diff --git a/BoardConfig.mk b/BoardConfig.mk
index 46ef90f..d722239 100755
--- a/BoardConfig.mk
+++ b/BoardConfig.mk
@@ -55,6 +55,9 @@
 # Api
 PRODUCT_SHIPPING_API_LEVEL := 27
 
+# Audio
+USE_CUSTOM_AUDIO_POLICY := 1
+
 # Bluetooth
 BOARD_HAVE_BLUETOOTH := true
 BOARD_HAVE_BLUETOOTH_QCOM := true
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
new file mode 100644
index 0000000..76c3847
--- /dev/null
+++ b/policy_hal/Android.mk
@@ -0,0 +1,84 @@
+ifeq ($(TARGET_DEVICE),zenfone6)
+ifeq ($(USE_CUSTOM_AUDIO_POLICY), 1)
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := AudioPolicyManager.cpp
+
+LOCAL_C_INCLUDES := $(TOPDIR)frameworks/av/services \
+                    $(TOPDIR)frameworks/av/services/audioflinger \
+                    $(call include-path-for, audio-effects) \
+                    $(call include-path-for, audio-utils) \
+                    $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
+                    $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \
+                    $(TOPDIR)frameworks/av/services/audiopolicy \
+                    $(TOPDIR)frameworks/av/services/audiopolicy/common/managerdefinitions/include \
+                    $(call include-path-for, avextension) \
+
+LOCAL_HEADER_LIBRARIES := \
+        libbase_headers
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    liblog \
+    libsoundtrigger \
+    libaudiopolicymanagerdefault \
+    libserviceutility
+
+LOCAL_STATIC_LIBRARIES := \
+    libmedia_helper \
+
+LOCAL_CFLAGS += -Wall -Werror
+
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_VOICE_CONCURRENCY)),true)
+LOCAL_CFLAGS += -DVOICE_CONCURRENCY
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_RECORD_PLAY_CONCURRENCY)),true)
+LOCAL_CFLAGS += -DRECORD_PLAY_CONCURRENCY
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PCM_OFFLOAD)),true)
+    LOCAL_CFLAGS += -DPCM_OFFLOAD_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PCM_OFFLOAD_24)),true)
+       LOCAL_CFLAGS += -DPCM_OFFLOAD_ENABLED_24
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_EXTN_FORMATS)),true)
+    LOCAL_CFLAGS += -DAUDIO_EXTN_FORMATS_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AAC_ADTS_OFFLOAD)),true)
+    LOCAL_CFLAGS += -DAAC_ADTS_OFFLOAD_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_SPK)),true)
+    LOCAL_CFLAGS += -DAUDIO_EXTN_HDMI_SPK_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PROXY_DEVICE)),true)
+    LOCAL_CFLAGS += -DAUDIO_EXTN_AFE_PROXY_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FM_POWER_OPT)),true)
+LOCAL_CFLAGS += -DFM_POWER_OPT
+endif
+
+ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
+LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_COMPRESS_VOIP)),true)
+    LOCAL_CFLAGS += -DCOMPRESS_VOIP_ENABLED
+endif
+
+LOCAL_MODULE := libaudiopolicymanager
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif
+endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
new file mode 100644
index 0000000..ab933a0
--- /dev/null
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -0,0 +1,2343 @@
+/*
+ * Copyright (c) 2013-2018 The Linux Foundation. All rights reserved.
+ * Not a contribution.
+ *
+ * Copyright (C) 2009 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.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyManagerCustom"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL  AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+// A device mask for all audio input and output devices where matching inputs/outputs on device
+// type alone is not enough: the address must match too
+#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
+                                            AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+#define SAMPLE_RATE_8000 8000
+#include <inttypes.h>
+#include <math.h>
+
+#include <cutils/properties.h>
+#include <utils/Log.h>
+#include <hardware/audio.h>
+#include <hardware/audio_effect.h>
+#include <media/AudioParameter.h>
+#include <soundtrigger/SoundTrigger.h>
+#include "AudioPolicyManager.h"
+#include <policy.h>
+
+namespace android {
+/*audio policy: workaround for truncated touch sounds*/
+//FIXME: workaround for truncated touch sounds
+// to be removed when the problem is handled by system UI
+#define TOUCH_SOUND_FIXED_DELAY_MS 100
+#ifdef VOICE_CONCURRENCY
+audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath()
+{
+    audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST;
+    char propValue[PROPERTY_VALUE_MAX];
+
+    if (property_get("vendor.voice.conc.fallbackpath", propValue, NULL)) {
+        if (!strncmp(propValue, "deep-buffer", 11)) {
+            flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+        }
+        else if (!strncmp(propValue, "fast", 4)) {
+            flag = AUDIO_OUTPUT_FLAG_FAST;
+        }
+        else {
+            ALOGD("voice_conc:not a recognised path(%s) in prop vendor.voice.conc.fallbackpath",
+                 propValue);
+        }
+    }
+    else {
+        ALOGD("voice_conc:prop vendor.voice.conc.fallbackpath not set");
+    }
+
+    ALOGD("voice_conc:picked up flag(0x%x) from prop vendor.voice.conc.fallbackpath",
+        flag);
+
+    return flag;
+}
+#endif /*VOICE_CONCURRENCY*/
+
+void AudioPolicyManagerCustom::moveGlobalEffect()
+{
+    audio_io_handle_t dstOutput = getOutputForEffect();
+    if (hasPrimaryOutput() && dstOutput != mPrimaryOutput->mIoHandle)
+        mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
+                                   mPrimaryOutput->mIoHandle, dstOutput);
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+extern "C" AudioPolicyInterface* createAudioPolicyManager(
+         AudioPolicyClientInterface *clientInterface)
+{
+     return new AudioPolicyManagerCustom(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+     delete interface;
+}
+
+status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t device,
+                                                         audio_policy_dev_state_t state,
+                                                         const char *device_address,
+                                                         const char *device_name)
+{
+    ALOGD("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
+            device, state, device_address, device_name);
+
+    // connect/disconnect only 1 device at a time
+    if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+    sp<DeviceDescriptor> devDesc =
+            mHwModules.getDeviceDescriptor(device, device_address, device_name);
+
+    // handle output devices
+    if (audio_is_output_device(device)) {
+        SortedVector <audio_io_handle_t> outputs;
+
+        ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+
+        // save a copy of the opened output descriptors before any output is opened or closed
+        // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+        mPreviousOutputs = mOutputs;
+        switch (state)
+        {
+        // handle output device connection
+        case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+            if (index >= 0) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+                if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                   if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                        mHdmiAudioDisabled = false;
+                    } else {
+                        mHdmiAudioEvent = true;
+                    }
+                }
+#endif
+                ALOGW("setDeviceConnectionState() device already connected: %x", device);
+                return INVALID_OPERATION;
+            }
+            ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+            // register new device as available
+            index = mAvailableOutputDevices.add(devDesc);
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+            if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                    mHdmiAudioDisabled = false;
+                } else {
+                    mHdmiAudioEvent = true;
+                }
+                if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
+                    mAvailableOutputDevices.remove(devDesc);
+                    ALOGW("HDMI sink not connected, do not route audio to HDMI out");
+                    return INVALID_OPERATION;
+                }
+            }
+#endif
+            if (index >= 0) {
+                sp<HwModule> module = mHwModules.getModuleForDevice(device);
+                if (module == 0) {
+                    ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
+                          device);
+                    mAvailableOutputDevices.remove(devDesc);
+                    return INVALID_OPERATION;
+                }
+                mAvailableOutputDevices[index]->attach(module);
+            } else {
+                return NO_MEMORY;
+            }
+
+            // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
+            // parameters on newly connected devices (instead of opening the outputs...)
+            broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+
+            if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
+                mAvailableOutputDevices.remove(devDesc);
+
+                broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+                                               devDesc->mAddress);
+                return INVALID_OPERATION;
+            }
+            // Propagate device availability to Engine
+            mEngine->setDeviceConnectionState(devDesc, state);
+            if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+                chkDpConnAndAllowedForVoice();
+            }
+
+            // outputs should never be empty here
+            ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
+                    "checkOutputsForDevice() returned no outputs but status OK");
+            ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
+                  outputs.size());
+
+            } break;
+        // handle output device disconnection
+        case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+            if (index < 0) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+                if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                    if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                        mHdmiAudioDisabled = true;
+                    } else {
+                        mHdmiAudioEvent = false;
+                    }
+                }
+#endif
+                ALOGW("setDeviceConnectionState() device not connected: %x", device);
+                return INVALID_OPERATION;
+            }
+
+            ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
+
+            // Send Disconnect to HALs
+            broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+
+            // remove device from available output devices
+            mAvailableOutputDevices.remove(devDesc);
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+            if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+                if (!strncmp(device_address, "hdmi_spkr", 9)) {
+                    mHdmiAudioDisabled = true;
+                } else {
+                    mHdmiAudioEvent = false;
+                }
+            }
+#endif
+            checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
+
+            // Propagate device availability to Engine
+            mEngine->setDeviceConnectionState(devDesc, state);
+            if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+                mEngine->setDpConnAndAllowedForVoice(false);
+            }
+            } break;
+
+        default:
+            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
+        // output is suspended before any tracks are moved to it
+        checkA2dpSuspend();
+
+        if (!outputs.isEmpty()) {
+            for (size_t i = 0; i < outputs.size(); i++) {
+                sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+                // close voip output before track invalidation to allow creation of
+                // new voip stream from restoreTrack
+                if((desc->mFlags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) != 0) {
+                    closeOutput(outputs[i]);
+                    outputs.remove(outputs[i]);
+                }
+            }
+        }
+
+        checkOutputForAllStrategies();
+        // outputs must be closed after checkOutputForAllStrategies() is executed
+        if (!outputs.isEmpty()) {
+            for (size_t i = 0; i < outputs.size(); i++) {
+                sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+                // close unused outputs after device disconnection or direct outputs that have been
+                // opened by checkOutputsForDevice() to query dynamic parameters
+                if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+                        (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+                         (desc->mDirectOpenCount == 0))) {
+                    closeOutput(outputs[i]);
+                }
+            }
+            // check again after closing A2DP output to reset mA2dpSuspended if needed
+            checkA2dpSuspend();
+        }
+
+#ifdef FM_POWER_OPT
+        // handle FM device connection state to trigger FM AFE loopback
+        if (device == AUDIO_DEVICE_OUT_FM && hasPrimaryOutput()) {
+           audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+           if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+               /*
+                 when mPrimaryOutput->start() is called for FM it would check if isActive() is true
+                 or not as mRefCount=0 so isActive() would return false and curActiveCount will be
+                 1 and then the mRefCount will be increased by 1 for FM case.Updating curActiveCount
+                 is important as in case of adding other tracks when FM is still active isActive()
+                 will always be true as mRefCount will always be > 0,Hence curActiveCount will never
+                 update for them. However ,when fm stops and the track stops too mRefCount will be 0
+                 isActive will false,it will check if curActiveCount < 1 as curActiveCount was never
+                 updated so LOG_FATAL will cause the AudioServer to die.Hence this start() call will
+                 ensure that curActiveCount is updated at least once when FM starts prior to other
+                 tracks and on calling of stop() LOG_FATAL is not called.
