diff --git a/Android.mk b/Android.mk
index 2f4054f..429c4cb 100644
--- a/Android.mk
+++ b/Android.mk
@@ -12,7 +12,6 @@
 endif
 include $(MY_LOCAL_PATH)/voice_processing/Android.mk
 include $(MY_LOCAL_PATH)/mm-audio/Android.mk
-include $(MY_LOCAL_PATH)/policy_hal/Android.mk
 include $(MY_LOCAL_PATH)/visualizer/Android.mk
 include $(MY_LOCAL_PATH)/post_proc/Android.mk
 include $(MY_LOCAL_PATH)/qahw_api/Android.mk
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
deleted file mode 100644
index 5b7f6f1..0000000
--- a/policy_hal/Android.mk
+++ /dev/null
@@ -1,96 +0,0 @@
-# This file was modified by Dolby Laboratories, Inc. The portions of the
-# code that are surrounded by "DOLBY..." are copyrighted and
-# licensed separately, as follows:
-#
-# (C)  2016 Dolby Laboratories, Inc.
-# All rights reserved.
-#
-# This program is protected under international and U.S. Copyright laws as
-# an unpublished work. This program is confidential and proprietary to the
-# copyright owners. Reproduction or disclosure, in whole or in part, or the
-# production of derivative works therefrom without the express permission of
-# the copyright owners is prohibited.
-#
-ifneq ($(USE_LEGACY_AUDIO_POLICY), 1)
-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) \
-                    $(TOPDIR)system/core/base/include
-
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    liblog \
-    libsoundtrigger \
-    libaudiopolicymanagerdefault \
-    libserviceutility
-
-LOCAL_STATIC_LIBRARIES := \
-    libmedia_helper \
-
-LOCAL_CFLAGS += -Wall -Werror
-
-ifeq ($(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 $(AUDIO_FEATURE_ENABLED_FM_POWER_OPT)),true)
-LOCAL_CFLAGS += -DFM_POWER_OPT
-endif
-# DOLBY_START
-ifeq ($(strip $(DOLBY_ENABLE)),true)
-LOCAL_CFLAGS += $(dolby_cflags)
-endif
-# DOLBY_END
-
-ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
-endif
-
-LOCAL_MODULE := libaudiopolicymanager
-
-include $(BUILD_SHARED_LIBRARY)
-
-endif
-endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
deleted file mode 100644
index 72bd644..0000000
--- a/policy_hal/AudioPolicyManager.cpp
+++ /dev/null
@@ -1,2246 +0,0 @@
-/*
- * Copyright (c) 2013-2017 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.
- *
- * This file was modified by Dolby Laboratories, Inc. The portions of the
- * code that are surrounded by "DOLBY..." are copyrighted and
- * licensed separately, as follows:
- *
- *  (C) 2015 Dolby Laboratories, Inc.
- *
- * 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>
-#ifdef DOLBY_ENABLE
-#include "DolbyAudioPolicy_impl.h"
-#endif // DOLBY_END
-
-#ifndef AUDIO_OUTPUT_FLAG_VOIP_RX
-#define AUDIO_OUTPUT_FLAG_VOIP_RX 0x800
-#endif
-
-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("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 voice.conc.fallbackpath",
-                propValue);
-        }
-    }
-    else {
-        ALOGD("voice_conc:prop voice.conc.fallbackpath not set");
-    }
-
-    ALOGD("voice_conc:picked up flag(0x%x) from prop voice.conc.fallbackpath",
-        flag);
-
-    return flag;
-}
-#endif /*VOICE_CONCURRENCY*/
-
-void AudioPolicyManagerCustom::moveGlobalEffect()
-{
-    audio_io_handle_t dstOutput = getOutputForEffect();
-    if (hasPrimaryOutput() && dstOutput != mPrimaryOutput->mIoHandle) {
-#ifdef DOLBY_ENABLE
-        status_t status =
-                        mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
-                                                    mPrimaryOutput->mIoHandle,
-                                                    dstOutput);
-        if (status == NO_ERROR) {
-            for (size_t i = 0; i < mEffects.size(); i++) {
-                 sp<EffectDescriptor> desc = mEffects.valueAt(i);
-                 if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) {
-                     // update the mIo member of EffectDescriptor
-                     // for the global effect
-                     ALOGV("%s updating mIo", __FUNCTION__);
-                     desc->mIo = dstOutput;
-                 }
-            }
-        } else {
-                ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__,
-                                     mPrimaryOutput->mIoHandle, dstOutput);
-        }
-#else // DOLBY_END
-        mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
-                                   mPrimaryOutput->mIoHandle, dstOutput);
-#endif
-    }
-}
-
-// ----------------------------------------------------------------------------
-// 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);
-
-            // 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);
-            } 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();
-        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) {
-               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);
-           }
-           AudioParameter param = AudioParameter();
-           param.addInt(String8("handle_fm"), (int)newDevice);
-           mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString());
-        }
-#endif /* FM_POWER_OPT end */
-
-        updateDevicesAndOutputs();
-#ifdef DOLBY_ENABLE
-        // Before closing the opened outputs, update endpoint property with device capabilities
-        audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true);
-        mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules);
-#endif // DOLBY_END
-        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;
-}
-
