diff --git a/Android.mk b/Android.mk
index 450d9a3..f856e9e 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,20 +1,22 @@
-ifneq ($(filter msm8960 msm8226 msm8x26 msm8974 msm8x74 msm8x84 msm8084 msm8992 msm8994,$(TARGET_BOARD_PLATFORM)),)
+# TODO:  Find a better way to separate build configs for ADP vs non-ADP devices
+ifneq ($(TARGET_BOARD_AUTO),true)
+  ifneq ($(filter msm8960 msm8226 msm8x26 msm8974 msm8x74 msm8x84 msm8084 msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
 
-MY_LOCAL_PATH := $(call my-dir)
+    MY_LOCAL_PATH := $(call my-dir)
 
-ifeq ($(BOARD_USES_LEGACY_ALSA_AUDIO),true)
-include $(MY_LOCAL_PATH)/legacy/Android.mk
-else
-include $(MY_LOCAL_PATH)/hal/Android.mk
-include $(MY_LOCAL_PATH)/voice_processing/Android.mk
-include $(MY_LOCAL_PATH)/visualizer/Android.mk
-include $(MY_LOCAL_PATH)/post_proc/Android.mk
-endif
+    ifeq ($(BOARD_USES_LEGACY_ALSA_AUDIO),true)
+      include $(MY_LOCAL_PATH)/legacy/Android.mk
+    else
+      include $(MY_LOCAL_PATH)/hal/Android.mk
+      include $(MY_LOCAL_PATH)/voice_processing/Android.mk
+      include $(MY_LOCAL_PATH)/visualizer/Android.mk
+      include $(MY_LOCAL_PATH)/post_proc/Android.mk
+    endif
+  else
+    ifneq ($(filter msm8909 ,$(TARGET_BOARD_PLATFORM)),)
+      #For msm8909 target
+      include $(call all-named-subdir-makefiles,msm8909)
+    endif
 
-else
-ifneq ($(filter msm8909 ,$(TARGET_BOARD_PLATFORM)),)
-#For msm8909 target
-include $(call all-named-subdir-makefiles,msm8909)
-
-endif
+  endif
 endif
diff --git a/hal/Android.mk b/hal/Android.mk
index 35dcbb9..d55e37a 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -10,10 +10,11 @@
 ifneq ($(filter msm8960,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="2"
 endif
-ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM = msm8974
 ifneq ($(filter msm8974,$(TARGET_BOARD_PLATFORM)),)
+  LOCAL_CFLAGS := -DPLATFORM_MSM8974
   LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="2"
 endif
 ifneq ($(filter msm8226,$(TARGET_BOARD_PLATFORM)),)
@@ -34,6 +35,12 @@
   LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
   LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
 endif
+ifneq ($(filter msm8996,$(TARGET_BOARD_PLATFORM)),)
+  LOCAL_CFLAGS := -DPLATFORM_MSM8996
+  LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
+  LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
+endif
+
 endif
 
 LOCAL_SRC_FILES := \
@@ -98,7 +105,7 @@
     LOCAL_SRC_FILES += audio_extn/dsm_feedback.c
 endif
 
-ifneq ($(filter msm8992 msm8994,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
   # push codec/mad calibration to HW dep node
   # applicable to msm8992/8994 or newer platforms
   LOCAL_CFLAGS += -DHWDEP_CAL_ENABLED
diff --git a/hal/audio_extn/ext_speaker.c b/hal/audio_extn/ext_speaker.c
index 55cbb4c..cb8cfc1 100644
--- a/hal/audio_extn/ext_speaker.c
+++ b/hal/audio_extn/ext_speaker.c
@@ -22,11 +22,7 @@
 #include <audio_hw.h>
 #include <dlfcn.h>
 
-#ifdef __LP64__
-        #define LIB_SPEAKER_BUNDLE "/system/lib64/soundfx/libspeakerbundle.so"
-#else
-        #define LIB_SPEAKER_BUNDLE "/system/lib/soundfx/libspeakerbundle.so"
-#endif
+#define LIB_SPEAKER_BUNDLE "soundfx/libspeakerbundle.so"
 
 typedef void (*set_mode_t)(int);
 typedef void (*set_speaker_on_t)(bool);
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index b5475a1..9909d50 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -31,6 +31,8 @@
 #define XSTR(x) STR(x)
 #define STR(x) #x
 
+#define SOUND_TRIGGER_LIBRARY_PATH "hw/sound_trigger.primary.%s.so"
+
 struct sound_trigger_info  {
     struct sound_trigger_session_info st_ses;
     bool lab_stopped;
@@ -193,7 +195,7 @@
             in->channel_mask = audio_channel_in_mask_from_count(in->config.channels);
             in->is_st_session = true;
             in->is_st_session_active = true;
-            ALOGD("%s: capture_handle %d is sound trigger", __func__, in->capture_handle);
+            ALOGV("%s: capture_handle %d is sound trigger", __func__, in->capture_handle);
             break;
         }
     }
@@ -222,7 +224,7 @@
         return;
     }
 
-    ALOGI("%s: device 0x%x of type %d for Event %d",
+    ALOGV("%s: device 0x%x of type %d for Event %d",
         __func__, snd_device, device_type, event);
     if (device_type == PCM_CAPTURE) {
         switch(event) {
@@ -293,7 +295,7 @@
     char sound_trigger_lib[100];
     void *lib_handle;
 
