xref: /hardware/libhardware_legacy/audio/AudioPolicyManagerBase.cpp

/*
 * 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

#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>

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++) {
                // 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) ||
                        (mOutputs.valueFor(outputs[i])->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
                    closeOutput(outputs[i]);
                }
            }
        }

        updateDevicesAndOutputs();
        for (size_t i = 0; i < mOutputs.size(); i++) {
            setOutputDevice(mOutputs.keyAt(i),
                            getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
                            true,
                            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;
    } 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;
    } 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();
    }

    // when changing from ring tone to in call mode, mute the ringing tone
    // immediately and delay the route change to avoid sending the ring tone
    // tail into the earpiece or headset.
    int delayMs = 0;
    if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) {
        // delay the device change command by twice the output latency to have some margin
        // and be sure that audio buffers not yet affected by the mute are out when
        // we actually apply the route change
        delayMs = hwOutputDesc->mLatency*2;
        setStreamMute(AudioSystem::RING, true, mPrimaryOutput);
    }

    if (isStateInCall(state)) {
        for (size_t i = 0; i < mOutputs.size(); i++) {
            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
            //take the biggest latency for all outputs
            if (delayMs < desc->mLatency*2) {
                delayMs = desc->mLatency*2;
            }
            //mute STRATEGY_MEDIA on all outputs
            if (desc->strategyRefCount(STRATEGY_MEDIA) != 0) {
                setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
                setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
                    getDeviceForStrategy(STRATEGY_MEDIA, 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);
        // unmute the ringing tone after a sufficient delay if it was muted before
        // setting output device above
        if (oldState == AudioSystem::MODE_RINGTONE) {
            setStreamMute(AudioSystem::RING, false, mPrimaryOutput, MUTE_TIME_MS);
        }
        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);
}

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 (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,
                                    uint32_t format,
                                    uint32_t channelMask,
                                    AudioSystem::output_flags flags)
{
    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() stream %d, samplingRate %d, format %d, channelMask %x, flags %x",
          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);
            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
    IOProfile *profile = getProfileForDirectOutput(device,
                                                   samplingRate,
                                                   format,
                                                   channelMask,
                                                   (audio_output_flags_t)flags);
    if (profile != NULL) {

        ALOGV("getOutput() opening direct output device %x", device);

        AudioOutputDescriptor *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)(flags | AUDIO_OUTPUT_FLAG_DIRECT);;
        outputDesc->mRefCount[stream] = 0;
        outputDesc->mStopTime[stream] = 0;
        output = mpClientInterface->openOutput(profile->mModule->mHandle,
                                        &outputDesc->mDevice,
                                        &outputDesc->mSamplingRate,
                                        &outputDesc->mFormat,
                                        &outputDesc->mChannelMask,
                                        &outputDesc->mLatency,
                                        outputDesc->mFlags);

        // 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;
        }
        addOutput(output, outputDesc);
        ALOGV("getOutput() returns direct output %d", output);
        return output;
    }

    // ignoring channel mask due to downmix capability in mixer

    // open a non direct output

    // 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.
                if (shouldWait && (desc->refCount() != 0) && (waitMs < desc->latency())) {
                    waitMs = desc->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