+               */
+               mPrimaryOutput->start();
+               mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1);
+               newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)|AUDIO_DEVICE_OUT_FM);
+               mFMIsActive = true;
+               mPrimaryOutput->mDevice = newDevice & ~AUDIO_DEVICE_OUT_FM;
+           } else {
+               newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false));
+               mFMIsActive = false;
+               mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, -1);
+               /*
+                 mPrimaryOutput->stop() is called as because of calling of start()
+                 in FM case curActiveCount is getting updated and hence stop() is
+                 called so that curActiveCount gets decremented and if any tracks
+                 are added after FM stops they may get curActiveCount=0 ,ouptput
+                 curActiveCount can be properly updated
+               */
+               mPrimaryOutput->stop();
+           }
+           AudioParameter param = AudioParameter();
+           param.addInt(String8("handle_fm"), (int)newDevice);
+           mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString());
+        }
+#endif /* FM_POWER_OPT end */
+
+        updateDevicesAndOutputs();
+        if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+            audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+            updateCallRouting(newDevice);
+        }
+
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
+                audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
+                // do not force device change on duplicated output because if device is 0, it will
+                // also force a device 0 for the two outputs it is duplicated to which may override
+                // a valid device selection on those outputs.
+                bool force = !desc->isDuplicated()
+                        && (!device_distinguishes_on_address(device)
+                                // always force when disconnecting (a non-duplicated device)
+                                || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
+                setOutputDevice(desc, newDevice, force, 0);
+            }
+        }
+
+        if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
+            cleanUpForDevice(devDesc);
+        }
+
+        mpClientInterface->onAudioPortListUpdate();
+        return NO_ERROR;
+    }  // end if is output device
+
+    // handle input devices
+    if (audio_is_input_device(device)) {
+        SortedVector <audio_io_handle_t> inputs;
+
+        ssize_t index = mAvailableInputDevices.indexOf(devDesc);
+        switch (state)
+        {
+        // handle input device connection
+        case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+            if (index >= 0) {
+                ALOGW("setDeviceConnectionState() device already connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            sp<HwModule> module = mHwModules.getModuleForDevice(device);
+            if (module == NULL) {
+                ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
+                      device);
+                return INVALID_OPERATION;
+            }
+
+            // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
+            // parameters on newly connected devices (instead of opening the inputs...)
+            broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+
+            if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
+                broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+                                               devDesc->mAddress);
+                return INVALID_OPERATION;
+            }
+
+            index = mAvailableInputDevices.add(devDesc);
+            if (index >= 0) {
+                mAvailableInputDevices[index]->attach(module);
+            } else {
+                return NO_MEMORY;
+            }
+
+            // Propagate device availability to Engine
+            mEngine->setDeviceConnectionState(devDesc, state);
+        } break;
+
+        // handle input device disconnection
+        case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+            if (index < 0) {
+                ALOGW("setDeviceConnectionState() device not connected: %d", device);
+                return INVALID_OPERATION;
+            }
+
+            ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
+
+            // Set Disconnect to HALs
+            broadcastDeviceConnectionState(device, state, devDesc->mAddress);
+
+            checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress);
+            mAvailableInputDevices.remove(devDesc);
+
+            // Propagate device availability to Engine
+            mEngine->setDeviceConnectionState(devDesc, state);
+        } break;
+
+        default:
+            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        closeAllInputs();
+        /*audio policy: fix call volume over USB*/
+        // As the input device list can impact the output device selection, update
+        // getDeviceForStrategy() cache
+        updateDevicesAndOutputs();
+
+        if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+            audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+            updateCallRouting(newDevice);
+        }
+
+        if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
+            cleanUpForDevice(devDesc);
+        }
+
+        mpClientInterface->onAudioPortListUpdate();
+        return NO_ERROR;
+    } // end if is input device
+
+    ALOGW("setDeviceConnectionState() invalid device: %x", device);
+    return BAD_VALUE;
+}
+
+void AudioPolicyManagerCustom::chkDpConnAndAllowedForVoice()
+{
+    String8 value;
+    bool connAndAllowed = false;
+    String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+                                                        String8("dp_for_voice"));
+
+    AudioParameter result = AudioParameter(valueStr);
+    if (result.get(String8("dp_for_voice"), value) == NO_ERROR) {
+        connAndAllowed = value.contains("true");
+    }
+    mEngine->setDpConnAndAllowedForVoice(connAndAllowed);
+}
+
+bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(routing_strategy strategy)
+{
+    if (strategy == STRATEGY_MEDIA) {
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<SwAudioOutputDescriptor> newOutputDesc = mOutputs.valueAt(i);
+            if (newOutputDesc->mFormat == AUDIO_FORMAT_DSD)
+                return false;
+        }
+    }
+    return true;
+}
+
+void AudioPolicyManagerCustom::checkOutputForStrategy(routing_strategy strategy)
+{
+    audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+    audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mOutputs);
+    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+    // also take into account external policy-related changes: add all outputs which are
+    // associated with policies in the "before" and "after" output vectors
+    ALOGV("checkOutputForStrategy(): policy related outputs");
+    for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
+        const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+        if (desc != 0 && desc->mPolicyMix != NULL) {
+            srcOutputs.add(desc->mIoHandle);
+            ALOGV(" previous outputs: adding %d", desc->mIoHandle);
+        }
+    }
+    for (size_t i = 0 ; i < mOutputs.size() ; i++) {
+        const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+        if (desc != 0 && desc->mPolicyMix != NULL) {
+            dstOutputs.add(desc->mIoHandle);
+            ALOGV(" new outputs: adding %d", desc->mIoHandle);
+        }
+    }
+
+    if (!vectorsEqual(srcOutputs,dstOutputs) && isInvalidationOfMusicStreamNeeded(strategy)) {
+        AudioPolicyManager::checkOutputForStrategy(strategy);
+    }
+}
+
+// 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 AudioPolicyManagerCustom::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+    ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+     " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
+     offloadInfo.sample_rate, offloadInfo.channel_mask,
+     offloadInfo.format,
+     offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+     offloadInfo.has_video);
+
+     if (mMasterMono) {
+        return false; // no offloading if mono is set.
+     }
+
+#ifdef VOICE_CONCURRENCY
+    char concpropValue[PROPERTY_VALUE_MAX];
+    if (property_get("vendor.voice.playback.conc.disabled", concpropValue, NULL)) {
+         bool propenabled = atoi(concpropValue) || !strncmp("true", concpropValue, 4);
+         if (propenabled) {
+            if (isInCall())
+            {
+                ALOGD("\n copl: blocking  compress offload on call mode\n");
+                return false;
+            }
+         }
+    }
+#endif
+    if (property_get_bool("vendor.voice.dsd.playback.conc.disabled", true) &&
+        isInCall() &&  (offloadInfo.format == AUDIO_FORMAT_DSD)) {
+        ALOGD("blocking DSD compress offload on call mode");
+        return false;
+    }
+#ifdef RECORD_PLAY_CONCURRENCY
+    char recConcPropValue[PROPERTY_VALUE_MAX];
+    bool prop_rec_play_enabled = false;
+
+    if (property_get("vendor.audio.rec.playback.conc.disabled", recConcPropValue, NULL)) {
+        prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
+    }
+
+    if ((prop_rec_play_enabled) &&
+         ((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCountOnDevices() > 0))) {
+        ALOGD("copl: blocking  compress offload for record concurrency");
+        return false;
+    }
+#endif
+    // 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;
+    }
+
+    // Check if offload has been disabled
+    bool offloadDisabled = property_get_bool("audio.offload.disable", false);
+    if (offloadDisabled) {
+        ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
+        return false;
+    }
+
+    //check if it's multi-channel AAC (includes sub formats) and FLAC format
+    if ((popcount(offloadInfo.channel_mask) > 2) &&
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+        ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
+        ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
+        return false;
+    }
+
+#ifdef AUDIO_EXTN_FORMATS_ENABLED
+    //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
+    if ((popcount(offloadInfo.channel_mask) > 2) &&
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) ||
+        ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
+            ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
+        return false;
+    }
+
+    // check against wma std bit rate restriction
+    if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) {
+        int32_t sr_id = -1;
+        uint32_t min_bitrate, max_bitrate;
+        for (int i = 0; i < WMA_STD_NUM_FREQ; i++) {
+            if (offloadInfo.sample_rate == wmaStdSampleRateTbl[i]) {
+                sr_id = i;
+                break;
+            }
+        }
+        if ((sr_id < 0) || (popcount(offloadInfo.channel_mask) > 2)
+                || (popcount(offloadInfo.channel_mask) <= 0)) {
+            ALOGE("invalid sample rate or channel count");
+            return false;
+        }
+
+        min_bitrate = wmaStdMinAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
+        max_bitrate = wmaStdMaxAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
+        if ((offloadInfo.bit_rate > max_bitrate) || (offloadInfo.bit_rate < min_bitrate)) {
+            ALOGD("offload disabled for WMA clips with unsupported bit rate");
+            ALOGD("bit_rate %d, max_bitrate %d, min_bitrate %d", offloadInfo.bit_rate, max_bitrate, min_bitrate);
+            return false;
+        }
+    }
+
+    // Safely choose the min bitrate as threshold and leave the restriction to NT decoder as we can't distinguish wma pro and wma lossless here.
+    if ((((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_PRO)) ||
+        (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_LOSSLESS))) {
+        ALOGD("offload disabled for WMA_PRO/WMA_LOSSLESS clips with bit rate over maximum supported value");
+        return false;
+    }
+#endif
+    //TODO: enable audio offloading with video when ready
+    const bool allowOffloadWithVideo =
+            property_get_bool("audio.offload.video", false /* default_value */);
+    if (offloadInfo.has_video && !allowOffloadWithVideo) {
+        ALOGV("isOffloadSupported: has_video == true, returning false");
+        return false;
+    }
+
+    const bool allowOffloadStreamingWithVideo =
+            property_get_bool("vendor.audio.av.streaming.offload.enable", false /*default value*/);
+    if (offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
+        ALOGW("offload disabled by vendor.audio.av.streaming.offload.enable %d",
+               allowOffloadStreamingWithVideo);
+        return false;
+    }
+
+    //If duration is less than minimum value defined in property, return false
+    char propValue[PROPERTY_VALUE_MAX];
+    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. AAC/AC3 and amrwb+ formats
+        if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))
+            return false;
+
+#ifdef AUDIO_EXTN_FORMATS_ENABLED
+        if (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DSD) ||
+            ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
+            return false;
+#endif
+    }
+
+    // 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 (mEffects.isNonOffloadableEffectEnabled()) {
+        return false;
+    }
+
+    // See if there is a profile to support this.