-    ALOGI("%s: Enter", __func__);
+    ALOGV("%s: Enter", __func__);
 
     st_dev = (struct sound_trigger_audio_device*)
                         calloc(1, sizeof(struct sound_trigger_audio_device));
@@ -303,7 +305,7 @@
     }
 
     snprintf(sound_trigger_lib, sizeof(sound_trigger_lib),
-             "/system/vendor/lib/hw/sound_trigger.primary.%s.so",
+             SOUND_TRIGGER_LIBRARY_PATH,
               XSTR(SOUND_TRIGGER_PLATFORM_NAME));
 
     st_dev->lib_handle = dlopen(sound_trigger_lib, RTLD_NOW);
@@ -314,7 +316,7 @@
         status = -EINVAL;
         goto cleanup;
     }
-    ALOGI("%s: DLOPEN successful for %s", __func__, sound_trigger_lib);
+    ALOGV("%s: DLOPEN successful for %s", __func__, sound_trigger_lib);
 
     st_dev->st_callback = (sound_trigger_hw_call_back_t)
               dlsym(st_dev->lib_handle, "sound_trigger_hw_call_back");
@@ -341,7 +343,7 @@
 
 void audio_extn_sound_trigger_deinit(struct audio_device *adev)
 {
-    ALOGI("%s: Enter", __func__);
+    ALOGV("%s: Enter", __func__);
     if (st_dev && (st_dev->adev == adev) && st_dev->lib_handle) {
         dlclose(st_dev->lib_handle);
         free(st_dev);
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 4d8b233..8afb0dc 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -33,6 +33,8 @@
 #include "audio_extn.h"
 #include <linux/msm_audio_calibration.h>
 
+#define THERMAL_CLIENT_LIBRARY_PATH "libthermalclient.so"
+
 #ifdef SPKR_PROT_ENABLED
 
 /*Range of spkr temparatures -30C to 80C*/
@@ -710,7 +712,7 @@
     pthread_mutex_init(&handle.mutex_spkr_prot, NULL);
     pthread_mutex_init(&handle.spkr_calib_cancelack_mutex, NULL);
     pthread_mutex_init(&handle.spkr_prot_thermalsync_mutex, NULL);
-    handle.thermal_handle = dlopen("/vendor/lib/libthermalclient.so",
+    handle.thermal_handle = dlopen(THERMAL_CLIENT_LIBRARY_PATH,
             RTLD_NOW);
     if (!handle.thermal_handle) {
         ALOGE("%s: DLOPEN for thermal client failed", __func__);
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 6747b05..a4ea34c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2013-2014 The Android Open Source Project
+ * Copyright (C) 2013-2016 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -172,6 +172,8 @@
     [USECASE_VOLTE_CALL] = "volte-call",
     [USECASE_QCHAT_CALL] = "qchat-call",
     [USECASE_VOWLAN_CALL] = "vowlan-call",
+    [USECASE_VOICEMMODE1_CALL] = "voicemmode1-call",
+    [USECASE_VOICEMMODE2_CALL] = "voicemmode2-call",
 
     [USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
     [USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",
@@ -270,7 +272,7 @@
 
     strcpy(mixer_path, use_case_table[usecase->id]);
     platform_add_backend_name(adev->platform, mixer_path, snd_device);
-    ALOGD("%s: apply and update mixer path: %s", __func__, mixer_path);
+    ALOGV("%s: apply and update mixer path: %s", __func__, mixer_path);
     audio_route_apply_and_update_path(adev->audio_route, mixer_path);
 
     ALOGV("%s: exit", __func__);
@@ -293,7 +295,7 @@
         snd_device = usecase->out_snd_device;
     strcpy(mixer_path, use_case_table[usecase->id]);
     platform_add_backend_name(adev->platform, mixer_path, snd_device);
-    ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path);
+    ALOGV("%s: reset and update mixer path: %s", __func__, mixer_path);
     audio_route_reset_and_update_path(adev->audio_route, mixer_path);
 
     ALOGV("%s: exit", __func__);
@@ -349,7 +351,7 @@
         platform_set_speaker_gain_in_combo(adev, snd_device, true);
     } else {
         const char * dev_path = platform_get_snd_device_name(snd_device);
-        ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, dev_path);
+        ALOGV("%s: snd_device(%d: %s)", __func__, snd_device, dev_path);
         audio_route_apply_and_update_path(adev->audio_route, dev_path);
     }
 
@@ -374,7 +376,7 @@
     adev->snd_dev_ref_cnt[snd_device]--;
     if (adev->snd_dev_ref_cnt[snd_device] == 0) {
         const char * dev_path = platform_get_snd_device_name(snd_device);
-        ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, dev_path);
+        ALOGV("%s: snd_device(%d: %s)", __func__, snd_device, dev_path);
 
         audio_extn_dsm_feedback_enable(adev, snd_device, false);
         if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
@@ -674,7 +676,7 @@
         return 0;
     }
 
-    ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__,
+    ALOGV("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__,
           out_snd_device, platform_get_snd_device_name(out_snd_device),
           in_snd_device,  platform_get_snd_device_name(in_snd_device));
 
@@ -817,7 +819,7 @@
     ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
           __func__, adev->snd_card, in->pcm_device_id, in->config.channels);
 
-    unsigned int flags = PCM_IN;
+    unsigned int flags = PCM_IN | PCM_MONOTONIC;
     unsigned int pcm_open_retry_count = 0;
 
     if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
@@ -844,8 +846,14 @@
         break;
     }
 
