audio/AudioPolicyManagerBase.cpp

/*
 * Copyright (C) 2009 The Android Open Source Project
 * Copyright (c) 2013, The Linux Foundation. All rights reserved.
 * Not a Contribution.
 *
 * 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;
        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 (checkOutputsForDevice(device, state, outputs) != 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 (!outputs.isEmpty()) {
                String8 paramStr;
                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();
                    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);
                    paramStr = mUsbCardAndDevice;
                }
                // not currently handling multiple simultaneous submixes: ignoring remote submix
                //   case and address
                if (!paramStr.isEmpty()) {
                    for (size_t i = 0; i < outputs.size(); i++) {
                        mpClientInterface->setParameters(outputs[i], paramStr);
                    }
                }
            }
            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);
            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,
                                                               uint32_t format,
                                                               uint32_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_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT | flags))) {
                    if (mAvailableOutputDevices & profile->mSupportedDevices) {
                        return mHwModules[i]->mOutputProfiles[j];
                    }
                }
            }
        }
    }
    return 0;
}

audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
                                    uint32_t samplingRate,
                                    uint32_t format,
                                    uint32_t channelMask,
                                    AudioSystem::output_flags flags,
                                    const audio_offload_info_t *offloadInfo)
{
    audio_io_handle_t output = 0;
    uint32_t latency = 0;
    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
    IOProfile *profile = NULL;
    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);
    }

#ifdef QCOM_HARDWARE
    if ((format == AudioSystem::PCM_16_BIT) &&(AudioSystem::popCount(channelMask) > 2)) {
        ALOGV("owerwrite flag(%x) for PCM16 multi-channel(CM:%x) playback", flags ,channelMask);
        flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_DIRECT;
    }
#endif

    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
    // creating an offloaded track and tearing it down immediately after start when audioflinger
    // detects there is an active non offloadable effect.
    // FIXME: We should check the audio session here but we do not have it in this context.
    // This may prevent offloading in rare situations where effects are left active by apps
    // in the background.
    if ((((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
            !isNonOffloadableEffectEnabled()) &&
            flags & AUDIO_OUTPUT_FLAG_DIRECT) {
        profile = getProfileForDirectOutput(device,
                                           samplingRate,
                                           format,
                                           channelMask,
                                           (audio_output_flags_t)flags);
    }

    if (profile != NULL) {
        AudioOutputDescriptor *outputDesc = NULL;

        for (size_t i = 0; i < mOutputs.size(); i++) {
            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
            if (!desc->isDuplicated() && (profile == desc->mProfile)) {
                outputDesc = desc;
                // reuse direct output if currently open and configured with same parameters
                if ((samplingRate == outputDesc->mSamplingRate) &&
                        (format == outputDesc->mFormat) &&
                        (channelMask == outputDesc->mChannelMask)) {
                    outputDesc->mDirectOpenCount++;
                    ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
                    return mOutputs.keyAt(i);
                }
            }
        }
        // close direct output if currently open and configured with different parameters
        if (outputDesc != NULL) {
            closeOutput(outputDesc->mId);
        }
        outputDesc = new AudioOutputDescriptor(profile);
        outputDesc->mDevice = device;
        outputDesc->mSamplingRate = samplingRate;
        outputDesc->mFormat = (audio_format_t)format;
        outputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
        outputDesc->mLatency = 0;
        outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
        outputDesc->mRefCount[stream] = 0;
        outputDesc->mStopTime[stream] = 0;
        outputDesc->mDirectOpenCount = 1;
        output = mpClientInterface->openOutput(profile->mModule->mHandle,
                                        &outputDesc->mDevice,
                                        &outputDesc->mSamplingRate,
                                        &outputDesc->mFormat,
                                        &outputDesc->mChannelMask,
                                        &outputDesc->mLatency,
                                        outputDesc->mFlags,
                                        offloadInfo);

        // only accept an output with the requested parameters
        if (output == 0 ||
            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
            (format != 0 && format != outputDesc->mFormat) ||
            (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
            ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
                    "format %d %d, channelMask %04x %04x", output, samplingRate,
                    outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
                    outputDesc->mChannelMask);
            if (output != 0) {
                mpClientInterface->closeOutput(output);
            }
            delete outputDesc;
            return 0;
        }
        audio_io_handle_t srcOutput = getOutputForEffect();
        addOutput(output, outputDesc);
        audio_io_handle_t dstOutput = getOutputForEffect();
        if (dstOutput == output) {
            mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
        }
        mPreviousOutputs = mOutputs;
        ALOGV("getOutput() returns new direct output %d", output);
        return output;
    }

    // ignoring channel mask due to downmix capability in mixer

    // open a non direct output

    // for non direct outputs, only PCM is supported
    if (audio_is_linear_pcm((audio_format_t)format)) {
        // get which output is suitable for the specified stream. The actual
        // routing change will happen when startOutput() will be called
        SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);

        output = selectOutput(outputs, flags);
    }
    ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
            "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);

    ALOGV("getOutput() returns output %d", output);

    return output;
}

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() unknow 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() unknow 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,
                                    uint32_t format,
                                    uint32_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 = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
        break;
    case AUDIO_SOURCE_VOICE_DOWNLINK:
        channelMask = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
        break;
    case AUDIO_SOURCE_VOICE_CALL:
        channelMask = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::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 = (audio_format_t)format;
    inputDesc->mChannelMask = (audio_channel_mask_t)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)) {
        ALOGV("getInput() failed opening input: samplingRate %d, format %d, channelMask %d",
                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() unknow 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() unknow 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());
#ifndef ICS_AUDIO_BLOB
        if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice))
#endif
        {
            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;
    }
#ifndef ICS_AUDIO_BLOB
    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);
#else
    *index =  mStreams[stream].mIndexCur.valueAt(0);
#endif
    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)
    {
        if(property_get("av.offload.enable", propValue, "false")) {
            bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
            if (!prop_enabled) {
               ALOGW("offload disabled by av.offload.enable = %s ", propValue );
               return false;
            }
        }
        if(offloadInfo.is_streaming &&
           property_get("av.streaming.offload.enable", propValue, "false")) {
            bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
            if (!prop_enabled) {
               ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
               return false;
            }
        }
        ALOGV("isOffloadSupported: has_video == true, property\
                set to enable offload");
    }

    //If duration is less than minimum value defined in property, return false
    if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
        if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
            ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
            return false;
        }
    } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
        ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
        //duration checks only valid for MP3/AAC formats,
        //do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
        if (offloadInfo.format == AUDIO_FORMAT_MP3 || offloadInfo.format == AUDIO_FORMAT_AAC)
            return false;
    }

    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
    // creating an offloaded track and tearing it down immediately after start when audioflinger
    // detects there is an active non offloadable effect.
    // FIXME: We should check the audio session here but we do not have it in this context.
    // This may prevent offloading in rare situations where effects are left active by apps
    // in the background.
    if (isNonOffloadableEffectEnabled()) {
        return false;
    }

    // See if there is a profile to support this.
    // AUDIO_DEVICE_NONE
    IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
                                            offloadInfo.sample_rate,
                                            offloadInfo.format,
                                            offloadInfo.channel_mask,
                                            AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
    ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
    return (profile != NULL);
}

// ----------------------------------------------------------------------------
// 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);
   }