diff --git a/audiod/Android.mk b/audiod/Android.mk
index c382c9d..c89b3cd 100644
--- a/audiod/Android.mk
+++ b/audiod/Android.mk
@@ -1,7 +1,10 @@
 LOCAL_PATH:= $(call my-dir)
 
 include $(CLEAR_VARS)
+
+ifneq (,$(findstring $(PLATFORM_VERSION), 5.0 5.1 5.1.1))
 include external/stlport/libstlport.mk
+endif
 
 LOCAL_SRC_FILES:= \
 	audiod_main.cpp \
@@ -13,8 +16,11 @@
 	libcutils \
 	libutils \
 	libbinder \
-	libmedia \
-	libstlport
+	libmedia
+
+ifneq (,$(findstring $(PLATFORM_VERSION), 5.0 5.1 5.1.1))
+LOCAL_SHARED_LIBRARIES += libstlport
+endif
 
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
 
diff --git a/audiod/AudioDaemon.cpp b/audiod/AudioDaemon.cpp
index b4857c4..6f38218 100644
--- a/audiod/AudioDaemon.cpp
+++ b/audiod/AudioDaemon.cpp
@@ -125,6 +125,7 @@
         struct dirent* in_file;
         int fd;
         String8 path;
+        String8 d_name;
 
         if ((dp = opendir(events_dir)) == NULL) {
             ALOGE("Cannot open switch directory to get list of audio events %s", events_dir);
@@ -148,8 +149,9 @@
             if (fd == -1) {
                 ALOGE("Open %s failed : %s", path.string(), strerror(errno));
             } else {
-                mAudioEvents.push_back(std::make_pair(in_file->d_name, fd));
-                mAudioEventsStatus.push_back(std::make_pair(in_file->d_name, 0));
+                d_name = in_file->d_name;
+                mAudioEvents.push_back(std::make_pair(d_name, fd));
+                mAudioEventsStatus.push_back(std::make_pair(d_name, 0));
                 ALOGD("event status mAudioEventsStatus= %s",
                           mAudioEventsStatus[0].first.string());
             }
diff --git a/hal/Android.mk b/hal/Android.mk
index e0149f9..315aab2 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -53,6 +53,7 @@
 LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
 
 ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_EDID)),true)
+    LOCAL_CFLAGS += -DHDMI_EDID
     LOCAL_SRC_FILES += edid.c
 endif
 
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 7bbe2f8..e443f06 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -46,6 +46,9 @@
 #define AUDIO_FORMAT_PCM_24_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_8_24_BIT)
 #define AUDIO_OFFLOAD_CODEC_FORMAT  "music_offload_codec_format"
 #define audio_is_offload_pcm(format) (0)
+#define OFFLOAD_USE_SMALL_BUFFER false
+#else
+#define OFFLOAD_USE_SMALL_BUFFER (info->use_small_bufs)
 #endif
 
 #ifndef AFE_PROXY_ENABLED
@@ -316,6 +319,9 @@
 void audio_extn_dolby_set_dmid(struct audio_device *adev);
 #else
 #define audio_extn_dolby_set_dmid(adev)                 (0)
+#define AUDIO_CHANNEL_OUT_PENTA (AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER)
+#define AUDIO_CHANNEL_OUT_SURROUND (AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT | \
+                                    AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_BACK_CENTER)
 #endif
 
 
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 3dbf159..7ad4811 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -71,7 +71,9 @@
 #ifdef COMPRESS_VOIP_ENABLED
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_VOIP_RX),
 #endif
+#ifdef HDMI_PASSTHROUGH_ENABLED
     STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH),
+#endif
 };
 
 const struct string_to_enum s_format_name_to_enum_table[] = {
@@ -566,8 +568,11 @@
     app_type_cfg[len++] = out->app_type_cfg.app_type;
     app_type_cfg[len++] = acdb_dev_id;
     if (((out->format == AUDIO_FORMAT_E_AC3) ||
-        (out->format == AUDIO_FORMAT_E_AC3_JOC)) &&
-        (out->flags  & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
+        (out->format == AUDIO_FORMAT_E_AC3_JOC))
+#ifdef HDMI_PASSTHROUGH_ENABLED
+        && (out->flags  & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)
+#endif
+        )
         app_type_cfg[len++] = sample_rate * 4;
     else
         app_type_cfg[len++] = sample_rate;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index ee6bab7..c67f210 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -2941,7 +2941,7 @@
         if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
             out->non_blocking = 1;
 