-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("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("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("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("av.streaming.offload.enable",
-                                                               false /*default value*/);
-    if (offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
-        ALOGW("offload disabled by 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. dolby 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("voice.playback.conc.disabled", propValue, NULL)) {
-        prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
-    }
-
-    if(property_get("voice.record.conc.disabled", propValue, NULL)) {
-        prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
-    }
-
-    if(property_get("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 effects where present on any of the above closed outputs,
-        // audioflinger moved them to the primary output by default
-        // move them back to the appropriate output.
-        moveGlobalEffect();
-    }
-
-    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("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("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));
-                }
-            }
-            // If effects where present on any of the above closed outputs,
-            // audioflinger moved them to the primary output by default
-            // move them back to the appropriate output.
-            moveGlobalEffect();
-        } 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 (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 = desc->device() != 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);
-        }
-        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 (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("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,
-        uint32_t samplingRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        audio_output_flags_t flags,
-        const audio_offload_info_t *offloadInfo)
-{
-    audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-    status_t status;
-
-#ifdef AUDIO_POLICY_TEST
-    if (mCurOutput != 0) {
-        ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
-                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
-        if (mTestOutputs[mCurOutput] == 0) {
-            ALOGV("getOutput() opening test output");
-            sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL,
-                                                                               mpClientInterface);
-            outputDesc->mDevice = mTestDevice;
-            outputDesc->mLatency = mTestLatencyMs;
-            outputDesc->mFlags =
-                    (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
-            outputDesc->mRefCount[stream] = 0;
-            audio_config_t config = AUDIO_CONFIG_INITIALIZER;
-            config.sample_rate = mTestSamplingRate;
-            config.channel_mask = mTestChannels;
-            config.format = mTestFormat;
-            if (offloadInfo != NULL) {
-                config.offload_info = *offloadInfo;
-            }
-            status = mpClientInterface->openOutput(0,
-                                                  &mTestOutputs[mCurOutput],
-                                                  &config,
-                                                  &outputDesc->mDevice,
-                                                  String8(""),
-                                                  &outputDesc->mLatency,
-                                                  outputDesc->mFlags);
-            if (status == NO_ERROR) {
-                outputDesc->mSamplingRate = config.sample_rate;
-                outputDesc->mFormat = config.format;
-                outputDesc->mChannelMask = config.channel_mask;
-                AudioParameter outputCmd = AudioParameter();
-                outputCmd.addInt(String8("set_id"),mCurOutput);
-                mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
-                addOutput(mTestOutputs[mCurOutput], outputDesc);
-            }
-        }
-        return mTestOutputs[mCurOutput];
-    }
-#endif //AUDIO_POLICY_TEST
-    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;
-       }
-
-    if ((stream == AUDIO_STREAM_VOICE_CALL) &&
-        (channelMask == 1) &&
-        (samplingRate == 8000 || samplingRate == 16000 ||
-         samplingRate == 32000 || samplingRate == 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 mode = 0, voipOutCount = 1, voipSampleRate = 1;
-        bool is_vr_mode_on = false;
-        String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
-                                                           String8("audio_mode"));
-        AudioParameter result = AudioParameter(valueStr);
-        if (result.getInt(String8("audio_mode"), value) == NO_ERROR) {
-            mode = value;
-        }
-
-        valueStr =  mpClientInterface->getParameters((audio_io_handle_t)0,
-                                              String8("voip_out_stream_count"));
-        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 ((mode == AUDIO_MODE_IN_COMMUNICATION) && (voipOutCount == 0) &&
-            ((voipSampleRate == 0) || (voipSampleRate == samplingRate))) {
-            if (audio_is_linear_pcm(format)) {
-                char propValue[PROPERTY_VALUE_MAX] = {0};
-                property_get("use.voice.path.for.pcm.voip", propValue, "0");
