <glink_channel name="g_glink_ctrl" latency_in_us="5000"
+        no_of_intents="1" intents_size="1024">
+    </glink_channel>
+    <!-- channel 2 configuration -->
+    <glink_channel name="g_glink_persistent_data_ild" latency_in_us="30000"
+        no_of_intents="0">
+    </glink_channel>
+    <!-- channel 3 configuration -->
+    <glink_channel name="g_glink_persistent_data_nild" latency_in_us="30000"
+        no_of_intents="0">
+    </glink_channel>
+    <!-- channel 4 configuration -->
+    <glink_channel name="g_glink_audio_data" latency_in_us="10000"
+        no_of_intents="2" intents_size="4096, 4096">
+    </glink_channel>
+</graphite_ipc_platform_info>
diff --git a/configs/msmcobalt/mixer_paths_dtp.xml b/configs/msmcobalt/mixer_paths_dtp.xml
index 9bcf15b..a6c61e4 100644
--- a/configs/msmcobalt/mixer_paths_dtp.xml
+++ b/configs/msmcobalt/mixer_paths_dtp.xml
@@ -138,6 +138,8 @@
     <ctl name="SLIMBUS_0_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_0_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
@@ -617,7 +619,7 @@
     </path>
 
     <path name="audio-ull-playback">
-        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-protected">
@@ -634,11 +636,11 @@
     </path>
 
     <path name="audio-ull-playback hdmi">
-        <ctl name="HDMI Mixer MultiMedia3" value="1" />
+        <ctl name="HDMI Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco">
-        <ctl name="AUX_PCM_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="AUX_PCM_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco-wb">
@@ -652,7 +654,7 @@
     </path>
 
     <path name="audio-ull-playback afe-proxy">
-        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
     </path>
     <path name="multi-channel-playback hdmi">
         <ctl name="HDMI Mixer MultiMedia2" value="1" />
@@ -1103,11 +1105,11 @@
     </path>
 
     <path name="low-latency-record">
-      <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco">
-      <ctl name="MultiMedia5 Mixer AUX_PCM_UL_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer AUX_PCM_UL_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco-wb">
@@ -1116,11 +1118,11 @@
     </path>
 
     <path name="low-latency-record usb-headset-mic">
-        <ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="1" />
+        <ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="1" />
     </path>
 
     <path name="low-latency-record capture-fm">
-      <ctl name="MultiMedia5 Mixer TERT_MI2S_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer TERT_MI2S_TX" value="1" />
     </path>
 
     <path name="fm-virtual-record capture-fm">
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 860d014..3c6f642 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -548,6 +548,11 @@
     <ctl name="LSM8 MUX" value="None" />
     <ctl name="SLIMBUS_5_TX LSM Function" value="None" />
     <!-- listen end-->
+    <!-- split a2dp -->
+    <ctl name="BT SampleRate" value="KHZ_8" />
+    <ctl name="AFE Input Channels" value="Zero" />
+    <ctl name="SLIM7_RX ADM Channels" value="Zero" />
+    <!-- split a2dp end-->
 
     <!-- ADSP testfwk -->
     <ctl name="SLIMBUS_DL_HL Switch" value="0" />
@@ -614,7 +619,7 @@
     </path>
 
     <path name="deep-buffer-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="deep-buffer-playback bt-sco" />
     </path>
 
@@ -657,7 +662,7 @@
     </path>
 
     <path name="low-latency-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="low-latency-playback bt-sco" />
     </path>
 
@@ -689,7 +694,7 @@
     </path>
 
     <path name="audio-ull-playback">
-        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-protected">
@@ -697,7 +702,7 @@
     </path>
 
     <path name="audio-ull-playback headphones">
-        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-and-headphones">
@@ -706,15 +711,15 @@
     </path>
 
     <path name="audio-ull-playback hdmi">
-        <ctl name="HDMI Mixer MultiMedia3" value="1" />
+        <ctl name="HDMI Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco">
-        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-ull-playback bt-sco" />
     </path>
 
@@ -724,11 +729,11 @@
     </path>
 
     <path name="audio-ull-playback afe-proxy">
-        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback usb-headphones">
-        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="multi-channel-playback hdmi">
@@ -760,7 +765,7 @@
     </path>
 
     <path name="compress-offload-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback bt-sco" />
     </path>
 
@@ -808,7 +813,7 @@
     </path>
 
     <path name="compress-offload-playback2 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback2 bt-sco" />
     </path>
 
@@ -856,7 +861,7 @@
     </path>
 
     <path name="compress-offload-playback3 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback3 bt-sco" />
     </path>
 
@@ -904,7 +909,7 @@
     </path>
 
     <path name="compress-offload-playback4 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback4 bt-sco" />
     </path>
 
@@ -952,7 +957,7 @@
     </path>
 
     <path name="compress-offload-playback5 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback5 bt-sco" />
     </path>
 
@@ -1000,7 +1005,7 @@
     </path>
 
     <path name="compress-offload-playback6 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback6 bt-sco" />
     </path>
 
@@ -1048,7 +1053,7 @@
     </path>
 
     <path name="compress-offload-playback7 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback7 bt-sco" />
     </path>
 
@@ -1096,7 +1101,7 @@
     </path>
 
     <path name="compress-offload-playback8 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback8 bt-sco" />
     </path>
 
@@ -1144,7 +1149,7 @@
     </path>
 
     <path name="compress-offload-playback9 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback9 bt-sco" />
     </path>
 
@@ -1192,7 +1197,7 @@
     </path>
 
     <path name="audio-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record bt-sco" />
     </path>
 
@@ -1209,7 +1214,7 @@
     </path>
 
     <path name="audio-record-compress bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record-compress bt-sco" />
     </path>
 
@@ -1218,24 +1223,24 @@
     </path>
 
     <path name="low-latency-record">
-      <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco">
-      <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="low-latency-record bt-sco" />
     </path>
 
     <path name="low-latency-record usb-headset-mic">
-        <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="1" />
+        <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
     <path name="low-latency-record capture-fm">
-      <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
     </path>
 
     <path name="fm-virtual-record capture-fm">
@@ -1393,12 +1398,12 @@
     </path>
 
    <path name="hfp-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="hfp-sco" />
    </path>
 
     <path name="hfp-sco-wb headphones">
-        <ctl name="AUX PCM SampleRate" value="16000" />
+        <ctl name="AUX PCM SampleRate" value="KHZ_16" />
         <path name="hfp-sco headphones" />
     </path>
 
@@ -1419,7 +1424,7 @@
     </path>
 
     <path name="compress-voip-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-voip-call bt-sco" />
     </path>
 
@@ -1459,7 +1464,7 @@
     </path>
 
     <path name="vowlan-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="vowlan-call bt-sco" />
     </path>
 
@@ -1499,7 +1504,7 @@
     </path>
 
     <path name="voicemmode1-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode1-call bt-sco" />
     </path>
 
@@ -1539,7 +1544,7 @@
     </path>
 
     <path name="voicemmode2-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode2-call bt-sco" />
     </path>
 
@@ -2376,4 +2381,122 @@
         <ctl name="SLIMBUS_DL_HL Switch" value="1" />
     </path>
 
+    <path name="bt-a2dp">
+        <ctl name="BT SampleRate" value="KHZ_48" />
+        <ctl name="AFE Input Channels" value="Two" />
+        <ctl name="SLIM7_RX ADM Channels" value="Two" />
+    </path>
+
+    <path name="speaker-and-bt-a2dp">
+        <path name="speaker" />
+        <path name="bt-a2dp" />
+    </path>
+
+    <path name="deep-buffer-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="low-latency-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="compress-offload-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="audio-ull-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-a2dp">
+        <path name="deep-buffer-playback bt-a2dp" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-a2dp">
+        <path name="compress-offload-playback bt-a2dp" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-a2dp">
+        <path name="low-latency-playback bt-a2dp" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback2 bt-a2dp" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback3 bt-a2dp" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback4 bt-a2dp" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback5 bt-a2dp" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback6 bt-a2dp" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback7 bt-a2dp" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback8 bt-a2dp" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback9 bt-a2dp" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-a2dp">
+        <path name="audio-ull-playback bt-a2dp" />
+        <path name="audio-ull-playback" />
+    </path>
 </mixer>
diff --git a/configs/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index 1c92421..3a188f9 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -50,6 +50,7 @@
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="0" />
@@ -67,6 +68,9 @@
     <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="0" />
     <ctl name="MultiMedia1 Mixer SLIM_4_TX" value="0" />
     <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer SLIM_4_TX" value="0" />
+    <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
     <ctl name="HDMI Mixer MultiMedia1" value="0" />
     <ctl name="HDMI Mixer MultiMedia2" value="0" />
     <ctl name="HDMI Mixer MultiMedia3" value="0" />
@@ -95,6 +99,9 @@
     <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 MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
+    <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
@@ -122,6 +129,7 @@
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="0" />
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="0" />
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="0" />
+    <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="0" />
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="0" />
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="0" />
     <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="0" />
@@ -262,6 +270,12 @@
     <ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="0" />
     <!-- audio record compress end-->
 
+    <!-- split a2dp -->
+    <ctl name="BT SampleRate" value="KHZ_8" />
+    <ctl name="AFE Input Channels" value="Zero" />
+    <ctl name="SLIM7_RX ADM Channels" value="0" />
+    <!-- split a2dp end-->
+
     <!-- ADSP testfwk -->
     <ctl name="SLIMBUS_DL_HL Switch" value="0" />
     <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
@@ -286,6 +300,12 @@
     <ctl name="SpkrRight VISENSE Switch" value="0" />
     <ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
     <ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
+    <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="ZERO" />
+    <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="ZERO" />
+    <ctl name="VI_FEED_TX Channels" value="Two" />
+    <ctl name="AIF4_VI Mixer SPKR_VI_1" value="0" />
+    <ctl name="AIF4_VI Mixer SPKR_VI_2" value="0" />
+    <ctl name="SLIM_4_TX Format" value="UNPACKED" />
 
     <ctl name="AIF1_CAP Mixer SLIM TX0" value="0" />
     <ctl name="AIF1_CAP Mixer SLIM TX2" value="0" />
@@ -357,7 +377,7 @@
     </path>
 
     <path name="deep-buffer-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="deep-buffer-playback bt-sco" />
     </path>
 
@@ -400,7 +420,7 @@
     </path>
 
     <path name="low-latency-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="low-latency-playback bt-sco" />
     </path>
 
@@ -432,7 +452,7 @@
     </path>
 
     <path name="audio-ull-playback">
-        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-protected">
@@ -440,7 +460,7 @@
     </path>
 
     <path name="audio-ull-playback headphones">
-        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback speaker-and-headphones">
@@ -449,15 +469,15 @@
     </path>
 
     <path name="audio-ull-playback hdmi">
-        <ctl name="HDMI Mixer MultiMedia3" value="1" />
+        <ctl name="HDMI Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco">
-        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-ull-playback bt-sco" />
     </path>
 
@@ -467,11 +487,11 @@
     </path>
 
     <path name="audio-ull-playback afe-proxy">
-        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="audio-ull-playback usb-headphones">
-        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia3" value="1" />
+        <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
     </path>
 
     <path name="multi-channel-playback hdmi">
@@ -503,7 +523,7 @@
     </path>
 
     <path name="compress-offload-playback bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback bt-sco" />
     </path>
 
@@ -551,7 +571,7 @@
     </path>
 
     <path name="compress-offload-playback2 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback2 bt-sco" />
     </path>
 
@@ -599,7 +619,7 @@
     </path>
 
     <path name="compress-offload-playback3 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback3 bt-sco" />
     </path>
 
@@ -647,7 +667,7 @@
     </path>
 
     <path name="compress-offload-playback4 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback4 bt-sco" />
     </path>
 
@@ -695,7 +715,7 @@
     </path>
 
     <path name="compress-offload-playback5 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback5 bt-sco" />
     </path>
 
@@ -743,7 +763,7 @@
     </path>
 
     <path name="compress-offload-playback6 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback6 bt-sco" />
     </path>
 
@@ -791,7 +811,7 @@
     </path>
 
     <path name="compress-offload-playback7 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback7 bt-sco" />
     </path>
 
@@ -839,7 +859,7 @@
     </path>
 
     <path name="compress-offload-playback8 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback8 bt-sco" />
     </path>
 
@@ -887,7 +907,7 @@
     </path>
 
     <path name="compress-offload-playback9 bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-offload-playback9 bt-sco" />
     </path>
 
@@ -935,7 +955,7 @@
     </path>
 
     <path name="audio-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record bt-sco" />
     </path>
 
@@ -952,7 +972,7 @@
     </path>
 
     <path name="audio-record-compress bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="audio-record-compress bt-sco" />
     </path>
 
@@ -961,24 +981,24 @@
     </path>
 
     <path name="low-latency-record">
-      <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco">
-      <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
     </path>
 
     <path name="low-latency-record bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="low-latency-record bt-sco" />
     </path>
 
     <path name="low-latency-record usb-headset-mic">
-        <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="1" />
+        <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
     </path>
 
     <path name="low-latency-record capture-fm">
-      <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="1" />
+      <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
     </path>
 
     <path name="fm-virtual-record capture-fm">
@@ -1150,7 +1170,7 @@
     </path>
 
     <path name="compress-voip-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="compress-voip-call bt-sco" />
     </path>
 
@@ -1190,7 +1210,7 @@
     </path>
 
     <path name="voicemmode1-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode1-call bt-sco" />
     </path>
 
@@ -1230,7 +1250,7 @@
     </path>
 
     <path name="voicemmode2-call bt-sco-wb">
-        <ctl name="BT_SCO SampleRate" value="16000" />
+        <ctl name="BT SampleRate" value="KHZ_16" />
         <path name="voicemmode2-call bt-sco" />
     </path>
 
