diff --git a/hal/Android.mk b/hal/Android.mk
index d25c926..a7e0a02 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -26,6 +26,7 @@
 LOCAL_SRC_FILES := \
 	audio_hw.c \
 	voice.c \
+	platform_info.c \
 	$(AUDIO_PLATFORM)/platform.c
 
 LOCAL_SRC_FILES += audio_extn/audio_extn.c
@@ -114,23 +115,19 @@
 	libtinyalsa \
 	libtinycompress \
 	libaudioroute \
-	libdl
+	libdl \
+	libexpat
 
 LOCAL_C_INCLUDES += \
 	external/tinyalsa/include \
 	external/tinycompress/include \
+	external/expat/lib \
 	$(call include-path-for, audio-route) \
 	$(call include-path-for, audio-effects) \
 	$(LOCAL_PATH)/$(AUDIO_PLATFORM) \
 	$(LOCAL_PATH)/audio_extn \
 	$(LOCAL_PATH)/voice_extn
 
-ifneq ($(filter msm8974,$(AUDIO_PLATFORM)),)
-    LOCAL_C_INCLUDES += external/expat/lib
-    LOCAL_SHARED_LIBRARIES += libexpat
-    LOCAL_SRC_FILES += $(AUDIO_PLATFORM)/platform_parser.c
-endif
-
 ifeq ($(strip $(AUDIO_FEATURE_ENABLED_LISTEN)),true)
     LOCAL_CFLAGS += -DAUDIO_LISTEN_ENABLED
     LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-listen
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index cfcc5a5..1a7b83e 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -152,8 +152,55 @@
 static unsigned int audio_device_ref_count;
 
 static int set_voice_volume_l(struct audio_device *adev, float volume);
-static uint32_t get_offload_buffer_size();
-static int set_gapless_mode(struct audio_device *adev);
+
+/* Read  offload buffer size from a property.
+ * If value is not power of 2  round it to
+ * power of 2.
+ */
+static uint32_t get_offload_buffer_size()
+{
+    char value[PROPERTY_VALUE_MAX] = {0};
+    uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+    if((property_get("audio.offload.buffer.size.kb", value, "")) &&
+            atoi(value)) {
+        fragment_size =  atoi(value) * 1024;
+        //ring buffer size needs to be 4k aligned.
+        CHECK(!(fragment_size * COMPRESS_OFFLOAD_NUM_FRAGMENTS % 4096));
+    }
+    if(fragment_size < MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
+        fragment_size = MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+    else if(fragment_size > MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
+        fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+    ALOGVV("%s: fragment_size %d", __func__, fragment_size);
+    return fragment_size;
+}
+
+static int check_and_set_gapless_mode(struct audio_device *adev) {
+
+
+    char value[PROPERTY_VALUE_MAX] = {0};
+    bool gapless_enabled = false;
+    const char *mixer_ctl_name = "Compress Gapless Playback";
+    struct mixer_ctl *ctl;
+
+    ALOGV("%s:", __func__);
+    property_get("audio.offload.gapless.enabled", value, NULL);
+    gapless_enabled = atoi(value) || !strncmp("true", value, 4);
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+                               __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+
+    if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
+        ALOGE("%s: Could not set gapless mode %d",
+                       __func__, gapless_enabled);
+         return -EINVAL;
+    }
+    return 0;
+}
 
 static bool is_supported_format(audio_format_t format)
 {
@@ -835,6 +882,7 @@
 {
     struct stream_out *out = (struct stream_out *) context;
     struct listnode *item;
+    int ret = 0;
 
     setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
     set_sched_policy(0, SP_FOREGROUND);
@@ -884,8 +932,14 @@
             event = STREAM_CBK_EVENT_WRITE_READY;
             break;
         case OFFLOAD_CMD_PARTIAL_DRAIN:
-            compress_next_track(out->compr);
-            compress_partial_drain(out->compr);
+            ret = compress_next_track(out->compr);
+            if(ret == 0)
+                compress_partial_drain(out->compr);
+            else if(ret == -ETIMEDOUT)
+                compress_drain(out->compr);
+            else
+                ALOGE("%s: Next track returned error %d",__func__, ret);
+
             send_callback = true;
             event = STREAM_CBK_EVENT_DRAIN_READY;
             break;
@@ -2127,7 +2181,7 @@
                 __func__, config->offload_info.version,
                 config->offload_info.bit_rate);
         //Decide if we need to use gapless mode by default
-        set_gapless_mode(adev);
+        check_and_set_gapless_mode(adev);
 
