audio policy: import audio policy manager from hardware legacy

Import AudioPolicyManagerBase from hardware/libhardware_legacy
to prepare move from android_audio_legacy name space.

Change-Id: I5d6682ccd2bfdeefbf2f6f81a557480a76aaf4fc

3 files changed, 4939 insertions, 0 deletions

diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
new file mode 100644
index 0000000..da03ee3
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -0,0 +1,261 @@
+/*
+ * 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.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYINTERFACE_H
+#define ANDROID_AUDIOPOLICYINTERFACE_H
+
+#include <media/AudioSystem.h>
+#include <media/ToneGenerator.h>
+#include <utils/String8.h>
+
+#include <hardware_legacy/AudioSystemLegacy.h>
+#include <hardware/audio_policy.h>
+
+namespace android_audio_legacy {
+    using android::Vector;
+    using android::String8;
+    using android::ToneGenerator;
+
+// ----------------------------------------------------------------------------
+
+// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
+// between the platform specific audio policy manager and Android generic audio policy manager.
+// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class.
+// This implementation makes use of the AudioPolicyClientInterface to control the activity and
+// configuration of audio input and output streams.
+//
+// The platform specific audio policy manager is in charge of the audio routing and volume control
+// policies for a given platform.
+// The main roles of this module are:
+//   - keep track of current system state (removable device connections, phone state, user requests...).
+//   System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface.
+//   - process getOutput() queries received when AudioTrack objects are created: Those queries
+//   return a handler on an output that has been selected, configured and opened by the audio policy manager and that
+//   must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method.
+//   When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide
+//   to close or reconfigure the output depending on other streams using this output and current system state.
+//   - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs.
+//   - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value
+//   applicable to each output as a function of platform specific settings and current output route (destination device). It
+//   also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries).
+//
+// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so)
+// and is linked with libaudioflinger.so
+
+
+//    Audio Policy Manager Interface
+class AudioPolicyInterface
+{
+
+public:
+    virtual ~AudioPolicyInterface() {}
+    //
+    // configuration functions
+    //
+
+    // indicate a change in device connection status
+    virtual status_t setDeviceConnectionState(audio_devices_t device,
+                                          AudioSystem::device_connection_state state,
+                                          const char *device_address) = 0;
+    // retrieve a device connection status
+    virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device,
+                                                                          const char *device_address) = 0;
+    // indicate a change in phone state. Valid phones states are defined by AudioSystem::audio_mode
+    virtual void setPhoneState(int state) = 0;
+    // force using a specific device category for the specified usage
+    virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0;
+    // retrieve current device category forced for a given usage
+    virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0;
+    // set a system property (e.g. camera sound always audible)
+    virtual void setSystemProperty(const char* property, const char* value) = 0;
+    // check proper initialization
+    virtual status_t initCheck() = 0;
+
+    //
+    // Audio routing query functions
+    //
+
+    // request an output appropriate for playback of the supplied stream type and parameters
+    virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                        uint32_t samplingRate,
+                                        audio_format_t format,
+                                        audio_channel_mask_t channelMask,
+                                        AudioSystem::output_flags flags,
+                                        const audio_offload_info_t *offloadInfo) = 0;
+    // indicates to the audio policy manager that the output starts being used by corresponding stream.
+    virtual status_t startOutput(audio_io_handle_t output,
+                                 AudioSystem::stream_type stream,
+                                 int session = 0) = 0;
+    // indicates to the audio policy manager that the output stops being used by corresponding stream.
+    virtual status_t stopOutput(audio_io_handle_t output,
+                                AudioSystem::stream_type stream,
+                                int session = 0) = 0;
+    // releases the output.
+    virtual void releaseOutput(audio_io_handle_t output) = 0;
+
+    // request an input appropriate for record from the supplied device with supplied parameters.
+    virtual audio_io_handle_t getInput(int inputSource,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    AudioSystem::audio_in_acoustics acoustics) = 0;
+    // indicates to the audio policy manager that the input starts being used.
+    virtual status_t startInput(audio_io_handle_t input) = 0;
+    // indicates to the audio policy manager that the input stops being used.
+    virtual status_t stopInput(audio_io_handle_t input) = 0;
+    // releases the input.
+    virtual void releaseInput(audio_io_handle_t input) = 0;
+
+    //
+    // volume control functions
+    //
+
+    // initialises stream volume conversion parameters by specifying volume index range.
+    virtual void initStreamVolume(AudioSystem::stream_type stream,
+                                      int indexMin,
+                                      int indexMax) = 0;
+
+    // sets the new stream volume at a level corresponding to the supplied index for the
+    // supplied device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means
+    // setting volume for all devices
+    virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream,
+                                          int index,
+                                          audio_devices_t device) = 0;
+
+    // retrieve current volume index for the specified stream and the
+    // specified device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means
+    // querying the volume of the active device.
+    virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream,
+                                          int *index,
+                                          audio_devices_t device) = 0;
+
+    // return the strategy corresponding to a given stream type
+    virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) = 0;
+
+    // return the enabled output devices for the given stream type
+    virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream) = 0;
+
+    // Audio effect management
+    virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc) = 0;
+    virtual status_t registerEffect(const effect_descriptor_t *desc,
+                                    audio_io_handle_t io,
+                                    uint32_t strategy,
+                                    int session,
+                                    int id) = 0;
+    virtual status_t unregisterEffect(int id) = 0;
+    virtual status_t setEffectEnabled(int id, bool enabled) = 0;
+
+    virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const = 0;
+    virtual bool isStreamActiveRemotely(int stream, uint32_t inPastMs = 0) const = 0;
+    virtual bool isSourceActive(audio_source_t source) const = 0;
+
+    //dump state
+    virtual status_t    dump(int fd) = 0;
+
+    virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo) = 0;
+};
+
+
+// Audio Policy client Interface
+class AudioPolicyClientInterface
+{
+public:
+    virtual ~AudioPolicyClientInterface() {}
+
+    //
+    // Audio HW module functions
+    //
+
+    // loads a HW module.
+    virtual audio_module_handle_t loadHwModule(const char *name) = 0;
+
+    //
+    // Audio output Control functions
+    //
+
+    // opens an audio output with the requested parameters. The parameter values can indicate to use the default values
+    // in case the audio policy manager has no specific requirements for the output being opened.
+    // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream.
+    // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
+    virtual audio_io_handle_t openOutput(audio_module_handle_t module,
+                                         audio_devices_t *pDevices,
+                                         uint32_t *pSamplingRate,
+                                         audio_format_t *pFormat,
+                                         audio_channel_mask_t *pChannelMask,
+                                         uint32_t *pLatencyMs,
+                                         audio_output_flags_t flags,
+                                         const audio_offload_info_t *offloadInfo = NULL) = 0;
+    // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
+    // a special mixer thread in the AudioFlinger.
+    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0;
+    // closes the output stream
+    virtual status_t closeOutput(audio_io_handle_t output) = 0;
+    // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in
+    // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded.
+    virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+    // restores a suspended output.
+    virtual status_t restoreOutput(audio_io_handle_t output) = 0;
+
+    //
+    // Audio input Control functions
+    //
+
+    // opens an audio input
+    virtual audio_io_handle_t openInput(audio_module_handle_t module,
+                                        audio_devices_t *pDevices,
+                                        uint32_t *pSamplingRate,
+                                        audio_format_t *pFormat,
+                                        audio_channel_mask_t *pChannelMask) = 0;
+    // closes an audio input
+    virtual status_t closeInput(audio_io_handle_t input) = 0;
+    //
+    // misc control functions
+    //
+
+    // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
+    // for each output (destination device) it is attached to.
+    virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0;
+
+    // invalidate a stream type, causing a reroute to an unspecified new output
+    virtual status_t invalidateStream(AudioSystem::stream_type stream) = 0;
+
+    // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
+    virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0) = 0;
+    // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
+    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
+
+    // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
+    // over a telephony device during a phone call.
+    virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) = 0;
+    virtual status_t stopTone() = 0;
+
+    // set down link audio volume.
+    virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0;
+
+    // move effect to the specified output
+    virtual status_t moveEffects(int session,
+                                     audio_io_handle_t srcOutput,
+                                     audio_io_handle_t dstOutput) = 0;
+
+};
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface);
+
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYINTERFACE_H
diff --git a/services/audiopolicy/AudioPolicyManagerBase.cpp b/services/audiopolicy/AudioPolicyManagerBase.cpp
new file mode 100644
index 0000000..6f58cf7
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyManagerBase.cpp
@@ -0,0 +1,4091 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyManagerBase"
+//#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 input devices that are considered "virtual" when evaluating
+// active inputs in getActiveInput()
+#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL  AUDIO_DEVICE_IN_REMOTE_SUBMIX
+// 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
+
+#include <utils/Log.h>
+#include <hardware_legacy/AudioPolicyManagerBase.h>
+#include <hardware/audio_effect.h>
+#include <hardware/audio.h>
+#include <math.h>
+#include <hardware_legacy/audio_policy_conf.h>
+#include <cutils/properties.h>
+
+namespace android_audio_legacy {
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManagerBase::setDeviceConnectionState(audio_devices_t device,
+                                                  AudioSystem::device_connection_state state,
+                                                  const char *device_address)
+{
+    SortedVector <audio_io_handle_t> outputs;
+
+    ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+
+    // connect/disconnect only 1 device at a time
+    if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+    if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+        ALOGE("setDeviceConnectionState() invalid address: %s", device_address);
+        return BAD_VALUE;
+    }
+
+    // handle output devices
+    if (audio_is_output_device(device)) {
+
+        if (!mHasA2dp && audio_is_a2dp_device(device)) {
+            ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
+            return BAD_VALUE;
+        }
+        if (!mHasUsb && audio_is_usb_device(device)) {
+            ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
+            return BAD_VALUE;
+        }
+        if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
+            ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
+            return BAD_VALUE;
+        }
+
+        // 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;
+        String8 paramStr;
+        switch (state)
+        {
+        // handle output device connection
+        case AudioSystem::DEVICE_STATE_AVAILABLE:
+            if (mAvailableOutputDevices & device) {
+                ALOGW("setDeviceConnectionState() device already connected: %x", device);
+                return INVALID_OPERATION;
+            }
+            ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+            if (mHasA2dp && audio_is_a2dp_device(device)) {
+                // handle A2DP device connection
+                AudioParameter param;
+                param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
+                paramStr = param.toString();
+            } else if (mHasUsb && audio_is_usb_device(device)) {
+                // handle USB device connection
+                paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+            }
+
+            if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) {
+                return INVALID_OPERATION;
+            }
+            ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs",
+                  outputs.size());
+            // register new device as available
+            mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
+
+            if (mHasA2dp && audio_is_a2dp_device(device)) {
+                // handle A2DP device connection
+                mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+                mA2dpSuspended = false;
+            } else if (audio_is_bluetooth_sco_device(device)) {
+                // handle SCO device connection
+                mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+            } else if (mHasUsb && audio_is_usb_device(device)) {
+                // handle USB device connection
+                mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+            }
+
+            break;
+        // handle output device disconnection
+        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableOutputDevices & device)) {
+                ALOGW("setDeviceConnectionState() device not connected: %x", device);
+                return INVALID_OPERATION;
+            }
+
+            ALOGV("setDeviceConnectionState() disconnecting device %x", device);
+            // remove device from available output devices
+            mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
+
+            checkOutputsForDevice(device, state, outputs, paramStr);
+            if (mHasA2dp && audio_is_a2dp_device(device)) {
+                // handle A2DP device disconnection
+                mA2dpDeviceAddress = "";
+                mA2dpSuspended = false;
+            } else if (audio_is_bluetooth_sco_device(device)) {
+                // handle SCO device disconnection
+                mScoDeviceAddress = "";
+            } else if (mHasUsb && audio_is_usb_device(device)) {
+                // handle USB device disconnection
+                mUsbCardAndDevice = "";
+            }
+            // not currently handling multiple simultaneous submixes: ignoring remote submix
+            //   case and address
+            } break;
+
+        default:
+            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        checkA2dpSuspend();
+        checkOutputForAllStrategies();
+        // outputs must be closed after checkOutputForAllStrategies() is executed
+        if (!outputs.isEmpty()) {
+            for (size_t i = 0; i < outputs.size(); i++) {
+                AudioOutputDescriptor *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 == AudioSystem::DEVICE_STATE_UNAVAILABLE) ||
+                        (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+                         (desc->mDirectOpenCount == 0))) {
+                    closeOutput(outputs[i]);
+                }
+            }
+        }
+
+        updateDevicesAndOutputs();
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            // 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.
+            setOutputDevice(mOutputs.keyAt(i),
+                            getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
+                            !mOutputs.valueAt(i)->isDuplicated(),
+                            0);
+        }
+
+        if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+        } else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO ||
+                   device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
+                   device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+            device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else {
+            return NO_ERROR;
+        }
+    }
+    // handle input devices
+    if (audio_is_input_device(device)) {
+
+        switch (state)
+        {
+        // handle input device connection
+        case AudioSystem::DEVICE_STATE_AVAILABLE: {
+            if (mAvailableInputDevices & device) {
+                ALOGW("setDeviceConnectionState() device already connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN);
+            }
+            break;
+
+        // handle input device disconnection
+        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableInputDevices & device)) {
+                ALOGW("setDeviceConnectionState() device not connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device);
+            } break;
+
+        default:
+            ALOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        audio_io_handle_t activeInput = getActiveInput();
+        if (activeInput != 0) {
+            AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+            audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+            if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+                ALOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
+                        inputDesc->mDevice, newDevice, activeInput);
+                inputDesc->mDevice = newDevice;
+                AudioParameter param = AudioParameter();
+                param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+                mpClientInterface->setParameters(activeInput, param.toString());
+            }
+        }
+
+        return NO_ERROR;
+    }
+
+    ALOGW("setDeviceConnectionState() invalid device: %x", device);
+    return BAD_VALUE;
+}
+
+AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(audio_devices_t device,
+                                                  const char *device_address)
+{
+    AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
+    String8 address = String8(device_address);
+    if (audio_is_output_device(device)) {
+        if (device & mAvailableOutputDevices) {
+            if (audio_is_a2dp_device(device) &&
+                (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
+                return state;
+            }
+            if (audio_is_bluetooth_sco_device(device) &&
+                address != "" && mScoDeviceAddress != address) {
+                return state;
+            }
+            if (audio_is_usb_device(device) &&
+                (!mHasUsb || (address != "" && mUsbCardAndDevice != address))) {
+                ALOGE("getDeviceConnectionState() invalid device: %x", device);
+                return state;
+            }
+            if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) {
+                return state;
+            }
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    } else if (audio_is_input_device(device)) {
+        if (device & mAvailableInputDevices) {
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    }
+
+    return state;
+}
+
+void AudioPolicyManagerBase::setPhoneState(int state)
+{
+    ALOGV("setPhoneState() state %d", state);
+    audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+    if (state < 0 || state >= AudioSystem::NUM_MODES) {
+        ALOGW("setPhoneState() invalid state %d", state);
+        return;
+    }
+
+    if (state == mPhoneState ) {
+        ALOGW("setPhoneState() setting same state %d", state);
+        return;
+    }
+
+    // if leaving call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isInCall()) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, false, true);
+        }
+    }
+
+    // store previous phone state for management of sonification strategy below
+    int oldState = mPhoneState;
+    mPhoneState = state;
+    bool force = false;
+
+    // are we entering or starting a call
+    if (!isStateInCall(oldState) && isStateInCall(state)) {
+        ALOGV("  Entering call in setPhoneState()");
+        // force routing command to audio hardware when starting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+        }
+    } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+        ALOGV("  Exiting call in setPhoneState()");
+        // force routing command to audio hardware when exiting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+        }
+    } else if (isStateInCall(state) && (state != oldState)) {
+        ALOGV("  Switching between telephony and VoIP in setPhoneState()");
+        // force routing command to audio hardware when switching between telephony and VoIP
+        // even if no device change is needed
+        force = true;
+    }
+
+    // check for device and output changes triggered by new phone state
+    newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+
+    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+    // force routing command to audio hardware when ending call
+    // even if no device change is needed
+    if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
+        newDevice = hwOutputDesc->device();
+    }
+
+    int delayMs = 0;
+    if (isStateInCall(state)) {
+        nsecs_t sysTime = systemTime();
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            // mute media and sonification strategies and delay device switch by the largest
+            // latency of any output where either strategy is active.