-                bool voipPcmSysPropEnabled = !strncmp("true", propValue, sizeof("true"));
-                if (voipPcmSysPropEnabled && (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");
-                }
-            }
-        }
-        //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;
-        valueStr =  mpClientInterface->getParameters((audio_io_handle_t)0,
-                                              String8("vr_audio_mode_on"));
-        result = AudioParameter(valueStr);
-        if (result.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("voice.playback.conc.disabled", propValue, NULL)) {
-       prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
-    }
-
-    if(property_get("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("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) &&
-        (offloadInfo != NULL) && !offload_disabled &&
-        ((offloadInfo->usage == AUDIO_USAGE_MEDIA) || (offloadInfo->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);
-    }
-
-    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 &&
-            property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
-            forced_deep = true;
-    }
-
-    if (stream == AUDIO_STREAM_TTS) {
-        flags = AUDIO_OUTPUT_FLAG_TTS;
-    }
-
-    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(format) && samplingRate <= SAMPLE_RATE_HZ_MAX &&
-            audio_channel_count_from_out_mask(channelMask) <= 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,
-                                       samplingRate,
-                                       format,
-                                       channelMask,
-                                       (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;
-        }
-
-        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("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 ((samplingRate == outputDesc->mSamplingRate) &&
-                            audio_formats_match(format, outputDesc->mFormat) &&
-                            (channelMask == outputDesc->mChannelMask)) {
-                        if (session == outputDesc->mDirectClientSession) {
-                            outputDesc->mDirectOpenCount++;
-                            ALOGV("getOutput() reusing direct output %d for session %d",
-                            mOutputs.keyAt(i), session);
-                            return mOutputs.keyAt(i);
-                        } else {
-                            ALOGV("getOutput() do not reuse direct output because current client (%d) "
-                                  "is not the same as requesting client (%d)",
-                                  outputDesc->mDirectClientSession, session);
-                            goto non_direct_output;
-                        }
-                    }
-                }
-            }
-            // close direct output if currently open and configured with different parameters
-            if (outputDesc != NULL) {
-                closeOutput(outputDesc->mIoHandle);
-            }
-        }
-
-        // if the selected profile is offloaded and no offload info was specified,
-        // create a default one
-        audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER;
-        if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) {
-            flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
-            defaultOffloadInfo.sample_rate = samplingRate;
-            defaultOffloadInfo.channel_mask = channelMask;
-            defaultOffloadInfo.format = format;
-            defaultOffloadInfo.stream_type = stream;
-            defaultOffloadInfo.bit_rate = 0;
-            defaultOffloadInfo.duration_us = -1;
-            defaultOffloadInfo.has_video = true; // conservative
-            defaultOffloadInfo.is_streaming = true; // likely
-            offloadInfo = &defaultOffloadInfo;
-        }
-
-        outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface);
-        outputDesc->mDevice = device;
-        outputDesc->mLatency = 0;
-        outputDesc->mFlags = (audio_output_flags_t)(outputDesc->mFlags | flags);
-        audio_config_t config = AUDIO_CONFIG_INITIALIZER;
-        config.sample_rate = samplingRate;
-        config.channel_mask = channelMask;
-        config.format = format;
-        if (offloadInfo != NULL) {
-            config.offload_info = *offloadInfo;
-        }
-        status = mpClientInterface->openOutput(profile->getModuleHandle(),
-                                               &output,
-                                               &config,
-                                               &outputDesc->mDevice,
-                                               String8(""),
-                                               &outputDesc->mLatency,
-                                               outputDesc->mFlags);
-
-        // only accept an output with the requested parameters
-        if (status != NO_ERROR ||
-            (samplingRate != 0 && samplingRate != config.sample_rate) ||
-            (format != AUDIO_FORMAT_DEFAULT && !audio_formats_match(format, config.format)) ||
-            (channelMask != 0 && channelMask != config.channel_mask)) {
-            ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
-                    "format %d %d, channelMask %04x %04x", output, samplingRate,
-                    outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
-                    outputDesc->mChannelMask);
-            if (output != AUDIO_IO_HANDLE_NONE) {
