diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index b7a7a39..3d932b0 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -147,9 +147,9 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.deep_buffer.media=true
 
-#Default pcm audio sink buffer size in msec. This is used in calculating framecount
+#QC property used when calculating client heap size in audio flinger
 PRODUCT_PROPERTY_OVERRIDES += \
-media.stagefright.audio.sink=280
+audio.heap.size.multiplier=7
 
 #enable voice path for PCM VoIP by default
 PRODUCT_PROPERTY_OVERRIDES += \
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index 3106942..96c9a5d 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -148,9 +148,9 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.deep_buffer.media=true
 
-#Default pcm audio sink buffer size in msec. This is used in calculating framecount
+#QC property used when calculating client heap size in audio flinger
 PRODUCT_PROPERTY_OVERRIDES += \
-media.stagefright.audio.sink=280
+audio.heap.size.multiplier=7
 
 #enable voice path for PCM VoIP by default
 PRODUCT_PROPERTY_OVERRIDES += \
diff --git a/configs/msm8996/msm8996.mk b/configs/msm8996/msm8996.mk
index 5b240e9..6d3fa4b 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -143,9 +143,9 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.deep_buffer.media=true
 
-#Default pcm audio sink buffer size in msec. This is used in calculating framecount
+#QC property used when calculating client heap size in audio flinger
 PRODUCT_PROPERTY_OVERRIDES += \
-media.stagefright.audio.sink=280
+audio.heap.size.multiplier=7
 
 #enable voice path for PCM VoIP by default
 PRODUCT_PROPERTY_OVERRIDES += \
diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index a1bd9a1..a8bce46 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -28,6 +28,9 @@
     <acdb_ids>
         <device name="SND_DEVICE_OUT_SPEAKER" acdb_id="15"/>
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED" acdb_id="124"/>
+        <device name="SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE" acdb_id="131"/>
+        <device name="SND_DEVICE_IN_VOICE_REC_TMIC" acdb_id="131"/>
+        <device name="SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE" acdb_id="132"/>
     </acdb_ids>
     <bit_width_configs>
         <device name="SND_DEVICE_OUT_SPEAKER" bit_width="24"/>
diff --git a/configs/msmcobalt/msmcobalt.mk b/configs/msmcobalt/msmcobalt.mk
index 12a922c..7117bd8 100644
--- a/configs/msmcobalt/msmcobalt.mk
+++ b/configs/msmcobalt/msmcobalt.mk
@@ -152,9 +152,9 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.deep_buffer.media=true
 
-#Default pcm audio sink buffer size in msec. This is used in calculating framecount
+#QC property used when calculating client heap size in audio flinger
 PRODUCT_PROPERTY_OVERRIDES += \
-media.stagefright.audio.sink=280
+audio.heap.size.multiplier=7
 
 #enable voice path for PCM VoIP by default
 PRODUCT_PROPERTY_OVERRIDES += \
@@ -214,3 +214,7 @@
 #Disable FM a2dp concurrency
 PRODUCT_PROPERTY_OVERRIDES += \
 fm.a2dp.conc.disabled=true
+
+#audio becoming noisy intent broadcast delay
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.noisy.broadcast.delay=600
diff --git a/configs/msmfalcon/msmfalcon.mk b/configs/msmfalcon/msmfalcon.mk
index 554f32b..509c159 100644
--- a/configs/msmfalcon/msmfalcon.mk
+++ b/configs/msmfalcon/msmfalcon.mk
@@ -137,9 +137,9 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 audio.deep_buffer.media=true
 
-#Default pcm audio sink buffer size in msec. This is used in calculating framecount
+#QC property used when calculating client heap size in audio flinger
 PRODUCT_PROPERTY_OVERRIDES += \
-media.stagefright.audio.sink=280
+audio.heap.size.multiplier=7
 
 #enable voice path for PCM VoIP by default
 PRODUCT_PROPERTY_OVERRIDES += \
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 9542fbd..eb3213c 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -666,6 +666,18 @@
         if ((24 == usecase->stream.out->bit_width) &&
             (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
             usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        } else if ((snd_device == SND_DEVICE_OUT_HDMI ||
+                    snd_device == SND_DEVICE_OUT_USB_HEADSET ||
+                    snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
+                   (usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) {
+             /*
+              * To best utlize DSP, check if the stream sample rate is supported/multiple of
+              * configured device sample rate, if not update the COPP rate to be equal to the
+              * device sample rate, else open COPP at stream sample rate
+              */
+              platform_check_and_update_copp_sample_rate(adev->platform, snd_device,
+                                      usecase->stream.out->sample_rate,
+                                      &usecase->stream.out->app_type_cfg.sample_rate);
         } else if ((snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
             usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) ||
             (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 13e4ae9..bf2908f 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -796,8 +796,10 @@
             audio_extn_dev_arbi_release(snd_device);
             return -EINVAL;
         }
-    } else if (platform_can_split_snd_device(adev->platform, snd_device,
-            &num_devices, new_snd_devices)) {
+    } else if (platform_split_snd_device(adev->platform,
+                                         snd_device,
+                                         &num_devices,
+                                         new_snd_devices) == 0) {
         for (i = 0; i < num_devices; i++) {
             enable_snd_device(adev, new_snd_devices[i]);
         }
@@ -870,8 +872,10 @@
         if (platform_can_enable_spkr_prot_on_device(snd_device) &&
              audio_extn_spkr_prot_is_enabled()) {
             audio_extn_spkr_prot_stop_processing(snd_device);
-        } else if (platform_can_split_snd_device(adev->platform, snd_device,
-                    &num_devices, new_snd_devices)) {
+        } else if (platform_split_snd_device(adev->platform,
+                                             snd_device,
+                                             &num_devices,
+                                             new_snd_devices) == 0) {
             for (i = 0; i < num_devices; i++) {
                 disable_snd_device(adev, new_snd_devices[i]);
             }
@@ -908,6 +912,115 @@
     return 0;
 }
 