     } else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
         ret = voice_check_and_set_incall_music_usecase(adev, out);
@@ -2655,55 +2709,6 @@
     return 0;
 }
 
-/* Read  offload buffer size from a property.
- * If value is not power of 2  round it to
- * power of 2.
- */
-static uint32_t get_offload_buffer_size()
-{
-    char value[PROPERTY_VALUE_MAX] = {0};
-    uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
-    if((property_get("audio.offload.buffer.size.kb", value, "")) &&
-            atoi(value)) {
-        fragment_size =  atoi(value) * 1024;
-        //ring buffer size needs to be 4k aligned.
-        CHECK(!(fragment_size * COMPRESS_OFFLOAD_NUM_FRAGMENTS % 4096));
-    }
-    if(fragment_size < MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
-        fragment_size = MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
-    else if(fragment_size > MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
-        fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
-    ALOGVV("%s: fragment_size %d", __func__, fragment_size);
-    return fragment_size;
-}
-
-static int set_gapless_mode(struct audio_device *adev) {
-
-
-    char value[PROPERTY_VALUE_MAX] = {0};
-    bool gapless_enabled = false;
-    const char *mixer_ctl_name = "Compress Gapless Playback";
-    struct mixer_ctl *ctl;
-
-    ALOGV("%s:", __func__);
-    property_get("audio.offload.gapless.enabled", value, NULL);
-    gapless_enabled = atoi(value) || !strncmp("true", value, 4);
-
-    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
-    if (!ctl) {
-        ALOGE("%s: Could not get ctl for mixer cmd - %s",
-                               __func__, mixer_ctl_name);
-        return -EINVAL;
-    }
-
-    if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
-        ALOGE("%s: Could not set gapless mode %d",
-                       __func__, gapless_enabled);
-         return -EINVAL;
-    }
-    return 0;
-
-}
 static struct hw_module_methods_t hal_module_methods = {
     .open = adev_open,
 };
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 54bda5a..c1ba595 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -391,6 +391,11 @@
     return device_id;
 }
 
+int platform_get_snd_device_index(char *snd_device_index_name)
+{
+    return -ENODEV;
+}
+
 int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id)
 {
     return -ENODEV;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index b4d8c5d..a25991f 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -31,7 +31,6 @@
 #include "platform.h"
 #include "audio_extn.h"
 #include "voice_extn.h"
-#include "platform_parser.h"
 
 #define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
 #define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
@@ -295,6 +294,84 @@
     [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
 };
 
+struct snd_device_index {
+    char name[100];
+    unsigned int index;
+};
+
+#define TO_NAME_INDEX(X)   #X, X
+
+/* Used to get index from parsed sting */
+struct snd_device_index snd_device_name_index[SND_DEVICE_MAX] = {
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_AFE_PROXY)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_USB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_TRANSMISSION_FM)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_FB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_ANC_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HEADSET_MIC_FLUENCE)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HDMI_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_QMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_MIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_DMIC_STEREO)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_USB_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_FM)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_AANC_HANDSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_QUAD_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+};
+
 #define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
 #define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
 
@@ -673,6 +750,30 @@
     return device_id;
 }
 
+int platform_get_snd_device_index(char *snd_device_index_name)
+{
+    int ret = 0;
+    int i;
+
+    if (snd_device_index_name == NULL) {
+        ALOGE("%s: snd_device_index_name is NULL", __func__);
+        ret = -ENODEV;
+        goto done;
+    }
+
+    for (i=0; i < SND_DEVICE_MAX; i++) {
+        if(strcmp(snd_device_name_index[i].name, snd_device_index_name) == 0) {
+            ret = snd_device_name_index[i].index;
+            goto done;
+        }
+    }
+    ALOGE("%s: Could not find index for snd_device_index_name = %s",
+            __func__, snd_device_index_name);
+    ret = -ENODEV;
+done:
+    return ret;
+}
+
 int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id)
 {
     int ret = 0;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index bb1f787..ca8469a 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -147,7 +147,7 @@
  * the buffer size of an input/output stream
  */
 #define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
-#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 4
 #define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
 #define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
 