+    // AUDIO_DEVICE_NONE
+    sp<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 != 0 ? "" : "NOT ");
+    return (profile != 0);
+}
+
+void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
+{
+    ALOGD("setPhoneState() state %d", state);
+    // store previous phone state for management of sonification strategy below
+    audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+    int oldState = mEngine->getPhoneState();
+
+    if (mEngine->setPhoneState(state) != NO_ERROR) {
+        ALOGW("setPhoneState() invalid or same state %d", state);
+        return;
+    }
+    /// Opens: can these line be executed after the switch of volume curves???
+    // if leaving call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isStateInCall(oldState)) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (size_t j = 0; j < mOutputs.size(); j++) {
+            audio_io_handle_t curOutput = mOutputs.keyAt(j);
+            for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
+                handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
+            }
+        }
+
+        // force reevaluating accessibility routing when call stops
+        mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+    }
+
+    /**
+     * Switching to or from incall state or switching between telephony and VoIP lead to force
+     * routing command.
+     */
+    bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
+                  || (is_state_in_call(state) && (state != oldState)));
+
+    // check for device and output changes triggered by new phone state
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+
+    sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
+#ifdef VOICE_CONCURRENCY
+    char propValue[PROPERTY_VALUE_MAX];
+    bool prop_playback_enabled = false, prop_rec_enabled=false, prop_voip_enabled = false;
+
+    if(property_get("vendor.voice.playback.conc.disabled", propValue, NULL)) {
+        prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+    }
+
+    if(property_get("vendor.voice.record.conc.disabled", propValue, NULL)) {
+        prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+    }
+
+    if(property_get("vendor.voice.voip.conc.disabled", propValue, NULL)) {
+        prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+    }
+
+    if ((AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state)) {
+        ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ",
+            oldState, state);
+        mvoice_call_state = state;
+        if (prop_rec_enabled) {
+            //Close all active inputs
+            Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
+            if (activeInputs.size() != 0) {
+               for (size_t i = 0; i <  activeInputs.size(); i++) {
+                   sp<AudioInputDescriptor> activeInput = activeInputs[i];
+                   switch(activeInput->inputSource()) {
+                       case AUDIO_SOURCE_VOICE_UPLINK:
+                       case AUDIO_SOURCE_VOICE_DOWNLINK:
+                       case AUDIO_SOURCE_VOICE_CALL:
+                           ALOGD("voice_conc:FOUND active input during call active: %d",activeInput->inputSource());
+                       break;
+
+                       case  AUDIO_SOURCE_VOICE_COMMUNICATION:
+                            if(prop_voip_enabled) {
+                                ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeInput->inputSource());
+                                AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
+                                audio_session_t activeSession = activeSessions.keyAt(0);
+                                stopInput(activeInput->mIoHandle, activeSession);
+                                releaseInput(activeInput->mIoHandle, activeSession);
+                            }
+                       break;
+
+                       default:
+                           ALOGD("voice_conc:CLOSING input on call setup  for inputSource: %d",activeInput->inputSource());
+                           AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
+                           audio_session_t activeSession = activeSessions.keyAt(0);
+                           stopInput(activeInput->mIoHandle, activeSession);
+                           releaseInput(activeInput->mIoHandle, activeSession);
+                       break;
+                   }
+               }
+           }
+        } else if (prop_voip_enabled) {
+            Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
+            if (activeInputs.size() != 0) {
+                for (size_t i = 0; i <  activeInputs.size(); i++) {
+                    sp<AudioInputDescriptor> activeInput = activeInputs[i];
+                    if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeInput->inputSource()) {
+                        ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeInput->inputSource());
+                        AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
+                        audio_session_t activeSession = activeSessions.keyAt(0);
+                        stopInput(activeInput->mIoHandle, activeSession);
+                        releaseInput(activeInput->mIoHandle, activeSession);
+                    }
+                }
+            }
+        }
+        if (prop_playback_enabled) {
+            // Move tracks associated to this strategy from previous output to new output
+            for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
+                ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
+                if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+                    if ((AUDIO_STREAM_MUSIC == i) ||
+                        (AUDIO_STREAM_VOICE_CALL == i) ) {
+                        ALOGD("voice_conc:Invalidate stream type %d", i);
+                        mpClientInterface->invalidateStream((audio_stream_type_t)i);
+                    }
+                } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+                    ALOGD("voice_conc:Invalidate stream type %d", i);
+                    mpClientInterface->invalidateStream((audio_stream_type_t)i);
+                }
+            }
+        }
+
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+            if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+               ALOGD("voice_conc:ouput desc / profile is NULL");
+               continue;
+            }
+
+            bool isFastFallBackNeeded =
+               ((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & outputDesc->mProfile->getFlags());
+
+            if ((AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) && isFastFallBackNeeded) {
+                if (((!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY))
+                            && prop_playback_enabled) {
+                    ALOGD("voice_conc:calling suspendOutput on call mode for primary output");
+                    mpClientInterface->suspendOutput(mOutputs.keyAt(i));
+                } //Close compress all sessions
+                else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+                                &&  prop_playback_enabled) {
+                    ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
+                    closeOutput(mOutputs.keyAt(i));
+                }
+                else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX)
+                                && prop_voip_enabled) {
+                    ALOGD("voice_conc:calling closeOutput on call mode for DIRECT  output");
+                    closeOutput(mOutputs.keyAt(i));
+                }
+            } else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+                if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX) {
+                    if (prop_voip_enabled) {
+                        ALOGD("voice_conc:calling closeOutput on call mode for DIRECT  output");
+                        closeOutput(mOutputs.keyAt(i));
+                    }
+                }
+                else if (prop_playback_enabled
+                           && (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
+                    ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
+                    closeOutput(mOutputs.keyAt(i));
+                }
+            }
+        }
+    }
+
+    if ((AUDIO_MODE_IN_CALL == oldState || AUDIO_MODE_IN_COMMUNICATION == oldState) &&
+       (AUDIO_MODE_NORMAL == state) && prop_playback_enabled && mvoice_call_state) {
+        ALOGD("voice_conc:EXITING from call mode oldState :: %d state::%d \n",oldState, state);
+        mvoice_call_state = 0;
+        if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+            //restore PCM (deep-buffer) output after call termination
+            for (size_t i = 0; i < mOutputs.size(); i++) {
+                sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+                if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+                   ALOGD("voice_conc:ouput desc / profile is NULL");
+                   continue;
+                }
+                if (!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
+                    ALOGD("voice_conc:calling restoreOutput after call mode for primary output");
+                    mpClientInterface->restoreOutput(mOutputs.keyAt(i));
+                }
+           }
+        }
+       //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
+        for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
+            ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
+            if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+                if ((AUDIO_STREAM_MUSIC == i) ||
+                    (AUDIO_STREAM_VOICE_CALL == i) ) {
+                    mpClientInterface->invalidateStream((audio_stream_type_t)i);
+                }
+            } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+                mpClientInterface->invalidateStream((audio_stream_type_t)i);
+            }
+        }
+    }
+
+#endif
+
+    sp<SwAudioOutputDescriptor> outputDesc = NULL;
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        outputDesc = mOutputs.valueAt(i);
+        if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+            ALOGD("voice_conc:ouput desc / profile is NULL");
+            continue;
+        }
+
+        if (property_get_bool("vendor.voice.dsd.playback.conc.disabled", true) &&
+            (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+            (outputDesc->mFormat == AUDIO_FORMAT_DSD)) {
+            ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output");
+            closeOutput(mOutputs.keyAt(i));
+            // call invalidate for music, so that DSD compress will fallback to deep-buffer.
+            mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+        }
+
+    }
+
+#ifdef RECORD_PLAY_CONCURRENCY
+    char recConcPropValue[PROPERTY_VALUE_MAX];
+    bool prop_rec_play_enabled = false;
+
+    if (property_get("vendor.audio.rec.playback.conc.disabled", recConcPropValue, NULL)) {
+        prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
+    }
+    if (prop_rec_play_enabled) {
+        if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
+            ALOGD("phone state changed to MODE_IN_COMM invlaidating music and voice streams");
+            // call invalidate for voice streams, so that it can use deepbuffer with VoIP out device from HAL
+            mpClientInterface->invalidateStream(AUDIO_STREAM_VOICE_CALL);
+            // call invalidate for music, so that compress will fallback to deep-buffer with VoIP out device
+            mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+
+            // close compress output to make sure session will be closed before timeout(60sec)
+            for (size_t i = 0; i < mOutputs.size(); i++) {
+
+                sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+                if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+                   ALOGD("ouput desc / profile is NULL");
+                   continue;
+                }
+
+                if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+                    ALOGD("calling closeOutput on call mode for COMPRESS output");
+                    closeOutput(mOutputs.keyAt(i));
+                }
+            }
+        } else if ((oldState == AUDIO_MODE_IN_COMMUNICATION) &&
+                    (mEngine->getPhoneState() == AUDIO_MODE_NORMAL)) {
+            // call invalidate for music so that music can fallback to compress
+            mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+        }
+    }
+#endif
+    mPrevPhoneState = oldState;
+    int delayMs = 0;
+    if (isStateInCall(state)) {
+        nsecs_t sysTime = systemTime();
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<SwAudioOutputDescriptor> 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 ((isStrategyActive(desc, STRATEGY_MEDIA,
+                                  SONIFICATION_HEADSET_MUSIC_DELAY,
+                                  sysTime) ||
+                 isStrategyActive(desc, STRATEGY_SONIFICATION,
+                                  SONIFICATION_HEADSET_MUSIC_DELAY,
+                                  sysTime)) &&
+                    (delayMs < (int)desc->latency()*2)) {
+                delayMs = desc->latency()*2;
+            }
+            setStrategyMute(STRATEGY_MEDIA, true, desc);
+            setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+            setStrategyMute(STRATEGY_SONIFICATION, true, desc);
+            setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+        }
+    }
+
+    if (hasPrimaryOutput()) {
+        // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+        // the device returned is not necessarily reachable via this output
+        audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+        // force routing command to audio hardware when ending call
+        // even if no device change is needed
+        if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
+            rxDevice = mPrimaryOutput->device();
+        }
+
+        if (state == AUDIO_MODE_IN_CALL) {
+            updateCallRouting(rxDevice, delayMs);
+        } else if (oldState == AUDIO_MODE_IN_CALL) {
+            if (mCallRxPatch != 0) {
+                mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+                mCallRxPatch.clear();
+            }
+            if (mCallTxPatch != 0) {
+                mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+                mCallTxPatch.clear();
+            }
+            setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+        } else {
+            setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+        }
+    }
+    //update device for all non-primary outputs
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        audio_io_handle_t output = mOutputs.keyAt(i);
+        if (output != mPrimaryOutput->mIoHandle) {
+            newDevice = getNewOutputDevice(mOutputs.valueFor(output), false /*fromCache*/);
+            setOutputDevice(mOutputs.valueFor(output), newDevice, (newDevice != AUDIO_DEVICE_NONE));
+        }
+    }
+    // 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 (size_t j = 0; j < mOutputs.size(); j++) {
+            audio_io_handle_t curOutput = mOutputs.keyAt(j);
+            for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
+                handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
+           }
+        }
+
+       // force reevaluating accessibility routing when call starts
+       mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+    }
+
+    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+    if (state == AUDIO_MODE_RINGTONE &&
+        isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+        mLimitRingtoneVolume = true;
+    } else {
+        mLimitRingtoneVolume = false;
+    }
+}
+
+void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
+                                         audio_policy_forced_cfg_t config)
+{
+    ALOGD("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
+    if (config == mEngine->getForceUse(usage)) {
+        return;
+    }
+
+    if (mEngine->setForceUse(usage, config) != NO_ERROR) {
+        ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
+        return;
+    }
+    bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
+            (usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
+            (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
+
+    // check for device and output changes triggered by new force usage
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+
+    /*audio policy: workaround for truncated touch sounds*/
+    //FIXME: workaround for truncated touch sounds
+    // to be removed when the problem is handled by system UI
+    uint32_t delayMs = 0;
+    uint32_t waitMs = 0;
+    if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) {
+        delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
+    }
+    if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+        audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+        waitMs = updateCallRouting(newDevice, delayMs);
+    }
+    // Use reverse loop to make sure any low latency usecases (generally tones)
+    // are not routed before non LL usecases (generally music).