@@ -1364,11 +1384,21 @@
     </path>
 
     <path name="speaker-protected">
+        <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+        <ctl name="AIF4_VI Mixer SPKR_VI_2" value="1" />
+        <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
         <path name="speaker" />
+        <ctl name="VI_FEED_TX Channels" value="Two" />
+        <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
+        <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="SLIM4_TX" />
     </path>
 
     <path name="voice-speaker-protected">
+        <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+        <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
         <path name="speaker-mono" />
+        <ctl name="VI_FEED_TX Channels" value="One" />
+        <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
     </path>
 
     <path name="vi-feedback">
@@ -1715,4 +1745,122 @@
         <ctl name="SLIMBUS_DL_HL Switch" value="1" />
     </path>
 
+    <path name="bt-a2dp">
+        <ctl name="BT SampleRate" value="KHZ_48" />
+        <ctl name="AFE Input Channels" value="Two" />
+        <ctl name="SLIM7_RX ADM Channels" value="2" />
+    </path>
+
+    <path name="speaker-and-bt-a2dp">
+        <path name="speaker" />
+        <path name="bt-a2dp" />
+    </path>
+
+    <path name="deep-buffer-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+    </path>
+
+    <path name="low-latency-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+    </path>
+
+    <path name="compress-offload-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+    </path>
+
+    <path name="compress-offload-playback2 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+    </path>
+
+    <path name="compress-offload-playback3 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+    </path>
+
+    <path name="compress-offload-playback4 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+    </path>
+
+    <path name="compress-offload-playback5 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+    </path>
+
+    <path name="compress-offload-playback6 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+    </path>
+
+    <path name="compress-offload-playback7 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+    </path>
+
+    <path name="compress-offload-playback8 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+    </path>
+
+    <path name="compress-offload-playback9 bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+    </path>
+
+    <path name="audio-ull-playback bt-a2dp">
+        <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+    </path>
+
+    <path name="deep-buffer-playback speaker-and-bt-a2dp">
+        <path name="deep-buffer-playback bt-a2dp" />
+        <path name="deep-buffer-playback" />
+    </path>
+
+    <path name="compress-offload-playback speaker-and-bt-a2dp">
+        <path name="compress-offload-playback bt-a2dp" />
+        <path name="compress-offload-playback" />
+    </path>
+
+    <path name="low-latency-playback speaker-and-bt-a2dp">
+        <path name="low-latency-playback bt-a2dp" />
+        <path name="low-latency-playback" />
+    </path>
+
+    <path name="compress-offload-playback2 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback2 bt-a2dp" />
+        <path name="compress-offload-playback2" />
+    </path>
+
+    <path name="compress-offload-playback3 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback3 bt-a2dp" />
+        <path name="compress-offload-playback3" />
+    </path>
+
+    <path name="compress-offload-playback4 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback4 bt-a2dp" />
+        <path name="compress-offload-playback4" />
+    </path>
+
+    <path name="compress-offload-playback5 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback5 bt-a2dp" />
+        <path name="compress-offload-playback5" />
+    </path>
+
+    <path name="compress-offload-playback6 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback6 bt-a2dp" />
+        <path name="compress-offload-playback6" />
+    </path>
+
+    <path name="compress-offload-playback7 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback7 bt-a2dp" />
+        <path name="compress-offload-playback7" />
+    </path>
+
+    <path name="compress-offload-playback8 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback8 bt-a2dp" />
+        <path name="compress-offload-playback8" />
+    </path>
+
+    <path name="compress-offload-playback9 speaker-and-bt-a2dp">
+        <path name="compress-offload-playback9 bt-a2dp" />
+        <path name="compress-offload-playback9" />
+    </path>
+
+    <path name="audio-ull-playback speaker-and-bt-a2dp">
+        <path name="audio-ull-playback bt-a2dp" />
+        <path name="audio-ull-playback" />
+    </path>
 </mixer>
diff --git a/configs/msmcobalt/msmcobalt.mk b/configs/msmcobalt/msmcobalt.mk
index ff73287..aaf9db5 100644
--- a/configs/msmcobalt/msmcobalt.mk
+++ b/configs/msmcobalt/msmcobalt.mk
@@ -52,6 +52,7 @@
 AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
 AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
 AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
+AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
 ##AUDIO_FEATURE_FLAGS
 
 #Audio Specific device overlays
@@ -79,6 +80,7 @@
     hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths.xml:system/etc/sound_trigger_mixer_paths.xml \
     hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths_wcd9330.xml:system/etc/sound_trigger_mixer_paths_wcd9330.xml \
     hardware/qcom/audio/configs/msmcobalt/sound_trigger_platform_info.xml:system/etc/sound_trigger_platform_info.xml \
+    hardware/qcom/audio/configs/msmcobalt/graphite_ipc_platform_info.xml:system/etc/graphite_ipc_platform_info.xml \
     hardware/qcom/audio/configs/msmcobalt/audio_platform_info.xml:system/etc/audio_platform_info.xml
 
 #XML Audio configuration files
@@ -189,3 +191,7 @@
 #flac sw decoder 24 bit decode capability
 PRODUCT_PROPERTY_OVERRIDES += \
 flac.sw.decoder.24bit.support=true
+
+#split a2dp DSP supported encoder list
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.bt.a2dp_offload_cap=sbc-aptx
diff --git a/hal/Android.mk b/hal/Android.mk
index 83787e3..adee78b 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -243,6 +243,11 @@
     LOCAL_SRC_FILES += audio_extn/source_track.c
 endif
 
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SPLIT_A2DP)),true)
+    LOCAL_CFLAGS += -DSPLIT_A2DP_ENABLED
+    LOCAL_SRC_FILES += audio_extn/a2dp.c
+endif
+
 LOCAL_SHARED_LIBRARIES := \
 	liblog \
 	libcutils \
@@ -251,6 +256,7 @@
 	libaudioroute \
 	libdl \
 	libaudioutils \
+	libhardware \
 	libexpat
 