+/*
+  legend:
+  uc - existing usecase
+  new_uc - new usecase
+  d1, d11, d2 - SND_DEVICE enums
+  a1, a2 - corresponding ANDROID device enums
+  B1, B2 - backend strings
+
+case 1
+  uc->dev  d1 (a1)               B1
+  new_uc->dev d1 (a1), d2 (a2)   B1, B2
+
+  resolution: disable and enable uc->dev on d1
+
+case 2
+  uc->dev d1 (a1)        B1
+  new_uc->dev d11 (a1)   B1
+
+  resolution: need to switch uc since d1 and d11 are related
+  (e.g. speaker and voice-speaker)
+  use ANDROID_DEVICE_OUT enums to match devices since SND_DEVICE enums may vary
+
+case 3
+  uc->dev d1 (a1)        B1
+  new_uc->dev d2 (a2)    B2
+
+  resolution: no need to switch uc
+
+case 4
+  uc->dev d1 (a1)      B1
+  new_uc->dev d2 (a2)  B1
+
+  resolution: disable enable uc-dev on d2 since backends match
+  we cannot enable two streams on two different devices if they
+  share the same backend. e.g. if offload is on speaker device using
+  QUAD_MI2S backend and a low-latency stream is started on voice-handset
+  using the same backend, offload must also be switched to voice-handset.
+
+case 5
+  uc->dev  d1 (a1)                  B1
+  new_uc->dev d1 (a1), d2 (a2)      B1
+
+  resolution: disable enable uc-dev on d2 since backends match
+  we cannot enable two streams on two different devices if they
+  share the same backend.
+
+case 6
+  uc->dev  d1 (a1)    B1
+  new_uc->dev d2 (a1) B2
+
+  resolution: no need to switch
+
+case 7
+  uc->dev d1 (a1), d2 (a2)       B1, B2
+  new_uc->dev d1 (a1)            B1
+
+  resolution: no need to switch
+
+*/
+static snd_device_t derive_playback_snd_device(void * platform,
+                                               struct audio_usecase *uc,
+                                               struct audio_usecase *new_uc,
+                                               snd_device_t new_snd_device)
+{
+    audio_devices_t a1 = uc->stream.out->devices;
+    audio_devices_t a2 = new_uc->stream.out->devices;
+
+    snd_device_t d1 = uc->out_snd_device;
+    snd_device_t d2 = new_snd_device;
+
+    // Treat as a special case when a1 and a2 are not disjoint
+    if ((a1 != a2) && (a1 & a2)) {
+        snd_device_t d3[2];
+        int num_devices = 0;
+        int ret = platform_split_snd_device(platform,
+                                            popcount(a1) > 1 ? d1 : d2,
+                                            &num_devices,
+                                            d3);
+        if (ret < 0) {
+            if (ret != -ENOSYS) {
+                ALOGW("%s failed to split snd_device %d",
+                      __func__,
+                      popcount(a1) > 1 ? d1 : d2);
+            }
+            goto end;
+        }
+
+        // NB: case 7 is hypothetical and isn't a practical usecase yet.
+        // But if it does happen, we need to give priority to d2 if
+        // the combo devices active on the existing usecase share a backend.
+        // This is because we cannot have a usecase active on a combo device
+        // and a new usecase requests one device in this combo pair.
+        if (platform_check_backends_match(d3[0], d3[1])) {
+            return d2; // case 5
+        } else {
+            return d1; // case 1
+        }
+    } else {
+        if (platform_check_backends_match(d1, d2)) {
+            return d2; // case 2, 4
+        } else {
+            return d1; // case 6, 3
+        }
+    }
+
+end:
+    return d2; // return whatever was calculated before.
+}
+
 static void check_usecases_codec_backend(struct audio_device *adev,
                                               struct audio_usecase *uc_info,
                                               snd_device_t snd_device)
@@ -963,7 +1076,9 @@
               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) &&
+            (derive_playback_snd_device(adev->platform,
+                                        usecase, uc_info,
+                                        snd_device) != usecase->out_snd_device || force_routing) &&
             ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
              (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
              (usecase->devices & AUDIO_DEVICE_OUT_USB_DEVICE) ||
diff --git a/hal/edid.c b/hal/edid.c
index 8f183d0..e889530 100644
--- a/hal/edid.c
+++ b/hal/edid.c
@@ -86,57 +86,109 @@
     return format_str;
 }
 
-static int get_edid_sf(unsigned char byte)
+static bool is_supported_sr(unsigned char sr_byte, int sampling_rate )
+{
+    int result = 0;
+
+    ALOGV("%s: sr_byte: %d, sampling_freq: %d",__func__, sr_byte, sampling_rate);
+    switch (sampling_rate) {
+    case 192000:
+        result = (sr_byte & BIT(6));
+        break;
+    case 176400:
+        result = (sr_byte & BIT(5));
+        break;
+    case 96000:
+        result = (sr_byte & BIT(4));
+        break;
+    case 88200:
+        result = (sr_byte & BIT(3));
+        break;
+    case 48000:
+        result = (sr_byte & BIT(2));
+        break;
+    case 44100:
+        result = (sr_byte & BIT(1));
+        break;
+    case 32000:
+        result = (sr_byte & BIT(0));
+        break;
+     default:
+        break;
+    }
+
+    if (result)
+        return true;
+
+    return false;
+}
+
+static unsigned char get_edid_bps_byte(unsigned char byte,
+                        unsigned char format)
+{
+    if (format == 0) {
+        ALOGV("%s: not lpcm format, return 0",__func__);
+        return 0;
+    }
+    return byte;
+}
+
+static bool is_supported_bps(unsigned char bps_byte, int bps)
+{
+    int result = 0;
+
+    switch (bps) {
+    case 24:
+        ALOGV("24bit");
+        result = (bps_byte & BIT(2));
+        break;
+    case 20:
+        ALOGV("20bit");
+        result = (bps_byte & BIT(1));
+        break;
+    case 16:
+        ALOGV("16bit");
+        result = (bps_byte & BIT(0));
+        break;
+     default:
+        break;
+    }
+
+    if (result)
+        return true;
+
+    return false;
+}
+
+static int get_highest_edid_sf(unsigned char byte)
 {
     int nfreq = 0;
 