+    // We can safely assume that LL output would always have lower index,
+    // and use this work-around to avoid routing of output with music stream
+    // from the context of short lived LL output.
+    // Note: in case output's share backend(HAL sharing is implicit) all outputs
+    //       gets routing update while processing first output itself.
+    for (size_t i = mOutputs.size(); i > 0; i--) {
+        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
+        audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+        if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
+            waitMs = setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE),
+                                     delayMs);
+        }
+        if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+            applyStreamVolumes(outputDesc, newDevice, waitMs, true);
+        }
+    }
+
+    Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
+    for (size_t i = 0; i <  activeInputs.size(); i++) {
+        sp<AudioInputDescriptor> activeDesc = activeInputs[i];
+        audio_devices_t newDevice = getNewInputDevice(activeDesc);
+        // Force new input selection if the new device can not be reached via current input
+        if (activeDesc->mProfile->getSupportedDevices().types() &
+                (newDevice & ~AUDIO_DEVICE_BIT_IN)) {
+            setInputDevice(activeDesc->mIoHandle, newDevice);
+        } else {
+            closeInput(activeDesc->mIoHandle);
+        }
+    }
+}
+
+status_t AudioPolicyManagerCustom::stopSource(const sp<AudioOutputDescriptor>& outputDesc,
+                                            audio_stream_type_t stream,
+                                            bool forceDeviceUpdate)
+{
+    if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+        ALOGW("stopSource() invalid stream %d", stream);
+        return INVALID_OPERATION;
+    }
+    // always handle stream stop, check which stream type is stopping
+    handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
+
+    // handle special case for sonification while in call
+    if (isInCall()) {
+        if (outputDesc->isDuplicated()) {
+            handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle);
+            handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle);
+        }
+        handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
+    }
+
+    if (outputDesc->mRefCount[stream] > 0) {
+        // decrement usage count of this stream on the output
+        outputDesc->changeRefCount(stream, -1);
+
+        // store time at which the stream was stopped - see isStreamActive()
+        if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) {
+            outputDesc->mStopTime[stream] = systemTime();
+            audio_devices_t prevDevice = outputDesc->device();
+            audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
+            // delay the device switch by twice the latency because stopOutput() is executed when
+            // the track stop() command is received and at that time the audio track buffer can
+            // still contain data that needs to be drained. The latency only covers the audio HAL
+            // and kernel buffers. Also the latency does not always include additional delay in the
+            // audio path (audio DSP, CODEC ...)
+            setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2);
+
+            // force restoring the device selection on other active outputs if it differs from the
+            // one being selected for this output
+            for (size_t i = 0; i < mOutputs.size(); i++) {
+                audio_io_handle_t curOutput = mOutputs.keyAt(i);
+                sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+                if (desc != outputDesc &&
+                        desc->isActive() &&
+                        outputDesc->sharesHwModuleWith(desc) &&
+                        (newDevice != desc->device())) {
+                        audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
+                        bool force = prevDevice != dev;
+                        uint32_t delayMs;
+                        if (dev == prevDevice) {
+                            delayMs = 0;
+                        } else {
+                            delayMs = outputDesc->latency()*2;
+                        }
+                        setOutputDevice(desc,
+                                    dev,
+                                    force,
+                                    delayMs);
+                    /*audio policy: fix media volume after ringtone*/
+                    // re-apply device specific volume if not done by setOutputDevice()
+                     if (!force) {
+                         applyStreamVolumes(desc, dev, delayMs);
+                     }
+                }
+            }
+            // update the outputs if stopping one with a stream that can affect notification routing
+            handleNotificationRoutingForStream(stream);
+        }
+        if (stream == AUDIO_STREAM_MUSIC) {
+            selectOutputForMusicEffects();
+        }
+        return NO_ERROR;
+    } else {
+        ALOGW("stopOutput() refcount is already 0");
+        return INVALID_OPERATION;
+    }
+}
+
+status_t AudioPolicyManagerCustom::startSource(const sp<AudioOutputDescriptor>& outputDesc,
+                                             audio_stream_type_t stream,
+                                             audio_devices_t device,
+                                             const char *address,
+                                             uint32_t *delayMs)
+{
+    // cannot start playback of STREAM_TTS if any other output is being used
+    uint32_t beaconMuteLatency = 0;
+
+    if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+        ALOGW("startSource() invalid stream %d", stream);
+        return INVALID_OPERATION;
+    }
+
+    *delayMs = 0;
+    if (stream == AUDIO_STREAM_TTS) {
+        ALOGV("\t found BEACON stream");
+        if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
+            return INVALID_OPERATION;
+        } else {
+            beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
+        }
+    } else {
+        // some playback other than beacon starts
+        beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
+    }
+
+    // force device change if the output is inactive and no audio patch is already present.
+    // check active before incrementing usage count
+    bool force = !outputDesc->isActive() &&
+            (outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE);
+
+    // increment usage count for this stream on the requested output:
+    // NOTE that the usage count is the same for duplicated output and hardware output which is
+    // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+    outputDesc->changeRefCount(stream, 1);
+
+    if (stream == AUDIO_STREAM_MUSIC) {
+        selectOutputForMusicEffects();
+    }
+
+    if (outputDesc->mRefCount[stream] == 1 || device != AUDIO_DEVICE_NONE) {
+        // starting an output being rerouted?
+        if (device == AUDIO_DEVICE_NONE) {
+            device = getNewOutputDevice(outputDesc, false /*fromCache*/);
+        }
+        routing_strategy strategy = getStrategy(stream);
+        bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+                            (strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
+                            (beaconMuteLatency > 0);
+        uint32_t waitMs = beaconMuteLatency;
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            if (desc != outputDesc) {
+                // force a device change if any other output is:
+                // - managed by the same hw module
+                // - has a current device selection that differs from selected device.
+                // - supports currently selected device
+                // - has an active audio patch
+                // In this case, the audio HAL must receive the new device selection so that it can
+                // change the device currently selected by the other active output.
+                if (outputDesc->sharesHwModuleWith(desc) &&
+                        desc->device() != device &&
+                        desc->supportedDevices() & device &&
+                        desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) {
+                    force = true;
+                }
+                // wait for audio on other active outputs to be presented when starting
+                // a notification so that audio focus effect can propagate, or that a mute/unmute
+                // event occurred for beacon
+                uint32_t latency = desc->latency();
+                if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+                    waitMs = latency;
+                }
+            }
+        }
+        uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force, 0, NULL, address);
+
+        // handle special case for sonification while in call
+        if (isInCall()) {
+            handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
+        }
+
+        // apply volume rules for current stream and device if necessary
+        checkAndSetVolume(stream,
+                          mVolumeCurves->getVolumeIndex(stream, device),
+                          outputDesc,
+                          device);
+
+        // update the outputs if starting an output with a stream that can affect notification
+        // routing
+        handleNotificationRoutingForStream(stream);
+
+        // force reevaluating accessibility routing when ringtone or alarm starts
+        if (strategy == STRATEGY_SONIFICATION) {
+            mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+        }
+        if (waitMs > muteWaitMs) {
+            *delayMs = waitMs - muteWaitMs;
+        }
+
+    } else {
+        // handle special case for sonification while in call
+        if (isInCall()) {
+            handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
+        }
+    }
+    return NO_ERROR;
+}
+
+void AudioPolicyManagerCustom::handleIncallSonification(audio_stream_type_t stream,
+                                                      bool starting, bool stateChange,
+                                                      audio_io_handle_t output)
+{
+    if(!hasPrimaryOutput()) {
+        return;
+    }
+    // no action needed for AUDIO_STREAM_PATCH stream type, it's for internal flinger tracks
+    if (stream == AUDIO_STREAM_PATCH) {
+        return;
+    }
+    // if the stream pertains to sonification strategy and we are in call we must
+    // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+    // in the device used for phone strategy and play the tone if the selected device does not
+    // interfere with the device used for phone strategy
+    // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+    // many times as there are active tracks on the output
+    const routing_strategy stream_strategy = getStrategy(stream);
+    if ((stream_strategy == STRATEGY_SONIFICATION) ||
+            ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+        ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+                stream, starting, outputDesc->mDevice, stateChange);
+        if (outputDesc->mRefCount[stream]) {
+            int muteCount = 1;
+            if (stateChange) {
+                muteCount = outputDesc->mRefCount[stream];
+            }
+            if (audio_is_low_visibility(stream)) {
+                ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+                for (int i = 0; i < muteCount; i++) {
+                    setStreamMute(stream, starting, outputDesc);
+                }
+            } else {
+                ALOGV("handleIncallSonification() high visibility");
+                if (outputDesc->device() &
+                        getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
+                    ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+                    for (int i = 0; i < muteCount; i++) {
+                        setStreamMute(stream, starting, outputDesc);
+                    }
+                }
+                if (starting) {
+                    mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+                                                 AUDIO_STREAM_VOICE_CALL);
+                } else {
+                    mpClientInterface->stopTone();
+                }
+            }
+        }
+    }
+}
+
+void AudioPolicyManagerCustom::handleNotificationRoutingForStream(audio_stream_type_t stream) {
+    switch(stream) {
+    case AUDIO_STREAM_MUSIC:
+        checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+        updateDevicesAndOutputs();
+        break;
+    default:
+        break;
+    }
+}
+
+status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
+                                                   int index,
+                                                   const sp<AudioOutputDescriptor>& outputDesc,
+                                                   audio_devices_t device,
+                                                   int delayMs,
+                                                   bool force)
+{
+    if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+        ALOGW("checkAndSetVolume() invalid stream %d", stream);
+        return INVALID_OPERATION;
+    }
+    // do not change actual stream volume if the stream is muted
+    if (outputDesc->mMuteCount[stream] != 0) {
+        ALOGVV("checkAndSetVolume() stream %d muted count %d",
+              stream, outputDesc->mMuteCount[stream]);
+        return NO_ERROR;
+    }
+    audio_policy_forced_cfg_t forceUseForComm =
+            mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
+    // do not change in call volume if bluetooth is connected and vice versa
+    if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
+        (stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) {
+        ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+             stream, forceUseForComm);
+        return INVALID_OPERATION;
+    }
+
+    if (device == AUDIO_DEVICE_NONE) {
+        device = outputDesc->device();
+    }
+
+    float volumeDb = computeVolume(stream, index, device);
+    if (outputDesc->isFixedVolume(device)) {
+        volumeDb = 0.0f;
+    }
+
+    outputDesc->setVolume(volumeDb, stream, device, delayMs, force);
+
+    if (stream == AUDIO_STREAM_VOICE_CALL ||
+        stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+        float voiceVolume;
+        // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+        if (stream == AUDIO_STREAM_VOICE_CALL) {
+            voiceVolume = (float)index/(float)mVolumeCurves->getVolumeIndexMax(stream);
+        } else {
+            voiceVolume = 1.0;
+        }
+
+        if (voiceVolume != mLastVoiceVolume) {
+            mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+            mLastVoiceVolume = voiceVolume;
+        }
+#ifdef FM_POWER_OPT
+    } else if (stream == AUDIO_STREAM_MUSIC && hasPrimaryOutput() &&
+               outputDesc == mPrimaryOutput && mFMIsActive) {
+        /* Avoid unnecessary set_parameter calls as it puts the primary
+           outputs FastMixer in HOT_IDLE leading to breaks in audio */
+        if (volumeDb != mPrevFMVolumeDb) {
+            mPrevFMVolumeDb = volumeDb;
+            AudioParameter param = AudioParameter();
+            param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb));
+            //Double delayMs to avoid sound burst while device switch.