 LOCAL_C_INCLUDES += \
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
new file mode 100644
index 0000000..414fc79
--- /dev/null
+++ b/hal/audio_extn/a2dp.c
@@ -0,0 +1,733 @@
+/*
+* Copyright (c) 2015-16, The Linux Foundation. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions are
+* met:
+*     * Redistributions of source code must retain the above copyright
+*       notice, this list of conditions and the following disclaimer.
+*     * Redistributions in binary form must reproduce the above
+*       copyright notice, this list of conditions and the following
+*       disclaimer in the documentation and/or other materials provided
+*       with the distribution.
+*     * Neither the name of The Linux Foundation nor the names of its
+*       contributors may be used to endorse or promote products derived
+*       from this software without specific prior written permission.
+*
+* THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+* ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+* BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+* OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+#define LOG_TAG "split_a2dp"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+#include <errno.h>
+#include <cutils/log.h>
+#include <dlfcn.h>
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <stdlib.h>
+#include <cutils/str_parms.h>
+#include <hardware/audio.h>
+#include <hardware/hardware.h>
+#include <cutils/properties.h>
+
+#ifdef SPLIT_A2DP_ENABLED
+#define AUDIO_PARAMETER_A2DP_STARTED "A2dpStarted"
+#define BT_IPC_LIB_NAME  "libbthost_if.so"
+#define ENC_MEDIA_FMT_NONE                                     0
+#define ENC_MEDIA_FMT_AAC                                  0x00010DA6
+#define ENC_MEDIA_FMT_APTX                                 0x000131ff
+#define ENC_MEDIA_FMT_APTX_HD                              0x00013200
+#define ENC_MEDIA_FMT_SBC                                  0x00010BF2
+#define MEDIA_FMT_AAC_AOT_LC                               2
+#define MEDIA_FMT_AAC_AOT_SBR                              5
+#define MEDIA_FMT_AAC_AOT_PS                               29
+#define MEDIA_FMT_AAC_FORMAT_FLAG_ADTS                     0
+#define MEDIA_FMT_AAC_FORMAT_FLAG_RAW                      3
+#define PCM_CHANNEL_L                                      1
+#define PCM_CHANNEL_R                                      2
+#define PCM_CHANNEL_C                                      3
+#define MEDIA_FMT_SBC_CHANNEL_MODE_MONO                    1
+#define MEDIA_FMT_SBC_CHANNEL_MODE_STEREO                  2
+#define MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO               8
+#define MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO            9
+#define MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS           0
+#define MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR                1
+#define MIXER_ENC_CONFIG_BLOCK     "SLIM_7_RX Encoder Config"
+#define MIXER_ENC_FMT_SBC          "SBC"
+#define MIXER_ENC_FMT_AAC          "AAC"
+#define MIXER_ENC_FMT_APTX         "APTX"
+#define MIXER_ENC_FMT_APTXHD       "APTXHD"
+#define MIXER_ENC_FMT_NONE         "NONE"
+
+
+typedef int (*audio_stream_open_t)(void);
+typedef int (*audio_stream_close_t)(void);
+typedef int (*audio_start_stream_t)(void);
+typedef int (*audio_stop_stream_t)(void);
+typedef int (*audio_suspend_stream_t)(void);
+typedef void (*audio_handoff_triggered_t)(void);
+typedef void (*clear_a2dpsuspend_flag_t)(void);
+typedef void * (*audio_get_codec_config_t)(uint8_t *multicast_status,uint8_t *num_dev,
+                               audio_format_t *codec_type);
+
+enum A2DP_STATE {
+    A2DP_STATE_CONNECTED,
+    A2DP_STATE_STARTED,
+    A2DP_STATE_STOPPED,
+    A2DP_STATE_DISCONNECTED,
+};
+
+/* structure used to  update a2dp state machine
+ * to communicate IPC library
+ * to store DSP encoder configuration information
+ */
+struct a2dp_data {
+    struct audio_device *adev;
+    void *bt_lib_handle;
+    audio_stream_open_t audio_stream_open;
+    audio_stream_close_t audio_stream_close;
+    audio_start_stream_t audio_start_stream;
+    audio_stop_stream_t audio_stop_stream;
+    audio_suspend_stream_t audio_suspend_stream;
+    audio_handoff_triggered_t audio_handoff_triggered;
+    clear_a2dpsuspend_flag_t clear_a2dpsuspend_flag;
+    audio_get_codec_config_t audio_get_codec_config;
+    enum A2DP_STATE bt_state;
+    audio_format_t bt_encoder_format;
+    void *enc_config_data;
+    bool a2dp_started;
+    bool a2dp_suspended;
+    int  a2dp_total_active_session_request;
+    bool is_a2dp_offload_supported;
+    bool is_handoff_in_progress;
+};
+
+struct a2dp_data a2dp;
+
+/* START of DSP configurable structures
+ * These values should match with DSP interface defintion
+ */
+
+/* AAC encoder configuration structure. */
+typedef struct aac_enc_cfg_t aac_enc_cfg_t;
+
+/* supported enc_mode are AAC_LC, AAC_SBR, AAC_PS
+ * supported aac_fmt_flag are ADTS/RAW
+ * supported channel_cfg are Native mode, Mono , Stereo
+ */
+struct aac_enc_cfg_t {
+    uint32_t      enc_format;
+    uint32_t      bit_rate;
+    uint32_t      enc_mode;
+    uint16_t      aac_fmt_flag;
+    uint32_t      channel_cfg;
+    uint32_t      sample_rate;
+} ;
+
+/* SBC encoder configuration structure. */
+typedef struct sbc_enc_cfg_t sbc_enc_cfg_t;
+
+/* supported num_subbands are 4/8
+ * supported blk_len are 4, 8, 12, 16
+ * supported channel_mode are MONO, STEREO, DUAL_MONO, JOINT_STEREO
+ * supported alloc_method are LOUNDNESS/SNR
+ * supported bit_rate for mono channel is max 320kbps
+ * supported bit rate for stereo channel is max 512 kbps
+ */
+struct sbc_enc_cfg_t{
+    uint32_t      enc_format;
+    uint32_t      num_subbands;
+    uint32_t      blk_len;
+    uint32_t      channel_mode;
+    uint32_t      alloc_method;
+    uint32_t      bit_rate;
+    uint32_t      sample_rate;
+};
+
+
+/* supported num_channels are Mono/Stereo
+ * supported channel_mapping for mono is CHANNEL_C
+ * supported channel mapping for stereo is CHANNEL_L and CHANNEL_R
+ * custom size and reserved are not used(for future enhancement)
+  */
+struct custom_enc_cfg_aptx_t
+{
+    uint32_t      enc_format;
+    uint32_t      sample_rate;
+    uint16_t      num_channels;
+    uint16_t      reserved;
+    uint8_t       channel_mapping[8];
+    uint32_t      custom_size;
+};
+
+/*********** END of DSP configurable structures ********************/
+
+/* API to identify DSP encoder captabilities */
+static void a2dp_offload_codec_cap_parser(char *value)
+{
+    char *tok = NULL;
+
+    tok = strtok(value, "-");
+    while (tok != NULL) {
+        if (strcmp(tok, "sbc") == 0) {
+            ALOGD("%s: SBC offload supported\n",__func__);
+            a2dp.is_a2dp_offload_supported = true;
+            break;
+        } else if (strcmp(tok, "aptx") == 0) {
+            ALOGD("%s: aptx offload supported\n",__func__);
+            a2dp.is_a2dp_offload_supported = true;
+            break;
+        }
+        tok = strtok(NULL,"-");
+    };
+}
+
+static void update_offload_codec_capabilities()
+{
+    char value[PROPERTY_VALUE_MAX] = {'\0'};
+
+    property_get("persist.bt.a2dp_offload_cap", value, "false");
+    ALOGD("get_offload_codec_capabilities = %s",value);
+    a2dp.is_a2dp_offload_supported =
+            property_get_bool("persist.bt.a2dp_offload_cap", false);
+    if (strcmp(value, "false") != 0)
+        a2dp_offload_codec_cap_parser(value);
+    ALOGD("%s: codec cap = %s",__func__,value);
+}
+
+/* API to open BT IPC library to start IPC communication */
+static void open_a2dp_output()
+{
+    int ret = 0;
+
+    ALOGD(" Open A2DP output start ");
+    if (a2dp.bt_lib_handle == NULL){
+        ALOGD(" Requesting for BT lib handle");
+        a2dp.bt_lib_handle = dlopen(BT_IPC_LIB_NAME, RTLD_NOW);
+
+        if (a2dp.bt_lib_handle == NULL) {
+            ALOGE("%s: DLOPEN failed for %s", __func__, BT_IPC_LIB_NAME);
+            ret = -ENOSYS;
+            goto init_fail;
+        } else {
+            a2dp.audio_stream_open = (audio_stream_open_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_stream_open");
+            a2dp.audio_start_stream = (audio_start_stream_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_start_stream");
+            a2dp.audio_get_codec_config = (audio_get_codec_config_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_get_codec_config");
+            a2dp.audio_suspend_stream = (audio_suspend_stream_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_suspend_stream");
+            a2dp.audio_handoff_triggered = (audio_handoff_triggered_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_handoff_triggered");
+            a2dp.clear_a2dpsuspend_flag = (clear_a2dpsuspend_flag_t)
+                          dlsym(a2dp.bt_lib_handle, "clear_a2dpsuspend_flag");
+            a2dp.audio_stop_stream = (audio_stop_stream_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_stop_stream");
+            a2dp.audio_stream_close = (audio_stream_close_t)
+                          dlsym(a2dp.bt_lib_handle, "audio_stream_close");
+        }
+    }
+
+    if (a2dp.bt_lib_handle && a2dp.audio_stream_open) {
+        if (a2dp.bt_state == A2DP_STATE_DISCONNECTED) {
+            ALOGD("calling BT stream open");
+            ret = a2dp.audio_stream_open();
+            if(ret != 0) {
+                ALOGE("Failed to open output stream for a2dp: status %d", ret);
+                goto init_fail;
+            }
+            a2dp.bt_state = A2DP_STATE_CONNECTED;
+        } else {
+            ALOGD("Called a2dp open with improper state, Ignoring request state %d", a2dp.bt_state);
+        }
+    } else {
+        ALOGE("a2dp handle is not identified, Ignoring open request");
+        a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+        goto init_fail;
+    }
+
+init_fail:
+    if(ret != 0 && (a2dp.bt_lib_handle != NULL)) {
+        dlclose(a2dp.bt_lib_handle);
+        a2dp.bt_lib_handle = NULL;
+    }
+}
+
+static int close_a2dp_output()
+{
+    ALOGV("%s\n",__func__);
+    if (!(a2dp.bt_lib_handle && a2dp.audio_stream_close)) {
+        ALOGE("a2dp handle is not identified, Ignoring close request");
+        return -ENOSYS;
+    }
+    if ((a2dp.bt_state == A2DP_STATE_CONNECTED) &&
+        (a2dp.bt_state == A2DP_STATE_STARTED) &&
+        (a2dp.bt_state == A2DP_STATE_STOPPED)) {
+        ALOGD("calling BT stream close");
+        if(a2dp.audio_stream_close() == false)
+            ALOGE("failed close a2dp control path from BT library");
+        a2dp.a2dp_started = false;
+        a2dp.a2dp_total_active_session_request = 0;
+        a2dp.a2dp_suspended = false;
+        a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+        a2dp.enc_config_data = NULL;
+        a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+    } else {
+        ALOGD("close a2dp called in improper state");
+        a2dp.a2dp_started = false;
+        a2dp.a2dp_total_active_session_request = 0;
+        a2dp.a2dp_suspended = false;
+        a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+        a2dp.enc_config_data = NULL;
+        a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+    }
+
+    return 0;
+}
+
+/* API to configure SBC DSP encoder */
+bool configure_sbc_enc_format(audio_sbc_encoder_config *sbc_bt_cfg)
+{
+    struct mixer_ctl *ctl_enc_data;
+    struct sbc_enc_cfg_t sbc_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(sbc_bt_cfg == NULL)
+        return false;
+
+   ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    a2dp.bt_encoder_format = AUDIO_FORMAT_SBC;
+    memset(&sbc_dsp_cfg, 0x0, sizeof(struct sbc_enc_cfg_t));
+    sbc_dsp_cfg.enc_format = ENC_MEDIA_FMT_SBC;
+    sbc_dsp_cfg.num_subbands = sbc_bt_cfg->subband;
+    sbc_dsp_cfg.blk_len = sbc_bt_cfg->blk_len;
+    switch(sbc_bt_cfg->channels) {
+        case 0:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_MONO;
+            break;
+        case 1:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO;
+            break;
+        case 3:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO;
+            break;
+        case 2:
+        default:
+            sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_STEREO;
+            break;
+    }
+    if (sbc_bt_cfg->alloc)
+        sbc_dsp_cfg.alloc_method = MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS;
+    else
+        sbc_dsp_cfg.alloc_method = MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR;
+    sbc_dsp_cfg.bit_rate = sbc_bt_cfg->bitrate;
+    sbc_dsp_cfg.sample_rate = sbc_bt_cfg->sampling_rate;
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&sbc_dsp_cfg,
+                                    sizeof(struct sbc_enc_cfg_t));
+    if (ret != 0) {
+        ALOGE("%s: failed to set SBC encoder config", __func__);
+        is_configured = false;
+    } else
+        is_configured = true;
+fail:
+    return is_configured;
+}
+
+/* API to configure APTX DSP encoder */
+bool configure_aptx_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
+{
+    struct mixer_ctl *ctl_enc_data;
+    struct custom_enc_cfg_aptx_t aptx_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(aptx_bt_cfg == NULL)
+        return false;
+
+    ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    a2dp.bt_encoder_format = AUDIO_FORMAT_APTX;
+    memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_aptx_t));
+    aptx_dsp_cfg.enc_format = ENC_MEDIA_FMT_APTX;
+    aptx_dsp_cfg.sample_rate = aptx_bt_cfg->sampling_rate;
+    aptx_dsp_cfg.num_channels = aptx_bt_cfg->channels;
+    switch(aptx_dsp_cfg.num_channels) {
+        case 1:
+            aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_C;
+            break;
+        case 2:
+        default:
+            aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+            aptx_dsp_cfg.channel_mapping[1] = PCM_CHANNEL_R;
+            break;
+    }
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+                              sizeof(struct custom_enc_cfg_aptx_t));
+    if (ret != 0) {
+        ALOGE("%s: Failed to set APTX encoder config", __func__);
+        is_configured = false;
+        goto fail;
+    }
+    is_configured = true;
+fail:
+    return is_configured;
+}
+
+/* API to configure APTX HD DSP encoder
+ * TODO: ADD 24 bit configuration support
+ */
+bool configure_aptx_hd_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
+{
+    struct mixer_ctl *ctl_enc_data;
+    struct custom_enc_cfg_aptx_t aptx_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(aptx_bt_cfg == NULL)
+        return false;
+
+    ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    a2dp.bt_encoder_format = AUDIO_FORMAT_APTX_HD;
+    memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_aptx_t));
+    aptx_dsp_cfg.enc_format = ENC_MEDIA_FMT_APTX_HD;
+    aptx_dsp_cfg.sample_rate = aptx_bt_cfg->sampling_rate;
+    aptx_dsp_cfg.num_channels = aptx_bt_cfg->channels;
+    switch(aptx_dsp_cfg.num_channels) {
+        case 1:
+            aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_C;
+            break;
+        case 2:
+        default:
+            aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+            aptx_dsp_cfg.channel_mapping[1] = PCM_CHANNEL_R;
+            break;
+    }
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+                              sizeof(struct custom_enc_cfg_aptx_t));
+    if (ret != 0) {
+        ALOGE("%s: Failed to set APTX HD encoder config", __func__);
+        is_configured = false;
+        goto fail;
+    }
+    is_configured = true;
+fail:
+    return is_configured;
+}
+
+/* API to configure AAC DSP encoder */
+bool configure_aac_enc_format(audio_aac_encoder_config *aac_bt_cfg)
+{
+    struct mixer_ctl *ctl_enc_data;
+    struct aac_enc_cfg_t aac_dsp_cfg;
+    bool is_configured = false;
+    int ret = 0;
+
+    if(aac_bt_cfg == NULL)
+        return false;
+
+    ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+    if (!ctl_enc_data) {
+        ALOGE(" ERROR  a2dp encoder CONFIG data mixer control not identifed");
+        is_configured = false;
+        goto fail;
+    }
+    a2dp.bt_encoder_format = AUDIO_FORMAT_AAC;
+    memset(&aac_dsp_cfg, 0x0, sizeof(struct aac_enc_cfg_t));
+    aac_dsp_cfg.enc_format = ENC_MEDIA_FMT_AAC;
+    aac_dsp_cfg.bit_rate = aac_bt_cfg->bitrate;
+    switch(aac_bt_cfg->enc_mode) {
+        case 0:
+            aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_LC;
+            break;
+        case 2:
+            aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_PS;
+            break;
+        case 1:
+        default:
+            aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_SBR;
+            break;
+    }
+    if (aac_bt_cfg->format_flag)
+        aac_dsp_cfg.aac_fmt_flag = MEDIA_FMT_AAC_FORMAT_FLAG_RAW;
+    else
+        aac_dsp_cfg.aac_fmt_flag = MEDIA_FMT_AAC_FORMAT_FLAG_ADTS;
+    aac_dsp_cfg.channel_cfg = aac_bt_cfg->channels;
+    ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aac_dsp_cfg,
+                              sizeof(struct aac_enc_cfg_t));
+    if (ret != 0) {
+        ALOGE("%s: failed to set SBC encoder config", __func__);
+        is_configured = false;
+    } else
+        is_configured = true;
+fail:
+    return is_configured;
+}
+
+bool configure_a2dp_encoder_format()
+{
+    void *codec_info = NULL;
+    uint8_t multi_cast = 0, num_dev = 1;
+    audio_format_t codec_type = AUDIO_FORMAT_INVALID;
+    bool is_configured = false;
+
+    if (!a2dp.audio_get_codec_config) {
+        ALOGE(" a2dp handle is not identified, ignoring a2dp encoder config");
+        return false;
+    }
+    ALOGD("configure_a2dp_encoder_format start");
+    codec_info = a2dp.audio_get_codec_config(&multi_cast, &num_dev,
+                               &codec_type);
+
+    switch(codec_type) {
+        case AUDIO_FORMAT_SBC:
+            ALOGD(" Received SBC encoder supported BT device");
+            is_configured =
+               configure_sbc_enc_format((audio_sbc_encoder_config *)codec_info);
+            break;
+        case AUDIO_FORMAT_APTX:
+            ALOGD(" Received APTX encoder supported BT device");
+            is_configured =
+              configure_aptx_enc_format((audio_aptx_encoder_config *)codec_info);
+            break;
+        case AUDIO_FORMAT_APTX_HD:
+            ALOGD(" Received APTX HD encoder supported BT device");
+            is_configured =
+             configure_aptx_hd_enc_format((audio_aptx_encoder_config *)codec_info);
+            break;
+        case AUDIO_FORMAT_AAC:
+            ALOGD(" Received AAC encoder supported BT device");
+            is_configured =
+              configure_aac_enc_format((audio_aac_encoder_config *)codec_info);
+            break;
+        default:
+            ALOGD(" Received Unsupported encoder formar");
+            is_configured = false;
+            break;
+    }
+    return is_configured;
+}
+
+int audio_extn_a2dp_start_playback()
+{
+    int ret = 0;
+
+    ALOGD("audio_extn_a2dp_start_playback start");
+
+    if(!(a2dp.bt_lib_handle && a2dp.audio_start_stream
+       && a2dp.audio_get_codec_config)) {
+        ALOGE("a2dp handle is not identified, Ignoring start request");
+        return -ENOSYS;
+    }
+
+    if(a2dp.a2dp_suspended == true) {
+        //session will be restarted after suspend completion
+        ALOGD("a2dp start requested during suspend state");
+        a2dp.a2dp_total_active_session_request++;
+        return 0;
+    }
+
+    if (!a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+        ALOGD("calling BT module stream start");
+        /* This call indicates BT IPC lib to start playback */
+        ret =  a2dp.audio_start_stream();
+        ALOGE("BT controller start return = %d",ret);
+        if (ret != 0 ) {
+           ALOGE("BT controller start failed");
+           a2dp.a2dp_started = false;
+           ret = -ETIMEDOUT;
+        } else {
+           if(configure_a2dp_encoder_format() == true) {
+                a2dp.a2dp_started = true;
+                ret = 0;
+                ALOGD("Start playback successful to BT library");
+           } else {
+                ALOGD(" unable to configure DSP encoder");
+                a2dp.a2dp_started = false;
+                ret = -ETIMEDOUT;
+           }
+        }
+    }
+
+    if (a2dp.a2dp_started)
+        a2dp.a2dp_total_active_session_request++;
+
+    ALOGD("start A2DP playback total active sessions :%d",
+          a2dp.a2dp_total_active_session_request);
+    return ret;
+}
+
+int audio_extn_a2dp_stop_playback()
+{
+    int ret =0;
+
+    ALOGV("audio_extn_a2dp_stop_playback start");
+    if(!(a2dp.bt_lib_handle && a2dp.audio_stop_stream)) {
+        ALOGE("a2dp handle is not identified, Ignoring start request");
+        return -ENOSYS;
+    }
+
+    if(a2dp.a2dp_suspended == true) {
+        ALOGD("STOP playback is called during suspend state");
+
+        // sessions are already closed during suspend, just update active sessions counts
+         if(a2dp.a2dp_total_active_session_request > 0)
+            a2dp.a2dp_total_active_session_request--;
+         return 0;
+    }
+    if (a2dp.a2dp_started && (a2dp.a2dp_total_active_session_request > 0))
+        a2dp.a2dp_total_active_session_request--;
+
+    if ( a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+        struct mixer_ctl *ctl_enc_config;
+        struct sbc_enc_cfg_t dummy_reset_config;
+
+        ALOGV("calling BT module stream stop");
+        ret = a2dp.audio_stop_stream();
+        if (ret < 0)
+            ALOGE("stop stream to BT IPC lib failed");
+        else
+            ALOGV("stop steam to BT IPC lib successful");
+        a2dp.is_handoff_in_progress = false;
+
+         memset(&dummy_reset_config, 0x0, sizeof(struct sbc_enc_cfg_t));
+        ctl_enc_config = mixer_get_ctl_by_name(a2dp.adev->mixer,
+                                               MIXER_ENC_CONFIG_BLOCK);
+        if (!ctl_enc_config) {
+            ALOGE(" ERROR  a2dp encoder format mixer control not identifed");
+        } else {
+            ret = mixer_ctl_set_array(ctl_enc_config, (void *)&dummy_reset_config,
+                                            sizeof(struct sbc_enc_cfg_t));
+             a2dp.bt_encoder_format = ENC_MEDIA_FMT_NONE;
+        }
+    }
+    if(!a2dp.a2dp_total_active_session_request)
+       a2dp.a2dp_started = false;
+    ALOGD("Stop A2DP playback total active sessions :%d",
+          a2dp.a2dp_total_active_session_request);
+    return 0;
+}
+
+void audio_extn_a2dp_set_parameters(struct str_parms *parms)
+{
+     int ret, val;
+     char value[32]={0};
+
+     if(a2dp.is_a2dp_offload_supported == false) {
+        ALOGV("no supported encoders identified,ignoring a2dp setparam");
+        return;
+     }
+
+     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value,
+                            sizeof(value));
+     if( ret >= 0) {
+         val = atoi(value);
+         if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
+             ALOGV("Received device connect request for A2DP");
+             open_a2dp_output();
+         }
+         goto param_handled;
+     }
+
+     ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value,
+                         sizeof(value));
+
+     if( ret >= 0) {
+         val = atoi(value);
+         if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
+             ALOGV("Received device dis- connect request");
+             close_a2dp_output();
+         }
+         goto param_handled;
+     }
+
+     ret = str_parms_get_str(parms, "A2dpSuspended", value, sizeof(value));
+     if (ret >= 0) {
+         if (a2dp.bt_lib_handle && (a2dp.bt_state != A2DP_STATE_DISCONNECTED) ) {
+             if ((!strncmp(value,"true",sizeof(value)))) {
+                ALOGD("Setting a2dp to suspend state");
+                int active_sessions = a2dp.a2dp_total_active_session_request, count = 0;
+                //Force close all active sessions on suspend (if any)
+                for(count  = 0; count< active_sessions; count ++)
+                    audio_extn_a2dp_stop_playback();
+                a2dp.a2dp_total_active_session_request = active_sessions;
+                a2dp.a2dp_suspended = true;
+
+                if(a2dp.audio_suspend_stream)
+                   a2dp.audio_suspend_stream();
+            } else if (a2dp.a2dp_suspended == true) {
+                ALOGD("Resetting a2dp suspend state");
+                if(a2dp.clear_a2dpsuspend_flag)
+                    a2dp.clear_a2dpsuspend_flag();
+
+                a2dp.a2dp_suspended = false;
+                //Force restart all active sessions post suspend (if any)
+                if(a2dp.a2dp_total_active_session_request > 0){
+                    int active_sessions = a2dp.a2dp_total_active_session_request;
+                    a2dp.a2dp_total_active_session_request = 0;
+                    audio_extn_a2dp_start_playback();
+                    a2dp.a2dp_total_active_session_request = active_sessions;
+                }
+            }
+        }
+        goto param_handled;
+     }
+     ret = str_parms_get_str(parms,"reconfigA2dp", value, sizeof(value));
+     if (ret >= 0) {
+         if (a2dp.bt_lib_handle && (a2dp.bt_state != A2DP_STATE_DISCONNECTED)) {
+             if (!strncmp(value,"true",sizeof(value)))
+                 a2dp.is_handoff_in_progress = true;
+         }
+         goto param_handled;
+     }
+param_handled:
+     ALOGV("end of a2dp setparam");
+}
+
+bool audio_extn_a2dp_is_force_device_switch()
+{
+    //During encoder reconfiguration mode, force a2dp device switch
+    return a2dp.is_handoff_in_progress;
+}
+
+void audio_extn_a2dp_init (void *adev)
+{
+  a2dp.adev = (struct audio_device*)adev;
+  a2dp.bt_lib_handle = NULL;
+  a2dp.a2dp_started = false;
+  a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+  a2dp.a2dp_total_active_session_request = 0;
+  a2dp.a2dp_suspended = false;
+  a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+  a2dp.enc_config_data = NULL;
+  a2dp.is_a2dp_offload_supported = false;
+  a2dp.is_handoff_in_progress = false;
+  update_offload_codec_capabilities();
+}
+#endif // SPLIT_A2DP_ENABLED
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 49e649c..569b4b2 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -755,6 +755,7 @@
    audio_extn_ssr_set_parameters(adev, parms);
    audio_extn_hfp_set_parameters(adev, parms);
    audio_extn_dts_eagle_set_parameters(adev, parms);
+   audio_extn_a2dp_set_parameters(parms);
    audio_extn_ddp_set_parameters(adev, parms);
    audio_extn_ds2_set_parameters(adev, parms);
    audio_extn_customstereo_set_parameters(adev, parms);
@@ -762,6 +763,7 @@
    audio_extn_pm_set_parameters(parms);
    audio_extn_source_track_set_parameters(adev, parms);
    audio_extn_fbsp_set_parameters(parms);
+   audio_extn_keep_alive_set_parameters(adev, parms);
    check_and_set_hdmi_connection_status(parms);
    if (adev->offload_effects_set_parameters != NULL)
        adev->offload_effects_set_parameters(parms);
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index fe3fe95..d186a5f 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -171,6 +171,20 @@
                                      char *value, int len);
 #endif
 