-                mpClientInterface->closeOutput(output);
-            }
-            // fall back to mixer output if possible when the direct output could not be open
-            if (audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX) {
-                goto non_direct_output;
-            }
-            return AUDIO_IO_HANDLE_NONE;
-        }
-        outputDesc->mSamplingRate = config.sample_rate;
-        outputDesc->mChannelMask = config.channel_mask;
-        outputDesc->mFormat = config.format;
-        outputDesc->mRefCount[stream] = 0;
-        outputDesc->mStopTime[stream] = 0;
-        outputDesc->mDirectOpenCount = 1;
-        outputDesc->mDirectClientSession = session;
-
-        audio_io_handle_t srcOutput = getOutputForEffect();
-        addOutput(output, outputDesc);
-        audio_io_handle_t dstOutput = getOutputForEffect();
-        if (dstOutput == output) {
-#ifdef DOLBY_ENABLE
-            status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
-            if (status == NO_ERROR) {
-                for (size_t i = 0; i < mEffects.size(); i++) {
-                    sp<EffectDescriptor> desc = mEffects.valueAt(i);
-                    if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) {
-                        // update the mIo member of EffectDescriptor for the global effect
-                        ALOGV("%s updating mIo", __FUNCTION__);
-                        desc->mIo = dstOutput;
-                    }
-                }
-            } else {
-                ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__, srcOutput, dstOutput);
-            }
-#else // DOLBY_END
-            mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
-#endif // LINE_ADDED_BY_DOLBY
-        }
-        mPreviousOutputs = mOutputs;
-        ALOGV("getOutput() 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) != 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(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, format);
-    }
-    ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
-            "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
-
-    ALOGV("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("voice.record.conc.disabled", propValue, NULL)) {
-        prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
-    }
-
-    if(property_get("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);
-}
-
-
-status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input,
-                                        audio_session_t session,
-                                        concurrency_type__mask_t *concurrency)
-{
-    ALOGV("startInput() input %d", input);
-    *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;
-    }
-
-    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;
-    }
-#ifdef RECORD_PLAY_CONCURRENCY
-    mIsInputRequestOnProgress = true;
-
-    char getPropValue[PROPERTY_VALUE_MAX];
-    bool prop_rec_play_enabled = false;
-
-    if (property_get("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));
-            }
-        }
-        // If effects where present on any of the above closed outputs,
-        // audioflinger moved them to the primary output by default
-        // move them back to the appropriate output.
-        moveGlobalEffect();
-    }
-#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.hasRouteChanged(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 */);
-
-        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) &&
-                    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);
-#ifdef RECORD_PLAY_CONCURRENCY
-    char propValue[PROPERTY_VALUE_MAX];
-    bool prop_rec_play_enabled = false;
-
-    if (property_get("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;
-}
-
-void AudioPolicyManagerCustom::closeAllInputs() {
-    bool patchRemoved = false;
-
-    for(size_t input_index = mInputs.size(); input_index > 0; input_index--) {
-        sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index-1);
-        ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
-        if (patch_index >= 0) {
-            sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
-            (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
-            mAudioPatches.removeItemsAt(patch_index);
-            patchRemoved = true;
-        }
-        mpClientInterface->closeInput(mInputs.keyAt(input_index-1));
-    }
-    mInputs.clear();
-    SoundTrigger::setCaptureState(false);
-    nextAudioPortGeneration();
-
-    if (patchRemoved) {
-        mpClientInterface->onAudioPatchListUpdate();
-    }
-}
-
-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
deleted file mode 100644
index 433380b..0000000
--- a/policy_hal/AudioPolicyManager.h
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- * Copyright (c) 2013-2017, 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);
-        // 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,
-                                    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);
-
-        virtual void closeAllInputs();
-
-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,
-                uint32_t samplingRate,
-                audio_format_t format,
-                audio_channel_mask_t channelMask,
-                audio_output_flags_t flags,
-                const audio_offload_info_t *offloadInfo);
-        // 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);
-        // 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
-};
-};