+            mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs*2);
+        }
+#endif /* FM_POWER_OPT end */
+    }
+
+    return NO_ERROR;
+}
+
+bool AudioPolicyManagerCustom::isDirectOutput(audio_io_handle_t output) {
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        audio_io_handle_t curOutput = mOutputs.keyAt(i);
+        sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+        if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool static tryForDirectPCM(audio_output_flags_t flags)
+{
+    bool trackDirectPCM = false;  // Output request for track created by other apps
+
+    if (flags == AUDIO_OUTPUT_FLAG_NONE) {
+        trackDirectPCM = property_get_bool("vendor.audio.offload.track.enable", true);
+    }
+    return trackDirectPCM;
+}
+
+status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr,
+                                                    audio_io_handle_t *output,
+                                                    audio_session_t session,
+                                                    audio_stream_type_t *stream,
+                                                    uid_t uid,
+                                                    const audio_config_t *config,
+                                                    audio_output_flags_t *flags,
+                                                    audio_port_handle_t *selectedDeviceId,
+                                                    audio_port_handle_t *portId)
+{
+    audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
+    audio_config_t tConfig;
+
+    uint32_t bitWidth = (audio_bytes_per_sample(config->format) * 8);
+
+    memcpy(&tConfig, config, sizeof(audio_config_t));
+    if ((*flags == AUDIO_OUTPUT_FLAG_DIRECT || tryForDirectPCM(*flags)) &&
+        (!memcmp(&config->offload_info, &tOffloadInfo, sizeof(audio_offload_info_t)))) {
+        tConfig.offload_info.sample_rate  = config->sample_rate;
+        tConfig.offload_info.channel_mask = config->channel_mask;
+        tConfig.offload_info.format = config->format;
+        tConfig.offload_info.stream_type = *stream;
+        tConfig.offload_info.bit_width = bitWidth;
+        if (attr != NULL) {
+            ALOGV("found attribute .. setting usage %d ", attr->usage);
+            tConfig.offload_info.usage = attr->usage;
+        } else {
+            ALOGI("%s:: attribute is NULL .. no usage set", __func__);
+        }
+    }
+
+    return AudioPolicyManager::getOutputForAttr(attr, output, session, stream,
+                                                (uid_t)uid, &tConfig,
+                                                flags,
+                                                (audio_port_handle_t*)selectedDeviceId,
+                                                portId);
+}
+
+audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice(
+        audio_devices_t device,
+        audio_session_t session,
+        audio_stream_type_t stream,
+        const audio_config_t *config,
+        audio_output_flags_t *flags)
+{
+    audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+    status_t status;
+
+    if (stream < AUDIO_STREAM_MIN || stream >= AUDIO_STREAM_CNT) {
+        ALOGE("%s: invalid stream %d", __func__, stream);
+        return AUDIO_IO_HANDLE_NONE;
+    }
+
+    if (((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
+            (stream != AUDIO_STREAM_MUSIC)) {
+        // compress should not be used for non-music streams
+        ALOGE("Offloading only allowed with music stream");
+        return 0;
+       }
+
+#ifdef COMPRESS_VOIP_ENABLED
+    if ((mEngine->getPhoneState() == AUDIO_MODE_IN_COMMUNICATION) &&
+        (stream == AUDIO_STREAM_VOICE_CALL) &&
+        audio_is_linear_pcm(config->format)) {
+        // let voice stream to go with primary output by default
+        // in case direct voip is bypassed
+        bool use_primary_out = true;
+
+        if ((config->channel_mask == 1) &&
+                (config->sample_rate == 8000 || config->sample_rate == 16000 ||
+                config->sample_rate == 32000 || config->sample_rate == 48000)) {
+            // Allow Voip direct output only if:
+            // audio mode is MODE_IN_COMMUNCATION; AND
+            // voip output is not opened already; AND
+            // requested sample rate matches with that of voip input stream (if opened already)
+            int value = 0;
+            uint32_t voipOutCount = 1, voipSampleRate = 1;
+
+            String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+                                              String8("voip_out_stream_count"));
+            AudioParameter result = AudioParameter(valueStr);
+            if (result.getInt(String8("voip_out_stream_count"), value) == NO_ERROR) {
+                voipOutCount = value;
+            }
+
+            valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+                                              String8("voip_sample_rate"));
+            result = AudioParameter(valueStr);
+            if (result.getInt(String8("voip_sample_rate"), value) == NO_ERROR) {
+                voipSampleRate = value;
+            }
+
+            if ((voipOutCount == 0) &&
+                ((voipSampleRate == 0) || (voipSampleRate == config->sample_rate))) {
+                char propValue[PROPERTY_VALUE_MAX] = {0};
+                property_get("vendor.voice.path.for.pcm.voip", propValue, "0");
+                bool voipPcmSysPropEnabled = !strncmp("true", propValue, sizeof("true"));
+                if (voipPcmSysPropEnabled && (config->format == AUDIO_FORMAT_PCM_16_BIT)) {
+                    *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
+                                                 AUDIO_OUTPUT_FLAG_DIRECT);
+                    ALOGD("Set VoIP and Direct output flags for PCM format");
+                    use_primary_out = false;
+                }
+            }
+        }
+
+        if (use_primary_out) {
+            *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY);
+        }
+#else
+    if (mEngine->getPhoneState() == AUDIO_MODE_IN_COMMUNICATION &&
+        stream == AUDIO_STREAM_VOICE_CALL &&
+        audio_is_linear_pcm(config->format)) {
+        //check if VoIP output is not opened already
+        bool voip_pcm_already_in_use = false;
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+             sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+             if (desc->mFlags == (AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT)) {
+                 //close voip output if currently open by the same client with different device
+                 if (desc->mDirectClientSession == session &&
+                     desc->device() != device) {
+                     closeOutput(desc->mIoHandle);
+                 } else {
+                     voip_pcm_already_in_use = true;
+                     ALOGD("VoIP PCM already in use");
+                 }
+                 break;
+             }
+        }
+
+        if (!voip_pcm_already_in_use) {
+            *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
+                                           AUDIO_OUTPUT_FLAG_DIRECT);
+            ALOGV("Set VoIP and Direct output flags for PCM format");
+        }
+#endif
+        //IF VOIP is going to be started at the same time as when
+        //vr is enabled, get VOIP to fallback to low latency
+        String8 vr_value;
+        String8 value_Str;
+        bool is_vr_mode_on = false;
+        AudioParameter ret;
+
+        value_Str =  mpClientInterface->getParameters((audio_io_handle_t)0,
+                                              String8("vr_audio_mode_on"));
+        ret = AudioParameter(value_Str);
+        if (ret.get(String8("vr_audio_mode_on"), vr_value) == NO_ERROR) {
+            is_vr_mode_on = vr_value.contains("true");
+            ALOGI("VR mode is %d, switch to primary output if request is for fast|raw",
+                is_vr_mode_on);
+        }
+
+        if (is_vr_mode_on) {
+             //check the flags being requested for, and clear FAST|RAW
+            *flags = (audio_output_flags_t)(*flags &
+                (~(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW)));
+
+        }
+
+    }
+
+#ifdef VOICE_CONCURRENCY
+    char propValue[PROPERTY_VALUE_MAX];
+    bool prop_play_enabled=false, prop_voip_enabled = false;
+
+    if(property_get("vendor.voice.playback.conc.disabled", propValue, NULL)) {
+       prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+    }
+
+    if(property_get("vendor.voice.voip.conc.disabled", propValue, NULL)) {
+       prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+    }
+
+    bool isDeepBufferFallBackNeeded =
+        ((AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags);
+    bool isFastFallBackNeeded =
+        ((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags);
+
+    if (prop_play_enabled && mvoice_call_state) {
+        //check if voice call is active  / running in background
+        if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+             ((AUDIO_MODE_IN_CALL == mPrevPhoneState)
+                && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+        {
+            if(AUDIO_OUTPUT_FLAG_VOIP_RX  & *flags) {
+                if(prop_voip_enabled) {
+                   ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
+                        *flags );
+                   return 0;
+                }
+            }
+            else {
+                if (isFastFallBackNeeded &&
+                    (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag)) {
+                    ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", *flags );
+                    *flags = AUDIO_OUTPUT_FLAG_FAST;
+                } else if (isDeepBufferFallBackNeeded &&
+                           (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag)) {
+                    if (AUDIO_STREAM_MUSIC == stream) {
+                        *flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+                        ALOGD("voice_conc:IN call mode adding deep-buffer flags %x ", *flags );
+                    }
+                    else {
+                        *flags = AUDIO_OUTPUT_FLAG_FAST;
+                        ALOGD("voice_conc:IN call mode adding fast flags %x ", *flags );
+                    }
+                }
+            }
+        }
+    } else if (prop_voip_enabled && mvoice_call_state) {
+        //check if voice call is active  / running in background
+        //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+        //return only ULL ouput
+        if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+             ((AUDIO_MODE_IN_CALL == mPrevPhoneState)
+                && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+        {
+            if(AUDIO_OUTPUT_FLAG_VOIP_RX  & *flags) {
+                    ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
+                        *flags );
+               return 0;
+            }
+        }
+     }
+#endif
+#ifdef RECORD_PLAY_CONCURRENCY
+    char recConcPropValue[PROPERTY_VALUE_MAX];
+    bool prop_rec_play_enabled = false;
+
+    if (property_get("vendor.audio.rec.playback.conc.disabled", recConcPropValue, NULL)) {
+        prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
+    }
+    if ((prop_rec_play_enabled) &&
+            ((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCountOnDevices() > 0))) {
+        if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
+            if (AUDIO_OUTPUT_FLAG_VOIP_RX & *flags) {
+                // allow VoIP using voice path
+                // Do nothing
+            } else if((*flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+                ALOGD("voice_conc:MODE_IN_COMM is setforcing deep buffer output for non ULL... flags: %x", *flags);
+                // use deep buffer path for all non ULL outputs
+                *flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+            }
+        } else if ((*flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+            ALOGD("voice_conc:Record mode is on forcing deep buffer output for non ULL... flags: %x ", *flags);
+            // use deep buffer path for all non ULL outputs
+            *flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+        }
+    }
+    if (prop_rec_play_enabled &&
+            (stream == AUDIO_STREAM_ENFORCED_AUDIBLE)) {
+           ALOGD("Record conc is on forcing ULL output for ENFORCED_AUDIBLE");
+           *flags = AUDIO_OUTPUT_FLAG_FAST;
+    }
+#endif
+
+#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
+    /*
+    * WFD audio routes back to target speaker when starting a ringtone playback.