+#ifndef SPLIT_A2DP_ENABLED
+#define audio_extn_a2dp_init(adev)                       (0)
+#define audio_extn_a2dp_start_playback()                 (0)
+#define audio_extn_a2dp_stop_playback()                  (0)
+#define audio_extn_a2dp_set_parameters(parms)            (0)
+#define audio_extn_a2dp_is_force_device_switch()         (0)
+#else
+void audio_extn_a2dp_init(void *adev);
+int audio_extn_a2dp_start_playback();
+void audio_extn_a2dp_stop_playback();
+void audio_extn_a2dp_set_parameters(struct str_parms *parms);
+bool audio_extn_a2dp_is_force_device_switch();
+#endif
+
 #ifndef SSR_ENABLED
 #define audio_extn_ssr_check_and_set_usecase(in)      (0)
 #define audio_extn_ssr_init(in, num_out_chan)         (0)
diff --git a/hal/audio_extn/keep_alive.c b/hal/audio_extn/keep_alive.c
index 1a4f135..60e7eef 100644
--- a/hal/audio_extn/keep_alive.c
+++ b/hal/audio_extn/keep_alive.c
@@ -38,7 +38,7 @@
 
 #define SILENCE_MIXER_PATH "silence-playback hdmi"
 #define SILENCE_DEV_ID 32           /* index into machine driver */
-#define SILENCE_INTERVAL_US 2000000
+#define SILENCE_INTERVAL 2 /*In secs*/
 
 typedef enum {
     STATE_DEINIT = -1,
@@ -52,7 +52,9 @@
 
 typedef struct {
     pthread_mutex_t lock;
+    pthread_mutex_t sleep_lock;
     pthread_cond_t  cond;
+    pthread_cond_t  wake_up_cond;
     pthread_t thread;
     state_t state;
     struct listnode cmd_list;
@@ -88,6 +90,8 @@
     ka.pcm = NULL;
     pthread_mutex_init(&ka.lock, (const pthread_mutexattr_t *) NULL);
     pthread_cond_init(&ka.cond, (const pthread_condattr_t *) NULL);
+    pthread_cond_init(&ka.wake_up_cond, (const pthread_condattr_t *) NULL);
+    pthread_mutex_init(&ka.sleep_lock, (const pthread_mutexattr_t *) NULL);
     list_init(&ka.cmd_list);
     if (pthread_create(&ka.thread,  (const pthread_attr_t *) NULL,
                        keep_alive_loop, NULL) < 0) {
@@ -143,6 +147,27 @@
     return 0;
 }
 
+
+static int set_mixer_control(struct mixer *mixer,
+                             const char * mixer_ctl_name,
+                             const char *mixer_val)
+{
+    struct mixer_ctl *ctl;
+    if ((mixer == NULL) || (mixer_ctl_name == NULL) || (mixer_val == NULL)) {
+       ALOGE("%s: Invalid input", __func__);
+       return -EINVAL;
+    }
+    ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
+    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+
+    return mixer_ctl_set_enum_by_string(ctl, mixer_val);
+}
+
 /* must be called with adev lock held */
 void audio_extn_keep_alive_start()
 {
@@ -151,18 +176,20 @@
     int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT], len = 0, rc;
     struct mixer_ctl *ctl;
     int acdb_dev_id, snd_device;
+    struct listnode *node;
+    struct audio_usecase *usecase;
     int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
 
     pthread_mutex_lock(&ka.lock);
 
     if (ka.state == STATE_DEINIT) {
         ALOGE(" %s : Invalid state ",__func__);
-        return;
+        goto exit;
     }
 
     if (audio_extn_passthru_is_active()) {
         ALOGE(" %s : Pass through is already active", __func__);
-        return;
+        goto exit;
     }
 
     if (ka.state == STATE_ACTIVE) {
@@ -170,6 +197,14 @@
         goto exit;
     }
 
+    /* Dont start keep_alive if any other PCM session is routed to HDMI*/
+    list_for_each(node, &adev->usecase_list) {
+         usecase = node_to_item(node, struct audio_usecase, list);
+         if (usecase->type == PCM_PLAYBACK &&
+                 usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+             goto exit;
+    }
+
     ka.done = false;
 
     /*configure app type */
@@ -202,9 +237,15 @@
           platform_get_default_app_type(adev->platform),
           acdb_dev_id, sample_rate);
     mixer_ctl_set_array(ctl, app_type_cfg, len);
+    /*Configure HDMI Backend with default values, this as well
+     *helps reconfigure HDMI backend after passthrough
+     */
+    set_mixer_control(adev->mixer, "HDMI RX Format", "LPCM");
+    set_mixer_control(adev->mixer, "HDMI_RX SampleRate", "KHZ_48");
+    set_mixer_control(adev->mixer, "HDMI_RX Channels", "Two");
 
     /*send calibration*/
-    struct audio_usecase *usecase = calloc(1, sizeof(struct audio_usecase));
+    usecase = calloc(1, sizeof(struct audio_usecase));
     usecase->type = PCM_PLAYBACK;
     usecase->out_snd_device = SND_DEVICE_OUT_HDMI;
 
@@ -232,13 +273,13 @@
 
     pthread_mutex_lock(&ka.lock);
 
-    if (ka.state == STATE_DEINIT)
-        return;
-
-    if (ka.state == STATE_IDLE)
+    if ((ka.state == STATE_DEINIT) || (ka.state == STATE_IDLE))
         goto exit;
 
+    pthread_mutex_lock(&ka.sleep_lock);
     ka.done = true;
+    pthread_cond_signal(&ka.wake_up_cond);
+    pthread_mutex_unlock(&ka.sleep_lock);
     while (ka.state != STATE_IDLE) {
         pthread_cond_wait(&ka.cond, &ka.lock);
     }
@@ -290,6 +331,7 @@
     struct listnode *item;
     uint8_t * silence = NULL;
     int32_t bytes = 0;
+    struct timespec ts;
 
     while (true) {
         pthread_mutex_lock(&ka.lock);
@@ -328,9 +370,17 @@
              * Just something to keep the connection alive is sufficient.
              * Hence a short burst of silence periodically.
              */
-            usleep(SILENCE_INTERVAL_US);
-        }
+            pthread_mutex_lock(&ka.sleep_lock);
+            clock_gettime(CLOCK_REALTIME, &ts);
+            ts.tv_sec += SILENCE_INTERVAL;
+            ts.tv_nsec = 0;
 
+            if (!ka.done)
+              pthread_cond_timedwait(&ka.wake_up_cond,
+                            &ka.sleep_lock, &ts);
+
+            pthread_mutex_unlock(&ka.sleep_lock);
+        }
         pthread_mutex_lock(&ka.lock);
         ka.state = STATE_IDLE;
         pthread_cond_signal(&ka.cond);
diff --git a/hal/audio_extn/passthru.c b/hal/audio_extn/passthru.c
index e6ac4dd..eaa8c0a 100644
--- a/hal/audio_extn/passthru.c
+++ b/hal/audio_extn/passthru.c
@@ -82,8 +82,14 @@
  */
 bool audio_extn_passthru_should_drop_data(struct stream_out * out)
 {
-
-    if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+    /*Drop data only
+     *stream is routed to HDMI and
+     *stream has PCM format or
+     *if a compress offload (DSP decode) session
+     */
+    if ((out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) &&
+        (((out->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) ||
+        ((out->compr_config.codec != NULL) && (out->compr_config.codec->compr_passthr == LEGACY_PCM)))) {
         if (android_atomic_acquire_load(&compress_passthru_active) > 0) {
             ALOGI("drop data as pass thru is active");
             return true;
@@ -112,9 +118,6 @@
     ALOGV("inc pass thru count to notify other streams");
     android_atomic_inc(&compress_passthru_active);
 
-    ALOGV("keep_alive_stop");
-    audio_extn_keep_alive_stop();
-
     while (true) {
         /* find max period time among active playback use cases */
         list_for_each(node, &adev->usecase_list) {
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 7e37efc..6142e86 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -98,9 +98,9 @@
             status = -ENOMEM;
             break;
         }
-        memcpy(&st_ses_info->st_ses, &config->st_ses, sizeof (config->st_ses));
-        ALOGV("%s: add capture_handle %d pcm %p", __func__,
-              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+        memcpy(&st_ses_info->st_ses, &config->st_ses, sizeof (struct sound_trigger_session_info));
+        ALOGV("%s: add capture_handle %d st session opaque ptr %p", __func__,
+              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.p_ses);
         list_add_tail(&st_dev->st_ses_list, &st_ses_info->list);
         break;
 