-    ALOGV("%s: pcm_prepare start", __func__);
-    pcm_prepare(in->pcm);
+    ALOGV("%s: pcm_prepare", __func__);
+    ret = pcm_prepare(in->pcm);
+    if (ret < 0) {
+        ALOGE("%s: pcm_prepare returned %d", __func__, ret);
+        pcm_close(in->pcm);
+        in->pcm = NULL;
+        goto error_open;
+    }
 
     audio_extn_perf_lock_release();
 
@@ -859,7 +867,7 @@
 
 error_config:
     adev->active_input = NULL;
-    ALOGD("%s: exit: status(%d)", __func__, ret);
+    ALOGV("%s: exit: status(%d)", __func__, ret);
 
     return ret;
 }
@@ -1036,12 +1044,12 @@
              * max channels of remaining use cases.
              */
             if (usecase->id == USECASE_VOICE_CALL) {
-                ALOGD("%s: voice call is active, no change in HDMI channels",
+                ALOGV("%s: voice call is active, no change in HDMI channels",
                       __func__);
                 ret = false;
                 break;
             } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
-                ALOGD("%s: multi channel playback is active, "
+                ALOGV("%s: multi channel playback is active, "
                       "no change in HDMI channels", __func__);
                 ret = false;
                 break;
@@ -1062,7 +1070,7 @@
         return 0;
 
     if (channels == adev->cur_hdmi_channels) {
-        ALOGD("%s: Requested channels are same as current", __func__);
+        ALOGV("%s: Requested channels are same as current", __func__);
         return 0;
     }
 
@@ -1203,10 +1211,16 @@
             }
             break;
         }
-        ALOGV("%s: pcm_prepare start", __func__);
-        if (pcm_is_ready(out->pcm))
-            pcm_prepare(out->pcm);
-
+        ALOGV("%s: pcm_prepare", __func__);
+        if (pcm_is_ready(out->pcm)) {
+            ret = pcm_prepare(out->pcm);
+            if (ret < 0) {
+                ALOGE("%s: pcm_prepare returned %d", __func__, ret);
+                pcm_close(out->pcm);
+                out->pcm = NULL;
+                goto error_open;
+            }
+        }
     } else {
         out->pcm = NULL;
         out->compr = compress_open(adev->snd_card, out->pcm_device_id,
@@ -1431,7 +1445,7 @@
     bool select_new_device = false;
     int status = 0;
 
-    ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
+    ALOGV("%s: enter: usecase(%d: %s) kvpairs: %s",
           __func__, out->usecase, use_case_table[out->usecase], kvpairs);
     parms = str_parms_create_str(kvpairs);
     ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
@@ -1638,13 +1652,28 @@
             compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
             out->send_new_metadata = 0;
         }
+        unsigned int avail;
+        struct timespec tstamp;
+        ret = compress_get_hpointer(out->compr, &avail, &tstamp);
+        /* Do not limit write size if the available frames count is unknown */
+        if (ret != 0) {
+            avail = bytes;
+        }
+        if (avail == 0) {
+            ret = 0;
+        } else {
+            if (avail > bytes) {
+                avail = bytes;
+            }
+            ret = compress_write(out->compr, buffer, avail);
+            ALOGVV("%s: writing buffer (%d bytes) to compress device returned %zd",
+                   __func__, avail, ret);
+        }
 
-        ret = compress_write(out->compr, buffer, bytes);
-        ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret);
         if (ret >= 0 && ret < (ssize_t)bytes) {
             send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
         }
-        if (!out->playback_started) {
+        if (ret > 0 && !out->playback_started) {
             compress_start(out->compr);
             out->playback_started = 1;
             out->offload_state = OFFLOAD_STATE_PLAYING;
@@ -1695,8 +1724,10 @@
     if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) {
         lock_output_stream(out);
         if (out->compr != NULL) {
-            compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
-                    &out->sample_rate);
+            unsigned long frames = 0;
+            // TODO: check return value
+            compress_get_tstamp(out->compr, &frames, &out->sample_rate);
+            *dsp_frames = (uint32_t)frames;
             ALOGVV("%s rendered frames %d sample_rate %d",
                    __func__, *dsp_frames, out->sample_rate);
         }
@@ -1728,13 +1759,14 @@
                                    uint64_t *frames, struct timespec *timestamp)
 {
     struct stream_out *out = (struct stream_out *)stream;
-    int ret = -1;
+    int ret = -EINVAL;
     unsigned long dsp_frames;
 
     lock_output_stream(out);
 
     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
         if (out->compr != NULL) {
+            // TODO: check return value
             compress_get_tstamp(out->compr, &dsp_frames,
                     &out->sample_rate);
             ALOGVV("%s rendered frames %ld sample_rate %d",
@@ -1892,7 +1924,7 @@
     lock_input_stream(in);
 
     if (!in->standby && in->is_st_session) {
-        ALOGD("%s: sound trigger pcm stop lab", __func__);
+        ALOGV("%s: sound trigger pcm stop lab", __func__);
         audio_extn_sound_trigger_stop_lab(in);
         in->standby = true;
     }
@@ -2037,6 +2069,10 @@
         ALOGV("%s: read failed - sleeping for buffer duration", __func__);
         usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) /
                in_get_sample_rate(&in->stream.common));
+        memset(buffer, 0, bytes); // clear return data
+    }
+    if (bytes > 0) {
+        in->frames_read += bytes / audio_stream_in_frame_size(stream);
     }
     return bytes;
 }
@@ -2046,6 +2082,29 @@
     return 0;
 }
 