+    * This is because primary output is reused for ringtone, so output device is
+    * updated based on SONIFICATION strategy for both ringtone and music playback.
+    * The same issue is not seen on remoted_submix HAL based WFD audio because
+    * primary output is not reused and a new output is created for ringtone playback.
+    * Issue is fixed by updating output flag to AUDIO_OUTPUT_FLAG_FAST when there is
+    * a non-music stream playback on WFD, so primary output is not reused for ringtone.
+    */
+    audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
+    if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY)
+          && (stream != AUDIO_STREAM_MUSIC)) {
+        ALOGD("WFD audio: use OUTPUT_FLAG_FAST for non music stream. flags:%x", *flags );
+        //For voip paths
+        if (*flags & AUDIO_OUTPUT_FLAG_DIRECT)
+            *flags = AUDIO_OUTPUT_FLAG_DIRECT;
+        else //route every thing else to ULL path
+            *flags = AUDIO_OUTPUT_FLAG_FAST;
+    }
+#endif
+
+    // 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 = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT);
+    }
+    if ((*flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+        *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT);
+    }
+
+    // Do internal direct magic here
+    bool offload_disabled = property_get_bool("audio.offload.disable", false);
+    if ((*flags == AUDIO_OUTPUT_FLAG_NONE) &&
+        (stream == AUDIO_STREAM_MUSIC) &&
+        ( !offload_disabled) &&
+        ((config->offload_info.usage == AUDIO_USAGE_MEDIA) ||
+        (config->offload_info.usage == AUDIO_USAGE_GAME))) {
+        audio_output_flags_t old_flags = *flags;
+        *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT);
+        ALOGD("Force Direct Flag .. old flags(0x%x)", old_flags);
+    } else if (*flags == AUDIO_OUTPUT_FLAG_DIRECT &&
+                (offload_disabled || stream != AUDIO_STREAM_MUSIC)) {
+        ALOGD("Offloading is disabled or Stream is not music --> Force Remove Direct Flag");
+        *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE);
+    }
+
+    // check if direct output for pcm/track offload already exits
+    bool direct_pcm_already_in_use = false;
+    if (*flags == AUDIO_OUTPUT_FLAG_DIRECT) {
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+            if (desc->mFlags == AUDIO_OUTPUT_FLAG_DIRECT) {
+                direct_pcm_already_in_use = true;
+                ALOGD("Direct PCM already in use");
+                break;
+            }
+        }
+        // prevent direct pcm for non-music stream blindly if direct pcm already in use
+        // for other music stream concurrency is handled after checking direct ouput usage
+        // and checking client
+        if (direct_pcm_already_in_use == true && stream != AUDIO_STREAM_MUSIC) {
+            ALOGD("disabling offload for non music stream as direct pcm is already in use");
+            *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE);
+        }
+    }
+
+    bool forced_deep = false;
+    // only allow deep buffering for music stream type
+    if (stream != AUDIO_STREAM_MUSIC) {
+        *flags = (audio_output_flags_t)(*flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+    } else if (/* stream == AUDIO_STREAM_MUSIC && */
+            (*flags == AUDIO_OUTPUT_FLAG_NONE || *flags == AUDIO_OUTPUT_FLAG_DIRECT) &&
+            property_get_bool("audio.deep_buffer.media", false /* default_value */) && !isInCall()) {
+            forced_deep = true;
+    }
+
+    if (stream == AUDIO_STREAM_TTS) {
+        *flags = AUDIO_OUTPUT_FLAG_TTS;
+    } else if (stream == AUDIO_STREAM_VOICE_CALL &&
+               audio_is_linear_pcm(config->format)) {
+        *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
+                                       AUDIO_OUTPUT_FLAG_DIRECT);
+        ALOGV("Set VoIP and Direct output flags for PCM format");
+    } else if (device == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
+        stream == AUDIO_STREAM_MUSIC &&
+        audio_is_linear_pcm(config->format) &&
+        isInCall()) {
+        *flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
+    }
+
+    sp<IOProfile> profile;
+
+    // skip direct output selection if the request can obviously be attached to a mixed output
+    // and not explicitly requested
+    if (((*flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
+            audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX &&
+            audio_channel_count_from_out_mask(config->channel_mask) <= 2) {
+        goto non_direct_output;
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled.
+    // This prevents 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.
+    //
+    // Supplementary annotation:
+    // For sake of track offload introduced, we need a rollback for both compress offload
+    // and track offload use cases.
+    if ((*flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT)) &&
+                (mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
+        ALOGD("non offloadable effect is enabled, try with non direct output");
+        goto non_direct_output;
+    }
+
+    profile = getProfileForDirectOutput(device,
+                                        config->sample_rate,
+                                        config->format,
+                                        config->channel_mask,
+                                        (audio_output_flags_t)*flags);
+
+    if (profile != 0) {
+
+        if (!(*flags & AUDIO_OUTPUT_FLAG_DIRECT) &&
+             (profile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
+            ALOGI("got Direct without requesting ... reject ");
+            profile = NULL;
+            goto non_direct_output;
+        }
+
+        if ((*flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0 || output != AUDIO_IO_HANDLE_NONE) {
+            sp<SwAudioOutputDescriptor> outputDesc = NULL;
+            // if multiple concurrent offload decode is supported
+            // do no check for reuse and also don't close previous output if its offload
+            // previous output will be closed during track destruction
+            if (!(property_get_bool("vendor.audio.offload.multiple.enabled", false) &&
+                    ((*flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0))) {
+                for (size_t i = 0; i < mOutputs.size(); i++) {
+                    sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+                    if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+                        outputDesc = desc;
+                        // reuse direct output if currently open by the same client
+                        // and configured with same parameters
+                        if ((config->sample_rate == desc->mSamplingRate) &&
+                            audio_formats_match(config->format, desc->mFormat) &&
+                            (config->channel_mask == desc->mChannelMask) &&
+                            (session == desc->mDirectClientSession)) {
+                            desc->mDirectOpenCount++;
+                            ALOGV("getOutputForDevice() reusing direct output %d for session %d",
+                                mOutputs.keyAt(i), session);
+                            return mOutputs.keyAt(i);
+                        }
+                    }
+                }
+                if (outputDesc != NULL) {
+                    if (*flags == AUDIO_OUTPUT_FLAG_DIRECT &&
+                         direct_pcm_already_in_use == true &&
+                         session != outputDesc->mDirectClientSession) {
+                         ALOGV("getOutput() do not reuse direct pcm output because current client (%d) "
+                               "is not the same as requesting client (%d) for different output conf",
+                         outputDesc->mDirectClientSession, session);
+                         goto non_direct_output;
+                    }
+                    closeOutput(outputDesc->mIoHandle);
+                }
+
+            }
+            if (!profile->canOpenNewIo()) {
+                goto non_direct_output;
+            }
+
+            outputDesc =
+                    new SwAudioOutputDescriptor(profile, mpClientInterface);
+            DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromType(device);
+            String8 address = outputDevices.size() > 0 ? outputDevices.itemAt(0)->mAddress
+                    : String8("");
+            status = outputDesc->open(config, device, address, stream, *flags, &output);
+
+            // only accept an output with the requested parameters
+            if (status != NO_ERROR ||
+                (config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
+                (config->format != AUDIO_FORMAT_DEFAULT &&
+                         !audio_formats_match(config->format, outputDesc->mFormat)) ||
+                (config->channel_mask != 0 && config->channel_mask != outputDesc->mChannelMask)) {
+                ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d,"
+                        "format %d %d, channel mask %04x %04x", output, config->sample_rate,
+                        outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
+                        config->channel_mask, outputDesc->mChannelMask);
+                if (output != AUDIO_IO_HANDLE_NONE) {
+                    outputDesc->close();
+                }
+                // fall back to mixer output if possible when the direct output could not be open
+                if (audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX) {
+                    goto non_direct_output;
+                }
+                return AUDIO_IO_HANDLE_NONE;
+            }
+            outputDesc->mRefCount[stream] = 0;
+            outputDesc->mStopTime[stream] = 0;
+            outputDesc->mDirectOpenCount = 1;
+            outputDesc->mDirectClientSession = session;
+
+            addOutput(output, outputDesc);
+            mPreviousOutputs = mOutputs;
+            ALOGV("getOutputForDevice() returns new direct output %d", output);
+            mpClientInterface->onAudioPortListUpdate();
+            return output;
+        }
+    }
+
+non_direct_output:
+
+    // A request for HW A/V sync cannot fallback to a mixed output because time
+    // stamps are embedded in audio data
+    if ((*flags & (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)) != 0) {
+        return AUDIO_IO_HANDLE_NONE;
+    }
+
+    // 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(config->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);
+
+        // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
+        *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+
+        if (forced_deep) {
+            *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+            ALOGI("setting force DEEP buffer now ");
+        } else if(*flags == AUDIO_OUTPUT_FLAG_NONE) {
+            // no deep buffer playback is requested hence fallback to primary
+            *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_PRIMARY);
+            ALOGI("FLAG None hence request for a primary output");
+        }
+
+        output = selectOutput(outputs, *flags, config->format);
+    }
+
+    ALOGW_IF((output == 0), "getOutputForDevice() could not find output for stream %d, "
+            "sampling rate %d, format %#x, channels %#x, flags %#x",
+            stream, config->sample_rate, config->format, config->channel_mask, *flags);
+
+    ALOGV("getOutputForDevice() returns output %d", output);
+
+    return output;
+}
+
+status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *attr,
+                                         audio_io_handle_t *input,
+                                         audio_session_t session,
+                                         uid_t uid,
+                                         const audio_config_base_t *config,
+                                         audio_input_flags_t flags,
+                                         audio_port_handle_t *selectedDeviceId,
+                                         input_type_t *inputType,
+                                         audio_port_handle_t *portId)
+{
+    audio_source_t inputSource;
+    inputSource = attr->source;
+#ifdef VOICE_CONCURRENCY
+
+    char propValue[PROPERTY_VALUE_MAX];
+    bool prop_rec_enabled=false, prop_voip_enabled = false;
+
+    if(property_get("vendor.voice.record.conc.disabled", propValue, NULL)) {
+        prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+    }
+
+    if(property_get("vendor.voice.voip.conc.disabled", propValue, NULL)) {
+        prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+     }
+
+    if (prop_rec_enabled && mvoice_call_state) {
+         //check if voice call is active  / running in background
+         //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+         //Need to block input request