+            // This avoid sending the ring tone or music tail into the earpiece or headset.
+            if ((desc->isStrategyActive(STRATEGY_MEDIA,
+                                     SONIFICATION_HEADSET_MUSIC_DELAY,
+                                     sysTime) ||
+                    desc->isStrategyActive(STRATEGY_SONIFICATION,
+                                         SONIFICATION_HEADSET_MUSIC_DELAY,
+                                         sysTime)) &&
+                    (delayMs < (int)desc->mLatency*2)) {
+                delayMs = desc->mLatency*2;
+            }
+            setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+            setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+        }
+    }
+
+    // change routing is necessary
+    setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
+
+    // if entering in call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isStateInCall(state)) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, true, true);
+        }
+    }
+
+    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+    if (state == AudioSystem::MODE_RINGTONE &&
+        isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+        mLimitRingtoneVolume = true;
+    } else {
+        mLimitRingtoneVolume = false;
+    }
+}
+
+void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+    ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+    bool forceVolumeReeval = false;
+    switch(usage) {
+    case AudioSystem::FOR_COMMUNICATION:
+        if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
+            config != AudioSystem::FORCE_NONE) {
+            ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+            return;
+        }
+        forceVolumeReeval = true;
+        mForceUse[usage] = config;
+        break;
+    case AudioSystem::FOR_MEDIA:
+        if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
+            config != AudioSystem::FORCE_WIRED_ACCESSORY &&
+            config != AudioSystem::FORCE_ANALOG_DOCK &&
+            config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE &&
+            config != AudioSystem::FORCE_NO_BT_A2DP) {
+            ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+            return;
+        }
+        mForceUse[usage] = config;
+        break;
+    case AudioSystem::FOR_RECORD:
+        if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
+            config != AudioSystem::FORCE_NONE) {
+            ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+            return;
+        }
+        mForceUse[usage] = config;
+        break;
+    case AudioSystem::FOR_DOCK:
+        if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK &&
+            config != AudioSystem::FORCE_BT_DESK_DOCK &&
+            config != AudioSystem::FORCE_WIRED_ACCESSORY &&
+            config != AudioSystem::FORCE_ANALOG_DOCK &&
+            config != AudioSystem::FORCE_DIGITAL_DOCK) {
+            ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+        }
+        forceVolumeReeval = true;
+        mForceUse[usage] = config;
+        break;
+    case AudioSystem::FOR_SYSTEM:
+        if (config != AudioSystem::FORCE_NONE &&
+            config != AudioSystem::FORCE_SYSTEM_ENFORCED) {
+            ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
+        }
+        forceVolumeReeval = true;
+        mForceUse[usage] = config;
+        break;
+    default:
+        ALOGW("setForceUse() invalid usage %d", usage);
+        break;
+    }
+
+    // check for device and output changes triggered by new force usage
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        audio_io_handle_t output = mOutputs.keyAt(i);
+        audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
+        setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+        if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+            applyStreamVolumes(output, newDevice, 0, true);
+        }
+    }
+
+    audio_io_handle_t activeInput = getActiveInput();
+    if (activeInput != 0) {
+        AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+        audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+        if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+            ALOGV("setForceUse() changing device from %x to %x for input %d",
+                    inputDesc->mDevice, newDevice, activeInput);
+            inputDesc->mDevice = newDevice;
+            AudioParameter param = AudioParameter();
+            param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+            mpClientInterface->setParameters(activeInput, param.toString());
+        }
+    }
+
+}
+
+AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
+{
+    return mForceUse[usage];
+}
+
+void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
+{
+    ALOGV("setSystemProperty() property %s, value %s", property, value);
+}
+
+// Find a direct output profile compatible with the parameters passed, even if the input flags do
+// not explicitly request a direct output
+AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getProfileForDirectOutput(
+                                                               audio_devices_t device,
+                                                               uint32_t samplingRate,
+                                                               audio_format_t format,
+                                                               audio_channel_mask_t channelMask,
+                                                               audio_output_flags_t flags)
+{
+    for (size_t i = 0; i < mHwModules.size(); i++) {
+        if (mHwModules[i]->mHandle == 0) {
+            continue;
+        }
+        for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
+            IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+            if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+                if (profile->isCompatibleProfile(device, samplingRate, format,
+                                           channelMask,
+                                           AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
+                    if (mAvailableOutputDevices & profile->mSupportedDevices) {
+                        return mHwModules[i]->mOutputProfiles[j];
+                    }
+                }
+            } else {
+                if (profile->isCompatibleProfile(device, samplingRate, format,
+                                           channelMask,
+                                           AUDIO_OUTPUT_FLAG_DIRECT)) {
+                    if (mAvailableOutputDevices & profile->mSupportedDevices) {
+                        return mHwModules[i]->mOutputProfiles[j];
+                    }
+                }
+            }
+        }
+    }
+    return 0;
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    AudioSystem::output_flags flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    audio_io_handle_t output = 0;
+    uint32_t latency = 0;
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
+          device, stream, samplingRate, format, channelMask, flags);
+
+#ifdef AUDIO_POLICY_TEST
+    if (mCurOutput != 0) {
+        ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+        if (mTestOutputs[mCurOutput] == 0) {
+            ALOGV("getOutput() opening test output");
+            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+            outputDesc->mDevice = mTestDevice;
+            outputDesc->mSamplingRate = mTestSamplingRate;
+            outputDesc->mFormat = mTestFormat;
+            outputDesc->mChannelMask = mTestChannels;
+            outputDesc->mLatency = mTestLatencyMs;
+            outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+            outputDesc->mRefCount[stream] = 0;
+            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+                                            &outputDesc->mSamplingRate,
+                                            &outputDesc->mFormat,
+                                            &outputDesc->mChannelMask,
+                                            &outputDesc->mLatency,
+                                            outputDesc->mFlags,
+                                            offloadInfo);
+            if (mTestOutputs[mCurOutput]) {
+                AudioParameter outputCmd = AudioParameter();
+                outputCmd.addInt(String8("set_id"),mCurOutput);
+                mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+                addOutput(mTestOutputs[mCurOutput], outputDesc);
+            }
+        }
+        return mTestOutputs[mCurOutput];
+    }
+#endif //AUDIO_POLICY_TEST
+
+    // open a direct output if required by specified parameters
+    //force direct flag if offload flag is set: offloading implies a direct output stream
+    // and all common behaviors are driven by checking only the direct flag
+    // this should normally be set appropriately in the policy configuration file
+    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+    }
+
+    // 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.
+    IOProfile *profile = NULL;
+    if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+            !isNonOffloadableEffectEnabled()) {
+        profile = getProfileForDirectOutput(device,
+                                           samplingRate,
+                                           format,
+                                           channelMask,
+                                           (audio_output_flags_t)flags);
+    }
+
+    if (profile != NULL) {
+        AudioOutputDescriptor *outputDesc = NULL;
+
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+                outputDesc = desc;
+                // reuse direct output if currently open and configured with same parameters
+                if ((samplingRate == outputDesc->mSamplingRate) &&
+                        (format == outputDesc->mFormat) &&
+                        (channelMask == outputDesc->mChannelMask)) {
+                    outputDesc->mDirectOpenCount++;
+                    ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+                    return mOutputs.keyAt(i);
+                }
+            }
+        }
+        // close direct output if currently open and configured with different parameters
+        if (outputDesc != NULL) {
+            closeOutput(outputDesc->mId);
+        }
+        outputDesc = new AudioOutputDescriptor(profile);
+        outputDesc->mDevice = device;
+        outputDesc->mSamplingRate = samplingRate;
+        outputDesc->mFormat = format;
+        outputDesc->mChannelMask = channelMask;
+        outputDesc->mLatency = 0;
+        outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+        outputDesc->mRefCount[stream] = 0;
+        outputDesc->mStopTime[stream] = 0;
+        outputDesc->mDirectOpenCount = 1;
+        output = mpClientInterface->openOutput(profile->mModule->mHandle,
+                                        &outputDesc->mDevice,
+                                        &outputDesc->mSamplingRate,
+                                        &outputDesc->mFormat,
+                                        &outputDesc->mChannelMask,
+                                        &outputDesc->mLatency,
+                                        outputDesc->mFlags,
+                                        offloadInfo);
+
+        // only accept an output with the requested parameters
+        if (output == 0 ||
+            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+            (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) ||
+            (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+            ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+                    "format %d %d, channelMask %04x %04x", output, samplingRate,
+                    outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+                    outputDesc->mChannelMask);
+            if (output != 0) {
+                mpClientInterface->closeOutput(output);
+            }
+            delete outputDesc;
+            return 0;
+        }
+        audio_io_handle_t srcOutput = getOutputForEffect();
+        addOutput(output, outputDesc);
+        audio_io_handle_t dstOutput = getOutputForEffect();
+        if (dstOutput == output) {
+            mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+        }
+        mPreviousOutputs = mOutputs;
+        ALOGV("getOutput() returns new direct output %d", output);
+        return output;
+    }
+
+    // ignoring channel mask due to downmix capability in mixer
+
+    // open a non direct output
+
+    // for non direct outputs, only PCM is supported
+    if (audio_is_linear_pcm(format)) {
+        // get which output is suitable for the specified stream. The actual
+        // routing change will happen when startOutput() will be called
+        SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+        output = selectOutput(outputs, flags);
+    }
+    ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+            "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+    ALOGV("getOutput() returns output %d", output);
+
+    return output;
+}
+
+audio_io_handle_t AudioPolicyManagerBase::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+                                                       AudioSystem::output_flags flags)
+{
+    // select one output among several that provide a path to a particular device or set of
+    // devices (the list was previously build by getOutputsForDevice()).
+    // The priority is as follows:
+    // 1: the output with the highest number of requested policy flags
+    // 2: the primary output
+    // 3: the first output in the list
+
+    if (outputs.size() == 0) {
+        return 0;
+    }
+    if (outputs.size() == 1) {
+        return outputs[0];
+    }
+
+    int maxCommonFlags = 0;
+    audio_io_handle_t outputFlags = 0;
+    audio_io_handle_t outputPrimary = 0;
+
+    for (size_t i = 0; i < outputs.size(); i++) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
+        if (!outputDesc->isDuplicated()) {
+            int commonFlags = (int)AudioSystem::popCount(outputDesc->mProfile->mFlags & flags);
+            if (commonFlags > maxCommonFlags) {
+                outputFlags = outputs[i];
+                maxCommonFlags = commonFlags;
+                ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
+            }
+            if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+                outputPrimary = outputs[i];
+            }
+        }
+    }
+
+    if (outputFlags != 0) {
+        return outputFlags;
+    }
+    if (outputPrimary != 0) {
+        return outputPrimary;
+    }
+
+    return outputs[0];
+}
+
+status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
+                                             AudioSystem::stream_type stream,
+                                             int session)
+{
+    ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        ALOGW("startOutput() unknown output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+    // 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) {
+        audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+        routing_strategy strategy = getStrategy(stream);
+        bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+                            (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
+        uint32_t waitMs = 0;
+        bool force = false;
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            if (desc != outputDesc) {
+                // force a device change if any other output is managed by the same hw
+                // module and has a current device selection that differs from selected device.
+                // 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() != newDevice) {
+                    force = true;
+                }
+                // wait for audio on other active outputs to be presented when starting
+                // a notification so that audio focus effect can propagate.
+                uint32_t latency = desc->latency();
+                if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+                    waitMs = latency;
+                }
+            }
+        }
+        uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+
+        // handle special case for sonification while in call
+        if (isInCall()) {
+            handleIncallSonification(stream, true, false);
+        }
+
+        // apply volume rules for current stream and device if necessary
+        checkAndSetVolume(stream,
+                          mStreams[stream].getVolumeIndex(newDevice),
+                          output,
+                          newDevice);
+
+        // update the outputs if starting an output with a stream that can affect notification
+        // routing
+        handleNotificationRoutingForStream(stream);
+        if (waitMs > muteWaitMs) {
+            usleep((waitMs - muteWaitMs) * 2 * 1000);
+        }
+    }
+    return NO_ERROR;
+}
+
+
+status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
+                                            AudioSystem::stream_type stream,
+                                            int session)
+{
+    ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        ALOGW("stopOutput() unknown output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+    // handle special case for sonification while in call
+    if (isInCall()) {
+        handleIncallSonification(stream, false, false);
+    }
+
+    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) {
+            outputDesc->mStopTime[stream] = systemTime();
+            audio_devices_t newDevice = getNewDevice(output, 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(output, newDevice, false, outputDesc->mLatency*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);
+                AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+                if (curOutput != output &&
+                        desc->isActive() &&
+                        outputDesc->sharesHwModuleWith(desc) &&
+                        (newDevice != desc->device())) {
+                    setOutputDevice(curOutput,
+                                    getNewDevice(curOutput, false /*fromCache*/),
+                                    true,
+                                    outputDesc->mLatency*2);
+                }
+            }
+            // 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 for output %d", output);
+        return INVALID_OPERATION;
+    }
+}
+
+void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
+{
+    ALOGV("releaseOutput() %d", output);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        ALOGW("releaseOutput() releasing unknown output %d", output);
+        return;
+    }
+
+#ifdef AUDIO_POLICY_TEST
+    int testIndex = testOutputIndex(output);
+    if (testIndex != 0) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+        if (outputDesc->isActive()) {
+            mpClientInterface->closeOutput(output);
+            delete mOutputs.valueAt(index);
+            mOutputs.removeItem(output);
+            mTestOutputs[testIndex] = 0;
+        }
+        return;
+    }
+#endif //AUDIO_POLICY_TEST
+
+    AudioOutputDescriptor *desc = mOutputs.valueAt(index);
+    if (desc->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
+        if (desc->mDirectOpenCount <= 0) {
+            ALOGW("releaseOutput() invalid open count %d for output %d",
+                                                              desc->mDirectOpenCount, output);
+            return;
+        }
+        if (--desc->mDirectOpenCount == 0) {
+            closeOutput(output);
+            // If effects where present on the output, audioflinger moved them to the primary
+            // output by default: move them back to the appropriate output.
+            audio_io_handle_t dstOutput = getOutputForEffect();
+            if (dstOutput != mPrimaryOutput) {
+                mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
+            }
+        }
+    }
+}
+
+
+audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    AudioSystem::audio_in_acoustics acoustics)
+{
+    audio_io_handle_t input = 0;
+    audio_devices_t device = getDeviceForInputSource(inputSource);
+
+    ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
+          inputSource, samplingRate, format, channelMask, acoustics);
+
+    if (device == AUDIO_DEVICE_NONE) {
+        ALOGW("getInput() could not find device for inputSource %d", inputSource);
+        return 0;
+    }
+
+    // adapt channel selection to input source
+    switch(inputSource) {
+    case AUDIO_SOURCE_VOICE_UPLINK:
+        channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+        channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_CALL:
+        channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
+        break;
+    default:
+        break;
+    }
+
+    IOProfile *profile = getInputProfile(device,
+                                         samplingRate,
+                                         format,
+                                         channelMask);
+    if (profile == NULL) {
+        ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d, "
+                "channelMask %04x",
+                device, samplingRate, format, channelMask);
+        return 0;
+    }
+
+    if (profile->mModule->mHandle == 0) {
+        ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
+        return 0;
+    }
+
+    AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
+
+    inputDesc->mInputSource = inputSource;
+    inputDesc->mDevice = device;
+    inputDesc->mSamplingRate = samplingRate;
+    inputDesc->mFormat = format;
+    inputDesc->mChannelMask = channelMask;
+    inputDesc->mRefCount = 0;
+    input = mpClientInterface->openInput(profile->mModule->mHandle,
+                                    &inputDesc->mDevice,
+                                    &inputDesc->mSamplingRate,
+                                    &inputDesc->mFormat,
+                                    &inputDesc->mChannelMask);
+
+    // only accept input with the exact requested set of parameters
+    if (input == 0 ||
+        (samplingRate != inputDesc->mSamplingRate) ||
+        (format != inputDesc->mFormat) ||
+        (channelMask != inputDesc->mChannelMask)) {
+        ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask %x",
+                samplingRate, format, channelMask);
+        if (input != 0) {
+            mpClientInterface->closeInput(input);
+        }
+        delete inputDesc;
+        return 0;
+    }
+    mInputs.add(input, inputDesc);
+    return input;
+}
+
+status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
+{
+    ALOGV("startInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        ALOGW("startInput() unknown input %d", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+#ifdef AUDIO_POLICY_TEST
+    if (mTestInput == 0)
+#endif //AUDIO_POLICY_TEST
+    {
+        // refuse 2 active AudioRecord clients at the same time except if the active input
+        // uses AUDIO_SOURCE_HOTWORD in which case it is closed.
+        audio_io_handle_t activeInput = getActiveInput();
+        if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) {
+            AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+            if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
+                ALOGW("startInput() preempting already started low-priority input %d", activeInput);
+                stopInput(activeInput);
+                releaseInput(activeInput);
+            } else {
+                ALOGW("startInput() input %d failed: other input already started", input);
+                return INVALID_OPERATION;
+            }
+        }
+    }
+
+    audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+    if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+        inputDesc->mDevice = newDevice;
+    }
+
+    // automatically enable the remote submix output when input is started
+    if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+        setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                AudioSystem::DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+    }
+
+    AudioParameter param = AudioParameter();
+    param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
+
+    int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ?
+                                        AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource;
+
+    param.addInt(String8(AudioParameter::keyInputSource), aliasSource);
+    ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+
+    mpClientInterface->setParameters(input, param.toString());
+
+    inputDesc->mRefCount = 1;
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
+{
+    ALOGV("stopInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        ALOGW("stopInput() unknown input %d", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+    if (inputDesc->mRefCount == 0) {
+        ALOGW("stopInput() input %d already stopped", input);
+        return INVALID_OPERATION;
+    } else {
+        // automatically disable the remote submix output when input is stopped
+        if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+            setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                    AudioSystem::DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+        }
+
+        AudioParameter param = AudioParameter();
+        param.addInt(String8(AudioParameter::keyRouting), 0);
+        mpClientInterface->setParameters(input, param.toString());
+        inputDesc->mRefCount = 0;
+        return NO_ERROR;
+    }
+}
+
+void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
+{
+    ALOGV("releaseInput() %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        ALOGW("releaseInput() releasing unknown input %d", input);
+        return;
+    }
+    mpClientInterface->closeInput(input);
+    delete mInputs.valueAt(index);
+    mInputs.removeItem(input);
+    ALOGV("releaseInput() exit");
+}
+
+void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
+                                            int indexMin,
+                                            int indexMax)
+{
+    ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+    if (indexMin < 0 || indexMin >= indexMax) {
+        ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
+        return;
+    }
+    mStreams[stream].mIndexMin = indexMin;
+    mStreams[stream].mIndexMax = indexMax;
+}
+
+status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream,
+                                                      int index,
+                                                      audio_devices_t device)
+{
+
+    if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+        return BAD_VALUE;
+    }
+    if (!audio_is_output_device(device)) {
+        return BAD_VALUE;
+    }
+
+    // Force max volume if stream cannot be muted
+    if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+    ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
+          stream, device, index);
+
+    // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
+    // clear all device specific values
+    if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+        mStreams[stream].mIndexCur.clear();
+    }
+    mStreams[stream].mIndexCur.add(device, index);
+
+    // compute and apply stream volume on all outputs according to connected device
+    status_t status = NO_ERROR;
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        audio_devices_t curDevice =
+                getDeviceForVolume(mOutputs.valueAt(i)->device());
+        if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) {
+            status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+            if (volStatus != NO_ERROR) {
+                status = volStatus;
+            }
+        }
+    }
+    return status;
+}
+
+status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream,
+                                                      int *index,
+                                                      audio_devices_t device)
+{
+    if (index == NULL) {
+        return BAD_VALUE;
+    }
+    if (!audio_is_output_device(device)) {
+        return BAD_VALUE;
+    }
+    // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
+    // the strategy the stream belongs to.
+    if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+        device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+    }
+    device = getDeviceForVolume(device);
+
+    *index =  mStreams[stream].getVolumeIndex(device);
+    ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
+    return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManagerBase::selectOutputForEffects(
+                                            const SortedVector<audio_io_handle_t>& outputs)
+{
+    // select one output among several suitable for global effects.
+    // The priority is as follows:
+    // 1: An offloaded output. If the effect ends up not being offloadable,
+    //    AudioFlinger will invalidate the track and the offloaded output
+    //    will be closed causing the effect to be moved to a PCM output.
+    // 2: A deep buffer output
+    // 3: the first output in the list
+
+    if (outputs.size() == 0) {
+        return 0;
+    }
+
+    audio_io_handle_t outputOffloaded = 0;
+    audio_io_handle_t outputDeepBuffer = 0;
+
+    for (size_t i = 0; i < outputs.size(); i++) {
+        AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
+        ALOGV("selectOutputForEffects outputs[%d] flags %x", i, desc->mFlags);
+        if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+            outputOffloaded = outputs[i];
+        }
+        if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+            outputDeepBuffer = outputs[i];
+        }
+    }
+
+    ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
+          outputOffloaded, outputDeepBuffer);
+    if (outputOffloaded != 0) {
+        return outputOffloaded;
+    }
+    if (outputDeepBuffer != 0) {
+        return outputDeepBuffer;
+    }
+
+    return outputs[0];
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(const effect_descriptor_t *desc)
+{
+    // apply simple rule where global effects are attached to the same output as MUSIC streams
+
+    routing_strategy strategy = getStrategy(AudioSystem::MUSIC);
+    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
+
+    audio_io_handle_t output = selectOutputForEffects(dstOutputs);
+    ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
+          output, (desc == NULL) ? "unspecified" : desc->name,  (desc == NULL) ? 0 : desc->flags);
+
+    return output;
+}
+
+status_t AudioPolicyManagerBase::registerEffect(const effect_descriptor_t *desc,
+                                audio_io_handle_t io,
+                                uint32_t strategy,
+                                int session,
+                                int id)
+{
+    ssize_t index = mOutputs.indexOfKey(io);
+    if (index < 0) {
+        index = mInputs.indexOfKey(io);
+        if (index < 0) {
+            ALOGW("registerEffect() unknown io %d", io);
+            return INVALID_OPERATION;
+        }
+    }
+
+    if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
+        ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
+                desc->name, desc->memoryUsage);
+        return INVALID_OPERATION;
+    }
+    mTotalEffectsMemory += desc->memoryUsage;
+    ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
+            desc->name, io, strategy, session, id);
+    ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
+
+    EffectDescriptor *pDesc = new EffectDescriptor();
+    memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
+    pDesc->mIo = io;
+    pDesc->mStrategy = (routing_strategy)strategy;
+    pDesc->mSession = session;
+    pDesc->mEnabled = false;
+
+    mEffects.add(id, pDesc);
+
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::unregisterEffect(int id)
+{
+    ssize_t index = mEffects.indexOfKey(id);
+    if (index < 0) {
+        ALOGW("unregisterEffect() unknown effect ID %d", id);
+        return INVALID_OPERATION;
+    }
+
+    EffectDescriptor *pDesc = mEffects.valueAt(index);
+
+    setEffectEnabled(pDesc, false);
+
+    if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
+        ALOGW("unregisterEffect() memory %d too big for total %d",
+                pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+        pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
+    }
+    mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
+    ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
+            pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+
+    mEffects.removeItem(id);
+    delete pDesc;
+
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::setEffectEnabled(int id, bool enabled)
+{
+    ssize_t index = mEffects.indexOfKey(id);
+    if (index < 0) {
+        ALOGW("unregisterEffect() unknown effect ID %d", id);
+        return INVALID_OPERATION;
+    }
+
+    return setEffectEnabled(mEffects.valueAt(index), enabled);
+}
+
+status_t AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
+{
+    if (enabled == pDesc->mEnabled) {
+        ALOGV("setEffectEnabled(%s) effect already %s",
+             enabled?"true":"false", enabled?"enabled":"disabled");
+        return INVALID_OPERATION;
+    }
+
+    if (enabled) {
+        if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
+            ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
+                 pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
+            return INVALID_OPERATION;
+        }
+        mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
+        ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
+    } else {
+        if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
+            ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
+                    pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
+            pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
+        }
+        mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
+        ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
+    }
+    pDesc->mEnabled = enabled;
+    return NO_ERROR;
+}
+
+bool AudioPolicyManagerBase::isNonOffloadableEffectEnabled()
+{
+    for (size_t i = 0; i < mEffects.size(); i++) {
+        const EffectDescriptor * const pDesc = mEffects.valueAt(i);
+        if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) &&
+                ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
+            ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
+                  pDesc->mDesc.name, pDesc->mSession);
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const
+{
+    nsecs_t sysTime = systemTime();
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+        if (outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManagerBase::isStreamActiveRemotely(int stream, uint32_t inPastMs) const
+{
+    nsecs_t sysTime = systemTime();
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+        if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+                outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManagerBase::isSourceActive(audio_source_t source) const
+{
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
+        if ((inputDescriptor->mInputSource == (int)source ||
+                (source == (audio_source_t)AUDIO_SOURCE_VOICE_RECOGNITION &&
+                 inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
+             && (inputDescriptor->mRefCount > 0)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+
+status_t AudioPolicyManagerBase::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbCardAndDevice.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AudioSystem::FOR_SYSTEM]);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+
+    snprintf(buffer, SIZE, "\nHW Modules dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mHwModules.size(); i++) {
+        snprintf(buffer, SIZE, "- HW Module %d:\n", i + 1);
+        write(fd, buffer, strlen(buffer));
+        mHwModules[i]->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nOutputs dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mOutputs.valueAt(i)->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nInputs dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mInputs.valueAt(i)->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nStreams dump:\n");
+    write(fd, buffer, strlen(buffer));
+    snprintf(buffer, SIZE,
+             " Stream  Can be muted  Index Min  Index Max  Index Cur [device : index]...\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        snprintf(buffer, SIZE, " %02d      ", i);
+        write(fd, buffer, strlen(buffer));
+        mStreams[i].dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
+            (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
+    write(fd, buffer, strlen(buffer));
+
+    snprintf(buffer, SIZE, "Registered effects:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mEffects.size(); i++) {
+        snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mEffects.valueAt(i)->dump(fd);
+    }
+
+
+    return NO_ERROR;
+}
+
+// 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 AudioPolicyManagerBase::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+    ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+     " BitRate=%u, duration=%lld us, has_video=%d",
+     offloadInfo.sample_rate, offloadInfo.channel_mask,
+     offloadInfo.format,
+     offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+     offloadInfo.has_video);
+
+    // Check if offload has been disabled
+    char propValue[PROPERTY_VALUE_MAX];
+    if (property_get("audio.offload.disable", propValue, "0")) {
+        if (atoi(propValue) != 0) {
+            ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+            return false;
+        }
+    }
+
+    // Check if stream type is music, then only allow offload as of now.
+    if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+    {
+        ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+        return false;
+    }
+
+    //TODO: enable audio offloading with video when ready
+    if (offloadInfo.has_video)
+    {
+        ALOGV("isOffloadSupported: has_video == true, returning false");
+        return false;
+    }
+
+    //If duration is less than minimum value defined in property, return false
+    if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+        if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+            ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+            return false;
+        }
+    } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+        ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+        return false;
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+    // creating an offloaded track and tearing it down immediately after start when audioflinger
+    // detects there is an active non offloadable effect.
+    // FIXME: We should check the audio session here but we do not have it in this context.
+    // This may prevent offloading in rare situations where effects are left active by apps
+    // in the background.
+    if (isNonOffloadableEffectEnabled()) {
+        return false;
+    }
+
+    // See if there is a profile to support this.
+    // AUDIO_DEVICE_NONE
+    IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+                                            offloadInfo.sample_rate,
+                                            offloadInfo.format,
+                                            offloadInfo.channel_mask,
+                                            AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+    ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
+    return (profile != NULL);
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerBase
+// ----------------------------------------------------------------------------
+
+AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
+    :
+#ifdef AUDIO_POLICY_TEST
+    Thread(false),
+#endif //AUDIO_POLICY_TEST
+    mPrimaryOutput((audio_io_handle_t)0),
+    mAvailableOutputDevices(AUDIO_DEVICE_NONE),
+    mPhoneState(AudioSystem::MODE_NORMAL),
+    mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
+    mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
+    mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false),
+    mSpeakerDrcEnabled(false)
+{
+    mpClientInterface = clientInterface;
+
+    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
+        mForceUse[i] = AudioSystem::FORCE_NONE;
+    }
+
+    mA2dpDeviceAddress = String8("");
+    mScoDeviceAddress = String8("");
+    mUsbCardAndDevice = String8("");
+
+    if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
+        if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
+            ALOGE("could not load audio policy configuration file, setting defaults");
+            defaultAudioPolicyConfig();
+        }
+    }
+
+    // must be done after reading the policy
+    initializeVolumeCurves();
+
+    // open all output streams needed to access attached devices
+    for (size_t i = 0; i < mHwModules.size(); i++) {
+        mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
+        if (mHwModules[i]->mHandle == 0) {
+            ALOGW("could not open HW module %s", mHwModules[i]->mName);
+            continue;
+        }
+        // open all output streams needed to access attached devices
+        // except for direct output streams that are only opened when they are actually
+        // required by an app.
+        for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+        {
+            const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j];
+
+            if ((outProfile->mSupportedDevices & mAttachedOutputDevices) &&
+                    ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
+                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
+                outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice &
+                                                            outProfile->mSupportedDevices);
+                audio_io_handle_t output = mpClientInterface->openOutput(
+                                                outProfile->mModule->mHandle,
+                                                &outputDesc->mDevice,
+                                                &outputDesc->mSamplingRate,
+                                                &outputDesc->mFormat,
+                                                &outputDesc->mChannelMask,
+                                                &outputDesc->mLatency,
+                                                outputDesc->mFlags);
+                if (output == 0) {
+                    delete outputDesc;
+                } else {
+                    mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices |
+                                            (outProfile->mSupportedDevices & mAttachedOutputDevices));
+                    if (mPrimaryOutput == 0 &&
+                            outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+                        mPrimaryOutput = output;
+                    }
+                    addOutput(output, outputDesc);
+                    setOutputDevice(output,
+                                    (audio_devices_t)(mDefaultOutputDevice &
+                                                        outProfile->mSupportedDevices),
+                                    true);
+                }
+            }
+        }
+    }
+
+    ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices),
+             "Not output found for attached devices %08x",
+             (mAttachedOutputDevices & ~mAvailableOutputDevices));
+
+    ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
+
+    updateDevicesAndOutputs();
+
+#ifdef AUDIO_POLICY_TEST
+    if (mPrimaryOutput != 0) {
+        AudioParameter outputCmd = AudioParameter();
+        outputCmd.addInt(String8("set_id"), 0);
+        mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+
+        mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
+        mTestSamplingRate = 44100;
+        mTestFormat = AudioSystem::PCM_16_BIT;
+        mTestChannels =  AudioSystem::CHANNEL_OUT_STEREO;
+        mTestLatencyMs = 0;
+        mCurOutput = 0;
+        mDirectOutput = false;
+        for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+            mTestOutputs[i] = 0;
+        }
+
+        const size_t SIZE = 256;
+        char buffer[SIZE];
+        snprintf(buffer, SIZE, "AudioPolicyManagerTest");
+        run(buffer, ANDROID_PRIORITY_AUDIO);
+    }
+#endif //AUDIO_POLICY_TEST
+}
+
+AudioPolicyManagerBase::~AudioPolicyManagerBase()
+{
+#ifdef AUDIO_POLICY_TEST
+    exit();
+#endif //AUDIO_POLICY_TEST
+   for (size_t i = 0; i < mOutputs.size(); i++) {
+        mpClientInterface->closeOutput(mOutputs.keyAt(i));
+        delete mOutputs.valueAt(i);
+   }
+   for (size_t i = 0; i < mInputs.size(); i++) {
+        mpClientInterface->closeInput(mInputs.keyAt(i));
+        delete mInputs.valueAt(i);
+   }
+   for (size_t i = 0; i < mHwModules.size(); i++) {
+        delete mHwModules[i];
+   }
+}
+
+status_t AudioPolicyManagerBase::initCheck()
+{
+    return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManagerBase::threadLoop()
+{
+    ALOGV("entering threadLoop()");
+    while (!exitPending())
+    {
+        String8 command;
+        int valueInt;
+        String8 value;
+
+        Mutex::Autolock _l(mLock);
+        mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+
+        command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+        AudioParameter param = AudioParameter(command);
+
+        if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
+            valueInt != 0) {
+            ALOGV("Test command %s received", command.string());
+            String8 target;
+            if (param.get(String8("target"), target) != NO_ERROR) {
+                target = "Manager";
+            }
+            if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_output"));
+                mCurOutput = valueInt;
+            }
+            if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_direct"));
+                if (value == "false") {
+                    mDirectOutput = false;
+                } else if (value == "true") {
+                    mDirectOutput = true;
+                }
+            }
+            if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_input"));
+                mTestInput = valueInt;
+            }
+
+            if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_format"));
+                int format = AudioSystem::INVALID_FORMAT;
+                if (value == "PCM 16 bits") {
+                    format = AudioSystem::PCM_16_BIT;
+                } else if (value == "PCM 8 bits") {
+                    format = AudioSystem::PCM_8_BIT;
+                } else if (value == "Compressed MP3") {
+                    format = AudioSystem::MP3;
+                }
+                if (format != AudioSystem::INVALID_FORMAT) {
+                    if (target == "Manager") {
+                        mTestFormat = format;
+                    } else if (mTestOutputs[mCurOutput] != 0) {
+                        AudioParameter outputParam = AudioParameter();
+                        outputParam.addInt(String8("format"), format);
+                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+                    }
+                }
+            }
+            if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_channels"));
+                int channels = 0;
+
+                if (value == "Channels Stereo") {
+                    channels =  AudioSystem::CHANNEL_OUT_STEREO;
+                } else if (value == "Channels Mono") {
+                    channels =  AudioSystem::CHANNEL_OUT_MONO;
+                }
+                if (channels != 0) {
+                    if (target == "Manager") {
+                        mTestChannels = channels;
+                    } else if (mTestOutputs[mCurOutput] != 0) {
+                        AudioParameter outputParam = AudioParameter();
+                        outputParam.addInt(String8("channels"), channels);
+                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+                    }
+                }
+            }
+            if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_sampleRate"));
+                if (valueInt >= 0 && valueInt <= 96000) {
+                    int samplingRate = valueInt;
+                    if (target == "Manager") {
+                        mTestSamplingRate = samplingRate;
+                    } else if (mTestOutputs[mCurOutput] != 0) {
+                        AudioParameter outputParam = AudioParameter();
+                        outputParam.addInt(String8("sampling_rate"), samplingRate);
+                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+                    }
+                }
+            }
+
+            if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_reopen"));
+
+                AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+                mpClientInterface->closeOutput(mPrimaryOutput);
+
+                audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
+
+                delete mOutputs.valueFor(mPrimaryOutput);
+                mOutputs.removeItem(mPrimaryOutput);
+
+                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+                outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
+                mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
+                                                &outputDesc->mDevice,
+                                                &outputDesc->mSamplingRate,
+                                                &outputDesc->mFormat,
+                                                &outputDesc->mChannelMask,
+                                                &outputDesc->mLatency,
+                                                outputDesc->mFlags);
+                if (mPrimaryOutput == 0) {
+                    ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
+                            outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
+                } else {
+                    AudioParameter outputCmd = AudioParameter();
+                    outputCmd.addInt(String8("set_id"), 0);
+                    mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+                    addOutput(mPrimaryOutput, outputDesc);
+                }
+            }
+
+
+            mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+        }
+    }
+    return false;
+}
+
+void AudioPolicyManagerBase::exit()
+{
+    {
+        AutoMutex _l(mLock);
+        requestExit();
+        mWaitWorkCV.signal();
+    }
+    requestExitAndWait();
+}
+
+int AudioPolicyManagerBase::testOutputIndex(audio_io_handle_t output)
+{
+    for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+        if (output == mTestOutputs[i]) return i;
+    }
+    return 0;
+}
+#endif //AUDIO_POLICY_TEST
+
+// ---
+
+void AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
+{
+    outputDesc->mId = id;
+    mOutputs.add(id, outputDesc);
+}
+
+
+status_t AudioPolicyManagerBase::checkOutputsForDevice(audio_devices_t device,
+                                                       AudioSystem::device_connection_state state,
+                                                       SortedVector<audio_io_handle_t>& outputs,
+                                                       const String8 paramStr)
+{
+    AudioOutputDescriptor *desc;
+
+    if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
+        // first list already open outputs that can be routed to this device
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) {
+                ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
+                outputs.add(mOutputs.keyAt(i));
+            }
+        }
+        // then look for output profiles that can be routed to this device
+        SortedVector<IOProfile *> profiles;
+        for (size_t i = 0; i < mHwModules.size(); i++)
+        {
+            if (mHwModules[i]->mHandle == 0) {
+                continue;
+            }
+            for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+            {
+                if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) {
+                    ALOGV("checkOutputsForDevice(): adding profile %d from module %d", j, i);
+                    profiles.add(mHwModules[i]->mOutputProfiles[j]);
+                }
+            }
+        }
+
+        if (profiles.isEmpty() && outputs.isEmpty()) {
+            ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+            return BAD_VALUE;
+        }
+
+        // open outputs for matching profiles if needed. Direct outputs are also opened to
+        // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+        for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+            IOProfile *profile = profiles[profile_index];
+
+            // nothing to do if one output is already opened for this profile
+            size_t j;
+            for (j = 0; j < mOutputs.size(); j++) {
+                desc = mOutputs.valueAt(j);
+                if (!desc->isDuplicated() && desc->mProfile == profile) {
+                    break;
+                }
+            }
+            if (j != mOutputs.size()) {
+                continue;
+            }
+
+            ALOGV("opening output for device %08x with params %s", device, paramStr.string());
+            desc = new AudioOutputDescriptor(profile);
+            desc->mDevice = device;
+            audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
+            offloadInfo.sample_rate = desc->mSamplingRate;
+            offloadInfo.format = desc->mFormat;
+            offloadInfo.channel_mask = desc->mChannelMask;
+
+            audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle,
+                                                                       &desc->mDevice,
+                                                                       &desc->mSamplingRate,
+                                                                       &desc->mFormat,
+                                                                       &desc->mChannelMask,
+                                                                       &desc->mLatency,
+                                                                       desc->mFlags,
+                                                                       &offloadInfo);
+            if (output != 0) {
+                if (!paramStr.isEmpty()) {
+                    mpClientInterface->setParameters(output, paramStr);
+                }
+
+                if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+                    String8 reply;
+                    char *value;
+                    if (profile->mSamplingRates[0] == 0) {
+                        reply = mpClientInterface->getParameters(output,
+                                                String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+                        ALOGV("checkOutputsForDevice() direct output sup sampling rates %s",
+                                  reply.string());
+                        value = strpbrk((char *)reply.string(), "=");
+                        if (value != NULL) {
+                            loadSamplingRates(value + 1, profile);
+                        }
+                    }
+                    if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+                        reply = mpClientInterface->getParameters(output,
+                                                       String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+                        ALOGV("checkOutputsForDevice() direct output sup formats %s",
+                                  reply.string());
+                        value = strpbrk((char *)reply.string(), "=");
+                        if (value != NULL) {
+                            loadFormats(value + 1, profile);
+                        }
+                    }
+                    if (profile->mChannelMasks[0] == 0) {
+                        reply = mpClientInterface->getParameters(output,
+                                                      String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+                        ALOGV("checkOutputsForDevice() direct output sup channel masks %s",
+                                  reply.string());
+                        value = strpbrk((char *)reply.string(), "=");
+                        if (value != NULL) {
+                            loadOutChannels(value + 1, profile);
+                        }
+                    }
+                    if (((profile->mSamplingRates[0] == 0) &&
+                             (profile->mSamplingRates.size() < 2)) ||
+                         ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+                             (profile->mFormats.size() < 2)) ||
+                         ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+                             (profile->mChannelMasks.size() < 2))) {
+                        ALOGW("checkOutputsForDevice() direct output missing param");
+                        mpClientInterface->closeOutput(output);
+                        output = 0;
+                    } else {
+                        addOutput(output, desc);
+                    }
+                } else {
+                    audio_io_handle_t duplicatedOutput = 0;
+                    // add output descriptor
+                    addOutput(output, desc);
+                    // set initial stream volume for device
+                    applyStreamVolumes(output, device, 0, true);
+
+                    //TODO: configure audio effect output stage here
+
+                    // open a duplicating output thread for the new output and the primary output
+                    duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
+                                                                              mPrimaryOutput);
+                    if (duplicatedOutput != 0) {
+                        // add duplicated output descriptor
+                        AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL);
+                        dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
+                        dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
+                        dupOutputDesc->mSamplingRate = desc->mSamplingRate;
+                        dupOutputDesc->mFormat = desc->mFormat;
+                        dupOutputDesc->mChannelMask = desc->mChannelMask;
+                        dupOutputDesc->mLatency = desc->mLatency;
+                        addOutput(duplicatedOutput, dupOutputDesc);
+                        applyStreamVolumes(duplicatedOutput, device, 0, true);
+                    } else {
+                        ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
+                                mPrimaryOutput, output);
+                        mpClientInterface->closeOutput(output);
+                        mOutputs.removeItem(output);
+                        output = 0;
+                    }
+                }
+            }
+            if (output == 0) {
+                ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+                delete desc;
+                profiles.removeAt(profile_index);
+                profile_index--;
+            } else {
+                outputs.add(output);
+                ALOGV("checkOutputsForDevice(): adding output %d", output);
+            }
+        }
+
+        if (profiles.isEmpty()) {
+            ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+            return BAD_VALUE;
+        }
+    } else {
+        // check if one opened output is not needed any more after disconnecting one device
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() &&
+                    !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) {
+                ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i));
+                outputs.add(mOutputs.keyAt(i));
+            }
+        }
+        for (size_t i = 0; i < mHwModules.size(); i++)
+        {
+            if (mHwModules[i]->mHandle == 0) {
+                continue;
+            }
+            for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+            {
+                IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+                if ((profile->mSupportedDevices & device) &&
+                        (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
+                    ALOGV("checkOutputsForDevice(): clearing direct output profile %d on module %d",
+                          j, i);
+                    if (profile->mSamplingRates[0] == 0) {
+                        profile->mSamplingRates.clear();
+                        profile->mSamplingRates.add(0);
+                    }
+                    if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+                        profile->mFormats.clear();
+                        profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+                    }
+                    if (profile->mChannelMasks[0] == 0) {
+                        profile->mChannelMasks.clear();
+                        profile->mChannelMasks.add(0);
+                    }
+                }
+            }
+        }
+    }
+    return NO_ERROR;
+}
+
+void AudioPolicyManagerBase::closeOutput(audio_io_handle_t output)
+{
+    ALOGV("closeOutput(%d)", output);
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    if (outputDesc == NULL) {
+        ALOGW("closeOutput() unknown output %d", output);
+        return;
+    }
+
+    // look for duplicated outputs connected to the output being removed.
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
+        if (dupOutputDesc->isDuplicated() &&
+                (dupOutputDesc->mOutput1 == outputDesc ||
+                dupOutputDesc->mOutput2 == outputDesc)) {
+            AudioOutputDescriptor *outputDesc2;
+            if (dupOutputDesc->mOutput1 == outputDesc) {
+                outputDesc2 = dupOutputDesc->mOutput2;
+            } else {
+                outputDesc2 = dupOutputDesc->mOutput1;
+            }
+            // As all active tracks on duplicated output will be deleted,
+            // and as they were also referenced on the other output, the reference
+            // count for their stream type must be adjusted accordingly on
+            // the other output.
+            for (int j = 0; j < (int)AudioSystem::NUM_STREAM_TYPES; j++) {
+                int refCount = dupOutputDesc->mRefCount[j];
+                outputDesc2->changeRefCount((AudioSystem::stream_type)j,-refCount);
+            }
+            audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
+            ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
+
+            mpClientInterface->closeOutput(duplicatedOutput);
+            delete mOutputs.valueFor(duplicatedOutput);
+            mOutputs.removeItem(duplicatedOutput);
+        }
+    }
+
+    AudioParameter param;
+    param.add(String8("closing"), String8("true"));
+    mpClientInterface->setParameters(output, param.toString());
+
+    mpClientInterface->closeOutput(output);
+    delete outputDesc;
+    mOutputs.removeItem(output);
+    mPreviousOutputs = mOutputs;
+}
+
+SortedVector<audio_io_handle_t> AudioPolicyManagerBase::getOutputsForDevice(audio_devices_t device,
+                        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs)
+{
+    SortedVector<audio_io_handle_t> outputs;
+
+    ALOGVV("getOutputsForDevice() device %04x", device);
+    for (size_t i = 0; i < openOutputs.size(); i++) {
+        ALOGVV("output %d isDuplicated=%d device=%04x",
+                i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
+        if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
+            ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+            outputs.add(openOutputs.keyAt(i));
+        }
+    }
+    return outputs;
+}
+
+bool AudioPolicyManagerBase::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+                                   SortedVector<audio_io_handle_t>& outputs2)
+{
+    if (outputs1.size() != outputs2.size()) {
+        return false;
+    }
+    for (size_t i = 0; i < outputs1.size(); i++) {
+        if (outputs1[i] != outputs2[i]) {
+            return false;
+        }
+    }
+    return true;
+}
+
+void AudioPolicyManagerBase::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, mPreviousOutputs);
+    SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+    if (!vectorsEqual(srcOutputs,dstOutputs)) {
+        ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
+              strategy, srcOutputs[0], dstOutputs[0]);
+        // mute strategy while moving tracks from one output to another
+        for (size_t i = 0; i < srcOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]);
+            if (desc->isStrategyActive(strategy)) {
+                setStrategyMute(strategy, true, srcOutputs[i]);
+                setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
+            }
+        }
+
+        // Move effects associated to this strategy from previous output to new output
+        if (strategy == STRATEGY_MEDIA) {
+            audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
+            SortedVector<audio_io_handle_t> moved;
+            for (size_t i = 0; i < mEffects.size(); i++) {
+                EffectDescriptor *desc = mEffects.valueAt(i);
+                if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
+                        desc->mIo != fxOutput) {
+                    if (moved.indexOf(desc->mIo) < 0) {
+                        ALOGV("checkOutputForStrategy() moving effect %d to output %d",
+                              mEffects.keyAt(i), fxOutput);
+                        mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
+                                                       fxOutput);
+                        moved.add(desc->mIo);
+                    }
+                    desc->mIo = fxOutput;
+                }
+            }
+        }
+        // Move tracks associated to this strategy from previous output to new output
+        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+            if (getStrategy((AudioSystem::stream_type)i) == strategy) {
+                mpClientInterface->invalidateStream((AudioSystem::stream_type)i);
+            }
+        }
+    }
+}
+
+void AudioPolicyManagerBase::checkOutputForAllStrategies()
+{
+    checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+    checkOutputForStrategy(STRATEGY_PHONE);
+    checkOutputForStrategy(STRATEGY_SONIFICATION);
+    checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+    checkOutputForStrategy(STRATEGY_MEDIA);
+    checkOutputForStrategy(STRATEGY_DTMF);
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput()
+{
+    if (!mHasA2dp) {
+        return 0;
+    }
+
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+        if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+            return mOutputs.keyAt(i);
+        }
+    }
+
+    return 0;
+}
+
+void AudioPolicyManagerBase::checkA2dpSuspend()
+{
+    if (!mHasA2dp) {
+        return;
+    }
+    audio_io_handle_t a2dpOutput = getA2dpOutput();
+    if (a2dpOutput == 0) {
+        return;
+    }
+
+    // suspend A2DP output if:
+    //      (NOT already suspended) &&
+    //      ((SCO device is connected &&
+    //       (forced usage for communication || for record is SCO))) ||
+    //      (phone state is ringing || in call)
+    //
+    // restore A2DP output if:
+    //      (Already suspended) &&
+    //      ((SCO device is NOT connected ||
+    //       (forced usage NOT for communication && NOT for record is SCO))) &&
+    //      (phone state is NOT ringing && NOT in call)
+    //
+    if (mA2dpSuspended) {
+        if (((mScoDeviceAddress == "") ||
+             ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
+              (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) &&
+             ((mPhoneState != AudioSystem::MODE_IN_CALL) &&
+              (mPhoneState != AudioSystem::MODE_RINGTONE))) {
+
+            mpClientInterface->restoreOutput(a2dpOutput);
+            mA2dpSuspended = false;
+        }
+    } else {
+        if (((mScoDeviceAddress != "") &&
+             ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
+              (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) ||
+             ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
+              (mPhoneState == AudioSystem::MODE_RINGTONE))) {
+
+            mpClientInterface->suspendOutput(a2dpOutput);
+            mA2dpSuspended = true;
+        }
+    }
+}
+
+audio_devices_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
+{
+    audio_devices_t device = AUDIO_DEVICE_NONE;
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    // check the following by order of priority to request a routing change if necessary:
+    // 1: the strategy enforced audible is active on the output:
+    //      use device for strategy enforced audible
+    // 2: we are in call or the strategy phone is active on the output:
+    //      use device for strategy phone
+    // 3: the strategy sonification is active on the output:
+    //      use device for strategy sonification
+    // 4: the strategy "respectful" sonification is active on the output:
+    //      use device for strategy "respectful" sonification
+    // 5: the strategy media is active on the output:
+    //      use device for strategy media
+    // 6: the strategy DTMF is active on the output:
+    //      use device for strategy DTMF
+    if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
+        device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+    } else if (isInCall() ||
+                    outputDesc->isStrategyActive(STRATEGY_PHONE)) {
+        device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
+        device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
+        device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
+        device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+    } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
+        device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+    }
+
+    ALOGV("getNewDevice() selected device %x", device);
+    return device;
+}
+
+uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
+    return (uint32_t)getStrategy(stream);
+}
+
+audio_devices_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
+    audio_devices_t devices;
+    // By checking the range of stream before calling getStrategy, we avoid
+    // getStrategy's behavior for invalid streams.  getStrategy would do a ALOGE
+    // and then return STRATEGY_MEDIA, but we want to return the empty set.
+    if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+        devices = AUDIO_DEVICE_NONE;
+    } else {
+        AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
+        devices = getDeviceForStrategy(strategy, true /*fromCache*/);
+    }
+    return devices;
+}
+
+AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
+        AudioSystem::stream_type stream) {
+    // stream to strategy mapping
+    switch (stream) {
+    case AudioSystem::VOICE_CALL:
+    case AudioSystem::BLUETOOTH_SCO:
+        return STRATEGY_PHONE;
+    case AudioSystem::RING:
+    case AudioSystem::ALARM:
+        return STRATEGY_SONIFICATION;
+    case AudioSystem::NOTIFICATION:
+        return STRATEGY_SONIFICATION_RESPECTFUL;
+    case AudioSystem::DTMF:
+        return STRATEGY_DTMF;
+    default:
+        ALOGE("unknown stream type");
+    case AudioSystem::SYSTEM:
+        // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+        // while key clicks are played produces a poor result
+    case AudioSystem::TTS:
+    case AudioSystem::MUSIC:
+        return STRATEGY_MEDIA;
+    case AudioSystem::ENFORCED_AUDIBLE:
+        return STRATEGY_ENFORCED_AUDIBLE;
+    }
+}
+
+void AudioPolicyManagerBase::handleNotificationRoutingForStream(AudioSystem::stream_type stream) {
+    switch(stream) {
+    case AudioSystem::MUSIC:
+        checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+        updateDevicesAndOutputs();
+        break;
+    default:
+        break;
+    }
+}
+
+audio_devices_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
+                                                             bool fromCache)
+{
+    uint32_t device = AUDIO_DEVICE_NONE;
+
+    if (fromCache) {
+        ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+              strategy, mDeviceForStrategy[strategy]);
+        return mDeviceForStrategy[strategy];
+    }
+
+    switch (strategy) {
+
+    case STRATEGY_SONIFICATION_RESPECTFUL:
+        if (isInCall()) {
+            device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+        } else if (isStreamActiveRemotely(AudioSystem::MUSIC,
+                SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+            // while media is playing on a remote device, use the the sonification behavior.
+            // Note that we test this usecase before testing if media is playing because
+            //   the isStreamActive() method only informs about the activity of a stream, not
+            //   if it's for local playback. Note also that we use the same delay between both tests
+            device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+        } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+            // while media is playing (or has recently played), use the same device
+            device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+        } else {
+            // when media is not playing anymore, fall back on the sonification behavior
+            device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+        }
+
+        break;
+
+    case STRATEGY_DTMF:
+        if (!isInCall()) {
+            // when off call, DTMF strategy follows the same rules as MEDIA strategy
+            device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+            break;
+        }
+        // when in call, DTMF and PHONE strategies follow the same rules
+        // FALL THROUGH
+
+    case STRATEGY_PHONE:
+        // for phone strategy, we first consider the forced use and then the available devices by order
+        // of priority
+        switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
+        case AudioSystem::FORCE_BT_SCO:
+            if (!isInCall() || strategy != STRATEGY_DTMF) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+            if (device) break;
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+            if (device) break;
+            // if SCO device is requested but no SCO device is available, fall back to default case
+            // FALL THROUGH
+
+        default:    // FORCE_NONE
+            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+            if (mHasA2dp && !isInCall() &&
+                    (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+            if (device) break;
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+            if (device) break;
+            if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
+            if (device) break;
+            device = mDefaultOutputDevice;
+            if (device == AUDIO_DEVICE_NONE) {
+                ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+            }
+            break;
+
+        case AudioSystem::FORCE_SPEAKER:
+            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+            // A2DP speaker when forcing to speaker output
+            if (mHasA2dp && !isInCall() &&
+                    (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+                if (device) break;
+            }
+            if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+                if (device) break;
+                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+            if (device) break;
+            device = mDefaultOutputDevice;
+            if (device == AUDIO_DEVICE_NONE) {
+                ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+            }
+            break;
+        }
+    break;
+
+    case STRATEGY_SONIFICATION:
+
+        // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+        // handleIncallSonification().
+        if (isInCall()) {
+            device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+            break;
+        }
+        // FALL THROUGH
+
+    case STRATEGY_ENFORCED_AUDIBLE:
+        // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+        // except:
+        //   - when in call where it doesn't default to STRATEGY_PHONE behavior
+        //   - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+        if ((strategy == STRATEGY_SONIFICATION) ||
+                (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) {
+            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+            if (device == AUDIO_DEVICE_NONE) {
+                ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+            }
+        }
+        // The second device used for sonification is the same as the device used by media strategy
+        // FALL THROUGH
+
+    case STRATEGY_MEDIA: {
+        uint32_t device2 = AUDIO_DEVICE_NONE;
+        if (strategy != STRATEGY_SONIFICATION) {
+            // no sonification on remote submix (e.g. WFD)
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+        }
+        if ((device2 == AUDIO_DEVICE_NONE) &&
+                mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+                (getA2dpOutput() != 0) && !mA2dpSuspended) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+            if (device2 == AUDIO_DEVICE_NONE) {
+                device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+            }
+            if (device2 == AUDIO_DEVICE_NONE) {
+                device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+            }
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+        }
+        if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+            // no sonification on aux digital (e.g. HDMI)
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+        }
+        if ((device2 == AUDIO_DEVICE_NONE) &&
+                (mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+        }
+        if (device2 == AUDIO_DEVICE_NONE) {
+            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+        }
+
+        // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+        // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+        device |= device2;
+        if (device) break;
+        device = mDefaultOutputDevice;
+        if (device == AUDIO_DEVICE_NONE) {
+            ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+        }
+        } break;
+
+    default:
+        ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+        break;
+    }
+
+    ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+    return device;
+}
+
+void AudioPolicyManagerBase::updateDevicesAndOutputs()
+{
+    for (int i = 0; i < NUM_STRATEGIES; i++) {
+        mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+    }
+    mPreviousOutputs = mOutputs;
+}
+
+uint32_t AudioPolicyManagerBase::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+                                                       audio_devices_t prevDevice,
+                                                       uint32_t delayMs)
+{
+    // mute/unmute strategies using an incompatible device combination
+    // if muting, wait for the audio in pcm buffer to be drained before proceeding
+    // if unmuting, unmute only after the specified delay
+    if (outputDesc->isDuplicated()) {
+        return 0;
+    }
+
+    uint32_t muteWaitMs = 0;
+    audio_devices_t device = outputDesc->device();
+    bool shouldMute = outputDesc->isActive() && (AudioSystem::popCount(device) >= 2);
+    // temporary mute output if device selection changes to avoid volume bursts due to
+    // different per device volumes
+    bool tempMute = outputDesc->isActive() && (device != prevDevice);
+
+    for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+        audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+        bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+        bool doMute = false;
+
+        if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
+            doMute = true;
+            outputDesc->mStrategyMutedByDevice[i] = true;
+        } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
+            doMute = true;
+            outputDesc->mStrategyMutedByDevice[i] = false;
+        }
+        if (doMute || tempMute) {
+            for (size_t j = 0; j < mOutputs.size(); j++) {
+                AudioOutputDescriptor *desc = mOutputs.valueAt(j);
+                // skip output if it does not share any device with current output
+                if ((desc->supportedDevices() & outputDesc->supportedDevices())
+                        == AUDIO_DEVICE_NONE) {
+                    continue;
+                }
+                audio_io_handle_t curOutput = mOutputs.keyAt(j);
+                ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
+                      mute ? "muting" : "unmuting", i, curDevice, curOutput);
+                setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+                if (desc->isStrategyActive((routing_strategy)i)) {
+                    // do tempMute only for current output
+                    if (tempMute && (desc == outputDesc)) {
+                        setStrategyMute((routing_strategy)i, true, curOutput);
+                        setStrategyMute((routing_strategy)i, false, curOutput,
+                                            desc->latency() * 2, device);
+                    }
+                    if ((tempMute && (desc == outputDesc)) || mute) {
+                        if (muteWaitMs < desc->latency()) {
+                            muteWaitMs = desc->latency();
+                        }
+                    }
+                }
+            }
+        }
+    }
+
+    // FIXME: should not need to double latency if volume could be applied immediately by the
+    // audioflinger mixer. We must account for the delay between now and the next time
+    // the audioflinger thread for this output will process a buffer (which corresponds to
+    // one buffer size, usually 1/2 or 1/4 of the latency).
+    muteWaitMs *= 2;
+    // wait for the PCM output buffers to empty before proceeding with the rest of the command
+    if (muteWaitMs > delayMs) {
+        muteWaitMs -= delayMs;
+        usleep(muteWaitMs * 1000);
+        return muteWaitMs;
+    }
+    return 0;
+}
+
+uint32_t AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output,
+                                             audio_devices_t device,
+                                             bool force,
+                                             int delayMs)
+{
+    ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    AudioParameter param;
+    uint32_t muteWaitMs;
+
+    if (outputDesc->isDuplicated()) {
+        muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
+        muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
+        return muteWaitMs;
+    }
+    // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+    // output profile
+    if ((device != AUDIO_DEVICE_NONE) &&
+            ((device & outputDesc->mProfile->mSupportedDevices) == 0)) {
+        return 0;
+    }
+
+    // filter devices according to output selected
+    device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices);
+
+    audio_devices_t prevDevice = outputDesc->mDevice;
+
+    ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
+
+    if (device != AUDIO_DEVICE_NONE) {
+        outputDesc->mDevice = device;
+    }
+    muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+
+    // Do not change the routing if:
+    //  - the requested device is AUDIO_DEVICE_NONE
+    //  - the requested device is the same as current device and force is not specified.
+    // Doing this check here allows the caller to call setOutputDevice() without conditions
+    if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
+        ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
+        return muteWaitMs;
+    }
+
+    ALOGV("setOutputDevice() changing device");
+    // do the routing
+    param.addInt(String8(AudioParameter::keyRouting), (int)device);
+    mpClientInterface->setParameters(output, param.toString(), delayMs);
+
+    // update stream volumes according to new device
+    applyStreamVolumes(output, device, delayMs);
+
+    return muteWaitMs;
+}
+
+AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getInputProfile(audio_devices_t device,
+                                                   uint32_t samplingRate,
+                                                   audio_format_t format,
+                                                   audio_channel_mask_t channelMask)
+{
+    // Choose an input profile based on the requested capture parameters: select the first available
+    // profile supporting all requested parameters.
+
+    for (size_t i = 0; i < mHwModules.size(); i++)
+    {
+        if (mHwModules[i]->mHandle == 0) {
+            continue;
+        }
+        for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+        {
+            IOProfile *profile = mHwModules[i]->mInputProfiles[j];
+            if (profile->isCompatibleProfile(device, samplingRate, format,
+                                             channelMask, AUDIO_OUTPUT_FLAG_NONE)) {
+                return profile;
+            }
+        }
+    }
+    return NULL;
+}
+
+audio_devices_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
+{
+    uint32_t device = AUDIO_DEVICE_NONE;
+
+    switch (inputSource) {
+    case AUDIO_SOURCE_VOICE_UPLINK:
+      if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+          device = AUDIO_DEVICE_IN_VOICE_CALL;
+          break;
+      }
+      // FALL THROUGH
+
+    case AUDIO_SOURCE_DEFAULT:
+    case AUDIO_SOURCE_MIC:
+    case AUDIO_SOURCE_VOICE_RECOGNITION:
+    case AUDIO_SOURCE_HOTWORD:
+    case AUDIO_SOURCE_VOICE_COMMUNICATION:
+        if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
+            mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+            device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+        }
+        break;
+    case AUDIO_SOURCE_CAMCORDER:
+        if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
+            device = AUDIO_DEVICE_IN_BACK_MIC;
+        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+        }
+        break;
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+    case AUDIO_SOURCE_VOICE_CALL:
+        if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+            device = AUDIO_DEVICE_IN_VOICE_CALL;
+        }
+        break;
+    case AUDIO_SOURCE_REMOTE_SUBMIX:
+        if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+            device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+        }
+        break;
+    default:
+        ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+        break;
+    }
+    ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+    return device;
+}
+
+bool AudioPolicyManagerBase::isVirtualInputDevice(audio_devices_t device)
+{
+    if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+        device &= ~AUDIO_DEVICE_BIT_IN;
+        if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
+            return true;
+    }
+    return false;
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getActiveInput(bool ignoreVirtualInputs)
+{
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
+        if ((input_descriptor->mRefCount > 0)
+                && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
+            return mInputs.keyAt(i);
+        }
+    }
+    return 0;
+}
+
+
+audio_devices_t AudioPolicyManagerBase::getDeviceForVolume(audio_devices_t device)
+{
+    if (device == AUDIO_DEVICE_NONE) {
+        // this happens when forcing a route update and no track is active on an output.
+        // In this case the returned category is not important.
+        device =  AUDIO_DEVICE_OUT_SPEAKER;
+    } else if (AudioSystem::popCount(device) > 1) {
+        // Multiple device selection is either:
+        //  - speaker + one other device: give priority to speaker in this case.
+        //  - one A2DP device + another device: happens with duplicated output. In this case
+        // retain the device on the A2DP output as the other must not correspond to an active
+        // selection if not the speaker.
+        if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+            device = AUDIO_DEVICE_OUT_SPEAKER;
+        } else {
+            device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+        }
+    }
+
+    ALOGW_IF(AudioSystem::popCount(device) != 1,
+            "getDeviceForVolume() invalid device combination: %08x",
+            device);
+
+    return device;
+}
+
+AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(audio_devices_t device)
+{
+    switch(getDeviceForVolume(device)) {
+        case AUDIO_DEVICE_OUT_EARPIECE:
+            return DEVICE_CATEGORY_EARPIECE;
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+            return DEVICE_CATEGORY_HEADSET;
+        case AUDIO_DEVICE_OUT_SPEAKER:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+        case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+        case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+        case AUDIO_DEVICE_OUT_USB_DEVICE:
+        case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+        default:
+            return DEVICE_CATEGORY_SPEAKER;
+    }
+}
+
+float AudioPolicyManagerBase::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+        int indexInUi)
+{
+    device_category deviceCategory = getDeviceCategory(device);
+    const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+    // the volume index in the UI is relative to the min and max volume indices for this stream type
+    int nbSteps = 1 + curve[VOLMAX].mIndex -
+            curve[VOLMIN].mIndex;
+    int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+            (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+    // find what part of the curve this index volume belongs to, or if it's out of bounds
+    int segment = 0;
+    if (volIdx < curve[VOLMIN].mIndex) {         // out of bounds
+        return 0.0f;
+    } else if (volIdx < curve[VOLKNEE1].mIndex) {
+        segment = 0;
+    } else if (volIdx < curve[VOLKNEE2].mIndex) {
+        segment = 1;
+    } else if (volIdx <= curve[VOLMAX].mIndex) {
+        segment = 2;
+    } else {                                                               // out of bounds
+        return 1.0f;
+    }
+
+    // linear interpolation in the attenuation table in dB
+    float decibels = curve[segment].mDBAttenuation +
+            ((float)(volIdx - curve[segment].mIndex)) *
+                ( (curve[segment+1].mDBAttenuation -
+                        curve[segment].mDBAttenuation) /
+                    ((float)(curve[segment+1].mIndex -
+                            curve[segment].mIndex)) );
+
+    float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+    ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+            curve[segment].mIndex, volIdx,
+            curve[segment+1].mIndex,
+            curve[segment].mDBAttenuation,
+            decibels,
+            curve[segment+1].mDBAttenuation,
+            amplification);
+
+    return amplification;
+}
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+    AudioPolicyManagerBase::sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
+    {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManagerBase::VolumeCurvePoint
+            *AudioPolicyManagerBase::sVolumeProfiles[AUDIO_STREAM_CNT]
+                                                   [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = {
+    { // AUDIO_STREAM_VOICE_CALL
+        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVoiceVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_SYSTEM
+        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_RING
+        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_MUSIC
+        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_ALARM
+        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_NOTIFICATION
+        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_BLUETOOTH_SCO
+        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultVoiceVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_ENFORCED_AUDIBLE
+        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    {  // AUDIO_STREAM_DTMF
+        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+    { // AUDIO_STREAM_TTS
+        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EARPIECE
+    },
+};
+
+void AudioPolicyManagerBase::initializeVolumeCurves()
+{
+    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[i].mVolumeCurve[j] =
+                    sVolumeProfiles[i][j];
+        }
+    }
+
+    // Check availability of DRC on speaker path: if available, override some of the speaker curves
+    if (mSpeakerDrcEnabled) {
+        mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sDefaultSystemVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sSpeakerSonificationVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sSpeakerSonificationVolumeCurveDrc;
+        mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+                sSpeakerSonificationVolumeCurveDrc;
+    }
+}
+
+float AudioPolicyManagerBase::computeVolume(int stream,
+                                            int index,
+                                            audio_io_handle_t output,
+                                            audio_devices_t device)
+{
+    float volume = 1.0;
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    StreamDescriptor &streamDesc = mStreams[stream];
+
+    if (device == AUDIO_DEVICE_NONE) {
+        device = outputDesc->device();
+    }
+
+    // if volume is not 0 (not muted), force media volume to max on digital output
+    if (stream == AudioSystem::MUSIC &&
+        index != mStreams[stream].mIndexMin &&
+        (device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+         device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET ||
+         device == AUDIO_DEVICE_OUT_USB_ACCESSORY ||
+         device == AUDIO_DEVICE_OUT_USB_DEVICE)) {
+        return 1.0;
+    }
+
+    volume = volIndexToAmpl(device, streamDesc, index);
+
+    // if a headset is connected, apply the following rules to ring tones and notifications
+    // to avoid sound level bursts in user's ears:
+    // - always attenuate ring tones and notifications volume by 6dB
+    // - if music is playing, always limit the volume to current music volume,
+    // with a minimum threshold at -36dB so that notification is always perceived.
+    const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream);
+    if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+            AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+            AUDIO_DEVICE_OUT_WIRED_HEADSET |
+            AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
+        ((stream_strategy == STRATEGY_SONIFICATION)
+                || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
+                || (stream == AudioSystem::SYSTEM)
+                || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
+                    (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) &&
+        streamDesc.mCanBeMuted) {
+        volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+        // when the phone is ringing we must consider that music could have been paused just before
+        // by the music application and behave as if music was active if the last music track was
+        // just stopped
+        if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
+                mLimitRingtoneVolume) {
+            audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
+            float musicVol = computeVolume(AudioSystem::MUSIC,
+                               mStreams[AudioSystem::MUSIC].getVolumeIndex(musicDevice),
+                               output,
+                               musicDevice);
+            float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
+                                musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+            if (volume > minVol) {
+                volume = minVol;
+                ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+            }
+        }
+    }
+
+    return volume;
+}
+
+status_t AudioPolicyManagerBase::checkAndSetVolume(int stream,
+                                                   int index,
+                                                   audio_io_handle_t output,
+                                                   audio_devices_t device,
+                                                   int delayMs,
+                                                   bool force)
+{
+
+    // do not change actual stream volume if the stream is muted
+    if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+        ALOGVV("checkAndSetVolume() stream %d muted count %d",
+              stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+        return NO_ERROR;
+    }
+
+    // do not change in call volume if bluetooth is connected and vice versa
+    if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
+        (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
+        ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+             stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
+        return INVALID_OPERATION;
+    }
+
+    float volume = computeVolume(stream, index, output, device);
+    // We actually change the volume if:
+    // - the float value returned by computeVolume() changed
+    // - the force flag is set
+    if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
+            force) {
+        mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+        ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+        // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+        // enabled
+        if (stream == AudioSystem::BLUETOOTH_SCO) {
+            mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
+        }
+        mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
+    }
+
+    if (stream == AudioSystem::VOICE_CALL ||
+        stream == AudioSystem::BLUETOOTH_SCO) {
+        float voiceVolume;
+        // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+        if (stream == AudioSystem::VOICE_CALL) {
+            voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+        } else {
+            voiceVolume = 1.0;
+        }
+
+        if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+            mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+            mLastVoiceVolume = voiceVolume;
+        }
+    }
+
+    return NO_ERROR;
+}
+
+void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output,
+                                                audio_devices_t device,
+                                                int delayMs,
+                                                bool force)
+{
+    ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
+
+    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+        checkAndSetVolume(stream,
+                          mStreams[stream].getVolumeIndex(device),
+                          output,
+                          device,
+                          delayMs,
+                          force);
+    }
+}
+
+void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy,
+                                             bool on,
+                                             audio_io_handle_t output,
+                                             int delayMs,
+                                             audio_devices_t device)
+{
+    ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+        if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
+            setStreamMute(stream, on, output, delayMs, device);
+        }
+    }
+}
+
+void AudioPolicyManagerBase::setStreamMute(int stream,
+                                           bool on,
+                                           audio_io_handle_t output,
+                                           int delayMs,
+                                           audio_devices_t device)
+{
+    StreamDescriptor &streamDesc = mStreams[stream];
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    if (device == AUDIO_DEVICE_NONE) {
+        device = outputDesc->device();
+    }
+
+    ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
+          stream, on, output, outputDesc->mMuteCount[stream], device);
+
+    if (on) {
+        if (outputDesc->mMuteCount[stream] == 0) {
+            if (streamDesc.mCanBeMuted &&
+                    ((stream != AudioSystem::ENFORCED_AUDIBLE) ||
+                     (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) {
+                checkAndSetVolume(stream, 0, output, device, delayMs);
+            }
+        }
+        // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+        outputDesc->mMuteCount[stream]++;
+    } else {
+        if (outputDesc->mMuteCount[stream] == 0) {
+            ALOGV("setStreamMute() unmuting non muted stream!");
+            return;
+        }
+        if (--outputDesc->mMuteCount[stream] == 0) {
+            checkAndSetVolume(stream,
+                              streamDesc.getVolumeIndex(device),
+                              output,
+                              device,
+                              delayMs);
+        }
+    }
+}
+
+void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting, bool stateChange)
+{
+    // 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((AudioSystem::stream_type)stream);
+    if ((stream_strategy == STRATEGY_SONIFICATION) ||
+            ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+        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 (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
+                ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+                for (int i = 0; i < muteCount; i++) {
+                    setStreamMute(stream, starting, mPrimaryOutput);
+                }
+            } 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, mPrimaryOutput);
+                    }
+                }
+                if (starting) {
+                    mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
+                } else {
+                    mpClientInterface->stopTone();
+                }
+            }
+        }
+    }
+}
+
+bool AudioPolicyManagerBase::isInCall()
+{
+    return isStateInCall(mPhoneState);
+}
+
+bool AudioPolicyManagerBase::isStateInCall(int state) {
+    return ((state == AudioSystem::MODE_IN_CALL) ||
+            (state == AudioSystem::MODE_IN_COMMUNICATION));
+}
+
+uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
+{
+    return MAX_EFFECTS_CPU_LOAD;
+}
+
+uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
+{
+    return MAX_EFFECTS_MEMORY;
+}
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor(
+        const IOProfile *profile)
+    : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
+      mChannelMask(0), mLatency(0),
+    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE),
+    mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+    // clear usage count for all stream types
+    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        mRefCount[i] = 0;
+        mCurVolume[i] = -1.0;
+        mMuteCount[i] = 0;
+        mStopTime[i] = 0;
+    }
+    for (int i = 0; i < NUM_STRATEGIES; i++) {
+        mStrategyMutedByDevice[i] = false;
+    }
+    if (profile != NULL) {
+        mSamplingRate = profile->mSamplingRates[0];
+        mFormat = profile->mFormats[0];
+        mChannelMask = profile->mChannelMasks[0];
+        mFlags = profile->mFlags;
+    }
+}
+
+audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::device() const
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+    } else {
+        return mDevice;
+    }
+}
+
+uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::latency()
+{
+    if (isDuplicated()) {
+        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+    } else {
+        return mLatency;
+    }
+}
+
+bool AudioPolicyManagerBase::AudioOutputDescriptor::sharesHwModuleWith(
+        const AudioOutputDescriptor *outputDesc)
+{
+    if (isDuplicated()) {
+        return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+    } else if (outputDesc->isDuplicated()){
+        return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+    } else {
+        return (mProfile->mModule == outputDesc->mProfile->mModule);
+    }
+}
+
+void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
+{
+    // forward usage count change to attached outputs
+    if (isDuplicated()) {
+        mOutput1->changeRefCount(stream, delta);
+        mOutput2->changeRefCount(stream, delta);
+    }
+    if ((delta + (int)mRefCount[stream]) < 0) {
+        ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
+        mRefCount[stream] = 0;
+        return;
+    }
+    mRefCount[stream] += delta;
+    ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices()
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+    } else {
+        return mProfile->mSupportedDevices ;
+    }
+}
+
+bool AudioPolicyManagerBase::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+    return isStrategyActive(NUM_STRATEGIES, inPastMs);
+}
+
+bool AudioPolicyManagerBase::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
+                                                                       uint32_t inPastMs,
+                                                                       nsecs_t sysTime) const
+{
+    if ((sysTime == 0) && (inPastMs != 0)) {
+        sysTime = systemTime();
+    }
+    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        if (((getStrategy((AudioSystem::stream_type)i) == strategy) ||
+                (NUM_STRATEGIES == strategy)) &&
+                isStreamActive((AudioSystem::stream_type)i, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioPolicyManagerBase::AudioOutputDescriptor::isStreamActive(AudioSystem::stream_type stream,
+                                                                       uint32_t inPastMs,
+                                                                       nsecs_t sysTime) const
+{
+    if (mRefCount[stream] != 0) {
+        return true;
+    }
+    if (inPastMs == 0) {
+        return false;
+    }
+    if (sysTime == 0) {
+        sysTime = systemTime();
+    }
+    if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+        return true;
+    }
+    return false;
+}
+
+
+status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", device());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+    result.append(buffer);
+    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        snprintf(buffer, SIZE, " %02d     %.03f     %02d       %02d\n", i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+        result.append(buffer);
+    }
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- AudioInputDescriptor class implementation
+
+AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor(const IOProfile *profile)
+    : mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
+      mDevice(AUDIO_DEVICE_NONE), mRefCount(0),
+      mInputSource(0), mProfile(profile)
+{
+}
+
+status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- StreamDescriptor class implementation
+
+AudioPolicyManagerBase::StreamDescriptor::StreamDescriptor()
+    :   mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+    mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int AudioPolicyManagerBase::StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+    device = AudioPolicyManagerBase::getDeviceForVolume(device);