+static int in_get_capture_position(const struct audio_stream_in *stream,
+                                   int64_t *frames, int64_t *time)
+{
+    if (stream == NULL || frames == NULL || time == NULL) {
+        return -EINVAL;
+    }
+    struct stream_in *in = (struct stream_in *)stream;
+    int ret = -ENOSYS;
+
+    lock_input_stream(in);
+    if (in->pcm) {
+        struct timespec timestamp;
+        unsigned int avail;
+        if (pcm_get_htimestamp(in->pcm, &avail, &timestamp) == 0) {
+            *frames = in->frames_read + avail;
+            *time = timestamp.tv_sec * 1000000000LL + timestamp.tv_nsec;
+            ret = 0;
+        }
+    }
+    pthread_mutex_unlock(&in->lock);
+    return ret;
+}
+
 static int add_remove_audio_effect(const struct audio_stream *stream,
                                    effect_handle_t effect,
                                    bool enable)
@@ -2150,9 +2209,12 @@
             config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
         if (config->channel_mask == 0)
             config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+        if (config->format == AUDIO_FORMAT_DEFAULT)
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
 
         out->channel_mask = config->channel_mask;
         out->sample_rate = config->sample_rate;
+        out->format = config->format;
         out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH;
         out->config = pcm_config_hdmi_multi;
         out->config.rate = config->sample_rate;
@@ -2325,7 +2387,7 @@
 error_open:
     free(out);
     *stream_out = NULL;
-    ALOGD("%s: exit: ret %d", __func__, ret);
+    ALOGV("%s: exit: ret %d", __func__, ret);
     return ret;
 }
 
@@ -2363,7 +2425,7 @@
     int ret;
     int status = 0;
 
-    ALOGD("%s: enter: %s", __func__, kvpairs);
+    ALOGV("%s: enter: %s", __func__, kvpairs);
 
     pthread_mutex_lock(&adev->lock);
 
@@ -2492,7 +2554,7 @@
 
     pthread_mutex_lock(&adev->lock);
     if (adev->mode != mode) {
-        ALOGD("%s: mode %d\n", __func__, mode);
+        ALOGV("%s: mode %d\n", __func__, mode);
         adev->mode = mode;
         if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION) &&
                 voice_is_in_call(adev)) {
@@ -2512,7 +2574,7 @@
     int ret;
     struct audio_device *adev = (struct audio_device *)dev;
 
-    ALOGD("%s: state %d\n", __func__, state);
+    ALOGV("%s: state %d\n", __func__, state);
     pthread_mutex_lock(&adev->lock);
     ret = voice_set_mic_mute(adev, state);
     adev->mic_muted = state;
@@ -2576,6 +2638,7 @@
     in->stream.set_gain = in_set_gain;
     in->stream.read = in_read;
     in->stream.get_input_frames_lost = in_get_input_frames_lost;
+    in->stream.get_capture_position = in_get_capture_position;
 
     in->device = devices;
     in->source = source;
@@ -2584,6 +2647,7 @@
     in->channel_mask = config->channel_mask;
     in->capture_handle = handle;
     in->flags = flags;
+    // in->frames_read = 0;
 
     /* Update config params with the requested sample rate and channels */
     if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
@@ -2770,7 +2834,6 @@
                     ALOGV("%s: (%s) card %d  device %d", __func__,
                             dir ? "input" : "output", card_id, device_id);
                     pcm_params_to_string(*pparams, info, ARRAY_SIZE(info));
-                    ALOGV(info); /* print parameters */
                 } else {
                     ALOGV("%s: cannot locate card %d  device %d", __func__, card_id, device_id);
                 }
@@ -2845,7 +2908,7 @@
 {
     int i, ret;
 
-    ALOGD("%s: enter", __func__);
+    ALOGV("%s: enter", __func__);
     if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
     pthread_mutex_lock(&adev_init_lock);
     if (audio_device_ref_count != 0) {
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 8c07b6d..0fc26ff 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2013-2014 The Android Open Source Project
+ * Copyright (C) 2013-2016 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -27,9 +27,9 @@
 #include <audio_route/audio_route.h>
 #include "voice.h"
 
-#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
-#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"
-#define ADM_LIBRARY_PATH "/system/vendor/lib/libadm.so"
+#define VISUALIZER_LIBRARY_PATH "soundfx/libqcomvisualizer.so"
+#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "soundfx/libqcompostprocbundle.so"
+#define ADM_LIBRARY_PATH "libadm.so"
 
 /* Flags used to initialize acdb_settings variable that goes to ACDB library */
 #define DMIC_FLAG       0x00000002
@@ -69,13 +69,29 @@
     USECASE_AUDIO_RECORD,
     USECASE_AUDIO_RECORD_LOW_LATENCY,
 
-    USECASE_VOICE_CALL,
+    /* Voice extension usecases
+     *
+     * Following usecase are specific to voice session names created by
+     * MODEM and APPS on 8992/8994/8084/8974 platforms.
+     */
+    USECASE_VOICE_CALL,  /* Usecase setup for voice session on first subscription for DSDS/DSDA */
+    USECASE_VOICE2_CALL, /* Usecase setup for voice session on second subscription for DSDS/DSDA */
+    USECASE_VOLTE_CALL,  /* Usecase setup for VoLTE session on first subscription */
+    USECASE_QCHAT_CALL,  /* Usecase setup for QCHAT session */
+    USECASE_VOWLAN_CALL, /* Usecase setup for VoWLAN session */
 