+        if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+           ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
+             (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+        {
+            switch(inputSource) {
+                case AUDIO_SOURCE_VOICE_UPLINK:
+                case AUDIO_SOURCE_VOICE_DOWNLINK:
+                case AUDIO_SOURCE_VOICE_CALL:
+                    ALOGD("voice_conc:Creating input during incall mode for inputSource: %d",
+                        inputSource);
+                break;
+
+                case AUDIO_SOURCE_VOICE_COMMUNICATION:
+                    if(prop_voip_enabled) {
+                       ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
+                        inputSource);
+                       return NO_INIT;
+                    }
+                break;
+                default:
+                    ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
+                        inputSource);
+                return NO_INIT;
+            }
+        }
+    }//check for VoIP flag
+    else if(prop_voip_enabled && mvoice_call_state) {
+         //check if voice call is active  / running in background
+         //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+         //Need to block input request
+        if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+           ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
+             (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+        {
+            if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+                ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
+                return NO_INIT;
+            }
+        }
+    }
+
+#endif
+
+    return AudioPolicyManager::getInputForAttr(attr,
+                                               input,
+                                               session,
+                                               uid,
+                                               config,
+                                               flags,
+                                               selectedDeviceId,
+                                               inputType,
+                                               portId);
+}
+
+uint32_t AudioPolicyManagerCustom::activeNonSoundTriggerInputsCountOnDevices(audio_devices_t devices) const
+{
+    uint32_t count = 0;
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        const sp<AudioInputDescriptor>  inputDescriptor = mInputs.valueAt(i);
+        if (inputDescriptor->isActive() && !inputDescriptor->isSoundTrigger() &&
+                ((devices == AUDIO_DEVICE_IN_DEFAULT) ||
+                 ((inputDescriptor->mDevice & devices & ~AUDIO_DEVICE_BIT_IN) != 0))) {
+            count++;
+        }
+    }
+    return count;
+}
+
+status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input,
+                                        audio_session_t session,
+                                        bool silenced,
+                                        concurrency_type__mask_t *concurrency)
+{
+
+    ALOGV("startInput(input:%d, session:%d, silenced:%d, concurrency:%d)",
+            input, session, silenced, *concurrency);
+    *concurrency = API_INPUT_CONCURRENCY_NONE;
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        ALOGW("startInput() unknown input %d", input);
+        return BAD_VALUE;
+    }
+    sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+    sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
+    if (audioSession == 0) {
+        ALOGW("startInput() unknown session %d on input %d", session, input);
+        return BAD_VALUE;
+    }
+
+// FIXME: disable concurrent capture until UI is ready
+#if 0
+    if (!isConcurentCaptureAllowed(inputDesc, audioSession)) {
+        ALOGW("startInput(%d) failed: other input already started", input);
+        return INVALID_OPERATION;
+    }
+
+    if (isInCall()) {
+        *concurrency |= API_INPUT_CONCURRENCY_CALL;
+    }
+
+    if (mInputs.activeInputsCountOnDevices() != 0) {
+        *concurrency |= API_INPUT_CONCURRENCY_CAPTURE;
+    }
+#else
+    if (!is_virtual_input_device(inputDesc->mDevice)) {
+        if (mCallTxPatch != 0 &&
+            inputDesc->getModuleHandle() == mCallTxPatch->mPatch.sources[0].ext.device.hw_module) {
+            ALOGW("startInput(%d) failed: call in progress", input);
+            return INVALID_OPERATION;
+        }
+
+        Vector< sp<AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
+
+        // If a UID is idle and records silence and another not silenced recording starts
+        // from another UID (idle or active) we stop the current idle UID recording in
+        // favor of the new one - "There can be only one" TM
+        if (!silenced) {
+
+            for (const auto& activeDesc : activeInputs) {
+                if ((audioSession->flags() & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0 &&
+                        activeDesc->getId() == inputDesc->getId()) {
+                     continue;
+                }
+
+                AudioSessionCollection activeSessions = activeDesc->getAudioSessions(
+                        true /*activeOnly*/);
+                sp<AudioSession> activeSession = activeSessions.valueAt(0);
+                if (activeSession->isSilenced()) {
+                    audio_io_handle_t activeInput = activeDesc->mIoHandle;
+                    audio_session_t activeSessionId = activeSession->session();
+                    stopInput(activeInput, activeSessionId);
+                    releaseInput(activeInput, activeSessionId);
+                    ALOGV("startInput(%d) stopping silenced input %d", input, activeInput);
+                    activeInputs = mInputs.getActiveInputs();
+                }
+            }
+        }
+        for (const auto& activeDesc : activeInputs) {
+            if ((audioSession->flags() & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0 &&
+                    activeDesc->getId() == inputDesc->getId()) {
+                continue;
+            }
+            // Don't allow sound triggers streams to preempt one another.
+            if (inputDesc->isSoundTrigger() && activeDesc->isSoundTrigger()) {
+                continue;
+            }
+
+            audio_source_t activeSource = activeDesc->inputSource(true);
+            if (audioSession->inputSource() == AUDIO_SOURCE_HOTWORD) {
+                if (activeSource == AUDIO_SOURCE_HOTWORD) {
+                    if (activeDesc->hasPreemptedSession(session)) {
+                        ALOGW("startInput(%d) failed for HOTWORD: "
+                                "other input %d already started for HOTWORD",
+                              input, activeDesc->mIoHandle);
+                        return INVALID_OPERATION;
+                    }
+                } else {
+                    ALOGV("startInput(%d) failed for HOTWORD: other input %d already started",
+                          input, activeDesc->mIoHandle);
+                    return INVALID_OPERATION;
+                }
+            } else {
+                if (activeSource != AUDIO_SOURCE_HOTWORD) {
+                    ALOGW("startInput(%d) failed: other input %d already started",
+                          input, activeDesc->mIoHandle);
+                    return INVALID_OPERATION;
+                }
+            }
+        }
+
+        // We only need to check if the sound trigger session supports concurrent capture if the
+        // input is also a sound trigger input. Otherwise, we should preempt any hotword stream
+        // that's running.
+        const bool allowConcurrentWithSoundTrigger =
+            inputDesc->isSoundTrigger() ? soundTriggerSupportsConcurrentCapture() : false;
+
+        // if capture is allowed, preempt currently active HOTWORD captures
+        for (const auto& activeDesc : activeInputs) {
+            if (allowConcurrentWithSoundTrigger && activeDesc->isSoundTrigger()) {
+                continue;
+            }
+
+            audio_source_t activeSource = activeDesc->inputSource(true);
+            if (activeSource == AUDIO_SOURCE_HOTWORD) {
+                AudioSessionCollection activeSessions =
+                        activeDesc->getAudioSessions(true /*activeOnly*/);
+                audio_session_t activeSession = activeSessions.keyAt(0);
+                audio_io_handle_t activeHandle = activeDesc->mIoHandle;
+                SortedVector<audio_session_t> sessions = activeDesc->getPreemptedSessions();
+                sessions.add(activeSession);
+                inputDesc->setPreemptedSessions(sessions);
+                stopInput(activeHandle, activeSession);
+                releaseInput(activeHandle, activeSession);
+                ALOGV("startInput(%d) for HOTWORD preempting HOTWORD input %d",
+                      input, activeDesc->mIoHandle);
+            }
+        }
+    }
+#endif
+
+#ifdef RECORD_PLAY_CONCURRENCY
+    mIsInputRequestOnProgress = true;
+
+    char getPropValue[PROPERTY_VALUE_MAX];
+    bool prop_rec_play_enabled = false;
+
+    if (property_get("vendor.audio.rec.playback.conc.disabled", getPropValue, NULL)) {
+        prop_rec_play_enabled = atoi(getPropValue) || !strncmp("true", getPropValue, 4);
+    }
+
+    if ((prop_rec_play_enabled) && (mInputs.activeInputsCountOnDevices() == 0)){
+        // send update to HAL on record playback concurrency
+        AudioParameter param = AudioParameter();
+        param.add(String8("rec_play_conc_on"), String8("true"));
+        ALOGD("startInput() setParameters rec_play_conc is setting to ON ");
+        mpClientInterface->setParameters(0, param.toString());
+
+        // Call invalidate to reset all opened non ULL audio tracks
+        // Move tracks associated to this strategy from previous output to new output
+        for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
+            // Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder)
+            if (i != AUDIO_STREAM_ENFORCED_AUDIBLE) {
+               ALOGD("Invalidate on releaseInput for stream :: %d ", i);
+               //FIXME see fixme on name change
+               mpClientInterface->invalidateStream((audio_stream_type_t)i);
+            }
+        }
+        // close compress tracks
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+            if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+               ALOGD("ouput desc / profile is NULL");
+               continue;
+            }
+            if (outputDesc->mProfile->getFlags()
+                            & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+                // close compress  sessions
+                ALOGD("calling closeOutput on record conc for COMPRESS output");
+                closeOutput(mOutputs.keyAt(i));
+            }
+        }
+    }
+#endif
+
+    // increment activity count before calling getNewInputDevice() below as only active sessions
+    // are considered for device selection
+    audioSession->changeActiveCount(1);
+
+    // Routing?
+    mInputRoutes.incRouteActivity(session);
+
+    if (audioSession->activeCount() == 1 || mInputRoutes.getAndClearRouteChanged(session)) {
+        // indicate active capture to sound trigger service if starting capture from a mic on
+        // primary HW module
+        audio_devices_t device = getNewInputDevice(inputDesc);
+        setInputDevice(input, device, true /* force */);
+        status_t status = inputDesc->start();
+        if (status != NO_ERROR) {
+            mInputRoutes.decRouteActivity(session);
+            audioSession->changeActiveCount(-1);
+            return status;
+        }
+
+        if (inputDesc->getAudioSessionCount(true/*activeOnly*/) == 1) {
+            // if input maps to a dynamic policy with an activity listener, notify of state change
+            if ((inputDesc->mPolicyMix != NULL)
+                    && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
+                mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress,
+                        MIX_STATE_MIXING);
+            }
+
+            audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
+            if ((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) {
+                if (property_get_bool("persist.vendor.audio.va_concurrency_enabled", false)) {
+                    if (activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices) == 1)
+                        SoundTrigger::setCaptureState(true);
+                } else if (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1)
+                    SoundTrigger::setCaptureState(true);
+            }
+
+            // automatically enable the remote submix output when input is started if not
+            // used by a policy mix of type MIX_TYPE_RECORDERS
+            // For remote submix (a virtual device), we open only one input per capture request.