     if (byte & BIT(6)) {
-        ALOGV("192kHz");
+        ALOGV("Highest: 192kHz");
         nfreq = 192000;
     } else if (byte & BIT(5)) {
-        ALOGV("176kHz");
+        ALOGV("Highest: 176kHz");
         nfreq = 176000;
     } else if (byte & BIT(4)) {
-        ALOGV("96kHz");
+        ALOGV("Highest: 96kHz");
         nfreq = 96000;
     } else if (byte & BIT(3)) {
-        ALOGV("88.2kHz");
+        ALOGV("Highest: 88.2kHz");
         nfreq = 88200;
     } else if (byte & BIT(2)) {
-        ALOGV("48kHz");
+        ALOGV("Highest: 48kHz");
         nfreq = 48000;
     } else if (byte & BIT(1)) {
-        ALOGV("44.1kHz");
+        ALOGV("Highest: 44.1kHz");
         nfreq = 44100;
     } else if (byte & BIT(0)) {
-        ALOGV("32kHz");
+        ALOGV("Highest: 32kHz");
         nfreq = 32000;
     }
     return nfreq;
 }
 
-static int get_edid_bps(unsigned char byte,
-                        unsigned char format)
-{
-    int bits_per_sample = 0;
-    if (format == 1) {
-        if (byte & BIT(2)) {
-            ALOGV("24bit");
-            bits_per_sample = 24;
-        } else if (byte & BIT(1)) {
-            ALOGV("20bit");
-            bits_per_sample = 20;
-        } else if (byte & BIT(0)) {
-            ALOGV("16bit");
-            bits_per_sample = 16;
-        }
-    } else {
-        ALOGV("not lpcm format, return 0");
-        return 0;
-    }
-    return bits_per_sample;
-}
-
 static void update_channel_map(edid_audio_info* info)
 {
     /* HDMI Cable follows CEA standard so SAD is received in CEA
@@ -589,8 +641,8 @@
     for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
         ALOGV("%s:FormatId:%d rate:%d bps:%d channels:%d", __func__,
               info->audio_blocks_array[i].format_id,
-              info->audio_blocks_array[i].sampling_freq,
-              info->audio_blocks_array[i].bits_per_sample,
+              info->audio_blocks_array[i].sampling_freq_bitmask,
+              info->audio_blocks_array[i].bits_per_sample_bitmask,
               info->audio_blocks_array[i].channels);
     }
     ALOGV("%s:no of audio blocks:%d", __func__, info->audio_blocks);
@@ -670,16 +722,16 @@
         ALOGD("info->audio_blocks_array[i].format_id %s",
               edid_format_to_str(formats[i]));
 
-        ALOGV("Frequency Byte %d\n", frequency[i]);
-        info->audio_blocks_array[i].sampling_freq = get_edid_sf(frequency[i]);
-        ALOGV("info->audio_blocks_array[i].sampling_freq %d",
-              info->audio_blocks_array[i].sampling_freq);
+        ALOGV("Frequency Bitmask %d\n", frequency[i]);
+        info->audio_blocks_array[i].sampling_freq_bitmask = frequency[i];
+        ALOGV("info->audio_blocks_array[i].sampling_freq_bitmask %d",
+              info->audio_blocks_array[i].sampling_freq_bitmask);
 
-        ALOGV("BitsPerSample Byte %d\n", bitrate[i]);
-        info->audio_blocks_array[i].bits_per_sample =
-                   get_edid_bps(bitrate[i],formats[i]);
-        ALOGV("info->audio_blocks_array[i].bits_per_sample %d",
-              info->audio_blocks_array[i].bits_per_sample);
+        ALOGV("BitsPerSample Bitmask %d\n", bitrate[i]);
+        info->audio_blocks_array[i].bits_per_sample_bitmask =
+                   get_edid_bps_byte(bitrate[i],formats[i]);
+        ALOGV("info->audio_blocks_array[i].bits_per_sample_bitmask %d",
+              info->audio_blocks_array[i].bits_per_sample_bitmask);
     }
     dump_speaker_allocation(info);
     dump_edid_data(info);
@@ -691,7 +743,7 @@
     int i = 0;
     if (info != NULL && sr != 0) {
         for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
-            if (info->audio_blocks_array[i].sampling_freq == sr) {
+        if (is_supported_sr(info->audio_blocks_array[i].sampling_freq_bitmask , sr)) {
                 ALOGV("%s: returns true for sample rate [%d]",
                       __func__, sr);
                 return true;
@@ -715,7 +767,7 @@
 
     if (info != NULL && bps != 0) {
         for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
-            if (info->audio_blocks_array[i].bits_per_sample == bps) {
+            if (is_supported_bps(info->audio_blocks_array[i].bits_per_sample_bitmask, bps)) {
                 ALOGV("%s: returns true for bit width [%d]",
                       __func__, bps);
                 return true;
@@ -726,3 +778,24 @@
            __func__, bps);
     return false;
 }
+
+int edid_get_highest_supported_sr(edid_audio_info* info)
+{
+    int sr = 0;
+    int highest_sr = 0;
+    int i;
+
+    if (info != NULL) {
+        for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
+          sr = get_highest_edid_sf(info->audio_blocks_array[i].sampling_freq_bitmask);
+          if (sr > highest_sr)
+            highest_sr = sr;
+        }
+    }
+    else
+        ALOGE("%s: info is NULL", __func__);
+
+    ALOGV("%s: returns [%d] for highest supported sr",
+        __func__, highest_sr);
+    return highest_sr;
+}
\ No newline at end of file
diff --git a/hal/edid.h b/hal/edid.h
index 387b17e..6a82103 100644
--- a/hal/edid.h
+++ b/hal/edid.h
@@ -79,8 +79,8 @@
 
 typedef struct edid_audio_block_info {
     edid_audio_format_id format_id;
-    int sampling_freq;
-    int bits_per_sample;
+    int sampling_freq_bitmask;
+    int bits_per_sample_bitmask;
     int channels;
 } edid_audio_block_info;
 