-    /* Voice extension usecases */
-    USECASE_VOICE2_CALL,
-    USECASE_VOLTE_CALL,
-    USECASE_QCHAT_CALL,
-    USECASE_VOWLAN_CALL,
+    /*
+     * Following usecase are specific to voice session names created by
+     * MODEM and APPS on 8996 platforms.
+     */
+
+    USECASE_VOICEMMODE1_CALL, /* Usecase setup for Voice/VoLTE/VoWLAN sessions on first
+                               * subscription for DSDS/DSDA
+                               */
+    USECASE_VOICEMMODE2_CALL, /* Usecase setup for voice/VoLTE/VoWLAN sessions on second
+                               * subscription for DSDS/DSDA
+                               */
+
     USECASE_INCALL_REC_UPLINK,
     USECASE_INCALL_REC_DOWNLINK,
     USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
@@ -175,6 +191,7 @@
     audio_usecase_t usecase;
     bool enable_aec;
     bool enable_ns;
+    int64_t frames_read; /* total frames read, not cleared when entering standby */
 
     audio_io_handle_t capture_handle;
     audio_input_flags_t flags;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index ab836bd..49e6801 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2013-2014 The Android Open Source Project
+ * Copyright (C) 2013-2016 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -28,8 +28,11 @@
 #include "audio_extn.h"
 #include <linux/msm_audio.h>
 
-#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
-#define MIXER_XML_PATH_WCD9330 "/system/etc/mixer_paths_wcd9330.xml"
+#define MIXER_XML_DEFAULT_PATH "/system/etc/mixer_paths.xml"
+#define MIXER_XML_BASE_STRING "/system/etc/mixer_paths"
+#define TOMTOM_8226_SND_CARD_NAME "msm8226-tomtom-snd-card"
+#define TOMTOM_MIXER_FILE_SUFFIX "wcd9330"
+
 #define LIB_ACDB_LOADER "libacdbloader.so"
 #define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
 #define CVD_VERSION_MIXER_CTL "CVD Version"
@@ -94,7 +97,7 @@
 
 /* Audio calibration related functions */
 typedef void (*acdb_deallocate_t)();
-typedef int  (*acdb_init_v2_cvd_t)(char *, char *);
+typedef int  (*acdb_init_v2_cvd_t)(char *, char *, int);
 typedef int  (*acdb_init_v2_t)(char *);
 typedef int  (*acdb_init_t)();
 typedef void (*acdb_send_audio_cal_t)(int, int);
@@ -153,6 +156,11 @@
     [USECASE_VOLTE_CALL] = {VOLTE_CALL_PCM_DEVICE, VOLTE_CALL_PCM_DEVICE},
     [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE, QCHAT_CALL_PCM_DEVICE},
     [USECASE_VOWLAN_CALL] = {VOWLAN_CALL_PCM_DEVICE, VOWLAN_CALL_PCM_DEVICE},
+    [USECASE_VOICEMMODE1_CALL] = {VOICEMMODE1_CALL_PCM_DEVICE,
+                                  VOICEMMODE1_CALL_PCM_DEVICE},
+    [USECASE_VOICEMMODE2_CALL] = {VOICEMMODE2_CALL_PCM_DEVICE,
+                                  VOICEMMODE2_CALL_PCM_DEVICE},
+
     [USECASE_INCALL_REC_UPLINK] = {AUDIO_RECORD_PCM_DEVICE,
                                    AUDIO_RECORD_PCM_DEVICE},
     [USECASE_INCALL_REC_DOWNLINK] = {AUDIO_RECORD_PCM_DEVICE,
@@ -251,6 +259,9 @@
     [SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE] = "voice-rec-dmic-ef-fluence",
     [SND_DEVICE_IN_VOICE_REC_HEADSET_MIC] = "headset-mic",
 
+    [SND_DEVICE_IN_UNPROCESSED_MIC] = "voice-rec-mic",
+    [SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = "headset-mic",
+
     [SND_DEVICE_IN_VOICE_RX] = "voice-rx",
 
     [SND_DEVICE_IN_THREE_MIC] = "three-mic",
@@ -311,7 +322,7 @@
     [SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS] = 117,
     [SND_DEVICE_IN_SPEAKER_DMIC_STEREO] = 35,
 
-    [SND_DEVICE_IN_HEADSET_MIC] = 8,
+    [SND_DEVICE_IN_HEADSET_MIC] = ACDB_ID_HEADSET_MIC_AEC,
     [SND_DEVICE_IN_HEADSET_MIC_AEC] = ACDB_ID_HEADSET_MIC_AEC,
 
     [SND_DEVICE_IN_HDMI_MIC] = 4,
@@ -326,16 +337,19 @@
     [SND_DEVICE_IN_VOICE_SPEAKER_MIC] = 11,
     [SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP] = 11,
     [SND_DEVICE_IN_VOICE_SPEAKER_DMIC] = 43,
-    [SND_DEVICE_IN_VOICE_HEADSET_MIC] = 8,
+    [SND_DEVICE_IN_VOICE_HEADSET_MIC] = ACDB_ID_HEADSET_MIC_AEC,
     [SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC] = 16,
     [SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC] = 36,
     [SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC] = 16,
 