+            if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+                String8 address = String8("");
+                if (inputDesc->mPolicyMix == NULL) {
+                    address = String8("0");
+                } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+                    address = inputDesc->mPolicyMix->mDeviceAddress;
+                }
+                if (address != "") {
+                    setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                            AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+                            address, "remote-submix");
+                }
+            }
+        }
+    }
+
+    ALOGV("AudioPolicyManager::startInput() input source = %d", audioSession->inputSource());
+#ifdef RECORD_PLAY_CONCURRENCY
+    mIsInputRequestOnProgress = false;
+#endif
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerCustom::stopInput(audio_io_handle_t input,
+                                       audio_session_t session)
+{
+    status_t status;
+    status = AudioPolicyManager::stopInput(input, session);
+    if (property_get_bool("persist.vendor.audio.va_concurrency_enabled", false)) {
+        ssize_t index = mInputs.indexOfKey(input);
+        sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+        audio_devices_t device = inputDesc->mDevice;
+        audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
+        if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+                activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices) == 0) {
+                SoundTrigger::setCaptureState(false);
+        }
+    }
+#ifdef RECORD_PLAY_CONCURRENCY
+    char propValue[PROPERTY_VALUE_MAX];
+    bool prop_rec_play_enabled = false;
+
+    if (property_get("vendor.audio.rec.playback.conc.disabled", propValue, NULL)) {
+        prop_rec_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+    }
+
+    if ((prop_rec_play_enabled) && (mInputs.activeInputsCountOnDevices() == 0)) {
+
+        //send update to HAL on record playback concurrency
+        AudioParameter param = AudioParameter();
+        param.add(String8("rec_play_conc_on"), String8("false"));
+        ALOGD("stopInput() setParameters rec_play_conc is setting to OFF ");
+        mpClientInterface->setParameters(0, param.toString());
+
+        //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
+        for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
+            //Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder stop tone)
+            if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE) && (i != AUDIO_STREAM_PATCH)) {
+               ALOGD(" Invalidate on stopInput for stream :: %d ", i);
+               //FIXME see fixme on name change
+               mpClientInterface->invalidateStream((audio_stream_type_t)i);
+            }
+        }
+    }
+#endif
+    return status;
+}
+
+AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
+    : AudioPolicyManager(clientInterface),
+      mHdmiAudioDisabled(false),
+      mHdmiAudioEvent(false),
+#ifndef FM_POWER_OPT
+      mPrevPhoneState(0)
+#else
+      mPrevPhoneState(0),
+      mPrevFMVolumeDb(0.0f),
+      mFMIsActive(false)
+#endif
+{
+
+#ifdef USE_XML_AUDIO_POLICY_CONF
+    ALOGD("USE_XML_AUDIO_POLICY_CONF is TRUE");
+#else
+    ALOGD("USE_XML_AUDIO_POLICY_CONF is FALSE");
+#endif
+
+#ifdef RECORD_PLAY_CONCURRENCY
+    mIsInputRequestOnProgress = false;
+#endif
+
+
+#ifdef VOICE_CONCURRENCY
+    mFallBackflag = getFallBackPath();
+#endif
+}
+}
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
new file mode 100644
index 0000000..9e58dcf
--- /dev/null
+++ b/policy_hal/AudioPolicyManager.h
@@ -0,0 +1,203 @@
+/*
+ * Copyright (c) 2013-2018, The Linux Foundation. All rights reserved.
+ * Not a contribution.
+ *
+ * Copyright (C) 2009 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.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <audiopolicy/managerdefault/AudioPolicyManager.h>
+#include <audio_policy_conf.h>
+#include <Volume.h>
+
+
+namespace android {
+#ifndef AUDIO_EXTN_FORMATS_ENABLED
+#define AUDIO_FORMAT_WMA 0x12000000UL
+#define AUDIO_FORMAT_WMA_PRO 0x13000000UL
+#define AUDIO_FORMAT_FLAC 0x1B000000UL
+#define AUDIO_FORMAT_ALAC 0x1C000000UL
+#define AUDIO_FORMAT_APE 0x1D000000UL
+#endif
+
+#define WMA_STD_NUM_FREQ     7
+#define WMA_STD_NUM_CHANNELS 2
+static uint32_t wmaStdSampleRateTbl[WMA_STD_NUM_FREQ] =
+{
+    8000, 11025, 16000, 22050, 32000, 44100, 48000
+};
+
+static uint32_t wmaStdMinAvgByteRateTbl[WMA_STD_NUM_FREQ][WMA_STD_NUM_CHANNELS] =
+{
+    {128, 12000},
+    {8016, 8016},
+    {10000, 16000},
+    {16016, 20008},
+    {20000, 24000},
+    {20008, 31960},
+    {63000, 63000}
+};
+
+static uint32_t wmaStdMaxAvgByteRateTbl[WMA_STD_NUM_FREQ][WMA_STD_NUM_CHANNELS] =
+{
+    {8000, 12000},
+    {10168, 10168},
+    {16000, 20000},
+    {20008, 32048},
+    {20000, 48000},
+    {48024, 320032},
+    {256008, 256008}
+};
+
+#define MAX_BITRATE_WMA_PRO      1536000
+#define MAX_BITRATE_WMA_LOSSLESS 1152000
+
+#ifndef AAC_ADTS_OFFLOAD_ENABLED
+#define AUDIO_FORMAT_AAC_ADTS 0x1E000000UL
+#endif
+
+#ifndef AUDIO_EXTN_AFE_PROXY_ENABLED
+#define AUDIO_DEVICE_OUT_PROXY 0x1000000
+#endif
+
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManagerCustom: public AudioPolicyManager
+{
+
+public:
+        AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface);
+
+        virtual ~AudioPolicyManagerCustom() {}
+
+        status_t setDeviceConnectionStateInt(audio_devices_t device,
+                                          audio_policy_dev_state_t state,
+                                          const char *device_address,
+                                          const char *device_name);
+        virtual void setPhoneState(audio_mode_t state);
+        virtual void setForceUse(audio_policy_force_use_t usage,
+                                 audio_policy_forced_cfg_t config);
+
+        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+        virtual status_t getInputForAttr(const audio_attributes_t *attr,
+                                         audio_io_handle_t *input,
+                                         audio_session_t session,
+                                         uid_t uid,
+                                         const audio_config_base_t *config,
+                                         audio_input_flags_t flags,
+                                         audio_port_handle_t *selectedDeviceId,
+                                         input_type_t *inputType,
+                                         audio_port_handle_t *portId);
+        /* count active capture sessions (that are not sound trigger) using one of
+           the specified devices. Ignore devices if AUDIO_DEVICE_IN_DEFAULT is passed */
+        uint32_t activeNonSoundTriggerInputsCountOnDevices(
+            audio_devices_t devices = AUDIO_DEVICE_IN_DEFAULT) const;
+        // indicates to the audio policy manager that the input starts being used.
+        virtual status_t startInput(audio_io_handle_t input,
+                                    audio_session_t session,
+                                    bool silenced,
+                                    concurrency_type__mask_t *concurrency);
+        // indicates to the audio policy manager that the input stops being used.
+        virtual status_t stopInput(audio_io_handle_t input,
+                                   audio_session_t session);
+
+protected:
+
+         status_t checkAndSetVolume(audio_stream_type_t stream,
+                                                   int index,
+                                                   const sp<AudioOutputDescriptor>& outputDesc,
+                                                   audio_devices_t device,
+                                                   int delayMs = 0, bool force = false);
+
+        // avoid invalidation for active music stream on  previous outputs
+        // which is supported on the new device.
+        bool isInvalidationOfMusicStreamNeeded(routing_strategy strategy);
+
+        // Must be called before updateDevicesAndOutputs()
+        void checkOutputForStrategy(routing_strategy strategy);
+
+        // returns true if given output is direct output
+        bool isDirectOutput(audio_io_handle_t output);
+
+        // if argument "device" is different from AUDIO_DEVICE_NONE,  startSource() will force
+        // the re-evaluation of the output device.
+        status_t startSource(const sp<AudioOutputDescriptor>& outputDesc,
+                             audio_stream_type_t stream,
+                             audio_devices_t device,
+                             const char *address,
+                             uint32_t *delayMs);
+         status_t stopSource(const sp<AudioOutputDescriptor>& outputDesc,
+                            audio_stream_type_t stream,
+                            bool forceDeviceUpdate);
+        // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
+        // returns 0 if no mute/unmute event happened, the largest latency of the device where
+        //   the mute/unmute happened
+        uint32_t handleEventForBeacon(int){return 0;}
+        uint32_t setBeaconMute(bool){return 0;}
+#ifdef VOICE_CONCURRENCY
+        static audio_output_flags_t getFallBackPath();
+        int mFallBackflag;
+#endif /*VOICE_CONCURRENCY*/
+        void moveGlobalEffect();
+
+        // handle special cases for sonification strategy while in call: mute streams or replace by
+        // a special tone in the device used for communication
+        void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange, audio_io_handle_t output);
+        //parameter indicates of HDMI speakers disabled
+        bool mHdmiAudioDisabled;
+        //parameter indicates if HDMI plug in/out detected
+        bool mHdmiAudioEvent;
+private:
+        // updates device caching and output for streams that can influence the
+        //    routing of notifications
+        void handleNotificationRoutingForStream(audio_stream_type_t stream);
+        // internal method to return the output handle for the given device and format
+        audio_io_handle_t getOutputForDevice(
+                audio_devices_t device,
+                audio_session_t session,
+                audio_stream_type_t stream,
+                const audio_config_t *config,
+                audio_output_flags_t *flags);
+
+        // internal method to fill offload info in case of Direct PCM
+        status_t getOutputForAttr(const audio_attributes_t *attr,
+                audio_io_handle_t *output,
+                audio_session_t session,
+                audio_stream_type_t *stream,
+                uid_t uid,
+                const audio_config_t *config,
+                audio_output_flags_t *flags,
+                audio_port_handle_t *selectedDeviceId,
+                audio_port_handle_t *portId);
+        // internal method to query hal for whether display-port is connected
+        // and can be used for voip/voice call
+        void chkDpConnAndAllowedForVoice();
+        // Used for voip + voice concurrency usecase
+        int mPrevPhoneState;
+#ifdef VOICE_CONCURRENCY
+        int mvoice_call_state;
+#endif
+#ifdef RECORD_PLAY_CONCURRENCY
+        // Used for record + playback concurrency
+        bool mIsInputRequestOnProgress;
+#endif
+
+#ifdef FM_POWER_OPT
+        float mPrevFMVolumeDb;
+        bool mFMIsActive;
+#endif
+};
+};