@@ -112,12 +112,12 @@
         }
         st_ses_info = get_sound_trigger_info(config->st_ses.capture_handle);
         if (!st_ses_info) {
-            ALOGE("%s: pcm %p not in the list!", __func__, config->st_ses.pcm);
+            ALOGE("%s: st session opaque ptr %p not in the list!", __func__, config->st_ses.p_ses);
             status = -EINVAL;
             break;
         }
-        ALOGV("%s: remove capture_handle %d pcm %p", __func__,
-              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+        ALOGV("%s: remove capture_handle %d st session opaque ptr %p", __func__,
+              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.p_ses);
         list_remove(&st_ses_info->list);
         free(st_ses_info);
         break;
@@ -181,7 +181,7 @@
     pthread_mutex_unlock(&st_dev->lock);
     if (st_ses_info) {
         event.u.ses_info = st_ses_info->st_ses;
-        ALOGV("%s: AUDIO_EVENT_STOP_LAB pcm %p", __func__, st_ses_info->st_ses.pcm);
+        ALOGV("%s: AUDIO_EVENT_STOP_LAB st sess %p", __func__, st_ses_info->st_ses.p_ses);
         st_dev->st_callback(AUDIO_EVENT_STOP_LAB, &event);
         in->is_st_session_active = false;
     }
@@ -201,7 +201,6 @@
     list_for_each(node, &st_dev->st_ses_list) {
         st_ses_info = node_to_item(node, struct sound_trigger_info , list);
         if (st_ses_info->st_ses.capture_handle == in->capture_handle) {
-            in->pcm = st_ses_info->st_ses.pcm;
             in->config = st_ses_info->st_ses.config;
             in->channel_mask = audio_channel_in_mask_from_count(in->config.channels);
             in->is_st_session = true;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 272d64b..69d5cce 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -716,6 +716,13 @@
          audio_extn_spkr_prot_calib_cancel(adev);
 
 
+    if (((SND_DEVICE_OUT_BT_A2DP == snd_device) ||
+       (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device))
+        && (audio_extn_a2dp_start_playback() < 0)) {
+           ALOGE(" fail to configure A2dp control path ");
+           return -EINVAL;
+    }
+
     if (platform_can_enable_spkr_prot_on_device(snd_device) &&
          audio_extn_spkr_prot_is_enabled()) {
        if (platform_get_spkr_prot_acdb_id(snd_device) < 0) {
@@ -791,6 +798,11 @@
 
     if (adev->snd_dev_ref_cnt[snd_device] == 0) {
         ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
+
+        if ((SND_DEVICE_OUT_BT_A2DP == snd_device) ||
+           (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device))
+            audio_extn_a2dp_stop_playback();
+
         if (platform_can_enable_spkr_prot_on_device(snd_device) &&
              audio_extn_spkr_prot_is_enabled()) {
             audio_extn_spkr_prot_stop_processing(snd_device);
@@ -832,7 +844,7 @@
     struct audio_usecase *usecase;
     bool switch_device[AUDIO_USECASE_MAX];
     int i, num_uc_to_switch = 0;
-
+    bool force_restart_session = false;
     /*
      * This function is to make sure that all the usecases that are active on
      * the hardware codec backend are always routed to any one device that is
@@ -852,7 +864,15 @@
      */
     bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info,
                          snd_device);
-
+    /* For a2dp device reconfigure all active sessions
+     * with new AFE encoder format based on a2dp state
+     */
+    if ((SND_DEVICE_OUT_BT_A2DP == snd_device ||
+         SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) &&
+         audio_extn_a2dp_is_force_device_switch()) {
+         force_routing = true;
+         force_restart_session = true;
+    }
     ALOGD("%s:becf: force routing %d", __func__, force_routing);
 
     /* Disable all the usecases on the shared backend other than the
@@ -871,8 +891,10 @@
               platform_check_backends_match(snd_device, usecase->out_snd_device));
         if (usecase->type != PCM_CAPTURE &&
             usecase != uc_info &&
-            (usecase->out_snd_device != snd_device || force_routing)  &&
-            usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND &&
+            (usecase->out_snd_device != snd_device || force_routing) &&
+            ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
+             (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
+             (force_restart_session)) &&
             platform_check_backends_match(snd_device, usecase->out_snd_device)) {
                 ALOGD("%s:becf: check_usecases (%s) is active on (%s) - disabling ..",
                     __func__, use_case_table[usecase->id],
@@ -1163,6 +1185,14 @@
         }
     }
 
+    // Force all a2dp output devices to reconfigure for proper AFE encode format
+    if((usecase->stream.out) &&
+       (usecase->stream.out->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
+       audio_extn_a2dp_is_force_device_switch()) {
+         ALOGD("Force a2dp device switch to update new encoder config");
+         ret = true;
+     }
+
     return ret;
 }
 
@@ -1207,6 +1237,8 @@
                                                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)) ||
+                                 ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+                                 (usecase->devices & AUDIO_DEVICE_IN_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;
@@ -1214,7 +1246,8 @@
         } else if (voice_extn_compress_voip_is_active(adev)) {
             voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
             if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
-                (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+                ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
+                  ((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) &&
                  (voip_usecase->stream.out != adev->primary_output))) {
                     in_snd_device = voip_usecase->in_snd_device;
                     out_snd_device = voip_usecase->out_snd_device;
@@ -1770,193 +1803,6 @@
     return 0;
 }
 
-static bool allow_hdmi_channel_config(struct audio_device *adev,
-                                      bool enable_passthru)
-{
-    struct listnode *node;
-    struct audio_usecase *usecase;
-    bool ret = true;
-
-    if (enable_passthru && !audio_extn_passthru_is_enabled()) {
-        ret = false;
-        goto exit;
-    }
-
-    if (audio_extn_passthru_is_active()) {
-        ALOGI("%s: Compress audio passthrough is active,"
-              "no HDMI config change allowed", __func__);
-        ret = false;
-        goto exit;
-    }
-
-    list_for_each(node, &adev->usecase_list) {
-        usecase = node_to_item(node, struct audio_usecase, list);
-        if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-            /*
-             * If voice call is already existing, do not proceed further to avoid
-             * disabling/enabling both RX and TX devices, CSD calls, etc.
-             * Once the voice call done, the HDMI channels can be configured to
-             * max channels of remaining use cases.
-             */
-            if (usecase->id == USECASE_VOICE_CALL) {
-                ALOGV("%s: voice call is active, no change in HDMI channels",
-                      __func__);
-                ret = false;
-                break;
-            } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
-                if (!enable_passthru) {
-                    ALOGV("%s: multi channel playback is active, "
-                          "no change in HDMI channels", __func__);
-                    ret = false;
-                    break;
-                }
-            } else if (is_offload_usecase(usecase->id) &&
-                       audio_channel_count_from_out_mask(usecase->stream.out->channel_mask) > 2) {
-                if (!enable_passthru) {
-                    ALOGD("%s:multi-channel(%x) compress offload playback is active"
-                        ", no change in HDMI channels", __func__,
-                        usecase->stream.out->channel_mask);
-                    ret = false;
-                    break;
-                }
-            }
-        }
-    }
-    ALOGV("allow hdmi config %d", ret);
-exit:
-    return ret;
-}
-
-static int check_and_set_hdmi_config(struct audio_device *adev,
-                                     uint32_t channels,
-                                     uint32_t sample_rate,
-                                     audio_format_t format,
-                                     bool enable_passthru)
-{
-    struct listnode *node;
-    struct audio_usecase *usecase;
-    int32_t factor = 1;
-    bool config = false;
-
-    ALOGV("%s channels %d sample_rate %d format:%x enable_passthru:%d",
-         __func__, channels, sample_rate, format, enable_passthru);
-
-    if (channels != adev->cur_hdmi_channels) {
-        ALOGV("channel does not match current hdmi channels");
-        config = true;
-    }
-
-    if (sample_rate != adev->cur_hdmi_sample_rate) {
-        ALOGV("sample rate does not match current hdmi sample rate");
-        config = true;
-    }
-
-    if (format != adev->cur_hdmi_format) {
-        ALOGV("format does not match current hdmi format");
-        config = true;
-    }
-
-    /* TBD - add check for bit width */
-    if (!config) {
-        ALOGV("No need to config hdmi");
-        return 0;
-    }
-
-    if (enable_passthru &&
-        (format == AUDIO_FORMAT_E_AC3)) {
-        ALOGV("factor 4 for E_AC3 passthru");
-        factor = 4;
-    }
-
-    platform_set_hdmi_config(adev->platform, channels, factor * sample_rate,
-                             enable_passthru);
-    adev->cur_hdmi_channels = channels;
-    adev->cur_hdmi_format = format;
-    adev->cur_hdmi_sample_rate = sample_rate;
-
-    /*
-     * Deroute all the playback streams routed to HDMI so that
-     * the back end is deactivated. Note that backend will not
-     * be deactivated if any one stream is connected to it.
-     */
-    list_for_each(node, &adev->usecase_list) {
-        usecase = node_to_item(node, struct audio_usecase, list);
-        if (usecase->type == PCM_PLAYBACK &&
-                usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-            disable_audio_route(adev, usecase);
-        }
-    }
-
-    bool was_active = audio_extn_keep_alive_is_active();
-    if (was_active)
-        audio_extn_keep_alive_stop();
-
-    /*
-     * Enable all the streams disabled above. Now the HDMI backend
-     * will be activated with new channel configuration
-     */
-    list_for_each(node, &adev->usecase_list) {
-        usecase = node_to_item(node, struct audio_usecase, list);
-        if (usecase->type == PCM_PLAYBACK &&
-                usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-            enable_audio_route(adev, usecase);
-        }
-    }
-
-    if (was_active)
-        audio_extn_keep_alive_start();
-
-    return 0;
-}
-
-/* called with out lock taken */
-static int check_and_set_hdmi_backend(struct stream_out *out)
-{
-    struct audio_device *adev = out->dev;
-    int ret;
-    bool enable_passthru = false;
-
-    if (!(out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL))
-        return -1;
-
-    ALOGV("%s usecase %s out->format:%x out->bit_width:%d", __func__, use_case_table[out->usecase],out->format,out->bit_width);
-
-    if (is_offload_usecase(out->usecase) &&
-        audio_extn_passthru_is_passthrough_stream(out)) {
-        enable_passthru = true;
-        ALOGV("%s : enable_passthru is set to true", __func__);
-    }
-
-    /* Check if change in HDMI channel config is allowed */
-    if (!allow_hdmi_channel_config(adev, enable_passthru)) {
-        return -EPERM;
-    }
-
-    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
-        uint32_t channels;
-        ALOGV("Offload usecase, enable passthru %d", enable_passthru);
-
-        if (enable_passthru) {
-            audio_extn_passthru_on_start(out);
-            audio_extn_passthru_update_stream_configuration(adev, out);
-        }
-
-        /* For pass through case, the backend should be configured as stereo */
-        channels = enable_passthru ? DEFAULT_HDMI_OUT_CHANNELS :
-                                     out->compr_config.codec->ch_in;
-
-        ret = check_and_set_hdmi_config(adev, channels,
-                                        out->sample_rate, out->format,
-                                        enable_passthru);
-    } else
-        ret = check_and_set_hdmi_config(adev, out->config.channels,
-                                        out->config.rate,
-                                        out->format,
-                                        false);
-    return ret;
-}
-
-
 static int stop_output_stream(struct stream_out *out)
 {
     int ret = 0;
@@ -1997,17 +1843,14 @@
         ALOGV("Disable passthrough , reset mixer to pcm");
         /* NO_PASSTHROUGH */
         out->compr_config.codec->compr_passthr = 0;
-
         audio_extn_passthru_on_stop(out);
         audio_extn_dolby_set_dap_bypass(adev, DAP_STATE_ON);
     }
 
     /* Must be called after removing the usecase from list */
     if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
-        check_and_set_hdmi_config(adev, DEFAULT_HDMI_OUT_CHANNELS,
-                                  DEFAULT_HDMI_OUT_SAMPLE_RATE,
-                                  DEFAULT_HDMI_OUT_FORMAT,
-                                  false);
+        audio_extn_keep_alive_start();
+
     ALOGV("%s: exit: status(%d)", __func__, ret);
     return ret;
 }
@@ -2049,12 +1892,6 @@
         goto error_config;
     }
 
-    /* This must be called before adding this usecase to the list */
-    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
-        /* This call can fail if compress pass thru is already active */
-        check_and_set_hdmi_backend(out);
-    }
-
     uc_info->id = out->usecase;
     uc_info->type = PCM_PLAYBACK;
     uc_info->stream.out = out;
@@ -2066,6 +1903,16 @@
     audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
                                  adev->perf_lock_opts,
                                  adev->perf_lock_opts_size);
+
+    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+        audio_extn_keep_alive_stop();
+        if (audio_extn_passthru_is_enabled() &&
+            audio_extn_passthru_is_passthrough_stream(out)) {
+            audio_extn_passthru_on_start(out);
+            audio_extn_passthru_update_stream_configuration(adev, out);
+        }
+    }
+
     select_devices(adev, out->usecase);
 
     ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
@@ -2451,6 +2298,17 @@
                 (platform_get_edid_info(adev->platform) != 0) /* HDMI disconnected */) {
             val = AUDIO_DEVICE_OUT_SPEAKER;
         }
+        /*
+         * When A2DP is disconnected the
+         * music playback is paused and the policy manager sends routing=0
+         * But the audioflingercontinues to write data until standby time
+         * (3sec). As BT is turned off, the write gets blocked.
+         * Avoid this by routing audio to speaker until standby.
+         */
+        if ((out->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
+                (val == AUDIO_DEVICE_NONE)) {
+                val = AUDIO_DEVICE_OUT_SPEAKER;
+        }
 