-    [SND_DEVICE_IN_VOICE_REC_MIC] = 62,
+    [SND_DEVICE_IN_VOICE_REC_MIC] = ACDB_ID_VOICE_REC_MIC,
     [SND_DEVICE_IN_VOICE_REC_MIC_NS] = 113,
     [SND_DEVICE_IN_VOICE_REC_DMIC_STEREO] = 35,
     [SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE] = 43,
-    [SND_DEVICE_IN_VOICE_REC_HEADSET_MIC] = 8,
+    [SND_DEVICE_IN_VOICE_REC_HEADSET_MIC] = ACDB_ID_HEADSET_MIC_AEC,
+
+    [SND_DEVICE_IN_UNPROCESSED_MIC] = ACDB_ID_VOICE_REC_MIC,
+    [SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = ACDB_ID_HEADSET_MIC_AEC,
 
     [SND_DEVICE_IN_VOICE_RX] = 44,
 
@@ -430,6 +444,9 @@
     {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE)},
     {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_HEADSET_MIC)},
 
+    {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC)},
+
     {TO_NAME_INDEX(SND_DEVICE_IN_THREE_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_QUAD_MIC)},
     {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
@@ -472,7 +489,7 @@
     int mccmnc;
     property_get("gsm.sim.operator.numeric",value,"0");
     mccmnc = atoi(value);
-    ALOGD("%s: tmus mccmnc %d", __func__, mccmnc);
+    ALOGV("%s: tmus mccmnc %d", __func__, mccmnc);
     switch(mccmnc) {
     /* TMUS MCC(310), MNC(490, 260, 026) */
     case 310490:
@@ -638,7 +655,7 @@
             platform_add_backend_name(adev->platform, my_data->ec_ref_mixer_path, snd_device);
         }
 
-        ALOGD("%s: enabling %s", __func__, my_data->ec_ref_mixer_path);
+        ALOGV("%s: enabling %s", __func__, my_data->ec_ref_mixer_path);
         audio_route_apply_and_update_path(adev->audio_route, my_data->ec_ref_mixer_path);
     }
 }
@@ -815,7 +832,7 @@
         goto done;
     }
 
-    ALOGD("%s: num_modems %d\n", __func__, modems);
+    ALOGV("%s: num_modems %d\n", __func__, modems);
     if (modems > 0)
         my_data->csd = open_csd_client(false /*is_i2s_ext_modem*/);
 
@@ -907,11 +924,14 @@
 void *platform_init(struct audio_device *adev)
 {
     char value[PROPERTY_VALUE_MAX];
-    struct platform_data *my_data;
-    int retry_num = 0, snd_card_num = 0;
-    bool dual_mic_config = false;
+    struct platform_data *my_data = NULL;
+    int retry_num = 0, snd_card_num = 0, key = 0, ret = 0;
+    bool dual_mic_config = false, use_default_mixer_path = true;
     const char *snd_card_name;
     char *cvd_version = NULL;
+    char *snd_internal_name = NULL;
+    char *tmp = NULL;
+    char mixer_xml_file[MIXER_PATH_MAX_LENGTH]= {0};
 
     my_data = calloc(1, sizeof(struct platform_data));
 
@@ -953,23 +973,51 @@
             continue;
         }
 
-        ALOGD("%s: snd_card_name: %s", __func__, snd_card_name);
+        if ((snd_internal_name = strtok_r(snd_card_name, "-", &tmp)) != NULL) {
+           /* Get the codec internal name from the sound card name
+            * and form the mixer paths file name dynamically. This
+            * is generic way of picking any codec name based mixer
+            * files in future with no code change. This code
+            * assumes mixer files are formed with format as
+            * mixer_paths_internalcodecname.xml
 
-        if (!strncmp(snd_card_name, "msm8226-tomtom-snd-card",
-                     sizeof("msm8226-tomtom-snd-card"))) {
-            ALOGD("%s: Call MIXER_XML_PATH_WCD9330", __func__);
-            adev->audio_route = audio_route_init(snd_card_num,
-                                                 MIXER_XML_PATH_WCD9330);
-        } else {
-            adev->audio_route = audio_route_init(snd_card_num, MIXER_XML_PATH);
+            * If this dynamically read mixer files fails to open then it
+            * falls back to default mixer file i.e mixer_paths.xml. This is
+            * done to preserve backward compatibility but not mandatory as
+            * long as the mixer files are named as per above assumption.
+            */
+
+            if ((snd_internal_name = strtok_r(NULL, "-", &tmp)) != NULL) {
+                // need to carryforward old file name
+                if (!strncmp(snd_card_name, TOMTOM_8226_SND_CARD_NAME,
+                             sizeof(TOMTOM_8226_SND_CARD_NAME))) {
+                    snprintf(mixer_xml_file, sizeof(mixer_xml_file), "%s_%s.xml",
+                             MIXER_XML_BASE_STRING, TOMTOM_MIXER_FILE_SUFFIX );
+                } else {
+                    snprintf(mixer_xml_file, sizeof(mixer_xml_file), "%s_%s.xml",
+                             MIXER_XML_BASE_STRING, snd_internal_name);
+                }
+
+                if (F_OK == access(mixer_xml_file, 0)) {
+                    use_default_mixer_path = false;
+                }
+            }
         }
 