diff --git a/hal/msm8974/platform_parser.h b/hal/msm8974/platform_parser.h
deleted file mode 100644
index 3e91934..0000000
--- a/hal/msm8974/platform_parser.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are
- * met:
- *     * Redistributions of source code must retain the above copyright
- *       notice, this list of conditions and the following disclaimer.
- *     * Redistributions in binary form must reproduce the above
- *       copyright notice, this list of conditions and the following
- *       disclaimer in the documentation and/or other materials provided
- *       with the distribution.
- *     * Neither the name of The Linux Foundation nor the names of its
- *       contributors may be used to endorse or promote products derived
- *       from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
- * ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
- * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
- * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
- * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
- * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
- * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
- * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef AUDIO_PLATFORM_PARSER_H
-#define AUDIO_PLATFORM_PARSER_H
-
-int platform_info_init(void);
-
-#endif // AUDIO_PLATFORM_PARSER_H
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 0b2c435..a5f5074 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -27,6 +27,7 @@
                                       char *device_name);
 void platform_add_backend_name(char *mixer_path, snd_device_t snd_device);
 int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type);
+int platform_get_snd_device_index(char *snd_device_index_name);
 int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id);
 int platform_send_audio_calibration(void *platform, snd_device_t snd_device);
 int platform_switch_voice_call_device_pre(void *platform);
@@ -58,4 +59,7 @@
 
 bool platform_listen_update_status(snd_device_t snd_device);
 
+/* From platform_info_parser.c */
+int platform_info_init(void);
+
 #endif // AUDIO_PLATFORM_API_H
diff --git a/hal/msm8974/platform_parser.c b/hal/platform_info.c
similarity index 70%
rename from hal/msm8974/platform_parser.c
rename to hal/platform_info.c
index 8f86d97..8f56107 100644
--- a/hal/msm8974/platform_parser.c
+++ b/hal/platform_info.c
@@ -27,7 +27,7 @@
  * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#define LOG_TAG "platform_parser"
+#define LOG_TAG "platform_info"
 #define LOG_NDDEBUG 0
 
 #include <errno.h>
@@ -35,41 +35,40 @@
 #include <expat.h>
 #include <cutils/log.h>
 #include <audio_hw.h>
-#include <platform_api.h>
-#include "platform.h"
-#include "platform_parser.h"
+#include "platform_api.h"
+#include <platform.h>
 
-#define PLATFORM_XML_PATH        "/system/etc/platform_info.xml"
+#define PLATFORM_INFO_XML_PATH      "/system/etc/audio_platform_info.xml"
 #define BUF_SIZE                    1024
 
-static void process_device(void *userdata, const XML_Char **attr)
+static void process_device(const XML_Char **attr)
 {
-    unsigned int    *snd_device_index = userdata;
+    int index;
 
-    if (strcmp(attr[0], "name") != 0)
+    if (strcmp(attr[0], "name") != 0) {
+        ALOGE("%s: 'name' not found, no ACDB ID set!", __func__);
         goto done;
+    }
 
-    if (platform_get_snd_device_name(*snd_device_index) == NULL)
-        goto next;
-    if (strcmp(attr[1], platform_get_snd_device_name(*snd_device_index)) != 0) {
-        ALOGE("%s: %s in platform.h at index %d does not match %s, from %s no ACDB ID set!",
-            __func__, platform_get_snd_device_name(*snd_device_index),
-            *snd_device_index, attr[1], PLATFORM_XML_PATH);
+    index = platform_get_snd_device_index((char *)attr[1]);
+    if (index < 0) {
+        ALOGE("%s: Device %s in %s not found, no ACDB ID set!",
+              __func__, attr[1], PLATFORM_INFO_XML_PATH);
         goto done;
     }
 
     if (strcmp(attr[2], "acdb_id") != 0) {
-        ALOGE("%s: Device %s at index %d in %s has no acdb_id, no ACDB ID set!",
-              __func__, attr[1], *snd_device_index, PLATFORM_XML_PATH);
+        ALOGE("%s: Device %s in %s has no acdb_id, no ACDB ID set!",
+              __func__, attr[1], PLATFORM_INFO_XML_PATH);
         goto done;
     }
 
-    if(platform_set_snd_device_acdb_id(*snd_device_index,
-                                atoi((char *)attr[3])) != 0)
+    if(platform_set_snd_device_acdb_id(index, atoi((char *)attr[3])) < 0) {
+        ALOGE("%s: Device %s in %s, ACDB ID %d was not set!",
+              __func__, attr[1], PLATFORM_INFO_XML_PATH, atoi((char *)attr[3]));
         goto done;
+    }
 
-next:
-     (*snd_device_index)++;
 done:
     return;
 }
@@ -82,7 +81,7 @@
     unsigned int                i;
 
     if (strcmp(tag_name, "device") == 0)
-        process_device(userdata, attr);
+        process_device(attr);
 
     return;
 }
@@ -98,13 +97,12 @@
     FILE            *file;
     int             ret = 0;
     int             bytes_read;
-    unsigned int    snd_device_index = SND_DEVICE_MIN;
     void            *buf;
 
-    file = fopen(PLATFORM_XML_PATH, "r");
+    file = fopen(PLATFORM_INFO_XML_PATH, "r");
     if (!file) {
         ALOGD("%s: Failed to open %s, using defaults.",
-            __func__, PLATFORM_XML_PATH);
+            __func__, PLATFORM_INFO_XML_PATH);
         ret = -ENODEV;
         goto done;
     }
@@ -116,7 +114,6 @@
         goto err_close_file;
     }
 
-    XML_SetUserData(parser, &snd_device_index);
     XML_SetElementHandler(parser, start_tag, end_tag);
 
     while (1) {
@@ -137,7 +134,7 @@
         if (XML_ParseBuffer(parser, bytes_read,
                             bytes_read == 0) == XML_STATUS_ERROR) {
             ALOGE("%s: XML_ParseBuffer failed, for %s",
-                __func__, PLATFORM_XML_PATH);
+                __func__, PLATFORM_INFO_XML_PATH);
             ret = -EINVAL;
             goto err_free_parser;
         }
@@ -146,12 +143,6 @@
             break;
     }
 
-    if (snd_device_index != SND_DEVICE_MAX) {
-        ALOGE("%s: Only %d/%d ACDB ID's set! Fix %s!",
-            __func__, snd_device_index, SND_DEVICE_MAX, PLATFORM_XML_PATH);
-        ret = -EINVAL;
-    }
-
 err_free_parser:
     XML_ParserFree(parser);
 err_close_file:
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
index b6a06e4..c68ab6e 100644
--- a/policy_hal/Android.mk
+++ b/policy_hal/Android.mk
@@ -30,6 +30,14 @@
 LOCAL_CFLAGS += -DAUDIO_EXTN_INCALL_MUSIC_ENABLED
 endif
 
+
+ifeq ($(strip $(TARGET_BOARD_PLATFORM)),msm8916)
+LOCAL_CFLAGS += -DVOICE_CONCURRENCY
+LOCAL_CFLAGS += -DWFD_CONCURRENCY
+endif
+
+
+
 include $(BUILD_SHARED_LIBRARY)
 
 endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index f64bbfe..5142353 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -418,12 +418,36 @@
 
 AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(AudioSystem::stream_type stream)
 {
-#ifdef QCOM_INCALL_MUSIC_ENABLED
-    if (stream == AudioSystem::INCALL_MUSIC)
-        return STRATEGY_MEDIA;
+       // stream to strategy mapping
+        switch (stream) {
+        case AudioSystem::VOICE_CALL:
+        case AudioSystem::BLUETOOTH_SCO:
+            return STRATEGY_PHONE;
+        case AudioSystem::RING:
+        case AudioSystem::ALARM:
+            return STRATEGY_SONIFICATION;
+        case AudioSystem::NOTIFICATION:
+            return STRATEGY_SONIFICATION_RESPECTFUL;
+        case AudioSystem::DTMF:
+            return STRATEGY_DTMF;
+        default:
+            ALOGE("unknown stream type");
+        case AudioSystem::SYSTEM:
+            // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+            // while key clicks are played produces a poor result
+        case AudioSystem::TTS:
+        case AudioSystem::MUSIC:
+#ifdef AUDIO_EXTN_INCALL_MUSIC_ENABLED
+        case AudioSystem::INCALL_MUSIC:
 #endif
+#ifdef QCOM_INCALL_MUSIC_ENABLED
+        case AudioSystem::INCALL_MUSIC:
+#endif
+             return STRATEGY_MEDIA;
+        case AudioSystem::ENFORCED_AUDIBLE:
+            return STRATEGY_ENFORCED_AUDIBLE;
+    }
 
-    return getStrategy(stream);
 }
 
 audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
@@ -632,7 +656,8 @@
         if (device2 == AUDIO_DEVICE_NONE) {
             device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
         }
-        if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+        if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+             && (device2 == AUDIO_DEVICE_NONE)) {
             // no sonification on aux digital (e.g. HDMI)
             device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
         }
@@ -642,12 +667,14 @@
             device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
         }
 #ifdef AUDIO_EXTN_FM_ENABLED
-            if ((strategy != STRATEGY_SONIFICATION) && (device2 == AUDIO_DEVICE_NONE)) {
+            if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+                 && (device2 == AUDIO_DEVICE_NONE)) {
                 device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM_TX;
             }
 #endif
 #ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
-            if ((strategy != STRATEGY_SONIFICATION) && (device2 == AUDIO_DEVICE_NONE)) {
+            if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+                 && (device2 == AUDIO_DEVICE_NONE)) {
                 // no sonification on WFD sink
                 device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY;
             }
@@ -878,6 +905,422 @@
 #endif
     return AudioPolicyManagerBase::computeVolume(stream, index, output, device);
 }
+
+
+audio_io_handle_t AudioPolicyManager::getOutput(AudioSystem::stream_type stream,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channelMask,
+                                    AudioSystem::output_flags flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    audio_io_handle_t output = 0;
+    uint32_t latency = 0;
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    IOProfile *profile = NULL;
+
+#ifdef VOICE_CONCURRENCY
+    if (isInCall()) {
+        ALOGV(" IN call mode adding ULL flags .. flags: %x ", flags );
+        //For voip paths
+        if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
+            flags = AudioSystem::OUTPUT_FLAG_DIRECT;
+        else //route every thing else to ULL path
+            flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+    }
+#endif
+
+#ifdef WFD_CONCURRENCY
+    if ((mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY)
+          && (stream != AudioSystem::MUSIC)) {
+        ALOGV(" WFD mode adding ULL flags for non music stream.. flags: %x ", flags );
+        //For voip paths
+        if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
+            flags = AudioSystem::OUTPUT_FLAG_DIRECT;
+        else //route every thing else to ULL path
+            flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+    }
+#endif
+
+    ALOGV(" getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x ",
+          device, stream, samplingRate, format, channelMask, flags);
+
+
+
+#ifdef AUDIO_POLICY_TEST
+    if (mCurOutput != 0) {
+        ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+        if (mTestOutputs[mCurOutput] == 0) {
+            ALOGV("getOutput() opening test output");
+            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+            outputDesc->mDevice = mTestDevice;
+            outputDesc->mSamplingRate = mTestSamplingRate;
+            outputDesc->mFormat = mTestFormat;
+            outputDesc->mChannelMask = mTestChannels;
+            outputDesc->mLatency = mTestLatencyMs;
+            outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+            outputDesc->mRefCount[stream] = 0;
+            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+                                            &outputDesc->mSamplingRate,
+                                            &outputDesc->mFormat,
+                                            &outputDesc->mChannelMask,
+                                            &outputDesc->mLatency,
+                                            outputDesc->mFlags,
+                                            offloadInfo);
+            if (mTestOutputs[mCurOutput]) {
+                AudioParameter outputCmd = AudioParameter();
+                outputCmd.addInt(String8("set_id"),mCurOutput);
+                mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+                addOutput(mTestOutputs[mCurOutput], outputDesc);
+            }
+        }
+        return mTestOutputs[mCurOutput];
+    }
+#endif //AUDIO_POLICY_TEST
+
+    // open a direct output if required by specified parameters
+    //force direct flag if offload flag is set: offloading implies a direct output stream
+    // and all common behaviors are driven by checking only the direct flag
+    // this should normally be set appropriately in the policy configuration file
+    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+    }
+
+    if ((format == AudioSystem::PCM_16_BIT) &&(AudioSystem::popCount(channelMask) > 2)) {
+        ALOGV("owerwrite flag(%x) for PCM16 multi-channel(CM:%x) playback", flags ,channelMask);
+        flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_DIRECT;
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+    // creating an offloaded track and tearing it down immediately after start when audioflinger
+    // detects there is an active non offloadable effect.
+    // FIXME: We should check the audio session here but we do not have it in this context.
+    // This may prevent offloading in rare situations where effects are left active by apps
+    // in the background.
+    if ((((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+            !isNonOffloadableEffectEnabled()) &&
+            flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        profile = getProfileForDirectOutput(device,
+                                           samplingRate,
+                                           format,
+                                           channelMask,
+                                           (audio_output_flags_t)flags);
+    }
+
+    if (profile != NULL) {
+        AudioOutputDescriptor *outputDesc = NULL;
+
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+                outputDesc = desc;
+                // reuse direct output if currently open and configured with same parameters
+                if ((samplingRate == outputDesc->mSamplingRate) &&
+                        (format == outputDesc->mFormat) &&
+                        (channelMask == outputDesc->mChannelMask)) {
+                    outputDesc->mDirectOpenCount++;
+                    ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+                    return mOutputs.keyAt(i);
+                }
+            }
+        }
+        // close direct output if currently open and configured with different parameters
+        if (outputDesc != NULL) {
+            closeOutput(outputDesc->mId);
+        }
+        outputDesc = new AudioOutputDescriptor(profile);
+        outputDesc->mDevice = device;
+        outputDesc->mSamplingRate = samplingRate;
+        outputDesc->mFormat = (audio_format_t)format;
+        outputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
+        outputDesc->mLatency = 0;
+        outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+        outputDesc->mRefCount[stream] = 0;
+        outputDesc->mStopTime[stream] = 0;
+        outputDesc->mDirectOpenCount = 1;
+        output = mpClientInterface->openOutput(profile->mModule->mHandle,
+                                        &outputDesc->mDevice,
+                                        &outputDesc->mSamplingRate,
+                                        &outputDesc->mFormat,
+                                        &outputDesc->mChannelMask,
+                                        &outputDesc->mLatency,
+                                        outputDesc->mFlags,
+                                        offloadInfo);
+
+        // only accept an output with the requested parameters
+        if (output == 0 ||
+            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+            (format != 0 && format != outputDesc->mFormat) ||
+            (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+            ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+                    "format %d %d, channelMask %04x %04x", output, samplingRate,
+                    outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+                    outputDesc->mChannelMask);
+            if (output != 0) {
+                mpClientInterface->closeOutput(output);
+            }
+            delete outputDesc;
+            return 0;
+        }
+        audio_io_handle_t srcOutput = getOutputForEffect();
+        addOutput(output, outputDesc);
+        audio_io_handle_t dstOutput = getOutputForEffect();
+        if (dstOutput == output) {
+            mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+        }
+        mPreviousOutputs = mOutputs;
+        ALOGV("getOutput() returns new direct output %d", output);
+        return output;
+    }
+
+    // ignoring channel mask due to downmix capability in mixer
+
+    // open a non direct output
+
+    // for non direct outputs, only PCM is supported
+    if (audio_is_linear_pcm((audio_format_t)format)) {
+        // get which output is suitable for the specified stream. The actual
+        // routing change will happen when startOutput() will be called
+        SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+        output = selectOutput(outputs, flags);
+    }
+    ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+            "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+    ALOGV("getOutput() returns output %d", output);
+
+    return output;
+}
+
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+    ALOGV(" isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+     " BitRate=%u, duration=%lld us, has_video=%d",
+     offloadInfo.sample_rate, offloadInfo.channel_mask,
+     offloadInfo.format,
+     offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+     offloadInfo.has_video);
+
+#ifdef VOICE_CONCURRENCY
+    if(isInCall())
+    {
+        ALOGD("\n  blocking  compress offload on call mode\n");
+        return false;
+    }
+#endif
+
+    // Check if offload has been disabled
+    char propValue[PROPERTY_VALUE_MAX];
+    if (property_get("audio.offload.disable", propValue, "0")) {
+        if (atoi(propValue) != 0) {
+            ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+            return false;
+        }
+    }
+
+    // Check if stream type is music, then only allow offload as of now.
+    if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+    {
+        ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+        return false;
+    }
+
+    //TODO: enable audio offloading with video when ready
+    if (offloadInfo.has_video)
+    {
+        if(property_get("av.offload.enable", propValue, NULL)) {
+            bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+            if (!prop_enabled) {
+               ALOGW("offload disabled by av.offload.enable = %s ", propValue );
+               return false;
+            }
+        }
+        if(offloadInfo.is_streaming &&
+           property_get("av.streaming.offload.enable", propValue, NULL)) {
+            bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+            if (!prop_enabled) {
+               ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
+               return false;
+            }
+        }
+        ALOGV("isOffloadSupported: has_video == true, property\
+                set to enable offload");
+    }
+
+    //If duration is less than minimum value defined in property, return false
+    if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+        if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+            ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+            return false;
+        }
+    } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+        ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+        //duration checks only valid for MP3/AAC formats,
+        //do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
+        if (offloadInfo.format == AUDIO_FORMAT_MP3 || offloadInfo.format == AUDIO_FORMAT_AAC)
+            return false;
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+    // creating an offloaded track and tearing it down immediately after start when audioflinger
+    // detects there is an active non offloadable effect.
+    // FIXME: We should check the audio session here but we do not have it in this context.
+    // This may prevent offloading in rare situations where effects are left active by apps
+    // in the background.
+    if (isNonOffloadableEffectEnabled()) {
+        return false;
+    }
+
+    // See if there is a profile to support this.
+    // AUDIO_DEVICE_NONE
+    IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+                                            offloadInfo.sample_rate,
+                                            offloadInfo.format,
+                                            offloadInfo.channel_mask,
+                                            AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+    ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
+    return (profile != NULL);
+}
+
+void AudioPolicyManager::setPhoneState(int state)
+
+{
+    ALOGV("setPhoneState() state %d", state);
+    audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+    if (state < 0 || state >= AudioSystem::NUM_MODES) {
+        ALOGW("setPhoneState() invalid state %d", state);
+        return;
+    }
+
+    if (state == mPhoneState ) {
+        ALOGW("setPhoneState() setting same state %d", state);
+        return;
+    }
+
+    // if leaving call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isInCall()) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, false, true);
+        }
+    }
+
+    // store previous phone state for management of sonification strategy below
+    int oldState = mPhoneState;
+    mPhoneState = state;
+    bool force = false;
+
+    // are we entering or starting a call
+    if (!isStateInCall(oldState) && isStateInCall(state)) {
+        ALOGV("  Entering call in setPhoneState()");
+        // force routing command to audio hardware when starting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+        }
+    } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+        ALOGV("  Exiting call in setPhoneState()");
+        // force routing command to audio hardware when exiting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+        }
+    } else if (isStateInCall(state) && (state != oldState)) {
+        ALOGV("  Switching between telephony and VoIP in setPhoneState()");
+        // force routing command to audio hardware when switching between telephony and VoIP
+        // even if no device change is needed
+        force = true;
+    }
+
+    // check for device and output changes triggered by new phone state
+    newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+
+    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+    // force routing command to audio hardware when ending call
+    // even if no device change is needed
+    if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
+        newDevice = hwOutputDesc->device();
+    }
+
+    int delayMs = 0;
+    if (isStateInCall(state)) {
+        nsecs_t sysTime = systemTime();
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            // mute media and sonification strategies and delay device switch by the largest
+            // latency of any output where either strategy is active.
+            // This avoid sending the ring tone or music tail into the earpiece or headset.
+            if ((desc->isStrategyActive(STRATEGY_MEDIA,
+                                     SONIFICATION_HEADSET_MUSIC_DELAY,
+                                     sysTime) ||
+                    desc->isStrategyActive(STRATEGY_SONIFICATION,
+                                         SONIFICATION_HEADSET_MUSIC_DELAY,
+                                         sysTime)) &&
+                    (delayMs < (int)desc->mLatency*2)) {
+                delayMs = desc->mLatency*2;
+            }
+            setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+            setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+        }
+    }
+
+    // change routing is necessary
+    setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
+
+    // if entering in call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isStateInCall(state)) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, true, true);
+        }
+    }
+
+    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+    if (state == AudioSystem::MODE_RINGTONE &&
+        isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+        mLimitRingtoneVolume = true;
+    } else {
+        mLimitRingtoneVolume = false;
+    }
+
+#ifdef VOICE_CONCURRENCY
+    //Call invalidate to reset all opened non ULL audio tracks
+    if(isInCall())
+    {
+        // Move tracks associated to this strategy from previous output to new output
+        for (int i = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                ALOGV("\n Invalidate on call mode for stream :: %d  \n", i);
+                //FIXME see fixme on name change
+                mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
+                                                  0 /* ignored */);
+        }
+    }
+#endif
+
+}
+
 extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
 {
     return new AudioPolicyManager(clientInterface);
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 7a8cfa9..34ca701 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -48,6 +48,17 @@
                                             uint32_t format,
                                             uint32_t channels,
                                             AudioSystem::audio_in_acoustics acoustics);
+        virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                            uint32_t samplingRate = 0,
+                                            uint32_t format = AudioSystem::FORMAT_DEFAULT,
+                                            uint32_t channels = 0,
+                                            AudioSystem::output_flags flags =
+                                                    AudioSystem::OUTPUT_FLAG_INDIRECT,
+                                            const audio_offload_info_t *offloadInfo = NULL);
+
+        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+        virtual void setPhoneState(int state);
 protected:
         // return the strategy corresponding to a given stream type
         static routing_strategy getStrategy(AudioSystem::stream_type stream);
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index c64ba6b..c724b58 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -239,6 +239,11 @@
     ALOGV("%s", __func__);
     bass_ctxt->ctl = output->ctl;
     ALOGV("output->ctl: %p", output->ctl);
+    if (offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass)))
+        if (bass_ctxt->ctl)
+            offload_bassboost_send_params(bass_ctxt->ctl, bass_ctxt->offload_bass,
+                                          OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG |
+                                          OFFLOAD_SEND_BASSBOOST_STRENGTH);
     return 0;
 }
 
diff --git a/post_proc/equalizer.c b/post_proc/equalizer.c
index e31d2b9..7c7ced2 100644
--- a/post_proc/equalizer.c
+++ b/post_proc/equalizer.c
@@ -491,6 +491,11 @@
 
     ALOGV("%s: %p", __func__, output->ctl);
     eq_ctxt->ctl = output->ctl;
+    if (offload_eq_get_enable_flag(&(eq_ctxt->offload_eq)))
+        if (eq_ctxt->ctl)
+            offload_eq_send_params(eq_ctxt->ctl, eq_ctxt->offload_eq,
+                                   OFFLOAD_SEND_EQ_ENABLE_FLAG |
+                                   OFFLOAD_SEND_EQ_BANDS_LEVEL);
     return 0;
 }
 
diff --git a/post_proc/reverb.c b/post_proc/reverb.c
index 4fc8c83..d104073 100644
--- a/post_proc/reverb.c
+++ b/post_proc/reverb.c
@@ -236,9 +236,14 @@
 
 void reverb_set_preset(reverb_context_t *context, int16_t preset)
 {
+    bool enable;
     ALOGV("%s: preset: %d", __func__, preset);
     context->next_preset = preset;
     offload_reverb_set_preset(&(context->offload_reverb), preset);
+
+    enable = (preset == REVERB_PRESET_NONE) ? false: true;
+    offload_reverb_set_enable_flag(&(context->offload_reverb), enable);
+
     if (context->ctl)
         offload_reverb_send_params(context->ctl, context->offload_reverb,
                                    OFFLOAD_SEND_REVERB_ENABLE_FLAG |
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 2f0ca6b..e9eb728 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -237,6 +237,11 @@
 
     ALOGV("%s", __func__);
     virt_ctxt->ctl = output->ctl;
+    if (offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt)))
+        if (virt_ctxt->ctl)
+            offload_virtualizer_send_params(virt_ctxt->ctl, virt_ctxt->offload_virt,
+                                          OFFLOAD_SEND_VIRTUALIZER_ENABLE_FLAG |
+                                          OFFLOAD_SEND_VIRTUALIZER_STRENGTH);
     return 0;
 }