-        if (config->offload_info.use_small_bufs) {
+        if (platform_use_small_buffer(&config->offload_info)) {
             //this flag is set from framework only if its for PCM formats
             //no need to check for PCM format again
             out->non_blocking = 0;
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 67f5279..5e61d36 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -38,6 +38,7 @@
 #ifndef QCOM_AUDIO_HW_H
 #define QCOM_AUDIO_HW_H
 
+#include <stdlib.h>
 #include <cutils/list.h>
 #include <hardware/audio.h>
 #include <tinyalsa/asoundlib.h>
diff --git a/hal/edid.h b/hal/edid.h
index ec83ec8..0d7fbe6 100644
--- a/hal/edid.h
+++ b/hal/edid.h
@@ -92,5 +92,9 @@
     int  channel_allocation;
 } edid_audio_info;
 
+#ifndef HDMI_EDID
+#define edid_get_sink_caps(info, edid_data) (0)
+#else
 bool edid_get_sink_caps(edid_audio_info* info, char *edid_data);
+#endif
 #endif /* EDID_H */
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 8d47a8b..c2805f9 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -3292,6 +3292,11 @@
     return fragment_size;
 }
 
+bool platform_use_small_buffer(audio_offload_info_t* info)
+{
+    return OFFLOAD_USE_SMALL_BUFFER;
+}
+
 int platform_set_codec_backend_cfg(struct audio_device* adev,
                          unsigned int bit_width, unsigned int sample_rate)
 {
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 3ce4c57..74c8e9b 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -189,8 +189,8 @@
  * We should take care of returning proper size when AudioFlinger queries for
  * the buffer size of an input/output stream
  */
-#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
-#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 5
+#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
 #define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
 #define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
 
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index cace634..861790e 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1098,6 +1098,11 @@
     return 0;
 }
 
+bool platform_use_small_buffer(audio_offload_info_t* info)
+{
+    return false;
+}
+
 int platform_get_edid_info(void *platform __unused)
 {
    return -ENOSYS;
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index c96d11e..7849644 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -134,6 +134,10 @@
     SND_DEVICE_IN_HANDSET_MIC_EXTERNAL
 };
 
+static const snd_device_t tasha_DB_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER
+};
+
 static const snd_device_t taiko_apq8084_sbc_variant_devices[] = {
     SND_DEVICE_IN_HANDSET_MIC,
     SND_DEVICE_IN_SPEAKER_MIC,
@@ -278,6 +282,12 @@
         hw_info->snd_devices = (snd_device_t *)tomtom_liquid_variant_devices;
         hw_info->num_snd_devices = ARRAY_SIZE(tomtom_liquid_variant_devices);
         strlcpy(hw_info->dev_extn, "-liquid", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "msm8996-tasha-db-snd-card")) {
+        strlcpy(hw_info->type, " dragon-board", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)tasha_DB_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(tasha_DB_variant_devices);
+        strlcpy(hw_info->dev_extn, "-db", sizeof(hw_info->dev_extn));
     } else {
         ALOGW("%s: Not a 8996 device", __func__);
     }
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index c7c43ba..3339f3f 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -3261,6 +3261,11 @@
     return fragment_size;
 }
 
+bool platform_use_small_buffer(audio_offload_info_t* info)
+{
+    return OFFLOAD_USE_SMALL_BUFFER;
+}
+
 int platform_set_codec_backend_cfg(struct audio_device* adev,
                          snd_device_t snd_device,
                          unsigned int bit_width, unsigned int sample_rate)
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 4b90f8b..2618bd8 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -185,8 +185,8 @@
  * We should take care of returning proper size when AudioFlinger queries for
  * the buffer size of an input/output stream
  */
-#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
-#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 5
+#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
 #define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
 #define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
 
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 1a3069f..f8749e7 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -93,6 +93,7 @@
 struct audio_offload_info_t;
 uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
 uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
+bool platform_use_small_buffer(audio_offload_info_t* info);
 uint32_t platform_get_compress_passthrough_buffer_size(audio_offload_info_t* info);
 
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
diff --git a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
index 6af9269..4cfee1b 100644
--- a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
+++ b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
@@ -4208,6 +4208,16 @@
                buffer->nOffset = 0;
                buffer->nTimeStamp = nTimestamp;
                frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
+
+               if (errno == ENETRESET)
+               {
+                   ALOGE("In SSR, return error to close the session");
+                   m_cb.EventHandler(&m_cmp,
+                       m_app_data,
+                      OMX_EventError,
+                      OMX_ErrorHardware,
+                      0, NULL );
+               }
                return OMX_ErrorNone;
            }
         }
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 71a210f..192320a 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -14,8 +14,12 @@
 	virtualizer.c \
 	reverb.c \
 	effect_api.c \
-	effect_util.c \
-        hw_accelerator.c
+	effect_util.c
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS)),true)
+    LOCAL_CFLAGS += -DHW_ACCELERATED_EFFECTS
+    LOCAL_SRC_FILES += hw_accelerator.c
+endif
 