@@ -100,5 +100,6 @@
 
 bool edid_is_supported_sr(edid_audio_info* info, int sr);
 bool edid_is_supported_bps(edid_audio_info* info, int bps);
+int edid_get_highest_supported_sr(edid_audio_info* info);
 
 #endif /* EDID_H */
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index a42f984..9416887 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -70,7 +70,8 @@
 #define LIB_ACDB_LOADER "libacdbloader.so"
 #define CVD_VERSION_MIXER_CTL "CVD Version"
 
-#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024 * 1024)
 #define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
@@ -255,6 +256,8 @@
     int metainfo_key;
     int source_mic_type;
     int max_mic_count;
+    bool is_dsd_supported;
+    bool is_asrc_supported;
 };
 
 static bool is_external_codec = false;
@@ -335,6 +338,7 @@
     [SND_DEVICE_OUT_SPEAKER_VBAT] = "vbat-speaker",
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
     [SND_DEVICE_OUT_HEADPHONES] = "headphones",
+    [SND_DEVICE_OUT_HEADPHONES_DSD] = "headphones-dsd",
     [SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
     [SND_DEVICE_OUT_LINE] = "line",
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
@@ -444,6 +448,7 @@
     [SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = "quad-mic",
     [SND_DEVICE_IN_THREE_MIC] = "three-mic",
     [SND_DEVICE_IN_HANDSET_TMIC] = "three-mic",
+    [SND_DEVICE_IN_VOICE_REC_TMIC] = "three-mic",
     [SND_DEVICE_IN_UNPROCESSED_MIC] = "unprocessed-mic",
     [SND_DEVICE_IN_UNPROCESSED_STEREO_MIC] = "voice-rec-dmic-ef",
     [SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = "three-mic",
@@ -466,6 +471,7 @@
     [SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
     [SND_DEVICE_OUT_LINE] = 10,
     [SND_DEVICE_OUT_HEADPHONES] = 10,
+    [SND_DEVICE_OUT_HEADPHONES_DSD] = 10,
     [SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
     [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
     [SND_DEVICE_OUT_SPEAKER_AND_LINE] = 10,
@@ -573,6 +579,7 @@
     [SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = 129,
     [SND_DEVICE_IN_THREE_MIC] = 46, /* for APSS Surround Sound Recording */
     [SND_DEVICE_IN_HANDSET_TMIC] = 125, /* for 3mic recording with fluence */
+    [SND_DEVICE_IN_VOICE_REC_TMIC] = 125,
     [SND_DEVICE_IN_UNPROCESSED_MIC] = 143,
     [SND_DEVICE_IN_UNPROCESSED_STEREO_MIC] = 144,
     [SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = 145,
@@ -597,6 +604,7 @@
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_VBAT)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_DSD)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_LINE)},
     {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
@@ -704,6 +712,7 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS)},
     {TO_NAME_INDEX(SND_DEVICE_IN_THREE_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_TMIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_STEREO_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_THREE_MIC)},
@@ -1245,12 +1254,15 @@
     backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
     backend_tag_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
     backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
+    backend_tag_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("headphones-dsd");
     backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
     backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("vbat-voice-speaker");
     backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("vbat-voice-speaker-2");
     backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
     backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
 
+    hw_interface_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("SLIMBUS_2_RX");
+    hw_interface_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("SLIMBUS_5_RX");
     hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
     hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
     hw_interface_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("DISPLAY_PORT_RX");
@@ -1935,6 +1947,10 @@
         my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
             strdup("SLIM_0_RX SampleRate");
 
+    my_data->current_backend_cfg[DSD_NATIVE_BACKEND].bitwidth_mixer_ctl =
+        strdup("SLIM_2_RX Format");
+    my_data->current_backend_cfg[DSD_NATIVE_BACKEND].samplerate_mixer_ctl =
+        strdup("SLIM_2_RX SampleRate");
         my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].bitwidth_mixer_ctl =
             strdup("SLIM_5_RX Format");
         my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
@@ -2008,6 +2024,12 @@
         }
     }
 
+    if(strstr(snd_card_name, "tavil")) {
+        ALOGD("%s:DSD playback is supported", __func__);
+        my_data->is_dsd_supported = true;
+        my_data->is_asrc_supported = true;
+        platform_set_native_support(NATIVE_AUDIO_MODE_MULTIPLE_44_1);
+    }
     my_data->edid_info = NULL;
     return my_data;
 }
@@ -2338,7 +2360,8 @@
 }
 int platform_set_native_support(int na_mode)
 {
-    if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode) {
+    if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode
+        || NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode) {
         na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled = true;
         na_props.na_mode = na_mode;
         ALOGD("%s:napb: native audio playback enabled in (%s) mode v2.0", __func__,
@@ -2351,6 +2374,16 @@
 
     return 0;
 }
+bool platform_check_codec_dsd_support(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    return my_data->is_dsd_supported;
+}
+bool platform_check_codec_asrc_support(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    return my_data->is_asrc_supported;
+}
 
 int platform_get_native_support()
 {
@@ -2404,6 +2437,8 @@
             mode = NATIVE_AUDIO_MODE_SRC;
         else if (value && !strncmp(value, "true", sizeof("true")))
             mode = NATIVE_AUDIO_MODE_TRUE_44_1;
+        else if (value && !strncmp(value, "multiple", sizeof("multiple")))
+            mode = NATIVE_AUDIO_MODE_MULTIPLE_44_1;
         else {
             mode = NATIVE_AUDIO_MODE_INVALID;
             ALOGE("%s:napb:native_audio_mode in platform info xml,invalid mode string",
@@ -2514,6 +2549,9 @@
                 if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
                             sizeof("headphones-44.1")) == 0)
                         port = HEADPHONE_44_1_BACKEND;
+                else if (strncmp(backend_tag_table[snd_device], "headphones-dsd",
+                            sizeof("headphones-dsd")) == 0)
+                        port = DSD_NATIVE_BACKEND;
                 else if (strncmp(backend_tag_table[snd_device], "headphones",
                             sizeof("headphones")) == 0)
                         port = HEADPHONE_BACKEND;
@@ -2556,7 +2594,8 @@
         snd_device = usecase->in_snd_device;
     acdb_dev_id = acdb_device_table[platform_get_spkr_prot_snd_device(snd_device)];
 