+    // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+    if (mIndexCur.indexOfKey(device) < 0) {
+        device = AUDIO_DEVICE_OUT_DEFAULT;
+    }
+    return mIndexCur.valueFor(device);
+}
+
+void AudioPolicyManagerBase::StreamDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%s         %02d         %02d         ",
+             mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+    result.append(buffer);
+    for (size_t i = 0; i < mIndexCur.size(); i++) {
+        snprintf(buffer, SIZE, "%04x : %02d, ",
+                 mIndexCur.keyAt(i),
+                 mIndexCur.valueAt(i));
+        result.append(buffer);
+    }
+    result.append("\n");
+
+    write(fd, result.string(), result.size());
+}
+
+// --- EffectDescriptor class implementation
+
+status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " I/O: %d\n", mIo);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Session: %d\n", mSession);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Name: %s\n",  mDesc.name);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " %s\n",  mEnabled ? "Enabled" : "Disabled");
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- IOProfile class implementation
+
+AudioPolicyManagerBase::HwModule::HwModule(const char *name)
+    : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0)
+{
+}
+
+AudioPolicyManagerBase::HwModule::~HwModule()
+{
+    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+         delete mOutputProfiles[i];
+    }
+    for (size_t i = 0; i < mInputProfiles.size(); i++) {
+         delete mInputProfiles[i];
+    }
+    free((void *)mName);
+}
+
+void AudioPolicyManagerBase::HwModule::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "  - name: %s\n", mName);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "  - handle: %d\n", mHandle);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    if (mOutputProfiles.size()) {
+        write(fd, "  - outputs:\n", strlen("  - outputs:\n"));
+        for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    output %d:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mOutputProfiles[i]->dump(fd);
+        }
+    }
+    if (mInputProfiles.size()) {
+        write(fd, "  - inputs:\n", strlen("  - inputs:\n"));
+        for (size_t i = 0; i < mInputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    input %d:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mInputProfiles[i]->dump(fd);
+        }
+    }
+}
+
+AudioPolicyManagerBase::IOProfile::IOProfile(HwModule *module)
+    : mFlags((audio_output_flags_t)0), mModule(module)
+{
+}
+
+AudioPolicyManagerBase::IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool AudioPolicyManagerBase::IOProfile::isCompatibleProfile(audio_devices_t device,
+                                                            uint32_t samplingRate,
+                                                            audio_format_t format,
+                                                            audio_channel_mask_t channelMask,
+                                                            audio_output_flags_t flags) const
+{
+    if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) {
+         return false;
+     }
+
+     if ((mSupportedDevices & device) != device) {
+         return false;
+     }
+     if ((mFlags & flags) != flags) {
+         return false;
+     }
+     size_t i;
+     for (i = 0; i < mSamplingRates.size(); i++)
+     {
+         if (mSamplingRates[i] == samplingRate) {
+             break;
+         }
+     }
+     if (i == mSamplingRates.size()) {
+         return false;
+     }
+     for (i = 0; i < mFormats.size(); i++)
+     {
+         if (mFormats[i] == format) {
+             break;
+         }
+     }
+     if (i == mFormats.size()) {
+         return false;
+     }
+     for (i = 0; i < mChannelMasks.size(); i++)
+     {
+         if (mChannelMasks[i] == channelMask) {
+             break;
+         }
+     }
+     if (i == mChannelMasks.size()) {
+         return false;
+     }
+     return true;
+}
+
+void AudioPolicyManagerBase::IOProfile::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "    - sampling rates: ");
+    result.append(buffer);
+    for (size_t i = 0; i < mSamplingRates.size(); i++) {
+        snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+        result.append(buffer);
+        result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", ");
+    }
+
+    snprintf(buffer, SIZE, "    - channel masks: ");
+    result.append(buffer);
+    for (size_t i = 0; i < mChannelMasks.size(); i++) {
+        snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+        result.append(buffer);
+        result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", ");
+    }
+
+    snprintf(buffer, SIZE, "    - formats: ");
+    result.append(buffer);
+    for (size_t i = 0; i < mFormats.size(); i++) {
+        snprintf(buffer, SIZE, "0x%08x", mFormats[i]);
+        result.append(buffer);
+        result.append(i == (mFormats.size() - 1) ? "\n" : ", ");
+    }
+
+    snprintf(buffer, SIZE, "    - devices: 0x%04x\n", mSupportedDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "    - flags: 0x%04x\n", mFlags);
+    result.append(buffer);
+
+    write(fd, result.string(), result.size());
+}
+
+// --- audio_policy.conf file parsing
+
+struct StringToEnum {
+    const char *name;
+    uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+const struct StringToEnum sDeviceNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+};
+
+const struct StringToEnum sFlagNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+};
+
+const struct StringToEnum sFormatNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+    STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+    STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+};
+
+const struct StringToEnum sOutChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const struct StringToEnum sInChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+
+uint32_t AudioPolicyManagerBase::stringToEnum(const struct StringToEnum *table,
+                                              size_t size,
+                                              const char *name)
+{
+    for (size_t i = 0; i < size; i++) {
+        if (strcmp(table[i].name, name) == 0) {
+            ALOGV("stringToEnum() found %s", table[i].name);
+            return table[i].value;
+        }
+    }
+    return 0;
+}
+
+bool AudioPolicyManagerBase::stringToBool(const char *value)
+{
+    return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+audio_output_flags_t AudioPolicyManagerBase::parseFlagNames(char *name)
+{
+    uint32_t flag = 0;
+
+    // it is OK to cast name to non const here as we are not going to use it after
+    // strtok() modifies it
+    char *flagName = strtok(name, "|");
+    while (flagName != NULL) {
+        if (strlen(flagName) != 0) {
+            flag |= stringToEnum(sFlagNameToEnumTable,
+                               ARRAY_SIZE(sFlagNameToEnumTable),
+                               flagName);
+        }
+        flagName = strtok(NULL, "|");
+    }
+    //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 ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+    }
+
+    return (audio_output_flags_t)flag;
+}
+
+audio_devices_t AudioPolicyManagerBase::parseDeviceNames(char *name)
+{
+    uint32_t device = 0;
+
+    char *devName = strtok(name, "|");
+    while (devName != NULL) {
+        if (strlen(devName) != 0) {
+            device |= stringToEnum(sDeviceNameToEnumTable,
+                                 ARRAY_SIZE(sDeviceNameToEnumTable),
+                                 devName);
+        }
+        devName = strtok(NULL, "|");
+    }
+    return device;
+}
+
+void AudioPolicyManagerBase::loadSamplingRates(char *name, IOProfile *profile)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+    // rates should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mSamplingRates.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        uint32_t rate = atoi(str);
+        if (rate != 0) {
+            ALOGV("loadSamplingRates() adding rate %d", rate);
+            profile->mSamplingRates.add(rate);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManagerBase::loadFormats(char *name, IOProfile *profile)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mFormats indicates the supported formats
+    // should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
+                                                             ARRAY_SIZE(sFormatNameToEnumTable),
+                                                             str);
+        if (format != AUDIO_FORMAT_DEFAULT) {
+            profile->mFormats.add(format);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManagerBase::loadInChannels(char *name, IOProfile *profile)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadInChannels() %s", name);
+
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sInChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+            profile->mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManagerBase::loadOutChannels(char *name, IOProfile *profile)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadOutChannels() %s", name);
+
+    // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+    // masks should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        profile->mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sOutChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            profile->mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+status_t AudioPolicyManagerBase::loadInput(cnode *root, HwModule *module)
+{
+    cnode *node = root->first_child;
+
+    IOProfile *profile = new IOProfile(module);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            loadSamplingRates((char *)node->value, profile);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            loadFormats((char *)node->value, profile);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            loadInChannels((char *)node->value, profile);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices = parseDeviceNames((char *)node->value);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
+            "loadInput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadInput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadInput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadInput() invalid supported formats");
+    if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadInput() adding input mSupportedDevices %04x", profile->mSupportedDevices);
+
+        module->mInputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        delete profile;
+        return BAD_VALUE;
+    }
+}
+
+status_t AudioPolicyManagerBase::loadOutput(cnode *root, HwModule *module)
+{
+    cnode *node = root->first_child;
+
+    IOProfile *profile = new IOProfile(module);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            loadSamplingRates((char *)node->value, profile);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            loadFormats((char *)node->value, profile);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            loadOutChannels((char *)node->value, profile);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices = parseDeviceNames((char *)node->value);
+        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+            profile->mFlags = parseFlagNames((char *)node->value);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
+            "loadOutput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadOutput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadOutput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadOutput() invalid supported formats");
+    if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x",
+              profile->mSupportedDevices, profile->mFlags);
+
+        module->mOutputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        delete profile;
+        return BAD_VALUE;
+    }
+}
+
+void AudioPolicyManagerBase::loadHwModule(cnode *root)
+{
+    cnode *node = config_find(root, OUTPUTS_TAG);
+    status_t status = NAME_NOT_FOUND;
+
+    HwModule *module = new HwModule(root->name);
+
+    if (node != NULL) {
+        if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
+            mHasA2dp = true;
+        } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
+            mHasUsb = true;
+        } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
+            mHasRemoteSubmix = true;
+        }
+
+        node = node->first_child;
+        while (node) {
+            ALOGV("loadHwModule() loading output %s", node->name);
+            status_t tmpStatus = loadOutput(node, module);
+            if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+                status = tmpStatus;
+            }
+            node = node->next;
+        }
+    }
+    node = config_find(root, INPUTS_TAG);
+    if (node != NULL) {
+        node = node->first_child;
+        while (node) {
+            ALOGV("loadHwModule() loading input %s", node->name);
+            status_t tmpStatus = loadInput(node, module);
+            if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+                status = tmpStatus;
+            }
+            node = node->next;
+        }
+    }
+    if (status == NO_ERROR) {
+        mHwModules.add(module);
+    } else {
+        delete module;
+    }
+}
+
+void AudioPolicyManagerBase::loadHwModules(cnode *root)
+{
+    cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
+    if (node == NULL) {
+        return;
+    }
+
+    node = node->first_child;
+    while (node) {
+        ALOGV("loadHwModules() loading module %s", node->name);
+        loadHwModule(node);
+        node = node->next;
+    }
+}
+
+void AudioPolicyManagerBase::loadGlobalConfig(cnode *root)
+{
+    cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
+    if (node == NULL) {
+        return;
+    }
+    node = node->first_child;
+    while (node) {
+        if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
+            mAttachedOutputDevices = parseDeviceNames((char *)node->value);
+            ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE,
+                    "loadGlobalConfig() no attached output devices");
+            ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices);
+        } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
+            mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
+                                              ARRAY_SIZE(sDeviceNameToEnumTable),
+                                              (char *)node->value);
+            ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE,
+                    "loadGlobalConfig() default device not specified");
+            ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice);
+        } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
+            mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN;
+            ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices);
+        } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
+            mSpeakerDrcEnabled = stringToBool((char *)node->value);
+            ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
+        }
+        node = node->next;
+    }
+}
+
+status_t AudioPolicyManagerBase::loadAudioPolicyConfig(const char *path)
+{
+    cnode *root;
+    char *data;
+
+    data = (char *)load_file(path, NULL);
+    if (data == NULL) {
+        return -ENODEV;
+    }
+    root = config_node("", "");
+    config_load(root, data);
+
+    loadGlobalConfig(root);
+    loadHwModules(root);
+
+    config_free(root);
+    free(root);
+    free(data);
+
+    ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
+
+    return NO_ERROR;
+}
+
+void AudioPolicyManagerBase::defaultAudioPolicyConfig(void)
+{
+    HwModule *module;
+    IOProfile *profile;
+
+    mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER;
+    mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER;
+    mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN;
+
+    module = new HwModule("primary");
+
+    profile = new IOProfile(module);
+    profile->mSamplingRates.add(44100);
+    profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+    profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
+    profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER;
+    profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
+    module->mOutputProfiles.add(profile);
+
+    profile = new IOProfile(module);
+    profile->mSamplingRates.add(8000);
+    profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+    profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
+    profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC;
+    module->mInputProfiles.add(profile);
+
+    mHwModules.add(module);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManagerBase.h b/services/audiopolicy/AudioPolicyManagerBase.h
new file mode 100644
index 0000000..1ff409e
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyManagerBase.h
@@ -0,0 +1,587 @@
+/*
+ * 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 <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+
+
+namespace android_audio_legacy {
+    using android::KeyedVector;
+    using android::DefaultKeyedVector;
+    using android::SortedVector;
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN  0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY  5000
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerBase implements audio policy manager behavior common to all platforms.
+// Each platform must implement an AudioPolicyManager class derived from AudioPolicyManagerBase
+// and override methods for which the platform specific behavior differs from the implementation
+// in AudioPolicyManagerBase. Even if no specific behavior is required, the AudioPolicyManager
+// class must be implemented as well as the class factory function createAudioPolicyManager()
+// and provided in a shared library libaudiopolicy.so.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManagerBase: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+    , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+                AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface);
+        virtual ~AudioPolicyManagerBase();
+
+        // AudioPolicyInterface
+        virtual status_t setDeviceConnectionState(audio_devices_t device,
+                                                          AudioSystem::device_connection_state state,
+                                                          const char *device_address);
+        virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device,
+                                                                              const char *device_address);
+        virtual void setPhoneState(int state);
+        virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+        virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
+        virtual void setSystemProperty(const char* property, const char* value);
+        virtual status_t initCheck();
+        virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            AudioSystem::output_flags flags,
+                                            const audio_offload_info_t *offloadInfo);
+        virtual status_t startOutput(audio_io_handle_t output,
+                                     AudioSystem::stream_type stream,
+                                     int session = 0);
+        virtual status_t stopOutput(audio_io_handle_t output,
+                                    AudioSystem::stream_type stream,
+                                    int session = 0);
+        virtual void releaseOutput(audio_io_handle_t output);
+        virtual audio_io_handle_t getInput(int inputSource,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            AudioSystem::audio_in_acoustics acoustics);
+
+        // indicates to the audio policy manager that the input starts being used.
+        virtual status_t startInput(audio_io_handle_t input);
+
+        // indicates to the audio policy manager that the input stops being used.
+        virtual status_t stopInput(audio_io_handle_t input);
+        virtual void releaseInput(audio_io_handle_t input);
+        virtual void initStreamVolume(AudioSystem::stream_type stream,
+                                                    int indexMin,
+                                                    int indexMax);
+        virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream,
+                                              int index,
+                                              audio_devices_t device);
+        virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream,
+                                              int *index,
+                                              audio_devices_t device);
+
+        // return the strategy corresponding to a given stream type
+        virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream);
+
+        // return the enabled output devices for the given stream type
+        virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream);
+
+        virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+        virtual status_t registerEffect(const effect_descriptor_t *desc,
+                                        audio_io_handle_t io,
+                                        uint32_t strategy,
+                                        int session,
+                                        int id);
+        virtual status_t unregisterEffect(int id);
+        virtual status_t setEffectEnabled(int id, bool enabled);
+
+        virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const;
+        // return whether a stream is playing remotely, override to change the definition of
+        //   local/remote playback, used for instance by notification manager to not make
+        //   media players lose audio focus when not playing locally
+        virtual bool isStreamActiveRemotely(int stream, uint32_t inPastMs = 0) const;
+        virtual bool isSourceActive(audio_source_t source) const;
+
+        virtual status_t dump(int fd);
+
+        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+protected:
+
+        enum routing_strategy {
+            STRATEGY_MEDIA,
+            STRATEGY_PHONE,
+            STRATEGY_SONIFICATION,
+            STRATEGY_SONIFICATION_RESPECTFUL,
+            STRATEGY_DTMF,
+            STRATEGY_ENFORCED_AUDIBLE,
+            NUM_STRATEGIES
+        };
+
+        // 4 points to define the volume attenuation curve, each characterized by the volume
+        // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+        // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+
+        enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
+
+        class VolumeCurvePoint
+        {
+        public:
+            int mIndex;
+            float mDBAttenuation;
+        };
+
+        // device categories used for volume curve management.