 LOCAL_CFLAGS+= -O2 -fvisibility=hidden
 
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index e38a41c..6f1bdfa 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -38,6 +38,7 @@
 #define LOG_TAG "offload_effect_bundle"
 //#define LOG_NDEBUG 0
 
+#include <stdlib.h>
 #include <cutils/list.h>
 #include <cutils/log.h>
 #include <system/thread_defs.h>
@@ -69,7 +70,9 @@
         &ins_env_reverb_descriptor,
         &aux_preset_reverb_descriptor,
         &ins_preset_reverb_descriptor,
+#ifdef HW_ACCELERATED_EFFECTS
         &hw_accelerator_descriptor,
+#endif
         NULL,
 };
 
@@ -302,9 +305,6 @@
         goto exit;
     }
 
-    if (out_ctxt->mixer)
-        mixer_close(out_ctxt->mixer);
-
     list_for_each(fx_node, &out_ctxt->effects_list) {
         effect_context_t *fx_ctxt = node_to_item(fx_node,
                                                  effect_context_t,
@@ -313,6 +313,9 @@
             fx_ctxt->ops.stop(fx_ctxt, out_ctxt);
     }
 
+    if (out_ctxt->mixer)
+        mixer_close(out_ctxt->mixer);
+
     list_remove(&out_ctxt->outputs_list_node);
 
 #ifdef DTS_EAGLE
@@ -604,6 +607,7 @@
             reverb_preset_init(reverb_ctxt);
         }
         reverb_ctxt->ctl = NULL;
+#ifdef HW_ACCELERATED_EFFECTS
     } else if (memcmp(uuid, &hw_accelerator_descriptor.uuid,
                sizeof(effect_uuid_t)) == 0) {
         hw_accelerator_context_t *hw_acc_ctxt = (hw_accelerator_context_t *)
@@ -625,6 +629,7 @@
         context->ops.process = hw_accelerator_process;
 
         context->desc = &hw_accelerator_descriptor;
+#endif
     } else {
         return -EINVAL;
     }
@@ -911,6 +916,7 @@
 
         } break;
 
+#ifdef HW_ACCELERATED_EFFECTS
     case EFFECT_CMD_HW_ACC: {
         ALOGV("EFFECT_CMD_HW_ACC cmdSize %d pCmdData %p, *replySize %d, pReplyData %p",
               cmdSize, pCmdData, *replySize, pReplyData);
@@ -925,6 +931,7 @@
         context->hw_acc_enabled = (value > 0) ? true : false;
         break;
     }
+#endif
     default:
         if (cmdCode >= EFFECT_CMD_FIRST_PROPRIETARY && context->ops.command)
             status = context->ops.command(context, cmdCode, cmdSize,
diff --git a/post_proc/effect_api.c b/post_proc/effect_api.c
index e15db17..16c370b 100644
--- a/post_proc/effect_api.c
+++ b/post_proc/effect_api.c
@@ -55,6 +55,7 @@
 #endif
 
 #include <stdbool.h>
+#include <errno.h>
 #include <cutils/log.h>
 #include <tinyalsa/asoundlib.h>
 #include <sound/audio_effects.h>
diff --git a/post_proc/effect_util.c b/post_proc/effect_util.c
index 8f7a604..52f623d 100644
--- a/post_proc/effect_util.c
+++ b/post_proc/effect_util.c
@@ -16,6 +16,7 @@
 
 #include <utils/Log.h>
 #include <stdlib.h>
+#include <string.h>
 #include "effect_util.h"
 
 #ifdef LOG_TAG
diff --git a/voice_processing/voice_processing.c b/voice_processing/voice_processing.c
index 1d18a3d..72e4d84 100644
--- a/voice_processing/voice_processing.c
+++ b/voice_processing/voice_processing.c
@@ -16,6 +16,7 @@
 
 #define LOG_TAG "voice_processing"
 /*#define LOG_NDEBUG 0*/
+#include <stdlib.h>
 #include <dlfcn.h>
 #include <cutils/log.h>
 #include <cutils/list.h>