+        if (use_default_mixer_path) {
+            memset(mixer_xml_file, 0, sizeof(mixer_xml_file));
+            strlcpy(mixer_xml_file, MIXER_XML_DEFAULT_PATH, MIXER_PATH_MAX_LENGTH);
+        }
+
+        ALOGD("%s: Loading mixer file: %s", __func__, mixer_xml_file);
+        adev->audio_route = audio_route_init(snd_card_num, mixer_xml_file);
+
         if (!adev->audio_route) {
             ALOGE("%s: Failed to init audio route controls, aborting.", __func__);
             goto init_failed;
         }
         adev->snd_card = snd_card_num;
-        ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
+        ALOGV("%s: Opened sound card:%d", __func__, snd_card_num);
         break;
     }
 
@@ -1084,7 +1132,7 @@
             ALOGV("%s: Could not find the symbol acdb_loader_send_gain_dep_cal from %s",
                   __func__, LIB_ACDB_LOADER);
 
-#if defined (PLATFORM_MSM8994)
+#if defined (PLATFORM_MSM8994) || (PLATFORM_MSM8996)
         acdb_init_v2_cvd_t acdb_init;
         acdb_init = (acdb_init_v2_cvd_t)dlsym(my_data->acdb_handle,
                                               "acdb_loader_init_v2");
@@ -1098,7 +1146,7 @@
         if (!cvd_version)
             ALOGE("failed to allocate cvd_version");
         else
-            acdb_init((char *)snd_card_name, cvd_version);
+            acdb_init((char *)snd_card_name, cvd_version, 0);
         free(cvd_version);
 #elif defined (PLATFORM_MSM8084)
         acdb_init_v2_t acdb_init;
@@ -1311,7 +1359,7 @@
 
     list_add_tail(operator_specific_device_table[snd_device], &device->list);
 
-    ALOGD("%s : deivce[%s] -> operator[%s] mixer_path[%s] acdb_id [%d]", __func__,
+    ALOGV("%s : deivce[%s] -> operator[%s] mixer_path[%s] acdb_id [%d]", __func__,
             platform_get_snd_device_name(snd_device), operator, mixer_path, acdb_id);
 
 }
@@ -1359,7 +1407,7 @@
         return -EINVAL;
     }
     if (my_data->acdb_send_audio_cal) {
-        ALOGD("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
+        ALOGV("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
               __func__, snd_device, acdb_dev_id);
         if (snd_device >= SND_DEVICE_OUT_BEGIN &&
                 snd_device < SND_DEVICE_OUT_END)
@@ -1968,8 +2016,14 @@
         } else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
             snd_device = SND_DEVICE_IN_VOICE_REC_HEADSET_MIC;
         }
+    } else if (source == AUDIO_SOURCE_UNPROCESSED) {
+        if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+            snd_device = SND_DEVICE_IN_UNPROCESSED_MIC;
+        } else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+            snd_device = SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC;
+        }
     } else if (source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
-            mode == AUDIO_MODE_IN_COMMUNICATION) {
+               mode == AUDIO_MODE_IN_COMMUNICATION) {
         if (out_device & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE))
             in_device = AUDIO_DEVICE_IN_BACK_MIC;
         if (adev->active_input) {
@@ -2356,7 +2410,7 @@
         info->mccmnc = strdup(str + strlen(name) + 1);
 
         list_add_tail(&operator_info_list, &info->list);
-        ALOGD("%s: add operator[%s] mccmnc[%s]", __func__, info->name, info->mccmnc);
+        ALOGV("%s: add operator[%s] mccmnc[%s]", __func__, info->name, info->mccmnc);
     }
 
     memset(value, 0, sizeof(value));
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index dcd763a..d6c9e8e 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2013-2014 The Android Open Source Project
+ * Copyright (C) 2013-2016 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -132,6 +132,9 @@
     SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
     SND_DEVICE_IN_VOICE_REC_HEADSET_MIC,
 
+    SND_DEVICE_IN_UNPROCESSED_MIC,
+    SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC,
+
     SND_DEVICE_IN_VOICE_RX,
 
     SND_DEVICE_IN_THREE_MIC,
@@ -158,6 +161,7 @@
 #define ACDB_ID_VOICE_HANDSET_TMUS 88
 #define ACDB_ID_VOICE_DMIC_EF_TMUS 89
 #define ACDB_ID_HEADSET_MIC_AEC 8
+#define ACDB_ID_VOICE_REC_MIC 62
 
 #define MAX_VOL_INDEX 5
 #define MIN_VOL_INDEX 0
@@ -218,6 +222,12 @@
 #define VOLTE_CALL_PCM_DEVICE 21
 #define QCHAT_CALL_PCM_DEVICE 33
 #define VOWLAN_CALL_PCM_DEVICE -1
+#elif PLATFORM_MSM8996
+#define VOICE_CALL_PCM_DEVICE 40
+#define VOICE2_CALL_PCM_DEVICE 41
+#define VOLTE_CALL_PCM_DEVICE 14
+#define QCHAT_CALL_PCM_DEVICE 20
+#define VOWLAN_CALL_PCM_DEVICE 33
 #else
 #define VOICE_CALL_PCM_DEVICE 2
 #define VOICE2_CALL_PCM_DEVICE 22
@@ -226,6 +236,14 @@
 #define VOWLAN_CALL_PCM_DEVICE 36
 #endif
 