-    if(!platform_can_split_snd_device(platform, snd_device, &num_devices, new_snd_device)) {
+    if (platform_split_snd_device(platform, snd_device, &num_devices,
+                                  new_snd_device) < 0) {
         new_snd_device[0] = snd_device;
     }
 
@@ -2858,22 +2897,21 @@
     return ret;
 }
 
-bool platform_can_split_snd_device(void *platform,
-                                   snd_device_t snd_device,
-                                   int *num_devices,
-                                   snd_device_t *new_snd_devices)
+int platform_split_snd_device(void *platform,
+                              snd_device_t snd_device,
+                              int *num_devices,
+                              snd_device_t *new_snd_devices)
 {
-    bool status = false;
+    int ret = -EINVAL;
     struct platform_data *my_data = (struct platform_data *)platform;
-
     if (NULL == num_devices || NULL == new_snd_devices) {
         ALOGE("%s: NULL pointer ..", __func__);
-        return false;
+        return -EINVAL;
     }
 
     /*
      * If wired headset/headphones/line devices share the same backend
-     * with speaker/earpiece this routine returns false.
+     * with speaker/earpiece this routine returns -EINVAL.
      */
     if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES &&
         !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_HEADPHONES)) {
@@ -2887,7 +2925,7 @@
              new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
 
         new_snd_devices[1] = SND_DEVICE_OUT_HEADPHONES;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_HDMI &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_HDMI)) {
         *num_devices = 2;
@@ -2900,7 +2938,7 @@
             new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
 
         new_snd_devices[1] = SND_DEVICE_OUT_HDMI;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_DISPLAY_PORT)) {
         *num_devices = 2;
@@ -2913,24 +2951,24 @@
             new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
 
         new_snd_devices[1] = SND_DEVICE_OUT_DISPLAY_PORT;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_USB_HEADSET)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_USB_HEADSET;
-        status = true;
+        ret = 0;
     } else if (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
-        status = true;
+        ret = 0;
     }
 
     ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
         snd_device, *num_devices, *new_snd_devices);
 
-    return status;
+    return ret;
 }
 
 int platform_get_ext_disp_type(void *platform)
@@ -3134,7 +3172,8 @@
     }
 
     if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
-        devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+        devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+        devices & AUDIO_DEVICE_OUT_LINE) {
         if (OUTPUT_SAMPLING_RATE_44100 == sample_rate &&
             NATIVE_AUDIO_MODE_SRC == na_mode &&
             !audio_extn_get_anc_enabled()) {
@@ -3155,7 +3194,16 @@
                 else
                     snd_device = SND_DEVICE_OUT_ANC_HEADSET;
             }
-        } else {
+        } else if (NATIVE_AUDIO_MODE_SRC == na_mode &&
+                   OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
+                snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+        } else if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode &&
+                   (sample_rate % OUTPUT_SAMPLING_RATE_44100 == 0) &&
+                   (out->format != AUDIO_FORMAT_DSD)) {
+                snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+        } else if (out->format == AUDIO_FORMAT_DSD) {
+                snd_device = SND_DEVICE_OUT_HEADPHONES_DSD;
+        }  else {
 #ifdef RECORD_PLAY_CONCURRENCY
             if (use_voip_out_devices)
                 snd_device = SND_DEVICE_OUT_VOIP_HEADPHONES;
@@ -3369,7 +3417,7 @@
                      snd_device = SND_DEVICE_IN_HANDSET_QMIC;
                 } else if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
                     (my_data->source_mic_type & SOURCE_THREE_MIC)) {
-                    snd_device = SND_DEVICE_IN_HANDSET_TMIC;
+                    snd_device = SND_DEVICE_IN_VOICE_REC_TMIC;
                 } else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
                     (my_data->source_mic_type & SOURCE_DUAL_MIC)) {
                     snd_device = SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE;
@@ -4217,18 +4265,21 @@
         fragment_size = info->offload_buffer_size;
     }
 
-    // For FLAC use max size since it is loss less, and has sampling rates
-    // upto 192kHZ
-    if (info != NULL && !info->has_video &&
-        info->format == AUDIO_FORMAT_FLAC) {
-       fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
-       ALOGV("FLAC fragment size %d", fragment_size);
-    }
-
-    if (info != NULL && info->has_video && info->is_streaming) {
-        fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
-        ALOGV("%s: offload fragment size reduced for AV streaming to %d",
-               __func__, fragment_size);
+    if (info != NULL && !info->has_video) {
+        if (info->is_streaming) {
+            fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
+            ALOGV("%s: offload fragment size reduced for AV streaming to %d",
+                   __func__, fragment_size);
+        } else if (info->format == AUDIO_FORMAT_FLAC) {
+            fragment_size = FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+            ALOGV("FLAC fragment size %d", fragment_size);
+        } else if (info->format == AUDIO_FORMAT_DSD) {
+            fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+            if((property_get("audio.native.dsd.buffer.size.kb", value, "")) &&
+                    atoi(value))
+                fragment_size =  atoi(value) * 1024;
+            ALOGV("DSD fragment size %d", fragment_size);
+        }
     }
 
     fragment_size = ALIGN( fragment_size, 1024);
@@ -4457,6 +4508,17 @@
             mixer_ctl_set_enum_by_string(ctl, "LPCM");
         }
     }
+    if (snd_device == SND_DEVICE_OUT_HEADPHONES || snd_device ==
+        SND_DEVICE_OUT_HEADPHONES_44_1) {
+        if (sample_rate > 48000 ||
+            (bit_width >= 24 && (sample_rate == 48000  || sample_rate == 44100))) {
+            ALOGV("%s: apply HPH HQ mode\n", __func__);
+            audio_route_apply_and_update_path(adev->audio_route, "hph-highquality-mode");
+        } else {
+            ALOGV("%s: apply HPH LP mode\n", __func__);
+            audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
+        }
+    }
 
     return ret;
 }
@@ -4500,8 +4562,12 @@
 
         //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 to see if current BE sample rate is supported by EDID
+            //else assign the highest sample rate supported by EDID
+            if (edid_is_supported_sr(edid_info,my_data->current_backend_cfg[backend_idx].sample_rate))
+                sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
+            else
+                sample_rate = edid_get_highest_supported_sr(edid_info);
         }
 