         /*
          * select_devices() call below switches all the usecases on the same
@@ -2748,9 +2606,10 @@
     }
 
     if (audio_extn_passthru_should_drop_data(out)) {
-        ALOGD(" %s : Drop data as compress passthrough session is going on", __func__);
-        usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
-                        out_get_sample_rate(&out->stream.common));
+        ALOGV(" %s : Drop data as compress passthrough session is going on", __func__);
+        if (audio_bytes_per_sample(out->format) != 0)
+            out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
+        ret = -EIO;
         goto exit;
     }
 
@@ -3125,7 +2984,6 @@
                 if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
                     pthread_mutex_lock(&out->dev->lock);
                     ALOGV("offload resume, check and set hdmi backend again");
-                    check_and_set_hdmi_backend(out);
                     pthread_mutex_unlock(&out->dev->lock);
                 }
                 status = compress_resume(out->compr);
@@ -3721,7 +3579,7 @@
         out->compr_config.codec->bit_rate =
                     config->offload_info.bit_rate;
         out->compr_config.codec->ch_in =
-                audio_channel_count_from_out_mask(config->channel_mask);
+                audio_channel_count_from_out_mask(out->channel_mask);
         out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
         out->bit_width = AUDIO_OUTPUT_BIT_WIDTH;
         /*TODO: Do we need to change it for passthrough */
@@ -4173,6 +4031,22 @@
     }
 
     audio_extn_set_parameters(adev, parms);
+    // reconfigure should be done only after updating a2dpstate in audio extn
+    ret = str_parms_get_str(parms,"reconfigA2dp", value, sizeof(value));
+    if (ret >= 0) {
+        struct audio_usecase *usecase;
+        struct listnode *node;
+        list_for_each(node, &adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            if ((usecase->type == PCM_PLAYBACK) &&
+                (usecase->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP)){
+                ALOGD("reconfigure a2dp... forcing device switch");
+                //force device switch to re configure encoder
+                select_devices(adev, usecase->id);
+                break;
+            }
+        }
+    }
 
 done:
     str_parms_destroy(parms);
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 9a87c0f..bf5b62b 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -192,8 +192,10 @@
 typedef struct codec_backend_cfg {
     uint32_t sample_rate;
     uint32_t bit_width;
+    uint32_t channels;
     char     *bitwidth_mixer_ctl;
     char     *samplerate_mixer_ctl;
+    char     *channels_mixer_ctl;
 } codec_backend_cfg_t;
 
 static native_audio_prop na_props = {0, 0, 0};
@@ -346,6 +348,8 @@
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
     [SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
     [SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+    [SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
@@ -465,6 +469,8 @@
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
     [SND_DEVICE_OUT_BT_SCO] = 22,
     [SND_DEVICE_OUT_BT_SCO_WB] = 39,
+    [SND_DEVICE_OUT_BT_A2DP] = 20,
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
@@ -586,6 +592,8 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
@@ -1206,6 +1214,8 @@
     backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
     backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
     backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("vbat-voice-speaker");
+    backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
+    backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
 
     hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
@@ -1848,6 +1858,9 @@
     /* init usb */
     audio_extn_usb_init(adev);
 
+    /*init a2dp*/
+    audio_extn_a2dp_init(adev);
+
     /* Read one time ssr property */
     audio_extn_ssr_update_enabled();
     audio_extn_spkr_prot_init(adev);
@@ -1916,6 +1929,8 @@
         strdup("HDMI_RX Bit Format");
     my_data->current_backend_cfg[HDMI_RX_BACKEND].samplerate_mixer_ctl =
         strdup("HDMI_RX SampleRate");
+    my_data->current_backend_cfg[HDMI_RX_BACKEND].channels_mixer_ctl =
+        strdup("HDMI_RX Channels");
 
     ret = audio_extn_utils_get_codec_version(snd_card_name,
                                              my_data->adev->snd_card,
@@ -2421,6 +2436,17 @@
     return ret;
 }
 
+int check_44100_support_device(audio_devices_t out_device)
+{
+    int ret = true;
+
+    if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+        out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+        out_device & AUDIO_DEVICE_OUT_LINE)
+        ret = false;
+
+    return ret;
+}
 
 static int platform_get_backend_index(snd_device_t snd_device)
 {
@@ -2876,6 +2902,9 @@
         } else if (devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
             snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
+        } else if ((devices & AUDIO_DEVICE_OUT_SPEAKER) &&
+                   (devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
+            snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP;
         } else {
             ALOGE("%s: Invalid combo device(%#x)", __func__, devices);
             goto exit;
@@ -2926,6 +2955,8 @@
                 snd_device = SND_DEVICE_OUT_BT_SCO_WB;
             else
                 snd_device = SND_DEVICE_OUT_BT_SCO;
+        } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+                snd_device = SND_DEVICE_OUT_BT_A2DP;
         } else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
                 if (my_data->is_vbat_speaker)
                     snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
@@ -3009,6 +3040,8 @@
             snd_device = SND_DEVICE_OUT_BT_SCO;
     } else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
         snd_device = SND_DEVICE_OUT_HDMI ;
+    } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+        snd_device = SND_DEVICE_OUT_BT_A2DP;
     } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
                devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
         ALOGD("%s: setting USB hadset channel capability(2) for Proxy", __func__);
@@ -3788,8 +3821,8 @@
                 !strncmp("true", propValue, 4);
         }
 
-        if (prop_playback_enabled && (voice_is_in_call(my_data->adev) ||
-             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev)))) {
+        if ((prop_playback_enabled && (voice_is_in_call(my_data->adev))) ||
+             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev))) {
             char *decoder_mime_type = value;
 
             //check if unsupported mime type or not
@@ -4017,16 +4050,21 @@
  * 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, audio_format_t format)
+                         snd_device_t snd_device, struct audio_backend_cfg backend_cfg)
 {
     int ret = 0;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
+    unsigned int bit_width = backend_cfg.bit_width;
+    unsigned int sample_rate = backend_cfg.sample_rate;
+    unsigned int channels = backend_cfg.channels;
+    audio_format_t format = backend_cfg.format;
+    bool passthrough_enabled = backend_cfg.passthrough_enabled;
 
     backend_idx = platform_get_backend_index(snd_device);
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
-          __func__, bit_width, sample_rate, backend_idx,
+
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d, backend_idx %d device (%s)",
+          __func__, bit_width, sample_rate, channels,backend_idx,
           platform_get_snd_device_name(snd_device));
 
     if (bit_width !=
@@ -4118,19 +4156,146 @@
             mixer_ctl_set_enum_by_string(ctl, rate_str);
             my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
     }
+    if ((backend_idx == HDMI_RX_BACKEND) &&
+        (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+        struct  mixer_ctl *ctl;
+        char *channel_cnt_str = NULL;
+
+        switch (channels) {
+        case 8:
+            channel_cnt_str = "Eight"; break;
+        case 7:
+            channel_cnt_str = "Seven"; break;
+        case 6:
+            channel_cnt_str = "Six"; break;
+        case 5:
+            channel_cnt_str = "Five"; break;
+        case 4:
+            channel_cnt_str = "Four"; break;
+        case 3:
+            channel_cnt_str = "Three"; break;
+        default:
+            channel_cnt_str = "Two"; break;
+        }
+
+        ctl = mixer_get_ctl_by_name(adev->mixer,
+           my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                   __func__,
+                   my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+            return -EINVAL;
+        }
+        mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
+        my_data->current_backend_cfg[backend_idx].channels = channels;
+        platform_set_edid_channels_configuration(adev->platform, channels);
+        ALOGD("%s:becf: afe: %s set to %s", __func__,
+               my_data->current_backend_cfg[backend_idx].channels_mixer_ctl, channel_cnt_str);
+    }
+
+    if (backend_idx == HDMI_RX_BACKEND) {
+        const char *hdmi_format_ctrl = "HDMI RX Format";
+        struct mixer_ctl *ctl;
+        ctl = mixer_get_ctl_by_name(adev->mixer,hdmi_format_ctrl);
+
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                   __func__, hdmi_format_ctrl);
+            return -EINVAL;
+        }
+
+        if (passthrough_enabled) {
+            ALOGD("%s:HDMI compress format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "Compr");
+        } else {
+            ALOGD("%s: HDMI PCM format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "LPCM");
+        }
+    }
 
     return ret;
 }
 
 /*
+ *Validate the selected bit_width, sample_rate and channels using the edid
+ *of the connected sink device.
+ */
+static void platform_check_hdmi_backend_cfg(struct audio_device* adev,
+                                   struct audio_usecase* usecase,
+                                   struct audio_backend_cfg *hdmi_backend_cfg)
+{
+    unsigned int bit_width;
+    unsigned int sample_rate;
+    unsigned int channels, max_supported_channels = 0;
+    struct platform_data *my_data = (struct platform_data *)adev->platform;
+    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool passthrough_enabled = false;
+
+    bit_width = hdmi_backend_cfg->bit_width;
+    sample_rate = hdmi_backend_cfg->sample_rate;
+    channels = hdmi_backend_cfg->channels;
+
+
+    ALOGI("%s:becf: HDMI: bitwidth %d, samplerate %d, channels %d"
+          ", usecase = %d", __func__, bit_width,
+          sample_rate, channels, usecase->id);
+
+    if (audio_extn_passthru_is_enabled() && audio_extn_passthru_is_active()
+        && (usecase->stream.out->compr_config.codec->compr_passthr != 0)) {
+        passthrough_enabled = true;
+        ALOGI("passthrough is enabled for this stream");
+    }
+
+    // For voice calls use default configuration i.e. 16b/48K, only applicable to
+    // default backend
+    if (!passthrough_enabled) {
+
+        max_supported_channels = platform_edid_get_max_channels(my_data);
+
+        //Check EDID info for supported samplerate
+        if (!edid_is_supported_sr(edid_info,sample_rate)) {
+            //reset to current sample rate
+            sample_rate = my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate;
+        }
+
+        //Check EDID info for supported bit width
+        if (!edid_is_supported_bps(edid_info,bit_width)) {
+            //reset to current sample rate
+            bit_width = my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width;
+        }
+
+        if (channels > max_supported_channels)
+            channels = max_supported_channels;
+
+    } else {
+        /*During pass through set default bit width and channels*/
+        channels = DEFAULT_HDMI_OUT_CHANNELS;
+        if ((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+            (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
+            sample_rate = sample_rate * 4 ;
+
+        bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        /* We force route so that the BE format can be set to Compr */
+    }
+
+    ALOGI("%s:becf: afe: HDMI backend: passthrough %d updated bit width: %d and sample rate: %d"
+           "channels %d", __func__, passthrough_enabled , bit_width,
+           sample_rate, channels);
+
+    hdmi_backend_cfg->bit_width = bit_width;
+    hdmi_backend_cfg->sample_rate = sample_rate;
+    hdmi_backend_cfg->channels = channels;
+    hdmi_backend_cfg->passthrough_enabled = passthrough_enabled;
+}
+
+/*
  * goes through all the current usecases and picks the highest
  * bitwidth & samplerate
  */
 static bool platform_check_codec_backend_cfg(struct audio_device* adev,
                                    struct audio_usecase* usecase,
                                    snd_device_t snd_device,
-                                   unsigned int* new_bit_width,
-                                   unsigned int* new_sample_rate)
+                                   struct audio_backend_cfg *backend_cfg)
 {
     bool backend_change = false;
     struct listnode *node;
@@ -4138,18 +4303,21 @@
     char value[PROPERTY_VALUE_MAX] = {0};
     unsigned int bit_width;
     unsigned int sample_rate;
+    unsigned int channels;
+    bool passthrough_enabled = false;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
     int na_mode = platform_get_native_support();
-    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool channels_updated = false;
 
     backend_idx = platform_get_backend_index(snd_device);
 
-    bit_width = *new_bit_width;
-    sample_rate = *new_sample_rate;
+    bit_width = backend_cfg->bit_width;
+    sample_rate = backend_cfg->sample_rate;
+    channels = backend_cfg->channels;
 
-    ALOGI("%s:becf: afe: Codec selected backend: %d current bit width: %d and sample rate: %d",
-          __func__, backend_idx, bit_width, sample_rate);
+    ALOGI("%s:becf: afe: Codec selected backend: %d current bit width: %d sample rate: %d channels: %d",
+          __func__, backend_idx, bit_width, sample_rate, channels);
 
     // For voice calls use default configuration i.e. 16b/48K, only applicable to
     // default backend
@@ -4175,12 +4343,13 @@
             struct audio_usecase *uc;
             uc = node_to_item(node, struct audio_usecase, list);
             struct stream_out *out = (struct stream_out*) uc->stream.out;
+            unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
             if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
 
                 ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
-                      "(%d) device %s", __func__, i++, use_case_table[uc->id],
+                      "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
                       uc->id, out->sample_rate,
-                      out->bit_width,
+                      out->bit_width, out_channels,
                       platform_get_snd_device_name(uc->out_snd_device));
 
                 if (platform_check_backends_match(snd_device, uc->out_snd_device)) {
@@ -4190,6 +4359,8 @@
                             sample_rate = out->sample_rate;
                         if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
                             sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                        if (channels < out_channels)
+                            channels = out_channels;
                 }
             }
         }
@@ -4218,14 +4389,12 @@
     }
 
     /*
-     * hifi playback not supported on spkr devices, limit the Sample Rate
+     * hifi playback not supported on non-44.1-support devices, limit the Sample Rate
      * to 48 khz.
      */
-    if (SND_DEVICE_OUT_SPEAKER == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_WSA == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_VBAT == snd_device) {
+    if (check_44100_support_device(usecase->devices)) {
         sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-        ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+        ALOGD("%s:becf: afe: playback on non-44.1-support device Configure afe to "
             "default Sample Rate(48k)", __func__);
     }
 