+#ifdef PLATFORM_MSM8996
+#define VOICEMMODE1_CALL_PCM_DEVICE 2
+#define VOICEMMODE2_CALL_PCM_DEVICE 22
+#else
+#define VOICEMMODE1_CALL_PCM_DEVICE 44
+#define VOICEMMODE2_CALL_PCM_DEVICE 45
+#endif
+
 #define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
 #define AFE_PROXY_RECORD_PCM_DEVICE 8
 
diff --git a/hal/voice.h b/hal/voice.h
index 23b9ee3..469a3b5 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2014 The Android Open Source Project
+ * Copyright (C) 2014-2016 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -21,7 +21,7 @@
 #define VOICE_SESS_IDX     (BASE_SESS_IDX)
 
 #ifdef MULTI_VOICE_SESSION_ENABLED
-#define MAX_VOICE_SESSIONS 5
+#define MAX_VOICE_SESSIONS 7
 #else
 #define MAX_VOICE_SESSIONS 1
 #endif
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 6e92da8..edf5523 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2014 The Android Open Source Project
+ * Copyright (C) 2014-2016 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -41,17 +41,21 @@
 
 #define VOICE_EXTN_PARAMETER_VALUE_MAX_LEN 256
 
-#define VOICE2_VSID 0x10DC1000
-#define VOLTE_VSID  0x10C02000
-#define QCHAT_VSID  0x10803000
-#define VOWLAN_VSID 0x10002000
-#define ALL_VSID    0xFFFFFFFF
+#define VOICE2_VSID              0x10DC1000
+#define VOLTE_VSID               0x10C02000
+#define QCHAT_VSID               0x10803000
+#define VOWLAN_VSID              0x10002000
+#define VOICEMMODE1_VSID         0x11C05000
+#define VOICEMMODE2_VSID         0x11DC5000
+#define ALL_VSID                 0xFFFFFFFF
 
 /* Voice Session Indices */
 #define VOICE2_SESS_IDX    (VOICE_SESS_IDX + 1)
 #define VOLTE_SESS_IDX     (VOICE_SESS_IDX + 2)
 #define QCHAT_SESS_IDX     (VOICE_SESS_IDX + 3)
 #define VOWLAN_SESS_IDX    (VOICE_SESS_IDX + 4)
+#define MMODE1_SESS_IDX    (VOICE_SESS_IDX + 5)
+#define MMODE2_SESS_IDX    (VOICE_SESS_IDX + 6)
 
 /* Call States */
 #define CALL_HOLD           (BASE_CALL_STATE + 2)
@@ -84,6 +88,8 @@
         vsid == VOICE2_VSID ||
         vsid == VOLTE_VSID ||
         vsid == QCHAT_VSID ||
+        vsid == VOICEMMODE1_VSID ||
+        vsid == VOICEMMODE2_VSID ||
         vsid == VOWLAN_VSID)
         return true;
     else
@@ -115,6 +121,14 @@
         usecase_id = USECASE_VOWLAN_CALL;
         break;
 
+    case MMODE1_SESS_IDX:
+        usecase_id = USECASE_VOICEMMODE1_CALL;
+        break;
+
+    case MMODE2_SESS_IDX:
+        usecase_id = USECASE_VOICEMMODE2_CALL;
+        break;
+
     default:
         ALOGE("%s: Invalid voice session index\n", __func__);
     }
@@ -356,6 +370,8 @@
     adev->voice.session[VOLTE_SESS_IDX].vsid =  VOLTE_VSID;
     adev->voice.session[QCHAT_SESS_IDX].vsid =  QCHAT_VSID;
     adev->voice.session[VOWLAN_SESS_IDX].vsid = VOWLAN_VSID;
+    adev->voice.session[MMODE1_SESS_IDX].vsid = VOICEMMODE1_VSID;
+    adev->voice.session[MMODE2_SESS_IDX].vsid = VOICEMMODE2_VSID;
 }
 
 int voice_extn_get_session_from_use_case(struct audio_device *adev,
@@ -385,6 +401,14 @@
         *session = &adev->voice.session[VOWLAN_SESS_IDX];
         break;
 
+    case USECASE_VOICEMMODE1_CALL:
+        *session = &adev->voice.session[MMODE1_SESS_IDX];
+        break;
+
+    case USECASE_VOICEMMODE2_CALL:
+        *session = &adev->voice.session[MMODE2_SESS_IDX];
+        break;
+
     default:
         ALOGE("%s: Invalid usecase_id:%d\n", __func__, usecase_id);
         *session = NULL;
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index b8aa9fc..1a8550c 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -1,4 +1,4 @@
-ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
 
 LOCAL_PATH:= $(call my-dir)
 
@@ -33,7 +33,7 @@
 
 ################################################################################
 
-ifneq ($(filter msm8992 msm8994,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
 
 include $(CLEAR_VARS)
 
diff --git a/post_proc/volume_listener.c b/post_proc/volume_listener.c
index ef63299..107a475 100644
--- a/post_proc/volume_listener.c
+++ b/post_proc/volume_listener.c
@@ -346,9 +346,6 @@
             memcpy(out_buffer->raw, in_buffer->raw, out_buffer->frameCount * 2 * sizeof(int16_t));
         }
 
-    } else {
-        ALOGW("%s: something wrong, didn't handle in_buffer and out_buffer same address case",
-              __func__);
     }
 
 exit: