diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 1b008db..04a7aba 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -53,8 +53,10 @@
 #include "voice_extn.h"
 
 #include "sound/compress_params.h"
+
 #define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
-#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (8 * 1024)
+#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
+#define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
 #define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
 /* ToDo: Check and update a proper value in msec */
@@ -157,7 +159,7 @@
  * If value is not power of 2  round it to
  * power of 2.
  */
-static uint32_t get_offload_buffer_size()
+static uint32_t get_offload_buffer_size(audio_offload_info_t* info)
 {
     char value[PROPERTY_VALUE_MAX] = {0};
     uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
@@ -167,6 +169,13 @@
         //ring buffer size needs to be 4k aligned.
         CHECK(!(fragment_size * COMPRESS_OFFLOAD_NUM_FRAGMENTS % 4096));
     }
+
+    if (info != NULL && info->has_video && info->is_streaming) {
+        fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
+        ALOGV("%s: offload fragment size reduced for AV streaming to %d",
+               __func__, out->compr_config.fragment_size);
+    }
+
     if(fragment_size < MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
         fragment_size = MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
     else if(fragment_size > MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
@@ -1347,11 +1356,21 @@
     struct compr_gapless_mdata tmp_mdata;
     tmp_mdata.encoder_delay = 0;
     tmp_mdata.encoder_padding = 0;
+
     if (!out || !parms) {
         ALOGE("%s: return invalid ",__func__);
         return -EINVAL;
     }
 
+    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FORMAT, value, sizeof(value));
+    if (ret >= 0) {
+        if (atoi(value) == SND_AUDIOSTREAMFORMAT_MP4ADTS) {
+            out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
+            ALOGV("ADTS format is set in offload mode");
+        }
+        out->send_new_metadata = 1;
+    }
+
     ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_SAMPLE_RATE, value, sizeof(value));
     if(ret >= 0)
         is_meta_data_params = true;
@@ -1512,7 +1531,8 @@
         voice_extn_out_get_parameters(out, query, reply);
         str = str_parms_to_str(reply);
         if (!strncmp(str, "", sizeof(""))) {
-           str = strdup(keys);
+            free(str);
+            str = strdup(keys);
         }
     }
     str_parms_destroy(query);
@@ -1889,6 +1909,18 @@
         /* no audio source uses val == 0 */
         if ((in->source != val) && (val != 0)) {
             in->source = val;
+            if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
+                (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+                (voice_extn_compress_voip_is_format_supported(in->format)) &&
+                (in->config.rate == 8000 || in->config.rate == 16000) &&
+                (popcount(in->channel_mask) == 1)) {
+                ret = voice_extn_compress_voip_open_input_stream(in);
+                if (ret != 0) {
+                    ALOGE("%s: Compress voip input cannot be opened, error:%d",
+                          __func__, ret);
+                    goto done;
+                }
+            }
         }
     }
 
@@ -1903,6 +1935,7 @@
         }
     }
 
+done:
     pthread_mutex_unlock(&adev->lock);
     pthread_mutex_unlock(&in->lock);
 
@@ -2146,7 +2179,7 @@
         else
             out->compr_config.codec->id =
                 get_snd_codec_id(config->offload_info.format);
-        out->compr_config.fragment_size = get_offload_buffer_size();
+        out->compr_config.fragment_size = get_offload_buffer_size(&config->offload_info);
         out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
         out->compr_config.codec->sample_rate =
                     compress_get_alsa_rate(config->offload_info.sample_rate);
@@ -2155,6 +2188,7 @@
         out->compr_config.codec->ch_in =
                     popcount(config->channel_mask);
         out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
+        out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
 
         if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
             out->non_blocking = 1;
@@ -2502,16 +2536,7 @@
     in->config.rate = config->sample_rate;
     in->format = config->format;
 
-    if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
-        (voice_extn_compress_voip_is_config_supported(config))) {
-        ret = voice_extn_compress_voip_open_input_stream(in);
-        if (ret != 0)
-        {
-            ALOGE("%s: Compress voip input cannot be opened, error:%d",
-                  __func__, ret);
-            goto err_open;
-        }
-    } else if (channel_count == 6) {
+    if (channel_count == 6) {
         if(audio_extn_ssr_get_enabled()) {
             if(audio_extn_ssr_init(adev, in)) {
                 ALOGE("%s: audio_extn_ssr_init failed", __func__);
@@ -2523,7 +2548,8 @@
             goto err_open;
         }
     } else if (audio_extn_compr_cap_enabled() &&
-            audio_extn_compr_cap_format_supported(config->format)) {
+            audio_extn_compr_cap_format_supported(config->format) &&
+            (in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) {
         audio_extn_compr_cap_init(adev, in);
     } else {
         in->config.channels = channel_count;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 4648dbb..c9719bb 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -76,7 +76,7 @@
 
 /* Audio calibration related functions */
 typedef void (*acdb_deallocate_t)();
-typedef int  (*acdb_init_t)();
+typedef int  (*acdb_init_t)(char *);
 typedef void (*acdb_send_audio_cal_t)(int, int);
 typedef void (*acdb_send_voice_cal_t)(int, int);
 typedef int (*acdb_reload_vocvoltable_t)(int);
@@ -643,11 +643,11 @@
                   __func__, LIB_ACDB_LOADER);
 
         my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
-                                                    "acdb_loader_init_ACDB");
+                                                    "acdb_loader_init_v2");
         if (my_data->acdb_init == NULL)
-            ALOGE("%s: dlsym error %s for acdb_loader_init_ACDB", __func__, dlerror());
+            ALOGE("%s: dlsym error %s for acdb_loader_init_v2", __func__, dlerror());
         else
-            my_data->acdb_init();
+            my_data->acdb_init(snd_card_name);
     }
 
     /* Initialize ACDB ID's */
@@ -1296,7 +1296,8 @@
         if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
             if (audio_extn_ssr_get_enabled() && channel_count == 6)
                 snd_device = SND_DEVICE_IN_QUAD_MIC;
-            else if (channel_count == 2)
+            else if (my_data->fluence_type & (FLUENCE_DUAL_MIC | FLUENCE_QUAD_MIC) &&
+                    channel_count == 2)
                 snd_device = SND_DEVICE_IN_HANDSET_STEREO_DMIC;
             else
                 snd_device = SND_DEVICE_IN_HANDSET_MIC;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index ca8469a..3ea068d 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -174,7 +174,7 @@
 #define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
 #define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
 #define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
-#define SPKR_PROT_CALIB_TX_PCM_DEVICE 22
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 25
 #define PLAYBACK_OFFLOAD_DEVICE 9
 #define COMPRESS_VOIP_CALL_PCM_DEVICE 3
 
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 69587dc..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,
@@ -881,6 +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);