@@ -4249,16 +4418,21 @@
     }
 
     if (backend_idx == HDMI_RX_BACKEND) {
-        //Check EDID info for supported samplerate
-        if (!edid_is_supported_sr(edid_info,sample_rate)) {
-            //reset to current sample rate
-            sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
-        }
-        //Check EDID info for supported bit widhth
-        if (!edid_is_supported_bps(edid_info,bit_width)) {
-            //reset to current sample rate
-            bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
-        }
+        struct audio_backend_cfg hdmi_backend_cfg;
+        hdmi_backend_cfg.bit_width = bit_width;
+        hdmi_backend_cfg.sample_rate = sample_rate;
+        hdmi_backend_cfg.channels = channels;
+        hdmi_backend_cfg.passthrough_enabled = false;
+
+        platform_check_hdmi_backend_cfg(adev, usecase, &hdmi_backend_cfg);
+
+        bit_width = hdmi_backend_cfg.bit_width;
+        sample_rate = hdmi_backend_cfg.sample_rate;
+        channels = hdmi_backend_cfg.channels;
+        passthrough_enabled = hdmi_backend_cfg.passthrough_enabled;
+
+        if (channels != my_data->current_backend_cfg[backend_idx].channels)
+            channels_updated = true;
     }
 
     //check if mulitchannel clip needs to be down sampled to 48k
@@ -4289,13 +4463,16 @@
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
     if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
-        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
-        *new_bit_width = bit_width;
-        *new_sample_rate = sample_rate;
+        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+         passthrough_enabled || channels_updated) {
+        backend_cfg->bit_width = bit_width;
+        backend_cfg->sample_rate = sample_rate;
+        backend_cfg->channels = channels;
+        backend_cfg->passthrough_enabled = passthrough_enabled;
         backend_change = true;
-        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d new sample rate: %d",
-              __func__,
-             *new_bit_width, *new_sample_rate);
+        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d"
+              " new sample rate: %d new channels %d",__func__,
+               backend_cfg->bit_width, backend_cfg->sample_rate, backend_cfg->channels);
     }
 
     return backend_change;
@@ -4304,23 +4481,24 @@
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
     struct audio_usecase *usecase, snd_device_t snd_device)
 {
-    unsigned int new_bit_width;
-    unsigned int new_sample_rate;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     int new_snd_devices[SND_DEVICE_OUT_END];
     int i, num_devices = 1;
+    struct audio_backend_cfg backend_cfg;
     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;
+    backend_cfg.bit_width = usecase->stream.out->bit_width;
+    backend_cfg.sample_rate = usecase->stream.out->sample_rate;
+    backend_cfg.format = usecase->stream.out->format;
+    backend_cfg.channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
+    /*this is populated by check_codec_backend_cfg hence set default value to false*/
+    backend_cfg.passthrough_enabled = false;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
-          new_sample_rate, backend_idx, usecase->id,
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+          ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
+          backend_cfg.sample_rate,  backend_cfg.channels, backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
 
     if (!platform_can_split_snd_device(adev->platform, snd_device,
@@ -4331,9 +4509,9 @@
         ALOGI("%s: becf: new_snd_devices[%d] is %s", __func__, i,
             platform_get_snd_device_name(new_snd_devices[i]));
         if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
-                                             &new_bit_width, &new_sample_rate)) {
+                                             &backend_cfg)) {
                 platform_set_codec_backend_cfg(adev, new_snd_devices[i],
-                                               new_bit_width, new_sample_rate, format);
+                                               backend_cfg);
                 ret = true;
         }
     }
@@ -5003,6 +5181,7 @@
     //reset HDMI_RX_BACKEND to default values
     my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
     my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+    my_data->current_backend_cfg[HDMI_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
 }
 
 int platform_set_mixer_control(struct stream_out *out, const char * mixer_ctl_name,
@@ -5021,91 +5200,6 @@
     return mixer_ctl_set_enum_by_string(ctl, mixer_val);
 }
 
-static int set_mixer_control(struct mixer *mixer,
-                             const char * mixer_ctl_name,
-                             const char *mixer_val)
-{
-    struct mixer_ctl *ctl;
-    ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
-    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
-    if (!ctl) {
-        ALOGE("%s: could not get ctl for mixer cmd - %s",
-              __func__, mixer_ctl_name);
-        return -EINVAL;
-    }
-
-    return mixer_ctl_set_enum_by_string(ctl, mixer_val);
-}
-
-int platform_set_hdmi_config(void *platform, uint32_t channel_count,
-                             uint32_t sample_rate, bool enable_passthrough)
-{
-    struct platform_data *my_data = (struct platform_data *)platform;
-    struct audio_device *adev = my_data->adev;
-    const char *hdmi_format_ctrl = "HDMI RX Format";
-    const char *hdmi_rate_ctrl   = "HDMI_RX SampleRate";
-    const char *hdmi_chans_ctrl  = "HDMI_RX Channels";
-    const char *channel_cnt_str  = NULL;
-
-    ALOGI("%s ch[%d] sr[%d], pthru[%d]", __func__,
-        channel_count, sample_rate, enable_passthrough);
-
-    switch (channel_count) {
-    case 8:
-        channel_cnt_str = "Eight"; break;
-    case 7:
-        channel_cnt_str = "Seven"; break;
-    case 6:
-        channel_cnt_str = "Six"; break;
-    case 5:
-        channel_cnt_str = "Five"; break;
-    case 4:
-        channel_cnt_str = "Four"; break;
-    case 3:
-        channel_cnt_str = "Three"; break;
-    default:
-        channel_cnt_str = "Two"; break;
-    }
-    ALOGV("%s: HDMI channel count: %s", __func__, channel_cnt_str);
-    set_mixer_control(adev->mixer, hdmi_chans_ctrl, channel_cnt_str);
-
-    if (enable_passthrough) {
-        ALOGD("%s:HDMI compress format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "Compr");
-    } else {
-        ALOGD("%s: HDMI PCM format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "LPCM");
-    }
-
-    switch (sample_rate) {
-    case 32000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_32");
-        break;
-    case 44100:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_44P1");
-        break;
-    case 96000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_96");
-        break;
-    case 128000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_128");
-        break;
-    case 176400:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_176_4");
-        break;
-    case 192000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_192");
-        break;
-    default:
-    case 48000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_48");
-        break;
-    }
-
-    return 0;
-}
-
-
 int platform_set_device_params(struct stream_out *out, int param, int value)
 {
     struct audio_device *adev = out->dev;
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 756c749..dcd351a 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -98,6 +98,8 @@
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
     SND_DEVICE_OUT_BT_SCO,
     SND_DEVICE_OUT_BT_SCO_WB,
+    SND_DEVICE_OUT_BT_A2DP,
+    SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -370,4 +372,13 @@
     char device_name[100];
     char interface_name[100];
 };
+
+struct audio_backend_cfg {
+    unsigned int   sample_rate;
+    unsigned int   channels;
+    unsigned int   bit_width;
+    bool           passthrough_enabled;
+    audio_format_t format;
+};
+
 #endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index aab5436..e42af8c 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2013, 2015 The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013, 2016 The Linux Foundation. All rights reserved.
  * Not a contribution.
  *
  * Copyright (C) 2013 The Android Open Source Project
@@ -64,6 +64,8 @@
     SND_DEVICE_OUT_HDMI,
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
     SND_DEVICE_OUT_BT_SCO,
+    SND_DEVICE_OUT_BT_A2DP,
+    SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
     SND_DEVICE_OUT_BT_SCO_WB,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 44bcbc8..f0f507d 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -188,8 +188,10 @@
 typedef struct codec_backend_cfg {
     uint32_t sample_rate;
     uint32_t bit_width;
+    uint32_t channels;
     char     *bitwidth_mixer_ctl;
     char     *samplerate_mixer_ctl;
+    char     *channels_mixer_ctl;
 } codec_backend_cfg_t;
 
 static native_audio_prop na_props = {0, 0, 0};
@@ -347,6 +349,8 @@
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
     [SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
     [SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+    [SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
@@ -461,6 +465,8 @@
     [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
     [SND_DEVICE_OUT_BT_SCO] = 22,
     [SND_DEVICE_OUT_BT_SCO_WB] = 39,
+    [SND_DEVICE_OUT_BT_A2DP] = 20,
+    [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
     [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
@@ -577,6 +583,8 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
@@ -1055,7 +1063,8 @@
                  sizeof("apq8084-taiko-i2s-cdp-snd-card"))) {
         plat_data->is_i2s_ext_modem = true;
     }
-    ALOGV("%s, is_i2s_ext_modem:%d",__func__, plat_data->is_i2s_ext_modem);
+    ALOGV("%s, is_i2s_ext_modem:%d soundcard name is %s",__func__,
+           plat_data->is_i2s_ext_modem, snd_card_name);
 
     return plat_data->is_i2s_ext_modem;
 }
@@ -1095,6 +1104,8 @@
     backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
     backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
     backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
+    backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
+    backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
 
     hw_interface_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("SLIMBUS_5_RX");
     hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
@@ -1676,6 +1687,9 @@
     /* init usb */
     audio_extn_usb_init(adev);
 
+    /*init a2dp*/
+    audio_extn_a2dp_init(adev);
+
     /* init dap hal */
     audio_extn_dap_hal_init(adev->snd_card);
 
@@ -1739,6 +1753,8 @@
         strdup("HDMI_RX Bit Format");
     my_data->current_backend_cfg[HDMI_RX_BACKEND].samplerate_mixer_ctl =
         strdup("HDMI_RX SampleRate");
+    my_data->current_backend_cfg[HDMI_RX_BACKEND].channels_mixer_ctl =
+        strdup("HDMI_RX Channels");
 
     my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].bitwidth_mixer_ctl =
         strdup("USB_AUDIO_RX Format");
@@ -2209,6 +2225,19 @@
 
     return ret;
 }
+
+int check_44100_support_device(audio_devices_t out_device)
+{
+    int ret = true;
+
+    if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+        out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+        out_device & AUDIO_DEVICE_OUT_LINE)
+        ret = false;
+
+    return ret;
+}
+
 static int platform_get_backend_index(snd_device_t snd_device)
 {
     int32_t port = DEFAULT_CODEC_BACKEND;
@@ -2646,6 +2675,9 @@
         } else if (devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
                                AUDIO_DEVICE_OUT_SPEAKER)) {
             snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
+        } else if ((devices & AUDIO_DEVICE_OUT_SPEAKER) &&
+                   (devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
+            snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP;
         } else {
             ALOGE("%s: Invalid combo device(%#x)", __func__, devices);
             goto exit;
@@ -2701,6 +2733,8 @@
                     snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
                 else
                     snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+        } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+            snd_device = SND_DEVICE_OUT_BT_A2DP;
         } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
                    devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
             snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -2750,6 +2784,8 @@
             snd_device = SND_DEVICE_OUT_BT_SCO_WB;
         else
             snd_device = SND_DEVICE_OUT_BT_SCO;
+    } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+        snd_device = SND_DEVICE_OUT_BT_A2DP;
     } else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
         snd_device = SND_DEVICE_OUT_HDMI ;
     } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
@@ -3854,8 +3890,8 @@
                 !strncmp("true", propValue, 4);
         }
 
-        if (prop_playback_enabled && (voice_is_in_call(my_data->adev) ||
-             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev)))) {
+        if ((prop_playback_enabled && (voice_is_in_call(my_data->adev))) ||
+             (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev))) {
             char *decoder_mime_type = value;
 
             //check if unsupported mime type or not
@@ -3990,17 +4026,20 @@
  * 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, audio_format_t format)
+                         snd_device_t snd_device, struct audio_backend_cfg backend_cfg)
 {
     int ret = 0;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
-
     backend_idx = platform_get_backend_index(snd_device);
+    unsigned int bit_width = backend_cfg.bit_width;
+    unsigned int sample_rate = backend_cfg.sample_rate;
+    unsigned int channels = backend_cfg.channels;
+    audio_format_t format = backend_cfg.format;
+    bool passthrough_enabled = backend_cfg.passthrough_enabled;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d device (%s)", __func__,  bit_width, sample_rate, backend_idx,
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+          ", backend_idx %d device (%s)", __func__,  bit_width, sample_rate, channels, backend_idx,
           platform_get_snd_device_name(snd_device));
 
     if (bit_width !=
@@ -4084,37 +4123,167 @@
             mixer_ctl_set_enum_by_string(ctl, rate_str);
             my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
     }
+    if ((backend_idx == HDMI_RX_BACKEND) &&
+        (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+        struct  mixer_ctl *ctl;
+        char *channel_cnt_str = NULL;
+
+        switch (channels) {
+        case 8:
+            channel_cnt_str = "Eight"; break;
+        case 7:
+            channel_cnt_str = "Seven"; break;
+        case 6:
+            channel_cnt_str = "Six"; break;
+        case 5:
+            channel_cnt_str = "Five"; break;
+        case 4:
+            channel_cnt_str = "Four"; break;
+        case 3:
+            channel_cnt_str = "Three"; break;
+        default:
+            channel_cnt_str = "Two"; break;
+        }
+
+        ctl = mixer_get_ctl_by_name(adev->mixer,
+           my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                   __func__,
+                   my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+            return -EINVAL;
+        }
+        mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
+        my_data->current_backend_cfg[backend_idx].channels = channels;
+        platform_set_edid_channels_configuration(adev->platform, channels);
+        ALOGD("%s:becf: afe: %s set to %s", __func__,
+               my_data->current_backend_cfg[backend_idx].channels_mixer_ctl, channel_cnt_str);
+    }
+
+    if (backend_idx == HDMI_RX_BACKEND) {
+        const char *hdmi_format_ctrl = "HDMI RX Format";
+        struct mixer_ctl *ctl;
+        ctl = mixer_get_ctl_by_name(adev->mixer,hdmi_format_ctrl);
+
+        if (!ctl) {
+            ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+                   __func__, hdmi_format_ctrl);
+            return -EINVAL;
+        }
+
+        if (passthrough_enabled) {
+            ALOGD("%s:HDMI compress format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "Compr");
+        } else {
+            ALOGD("%s: HDMI PCM format", __func__);
+            mixer_ctl_set_enum_by_string(ctl, "LPCM");
+        }
+    }
 