+        enum device_category {
+            DEVICE_CATEGORY_HEADSET,
+            DEVICE_CATEGORY_SPEAKER,
+            DEVICE_CATEGORY_EARPIECE,
+            DEVICE_CATEGORY_CNT
+        };
+
+        class IOProfile;
+
+        class HwModule {
+        public:
+                    HwModule(const char *name);
+                    ~HwModule();
+
+            void dump(int fd);
+
+            const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+            audio_module_handle_t mHandle;
+            Vector <IOProfile *> mOutputProfiles; // output profiles exposed by this module
+            Vector <IOProfile *> mInputProfiles;  // input profiles exposed by this module
+        };
+
+        // the IOProfile class describes the capabilities of an output or input stream.
+        // It is currently assumed that all combination of listed parameters are supported.
+        // It is used by the policy manager to determine if an output or input is suitable for
+        // a given use case,  open/close it accordingly and connect/disconnect audio tracks
+        // to/from it.
+        class IOProfile
+        {
+        public:
+            IOProfile(HwModule *module);
+            ~IOProfile();
+
+            bool isCompatibleProfile(audio_devices_t device,
+                                     uint32_t samplingRate,
+                                     audio_format_t format,
+                                     audio_channel_mask_t channelMask,
+                                     audio_output_flags_t flags) const;
+
+            void dump(int fd);
+
+            // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+            // indicates the supported parameters should be read from the output stream
+            // after it is opened for the first time
+            Vector <uint32_t> mSamplingRates; // supported sampling rates
+            Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+            Vector <audio_format_t> mFormats; // supported audio formats
+            audio_devices_t mSupportedDevices; // supported devices (devices this output can be
+                                               // routed to)
+            audio_output_flags_t mFlags; // attribute flags (e.g primary output,
+                                                // direct output...). For outputs only.
+            HwModule *mModule;                     // audio HW module exposing this I/O stream
+        };
+
+        // default volume curve
+        static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // default volume curve for media strategy
+        static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // volume curve for media strategy on speakers
+        static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // volume curve for sonification strategy on speakers
+        static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // default volume curves per stream and device category. See initializeVolumeCurves()
+        static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
+
+        // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+        // and keep track of the usage of this output by each audio stream type.
+        class AudioOutputDescriptor
+        {
+        public:
+            AudioOutputDescriptor(const IOProfile *profile);
+
+            status_t    dump(int fd);
+
+            audio_devices_t device() const;
+            void changeRefCount(AudioSystem::stream_type stream, int delta);
+
+            bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+            audio_devices_t supportedDevices();
+            uint32_t latency();
+            bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc);
+            bool isActive(uint32_t inPastMs = 0) const;
+            bool isStreamActive(AudioSystem::stream_type stream,
+                                uint32_t inPastMs = 0,
+                                nsecs_t sysTime = 0) const;
+            bool isStrategyActive(routing_strategy strategy,
+                             uint32_t inPastMs = 0,
+                             nsecs_t sysTime = 0) const;
+
+            audio_io_handle_t mId;              // output handle
+            uint32_t mSamplingRate;             //
+            audio_format_t mFormat;             //
+            audio_channel_mask_t mChannelMask;     // output configuration
+            uint32_t mLatency;                  //
+            audio_output_flags_t mFlags;   //
+            audio_devices_t mDevice;                   // current device this output is routed to
+            uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
+            nsecs_t mStopTime[AudioSystem::NUM_STREAM_TYPES];
+            AudioOutputDescriptor *mOutput1;    // used by duplicated outputs: first output
+            AudioOutputDescriptor *mOutput2;    // used by duplicated outputs: second output
+            float mCurVolume[AudioSystem::NUM_STREAM_TYPES];   // current stream volume
+            int mMuteCount[AudioSystem::NUM_STREAM_TYPES];     // mute request counter
+            const IOProfile *mProfile;          // I/O profile this output derives from
+            bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+                                                // device selection. See checkDeviceMuteStrategies()
+            uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+        };
+
+        // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+        // and keep track of the usage of this input.
+        class AudioInputDescriptor
+        {
+        public:
+            AudioInputDescriptor(const IOProfile *profile);
+
+            status_t    dump(int fd);
+
+            uint32_t mSamplingRate;                     //
+            audio_format_t mFormat;                     // input configuration
+            audio_channel_mask_t mChannelMask;             //
+            audio_devices_t mDevice;                    // current device this input is routed to
+            uint32_t mRefCount;                         // number of AudioRecord clients using this output
+            int      mInputSource;                      // input source selected by application (mediarecorder.h)
+            const IOProfile *mProfile;                  // I/O profile this output derives from
+        };
+
+        // stream descriptor used for volume control
+        class StreamDescriptor
+        {
+        public:
+            StreamDescriptor();
+
+            int getVolumeIndex(audio_devices_t device);
+            void dump(int fd);
+
+            int mIndexMin;      // min volume index
+            int mIndexMax;      // max volume index
+            KeyedVector<audio_devices_t, int> mIndexCur;   // current volume index per device
+            bool mCanBeMuted;   // true is the stream can be muted
+
+            const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
+        };
+
+        // stream descriptor used for volume control
+        class EffectDescriptor
+        {
+        public:
+
+            status_t dump(int fd);
+
+            int mIo;                // io the effect is attached to
+            routing_strategy mStrategy; // routing strategy the effect is associated to
+            int mSession;               // audio session the effect is on
+            effect_descriptor_t mDesc;  // effect descriptor
+            bool mEnabled;              // enabled state: CPU load being used or not
+        };
+
+        void addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc);
+
+        // return the strategy corresponding to a given stream type
+        static routing_strategy getStrategy(AudioSystem::stream_type stream);
+
+        // return appropriate device for streams handled by the specified strategy according to current
+        // phone state, connected devices...
+        // if fromCache is true, the device is returned from mDeviceForStrategy[],
+        // otherwise it is determine by current state
+        // (device connected,phone state, force use, a2dp output...)
+        // This allows to:
+        //  1 speed up process when the state is stable (when starting or stopping an output)
+        //  2 access to either current device selection (fromCache == true) or
+        // "future" device selection (fromCache == false) when called from a context
+        //  where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+        //  before updateDevicesAndOutputs() is called.
+        virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+                                                     bool fromCache);
+
+        // change the route of the specified output. Returns the number of ms we have slept to
+        // allow new routing to take effect in certain cases.
+        uint32_t setOutputDevice(audio_io_handle_t output,
+                             audio_devices_t device,
+                             bool force = false,
+                             int delayMs = 0);
+
+        // select input device corresponding to requested audio source
+        virtual audio_devices_t getDeviceForInputSource(int inputSource);
+
+        // return io handle of active input or 0 if no input is active
+        //    Only considers inputs from physical devices (e.g. main mic, headset mic) when
+        //    ignoreVirtualInputs is true.
+        audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+        // initialize volume curves for each strategy and device category
+        void initializeVolumeCurves();
+
+        // compute the actual volume for a given stream according to the requested index and a particular
+        // device
+        virtual float computeVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device);
+
+        // check that volume change is permitted, compute and send new volume to audio hardware
+        status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+        // apply all stream volumes to the specified output and device
+        void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+        // Mute or unmute all streams handled by the specified strategy on the specified output
+        void setStrategyMute(routing_strategy strategy,
+                             bool on,
+                             audio_io_handle_t output,
+                             int delayMs = 0,
+                             audio_devices_t device = (audio_devices_t)0);
+
+        // Mute or unmute the stream on the specified output
+        void setStreamMute(int stream,
+                           bool on,
+                           audio_io_handle_t output,
+                           int delayMs = 0,
+                           audio_devices_t device = (audio_devices_t)0);
+
+        // 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(int stream, bool starting, bool stateChange);
+
+        // true if device is in a telephony or VoIP call
+        virtual bool isInCall();
+
+        // true if given state represents a device in a telephony or VoIP call
+        virtual bool isStateInCall(int state);
+
+        // when a device is connected, checks if an open output can be routed
+        // to this device. If none is open, tries to open one of the available outputs.
+        // Returns an output suitable to this device or 0.
+        // when a device is disconnected, checks if an output is not used any more and
+        // returns its handle if any.
+        // transfers the audio tracks and effects from one output thread to another accordingly.
+        status_t checkOutputsForDevice(audio_devices_t device,
+                                       AudioSystem::device_connection_state state,
+                                       SortedVector<audio_io_handle_t>& outputs,
+                                       const String8 paramStr);
+
+        // close an output and its companion duplicating output.
+        void closeOutput(audio_io_handle_t output);
+
+        // checks and if necessary changes outputs used for all strategies.
+        // must be called every time a condition that affects the output choice for a given strategy
+        // changes: connected device, phone state, force use...
+        // Must be called before updateDevicesAndOutputs()
+        void checkOutputForStrategy(routing_strategy strategy);
+
+        // Same as checkOutputForStrategy() but for a all strategies in order of priority
+        void checkOutputForAllStrategies();
+
+        // manages A2DP output suspend/restore according to phone state and BT SCO usage
+        void checkA2dpSuspend();
+
+        // returns the A2DP output handle if it is open or 0 otherwise
+        audio_io_handle_t getA2dpOutput();
+
+        // selects the most appropriate device on output for current state
+        // must be called every time a condition that affects the device choice for a given output is
+        // changed: connected device, phone state, force use, output start, output stop..
+        // see getDeviceForStrategy() for the use of fromCache parameter
+
+        audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache);
+        // updates cache of device used by all strategies (mDeviceForStrategy[])
+        // must be called every time a condition that affects the device choice for a given strategy is
+        // changed: connected device, phone state, force use...
+        // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+         // Must be called after checkOutputForAllStrategies()
+
+        void updateDevicesAndOutputs();
+
+        virtual uint32_t getMaxEffectsCpuLoad();
+        virtual uint32_t getMaxEffectsMemory();
+#ifdef AUDIO_POLICY_TEST
+        virtual     bool        threadLoop();
+                    void        exit();
+        int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+        status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled);
+
+        // returns the category the device belongs to with regard to volume curve management
+        static device_category getDeviceCategory(audio_devices_t device);
+
+        // extract one device relevant for volume control from multiple device selection
+        static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+        SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+                        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs);
+        bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+                                           SortedVector<audio_io_handle_t>& outputs2);
+
+        // mute/unmute strategies using an incompatible device combination
+        // if muting, wait for the audio in pcm buffer to be drained before proceeding
+        // if unmuting, unmute only after the specified delay
+        // Returns the number of ms waited
+        uint32_t  checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+                                            audio_devices_t prevDevice,
+                                            uint32_t delayMs);
+
+        audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+                                       AudioSystem::output_flags flags);
+        IOProfile *getInputProfile(audio_devices_t device,
+                                   uint32_t samplingRate,
+                                   audio_format_t format,
+                                   audio_channel_mask_t channelMask);
+        IOProfile *getProfileForDirectOutput(audio_devices_t device,
+                                                       uint32_t samplingRate,
+                                                       audio_format_t format,
+                                                       audio_channel_mask_t channelMask,
+                                                       audio_output_flags_t flags);
+
+        audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+        bool isNonOffloadableEffectEnabled();
+
+        //
+        // Audio policy configuration file parsing (audio_policy.conf)
+        //
+        static uint32_t stringToEnum(const struct StringToEnum *table,
+                                     size_t size,
+                                     const char *name);
+        static bool stringToBool(const char *value);
+        static audio_output_flags_t parseFlagNames(char *name);
+        static audio_devices_t parseDeviceNames(char *name);
+        void loadSamplingRates(char *name, IOProfile *profile);
+        void loadFormats(char *name, IOProfile *profile);
+        void loadOutChannels(char *name, IOProfile *profile);
+        void loadInChannels(char *name, IOProfile *profile);
+        status_t loadOutput(cnode *root,  HwModule *module);
+        status_t loadInput(cnode *root,  HwModule *module);
+        void loadHwModule(cnode *root);
+        void loadHwModules(cnode *root);
+        void loadGlobalConfig(cnode *root);
+        status_t loadAudioPolicyConfig(const char *path);
+        void defaultAudioPolicyConfig(void);
+
+
+        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
+        audio_io_handle_t mPrimaryOutput;              // primary output handle
+        // list of descriptors for outputs currently opened
+        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;
+        // copy of mOutputs before setDeviceConnectionState() opens new outputs
+        // reset to mOutputs when updateDevicesAndOutputs() is called.
+        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
+        DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs;     // list of input descriptors
+        audio_devices_t mAvailableOutputDevices; // bit field of all available output devices
+        audio_devices_t mAvailableInputDevices; // bit field of all available input devices
+                                                // without AUDIO_DEVICE_BIT_IN to allow direct bit
+                                                // field comparisons
+        int mPhoneState;                                                    // current phone state
+        AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE];   // current forced use configuration
+
+        StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES];           // stream descriptors for volume control
+        String8 mA2dpDeviceAddress;                                         // A2DP device MAC address
+        String8 mScoDeviceAddress;                                          // SCO device MAC address
+        String8 mUsbCardAndDevice; // USB audio ALSA card and device numbers:
+                                   // card=<card_number>;device=<><device_number>
+        bool    mLimitRingtoneVolume;                                       // limit ringtone volume to music volume if headset connected
+        audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+        float   mLastVoiceVolume;                                           // last voice volume value sent to audio HAL
+
+        // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+        static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+        // Maximum memory allocated to audio effects in KB
+        static const uint32_t MAX_EFFECTS_MEMORY = 512;
+        uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+        uint32_t mTotalEffectsMemory;  // current memory used by effects
+        KeyedVector<int, EffectDescriptor *> mEffects;  // list of registered audio effects
+        bool    mA2dpSuspended;  // true if A2DP output is suspended
+        bool mHasA2dp; // true on platforms with support for bluetooth A2DP
+        bool mHasUsb; // true on platforms with support for USB audio
+        bool mHasRemoteSubmix; // true on platforms with support for remote presentation of a submix
+        audio_devices_t mAttachedOutputDevices; // output devices always available on the platform
+        audio_devices_t mDefaultOutputDevice; // output device selected by default at boot time
+                                              // (must be in mAttachedOutputDevices)
+        bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+                                // to boost soft sounds, used to adjust volume curves accordingly
+
+        Vector <HwModule *> mHwModules;
+
+#ifdef AUDIO_POLICY_TEST
+        Mutex   mLock;
+        Condition mWaitWorkCV;
+
+        int             mCurOutput;
+        bool            mDirectOutput;
+        audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+        int             mTestInput;
+        uint32_t        mTestDevice;
+        uint32_t        mTestSamplingRate;
+        uint32_t        mTestFormat;
+        uint32_t        mTestChannels;
+        uint32_t        mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+private:
+        static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+                int indexInUi);
+        // updates device caching and output for streams that can influence the
+        //    routing of notifications
+        void handleNotificationRoutingForStream(AudioSystem::stream_type stream);
+        static bool isVirtualInputDevice(audio_devices_t device);
+};
+
+};