         //Check EDID info for supported bit width
@@ -4553,6 +4619,7 @@
     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();
     bool channels_updated = false;
 
     backend_idx = platform_get_backend_index(snd_device);
@@ -4572,6 +4639,7 @@
               __func__);
         bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
         sample_rate =  CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        channels = CODEC_BACKEND_DEFAULT_CHANNELS;
     } else {
         /*
          * The backend should be configured at highest bit width and/or
@@ -4633,9 +4701,31 @@
                  ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
                        __func__);
             }
-        } else if (OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
-                 sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
-                 ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
+        } else if (na_mode != NATIVE_AUDIO_MODE_MULTIPLE_44_1) {
+            /*
+             * Map native sampling rates to upper limit range
+             * if multiple of native sampling rates are not supported.
+             * This check also indicates that this is not tavil codec
+             * And 32bit/384kHz is only supported on tavil
+             * Hence reset 32b/384kHz to 24b/192kHz.
+             */
+            switch (sample_rate) {
+                case 44100:
+                    sample_rate = 48000;
+                    break;
+                case 88200:
+                    sample_rate = 96000;
+                    break;
+                case 176400:
+                case 352800:
+                case 384000:
+                    sample_rate = 192000;
+                    break;
+            }
+            if (bit_width > 24)
+                bit_width = 24;
+
+            ALOGD("%s:becf: afe: napb not active - set non fractional rate",
                        __func__);
         }
     } else if ((usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) ||
@@ -4748,13 +4838,24 @@
     /*this is populated by check_codec_backend_cfg hence set default value to false*/
     backend_cfg.passthrough_enabled = false;
 
+    /* Set Backend sampling rate to 176.4 for DSD64 and
+     * 352.8Khz for DSD128.
+     * Set Bit Width to 16
+     */
+    if ((backend_idx == DSD_NATIVE_BACKEND) && (backend_cfg.format == AUDIO_FORMAT_DSD)) {
+        backend_cfg.bit_width = 16;
+        if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD64)
+            backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+        else if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD128)
+            backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+    }
     ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
           ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
           backend_cfg.sample_rate,  backend_cfg.channels, backend_idx, usecase->id,
           platform_get_snd_device_name(snd_device));
 
-    if (!platform_can_split_snd_device(adev->platform, snd_device,
-            &num_devices, new_snd_devices))
+    if (platform_split_snd_device(adev->platform, snd_device,
+                                  &num_devices, new_snd_devices) < 0)
         new_snd_devices[0] = snd_device;
 
     for (i = 0; i < num_devices; i++) {
@@ -5016,6 +5117,29 @@
     return ret;
 }
 
+void platform_check_and_update_copp_sample_rate(void* platform, snd_device_t snd_device,
+                                                unsigned int stream_sr, int* sample_rate)
+{
+    struct platform_data* my_data = (struct platform_data *)platform;
+    int backend_idx = platform_get_backend_index(snd_device);
+    int device_sr = my_data->current_backend_cfg[backend_idx].sample_rate;
+    /*Check if device SR is multiple of 8K or 11.025 Khz
+     *check if the stream SR is multiple of same base, if not set
+     *copp sample rate equal to device sample rate.
+     */
+     if (!(((sample_rate_multiple(device_sr, SAMPLE_RATE_8000)) &&
+                 (sample_rate_multiple(stream_sr, SAMPLE_RATE_8000))) ||
+           ((sample_rate_multiple(device_sr, SAMPLE_RATE_11025)) &&
+                 (sample_rate_multiple(stream_sr, SAMPLE_RATE_11025))))) {
+         *sample_rate = device_sr;
+     } else
+         *sample_rate = stream_sr;
+
+     ALOGI("sn_device %d device sr %d stream sr %d copp sr %d", snd_device, device_sr, stream_sr
+, *sample_rate);
+
+}
+
 int platform_get_edid_info(void *platform)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
@@ -5275,24 +5399,13 @@
 {
     struct platform_data *my_data = (struct platform_data *)platform;
     edid_audio_info *info = NULL;
-    int i, ret;
+    int ret = 0;
 
     ret = platform_get_edid_info(platform);
     info = (edid_audio_info *)my_data->edid_info;
     if (ret == 0 && info != NULL) {
-        for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
-             /*
-              * To check
-              *  is there any special for CONFIG_HDMI_PASSTHROUGH_CONVERT
-              *  & DOLBY_DIGITAL_PLUS
-              */
-            if (info->audio_blocks_array[i].sampling_freq == sample_rate) {
-                ALOGV("%s: returns true %d", __func__, sample_rate);
-                return true;
-            }
-        }
+        return edid_is_supported_sr(info, sample_rate);
     }
-    ALOGV("%s: returns false %d", __func__, sample_rate);
 