     return ret;
 }
 
 /*
+ *Validate the selected bit_width, sample_rate and channels using the edid
+ *of the connected sink device.
+ */
+static void platform_check_hdmi_backend_cfg(struct audio_device* adev,
+                                   struct audio_usecase* usecase,
+                                   struct audio_backend_cfg *hdmi_backend_cfg)
+{
+    unsigned int bit_width;
+    unsigned int sample_rate;
+    unsigned int channels, max_supported_channels = 0;
+    struct platform_data *my_data = (struct platform_data *)adev->platform;
+    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool passthrough_enabled = false;
+
+    bit_width = hdmi_backend_cfg->bit_width;
+    sample_rate = hdmi_backend_cfg->sample_rate;
+    channels = hdmi_backend_cfg->channels;
+
+
+    ALOGI("%s:becf: HDMI: bitwidth %d, samplerate %d, channels %d"
+          ", usecase = %d", __func__, bit_width,
+          sample_rate, channels, usecase->id);
+
+    if (audio_extn_passthru_is_enabled() && audio_extn_passthru_is_active()
+        && (usecase->stream.out->compr_config.codec->compr_passthr != 0)) {
+        passthrough_enabled = true;
+        ALOGI("passthrough is enabled for this stream");
+    }
+
+    // For voice calls use default configuration i.e. 16b/48K, only applicable to
+    // default backend
+    if (!passthrough_enabled) {
+
+        max_supported_channels = platform_edid_get_max_channels(my_data);
+
+        //Check EDID info for supported samplerate
+        if (!edid_is_supported_sr(edid_info,sample_rate)) {
+            //reset to current sample rate
+            sample_rate = my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate;
+        }
+
+        //Check EDID info for supported bit width
+        if (!edid_is_supported_bps(edid_info,bit_width)) {
+            //reset to current sample rate
+            bit_width = my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width;
+        }
+
+        if (channels > max_supported_channels)
+            channels = max_supported_channels;
+
+    } else {
+        /*During pass through set default bit width and channels*/
+        channels = DEFAULT_HDMI_OUT_CHANNELS;
+        if ((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+            (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
+            sample_rate = sample_rate * 4 ;
+
+        bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+        /* We force route so that the BE format can be set to Compr */
+    }
+
+    ALOGI("%s:becf: afe: HDMI backend: passthrough %d updated bit width: %d and sample rate: %d"
+           "channels %d", __func__, passthrough_enabled , bit_width,
+           sample_rate, channels);
+
+    hdmi_backend_cfg->bit_width = bit_width;
+    hdmi_backend_cfg->sample_rate = sample_rate;
+    hdmi_backend_cfg->channels = channels;
+    hdmi_backend_cfg->passthrough_enabled = passthrough_enabled;
+}
+
+/*
  * goes through all the current usecases and picks the highest
  * bitwidth & samplerate
  */
 static bool platform_check_codec_backend_cfg(struct audio_device* adev,
                                    struct audio_usecase* usecase,
                                    snd_device_t snd_device,
-                                   unsigned int* new_bit_width,
-                                   unsigned int* new_sample_rate)
+                                   struct audio_backend_cfg *backend_cfg)
 {
     bool backend_change = false;
     struct listnode *node;
     unsigned int bit_width;
     unsigned int sample_rate;
+    unsigned int channels;
+    bool passthrough_enabled = false;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
     int na_mode = platform_get_native_support();
-    edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+    bool channels_updated = false;
 
     backend_idx = platform_get_backend_index(snd_device);
 
-    bit_width = *new_bit_width;
-    sample_rate = *new_sample_rate;
+    bit_width = backend_cfg->bit_width;
+    sample_rate = backend_cfg->sample_rate;
+    channels = backend_cfg->channels;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
           ", backend_idx %d usecase = %d device (%s)", __func__, bit_width,
-          sample_rate, backend_idx, usecase->id,
+          sample_rate, channels, backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
 
     // For voice calls use default configuration i.e. 16b/48K, only applicable to
@@ -4141,12 +4310,13 @@
             struct audio_usecase *uc;
             uc = node_to_item(node, struct audio_usecase, list);
             struct stream_out *out = (struct stream_out*) uc->stream.out;
+            unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
             if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
 
                 ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
-                      "(%d) device %s", __func__, i++, use_case_table[uc->id],
+                      "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
                       uc->id, out->sample_rate,
-                      out->bit_width,
+                      out->bit_width, out_channels,
                       platform_get_snd_device_name(uc->out_snd_device));
 
                 if (platform_check_backends_match(snd_device, uc->out_snd_device)) {
@@ -4156,6 +4326,8 @@
                             sample_rate = out->sample_rate;
                         if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
                             sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+                        if (channels < out_channels)
+                            channels = out_channels;
                 }
             }
         }
@@ -4184,14 +4356,12 @@
     }
 
     /*
-     * hifi playback not supported on spkr devices, limit the Sample Rate
+     * hifi playback not supported on non-44.1-support devices, limit the Sample Rate
      * to 48 khz.
      */
-    if (SND_DEVICE_OUT_SPEAKER == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_WSA == snd_device ||
-        SND_DEVICE_OUT_SPEAKER_VBAT == snd_device) {
+    if (check_44100_support_device(usecase->devices)) {
         sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-        ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+        ALOGD("%s:becf: afe: playback on non-44.1-support device Configure afe to "
             "default Sample Rate(48k)", __func__);
     }
 
@@ -4213,29 +4383,39 @@
     }
 
     if (backend_idx == HDMI_RX_BACKEND) {
-        //Check EDID info for supported samplerate
-        if (!edid_is_supported_sr(edid_info,sample_rate)) {
-            //reset to current sample rate
-            sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
-        }
-        //Check EDID info for supported bit widhth
-        if (!edid_is_supported_bps(edid_info,bit_width)) {
-            //reset to current sample rate
-            bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
-        }
+        struct audio_backend_cfg hdmi_backend_cfg;
+        hdmi_backend_cfg.bit_width = bit_width;
+        hdmi_backend_cfg.sample_rate = sample_rate;
+        hdmi_backend_cfg.channels = channels;
+        hdmi_backend_cfg.passthrough_enabled = false;
+
+        platform_check_hdmi_backend_cfg(adev, usecase, &hdmi_backend_cfg);
+
+        bit_width = hdmi_backend_cfg.bit_width;
+        sample_rate = hdmi_backend_cfg.sample_rate;
+        channels = hdmi_backend_cfg.channels;
+        passthrough_enabled = hdmi_backend_cfg.passthrough_enabled;
+
+        if (channels != my_data->current_backend_cfg[backend_idx].channels)
+            channels_updated = true;
     }
+
     ALOGI("%s:becf: afe: Codec selected backend: %d updated bit width: %d and sample rate: %d",
           __func__, backend_idx , bit_width, sample_rate);
 
     // Force routing if the expected bitwdith or samplerate
     // is not same as current backend comfiguration
     if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
-        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
-        *new_bit_width = bit_width;
-        *new_sample_rate = sample_rate;
+        (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+         passthrough_enabled || channels_updated) {
+        backend_cfg->bit_width = bit_width;
+        backend_cfg->sample_rate = sample_rate;
+        backend_cfg->channels = channels;
+        backend_cfg->passthrough_enabled = passthrough_enabled;
         backend_change = true;
-        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d new sample rate: %d",
-              __func__, *new_bit_width, *new_sample_rate);
+        ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d"
+               "new sample rate: %d new channels: %d",
+              __func__, backend_cfg->bit_width, backend_cfg->sample_rate, backend_cfg->channels);
     }
 
     return backend_change;
@@ -4244,40 +4424,39 @@
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
     struct audio_usecase *usecase, snd_device_t snd_device)
 {
-    unsigned int new_bit_width;
-    unsigned int new_sample_rate;
     int backend_idx = DEFAULT_CODEC_BACKEND;
     int new_snd_devices[SND_DEVICE_OUT_END];
     int i, num_devices = 1;
     bool ret = false;
     struct platform_data *my_data = (struct platform_data *)adev->platform;
-    audio_format_t format;
+    struct audio_backend_cfg backend_cfg;
 
     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;
+    backend_cfg.bit_width = usecase->stream.out->bit_width;
+    backend_cfg.sample_rate = usecase->stream.out->sample_rate;
+    backend_cfg.format = usecase->stream.out->format;
+    backend_cfg.channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
+    /*this is populated by check_codec_backend_cfg hence set default value to false*/
+    backend_cfg.passthrough_enabled = false;
 
-    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
-          ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
-          new_sample_rate, backend_idx, usecase->id,
+    ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+          ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
+          backend_cfg.sample_rate, backend_cfg.channels, backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
 
-
     if (!platform_can_split_snd_device(my_data, snd_device, &num_devices, new_snd_devices))
         new_snd_devices[0] = snd_device;
 
     for (i = 0; i < num_devices; i++) {
         ALOGI("%s: new_snd_devices[%d] is %d", __func__, i, new_snd_devices[i]);
-        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, format);
-                ret = true;
+        if ((platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
+                                             &backend_cfg))) {
+            platform_set_codec_backend_cfg(adev, new_snd_devices[i],
+                                           backend_cfg);
+            ret = true;
         }
     }
-
     return ret;
 }
 
@@ -4942,6 +5121,7 @@
 
     //reset HDMI_RX_BACKEND to default values
     my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+    my_data->current_backend_cfg[HDMI_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
     my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
 }
 
@@ -4961,90 +5141,6 @@
     return mixer_ctl_set_enum_by_string(ctl, mixer_val);
 }
 
-static int set_mixer_control(struct mixer *mixer,
-                             const char * mixer_ctl_name,
-                             const char *mixer_val)
-{
-    struct mixer_ctl *ctl;
-    ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
-    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
-    if (!ctl) {
-        ALOGE("%s: could not get ctl for mixer cmd - %s",
-              __func__, mixer_ctl_name);
-        return -EINVAL;
-    }
-
-    return mixer_ctl_set_enum_by_string(ctl, mixer_val);
-}
-
-int platform_set_hdmi_config(void *platform, uint32_t channel_count,
-                             uint32_t sample_rate, bool enable_passthrough)
-{
-    struct platform_data *my_data = (struct platform_data *)platform;
-    struct audio_device *adev = my_data->adev;
-    const char *hdmi_format_ctrl = "HDMI RX Format";
-    const char *hdmi_rate_ctrl   = "HDMI_RX SampleRate";
-    const char *hdmi_chans_ctrl  = "HDMI_RX Channels";
-    const char *channel_cnt_str  = NULL;
-
-    ALOGI("%s ch[%d] sr[%d], pthru[%d]", __func__,
-        channel_count, sample_rate, enable_passthrough);
-
-    switch (channel_count) {
-    case 8:
-        channel_cnt_str = "Eight"; break;
-    case 7:
-        channel_cnt_str = "Seven"; break;
-    case 6:
-        channel_cnt_str = "Six"; break;
-    case 5:
-        channel_cnt_str = "Five"; break;
-    case 4:
-        channel_cnt_str = "Four"; break;
-    case 3:
-        channel_cnt_str = "Three"; break;
-    default:
-        channel_cnt_str = "Two"; break;
-    }
-    ALOGV("%s: HDMI channel count: %s", __func__, channel_cnt_str);
-    set_mixer_control(adev->mixer, hdmi_chans_ctrl, channel_cnt_str);
-
-    if (enable_passthrough) {
-        ALOGD("%s:HDMI compress format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "Compr");
-    } else {
-        ALOGD("%s: HDMI PCM format", __func__);
-        set_mixer_control(adev->mixer, hdmi_format_ctrl, "LPCM");
-    }
-
-    switch (sample_rate) {
-    case 32000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_32");
-        break;
-    case 44100:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_44P1");
-        break;
-    case 96000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_96");
-        break;
-    case 128000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_128");
-        break;
-    case 176400:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_176_4");
-        break;
-    case 192000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_192");
-        break;
-    default:
-    case 48000:
-        set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_48");
-        break;
-    }
-
-    return 0;
-}
-
 int platform_set_device_params(struct stream_out *out, int param, int value)
 {
     struct audio_device *adev = out->dev;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 24274c6..48bfb2b 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -94,6 +94,8 @@
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
     SND_DEVICE_OUT_BT_SCO,
     SND_DEVICE_OUT_BT_SCO_WB,
+    SND_DEVICE_OUT_BT_A2DP,
+    SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -461,4 +463,13 @@
     char device_name[100];
     char interface_name[100];
 };
+
+struct audio_backend_cfg {
+    unsigned int   sample_rate;
+    unsigned int   channels;
+    unsigned int   bit_width;
+    bool           passthrough_enabled;
+    audio_format_t format;
+};
+
 #endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 7293485..3222e0b 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -244,6 +244,7 @@
 {
     int ret = 0;
     struct audio_usecase *uc_info;
+    struct listnode *node;
 
     ALOGD("%s: enter, out_stream_count=%d, in_stream_count=%d",
            __func__, voip_data.out_stream_count, voip_data.in_stream_count);
@@ -277,6 +278,12 @@
 
         list_remove(&uc_info->list);
         free(uc_info);
+
+        // restore device for other active usecases
+        list_for_each(node, &adev->usecase_list) {
+            uc_info = node_to_item(node, struct audio_usecase, list);
+            select_devices(adev, uc_info->id);
+        }
     } else
         ALOGV("%s: NO-OP because out_stream_count=%d, in_stream_count=%d",
                __func__, voip_data.out_stream_count, voip_data.in_stream_count);