     return false;
 }
@@ -5631,16 +5744,6 @@
    return;
 }
 
-bool platform_check_codec_dsd_support(void *platform __unused)
-{
-    return false;
-}
-
-bool platform_check_codec_asrc_support(void *platform __unused)
-{
-    return false;
-}
-
 int platform_send_audio_cal(void* platform __unused,
         int acdb_dev_id __unused, int acdb_device_type __unused,
         int app_type __unused, int topology_id __unused,
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index cba9068..a0ae7d0 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -88,6 +88,7 @@
     SND_DEVICE_OUT_SPEAKER_VBAT,
     SND_DEVICE_OUT_LINE,
     SND_DEVICE_OUT_HEADPHONES,
+    SND_DEVICE_OUT_HEADPHONES_DSD,
     SND_DEVICE_OUT_HEADPHONES_44_1,
     SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
     SND_DEVICE_OUT_SPEAKER_AND_LINE,
@@ -202,6 +203,7 @@
     SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS,
     SND_DEVICE_IN_THREE_MIC,
     SND_DEVICE_IN_HANDSET_TMIC,
+    SND_DEVICE_IN_VOICE_REC_TMIC,
     SND_DEVICE_IN_UNPROCESSED_MIC,
     SND_DEVICE_IN_UNPROCESSED_STEREO_MIC,
     SND_DEVICE_IN_UNPROCESSED_THREE_MIC,
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index e025772..33bbc2f 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1237,12 +1237,14 @@
 int platform_spkr_prot_is_wsa_analog_mode(void *adev __unused)
 {
     return 0;
-bool platform_can_split_snd_device(void *platform __unused,
-                                   snd_device_t in_snd_device __unused,
-                                   int *num_devices __unused,
-                                   snd_device_t *out_snd_devices __unused)
+
+}
+
+int platform_can_split_snd_device(snd_device_t in_snd_device __unused,
+                                  int *num_devices __unused,
+                                  snd_device_t *out_snd_devices __unused)
 {
-    return false;
+    return -ENOSYS;
 }
 
 bool platform_check_backends_match(snd_device_t snd_device1 __unused,
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index e947f91..3551006 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -446,6 +446,7 @@
     [SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = "quad-mic",
     [SND_DEVICE_IN_THREE_MIC] = "three-mic",
     [SND_DEVICE_IN_HANDSET_TMIC] = "three-mic",
+    [SND_DEVICE_IN_VOICE_REC_TMIC] = "three-mic",
     [SND_DEVICE_IN_UNPROCESSED_MIC] = "unprocessed-mic",
     [SND_DEVICE_IN_UNPROCESSED_STEREO_MIC] = "voice-rec-dmic-ef",
     [SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = "three-mic",
@@ -570,6 +571,7 @@
     [SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = 125,
     [SND_DEVICE_IN_THREE_MIC] = 46, /* for APSS Surround Sound Recording */
     [SND_DEVICE_IN_HANDSET_TMIC] = 125, /* for 3mic recording with fluence */
+    [SND_DEVICE_IN_VOICE_REC_TMIC] = 125,
     [SND_DEVICE_IN_UNPROCESSED_MIC] = 143,
     [SND_DEVICE_IN_UNPROCESSED_STEREO_MIC] = 144,
     [SND_DEVICE_IN_UNPROCESSED_THREE_MIC] = 145,
@@ -693,6 +695,7 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE)},
     {TO_NAME_INDEX(SND_DEVICE_IN_THREE_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_TMIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_STEREO_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_THREE_MIC)},
@@ -2393,8 +2396,8 @@
         return -EINVAL;
     }
 
-    if(!platform_can_split_snd_device(my_data, snd_device,
-            &num_devices, new_snd_device)) {
+    if (platform_split_snd_device(my_data, snd_device,
+                                  &num_devices, new_snd_device) < 0) {
         new_snd_device[0] = snd_device;
     }
 
@@ -2697,58 +2700,58 @@
     return ret;
 }
 
-bool platform_can_split_snd_device(void *platform,
-                                   snd_device_t snd_device,
-                                   int *num_devices,
-                                   snd_device_t *new_snd_devices)
+int platform_split_snd_device(void *platform,
+                              snd_device_t snd_device,
+                              int *num_devices,
+                              snd_device_t *new_snd_devices)
 {
-    bool status = false;
+    int ret = -EINVAL;
     struct platform_data *my_data = (struct platform_data *)platform;
 
     if ( NULL == num_devices || NULL == new_snd_devices || NULL == my_data) {
         ALOGE("%s: NULL pointer ..", __func__);
-        return false;
+        return -EINVAL;
     }
 
     /*
      * If wired headset/headphones/line devices share the same backend
-     * with speaker/earpiece this routine returns false.
+     * with speaker/earpiece this routine returns -EINVAL.
      */
     if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES &&
         !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_HEADPHONES)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_HEADPHONES;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_HDMI &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_HDMI)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_HDMI;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_DISPLAY_PORT)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_DISPLAY_PORT;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_USB_HEADSET)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_USB_HEADSET;
-        status = true;
+        ret = 0;
     } else if (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
-        status = true;
+        ret = 0;
     }
 
     ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
         snd_device, *num_devices, *new_snd_devices);
 
-    return status;
+    return ret;
 }
 
 int platform_get_ext_disp_type(void *platform)
@@ -3138,7 +3141,7 @@
                      snd_device = SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE;
                 } else if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
                     (my_data->source_mic_type & SOURCE_THREE_MIC)) {
-                    snd_device = SND_DEVICE_IN_HANDSET_TMIC;
+                    snd_device = SND_DEVICE_IN_VOICE_REC_TMIC;
                 } else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
                     (my_data->source_mic_type & SOURCE_DUAL_MIC)) {
                     snd_device = SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE;
@@ -4562,8 +4565,12 @@
 
         //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 to see if current BE sample rate is supported by EDID
+            //else assign the highest sample rate supported by EDID
+            if (edid_is_supported_sr(edid_info,my_data->current_backend_cfg[backend_idx].sample_rate))
+                sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
+            else
+                sample_rate = edid_get_highest_supported_sr(edid_info);
         }
 
         //Check EDID info for supported bit width
@@ -4829,7 +4836,8 @@
           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))
+    if (platform_split_snd_device(my_data, snd_device, &num_devices,
+                                  new_snd_devices) < 0)
         new_snd_devices[0] = snd_device;
 
     for (i = 0; i < num_devices; i++) {
@@ -5299,6 +5307,32 @@
     return fragment_size;
 }
 
+void platform_check_and_update_copp_sample_rate(void* platform, snd_device_t snd_device,
+                                                unsigned int stream_sr, int* sample_rate)
+{
+    struct platform_data* my_data = (struct platform_data *)platform;
+    int backend_idx = platform_get_backend_index(snd_device);
+    int device_sr = my_data->current_backend_cfg[backend_idx].sample_rate;
+    /*
+     *Check if device SR is multiple of 8K or 11.025 Khz
+     *check if the stream SR is multiple of same base, if yes
+     *then have copp SR equal to stream SR, this ensures that
+     *post processing happens at stream SR, else have
+     *copp SR equal to device SR.
+     */
+     if (!(((sample_rate_multiple(device_sr, SAMPLE_RATE_8000)) &&
+                 (sample_rate_multiple(stream_sr, SAMPLE_RATE_8000))) ||
+           ((sample_rate_multiple(device_sr, SAMPLE_RATE_11025)) &&
+                 (sample_rate_multiple(stream_sr, SAMPLE_RATE_11025))))) {
+         *sample_rate = device_sr;
+     } else
+         *sample_rate = stream_sr;
+
+     ALOGI("sn_device %d device sr %d stream sr %d copp sr %d", snd_device, device_sr, stream_sr
+, *sample_rate);
+
+}
+
 void platform_reset_edid_info(void *platform) {
 
     ALOGV("%s:", __func__);
@@ -5347,24 +5381,13 @@
 {
     struct platform_data *my_data = (struct platform_data *)platform;
     edid_audio_info *info = NULL;
-    int i, ret;
+    int ret = 0;
 
     ret = platform_get_edid_info(platform);
     info = (edid_audio_info *)my_data->edid_info;
     if (ret == 0 && info != NULL) {
-        for (i = 0; i < info->audio_blocks && i < MAX_EDID_BLOCKS; i++) {
-             /*
-              * To check
-              *  is there any special for CONFIG_HDMI_PASSTHROUGH_CONVERT
-              *  & DOLBY_DIGITAL_PLUS
-              */
-            if (info->audio_blocks_array[i].sampling_freq == sample_rate) {
-                ALOGV("%s: returns true %d", __func__, sample_rate);
-                return true;
-            }
-        }
+        return edid_is_supported_sr(info, sample_rate);
     }
-    ALOGV("%s: returns false %d", __func__, sample_rate);
 
     return false;
 }
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index c231843..fcfe4d1 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -197,6 +197,7 @@
     SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE,
     SND_DEVICE_IN_THREE_MIC,
     SND_DEVICE_IN_HANDSET_TMIC,
+    SND_DEVICE_IN_VOICE_REC_TMIC,
     SND_DEVICE_IN_UNPROCESSED_MIC,
     SND_DEVICE_IN_UNPROCESSED_STEREO_MIC,
     SND_DEVICE_IN_UNPROCESSED_THREE_MIC,
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 61f42de..d3cb23f 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -25,7 +25,9 @@
 #define CODEC_BACKEND_DEFAULT_SAMPLE_RATE 48000
 #define CODEC_BACKEND_DEFAULT_CHANNELS 2
 #define CODEC_BACKEND_DEFAULT_TX_CHANNELS 1
-
+#define SAMPLE_RATE_8000 8000
+#define SAMPLE_RATE_11025 11025
+#define sample_rate_multiple(sr, base) ((sr % base)== 0?true:false)
 
 enum {
     NATIVE_AUDIO_MODE_SRC = 1,
@@ -147,10 +149,10 @@
 int platform_get_spkr_prot_snd_device(snd_device_t snd_device);
 int platform_get_vi_feedback_snd_device(snd_device_t snd_device);
 int platform_spkr_prot_is_wsa_analog_mode(void *adev);
-bool platform_can_split_snd_device(void *platform,
-                                   snd_device_t snd_device,
-                                   int *num_devices,
-                                   snd_device_t *new_snd_devices);
+int platform_split_snd_device(void *platform,
+                              snd_device_t snd_device,
+                              int *num_devices,
+                              snd_device_t *new_snd_devices);
 
 bool platform_check_backends_match(snd_device_t snd_device1, snd_device_t snd_device2);
 int platform_set_sidetone(struct audio_device *adev,
@@ -187,4 +189,6 @@
 
 unsigned char* platform_get_license(void* platform, int* size);
 int platform_get_max_mic_count(void *platform);
+void platform_check_and_update_copp_sample_rate(void *platform, snd_device_t snd_device,
+     unsigned int stream_sr,int *sample_rate);
 #endif // AUDIO_PLATFORM_API_H
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 089efc5..3396054 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -1409,6 +1409,7 @@
 {
     bool playerDirectPCM = false; // Output request for Track created by mediaplayer
     bool trackDirectPCM = false;  // Output request for track created by other apps
+    bool offloadDisabled = property_get_bool("audio.offload.disable", false);
 
     // Direct PCM is allowed only if
     // In case of mediaPlayer playback
@@ -1417,6 +1418,10 @@
     // In case of AudioTracks created by apps
     // track offload is enabled and FLAG requested is FLAG_NONE.
 
+    if (offloadDisabled) {
+        ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
+    }
+
     if (*flags == AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
        if (bitWidth == 24 || bitWidth == 32)
            playerDirectPCM =
@@ -1431,8 +1436,10 @@
         trackDirectPCM = property_get_bool("audio.offload.track.enable", true);
     }
 
-    ALOGI("%s for Direct PCM",trackDirectPCM || playerDirectPCM?"Check":"Dont check");
-    return trackDirectPCM || playerDirectPCM;
+    ALOGI("Direct PCM %s for this request",
+       (!offloadDisabled && (trackDirectPCM || playerDirectPCM))?"can be enabled":"is disabled");
+
+    return (!offloadDisabled && (trackDirectPCM || playerDirectPCM));
 }
 
 status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr,
@@ -1449,15 +1456,11 @@
 {
     audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
 
-    bool offloadDisabled = property_get_bool("audio.offload.disable", false);
     uint32_t bitWidth = (audio_bytes_per_sample(format) * 8);
 
-    if (offloadDisabled) {
-        ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
-    }
 
-    if (!offloadDisabled && (offloadInfo == NULL) &&
-        tryForDirectPCM(bitWidth, &flags)) {
+    if (tryForDirectPCM(bitWidth, &flags) &&
+        (offloadInfo == NULL)) {
 
         tOffloadInfo.sample_rate  = samplingRate;
         tOffloadInfo.channel_mask = channelMask;