C++ PositionalAudioStream类代码示例

int AudioMixer::prepareMixForListeningNode(Node* node) {
    AvatarAudioStream* nodeAudioStream = ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioStream();
    
    // zero out the client mix for this node
    memset(_clientSamples, 0, NETWORK_BUFFER_LENGTH_BYTES_STEREO);

    // loop through all other nodes that have sufficient audio to mix
    int streamsMixed = 0;
    foreach (const SharedNodePointer& otherNode, NodeList::getInstance()->getNodeHash()) {
        if (otherNode->getLinkedData()) {
            AudioMixerClientData* otherNodeClientData = (AudioMixerClientData*) otherNode->getLinkedData();

            // enumerate the ARBs attached to the otherNode and add all that should be added to mix

            const QHash<QUuid, PositionalAudioStream*>& otherNodeAudioStreams = otherNodeClientData->getAudioStreams();
            QHash<QUuid, PositionalAudioStream*>::ConstIterator i;
            for (i = otherNodeAudioStreams.constBegin(); i != otherNodeAudioStreams.constEnd(); i++) {
                PositionalAudioStream* otherNodeStream = i.value();
                
                if (*otherNode != *node || otherNodeStream->shouldLoopbackForNode()) {
                    streamsMixed += addStreamToMixForListeningNodeWithStream(otherNodeStream, nodeAudioStream);
                }
            }
        }
    }
    return streamsMixed;
}

float computeGain(const AudioMixerClientData& listenerNodeData, const AvatarAudioStream& listeningNodeStream,
        const PositionalAudioStream& streamToAdd, const glm::vec3& relativePosition, bool isEcho) {
    float gain = 1.0f;

    // injector: apply attenuation
    if (streamToAdd.getType() == PositionalAudioStream::Injector) {
        gain *= reinterpret_cast<const InjectedAudioStream*>(&streamToAdd)->getAttenuationRatio();

    // avatar: apply fixed off-axis attenuation to make them quieter as they turn away
    } else if (!isEcho && (streamToAdd.getType() == PositionalAudioStream::Microphone)) {
        glm::vec3 rotatedListenerPosition = glm::inverse(streamToAdd.getOrientation()) * relativePosition;

        // source directivity is based on angle of emission, in local coordinates
        glm::vec3 direction = glm::normalize(rotatedListenerPosition);
        float angleOfDelivery = fastAcosf(glm::clamp(-direction.z, -1.0f, 1.0f));   // UNIT_NEG_Z is "forward"

        const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
        const float OFF_AXIS_ATTENUATION_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
        float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION + (angleOfDelivery * (OFF_AXIS_ATTENUATION_STEP / PI_OVER_TWO));

        gain *= offAxisCoefficient;

        // apply master gain, only to avatars
        gain *= listenerNodeData.getMasterAvatarGain();
    }

    auto& audioZones = AudioMixer::getAudioZones();
    auto& zoneSettings = AudioMixer::getZoneSettings();

    // find distance attenuation coefficient
    float attenuationPerDoublingInDistance = AudioMixer::getAttenuationPerDoublingInDistance();
    for (int i = 0; i < zoneSettings.length(); ++i) {
        if (audioZones[zoneSettings[i].source].contains(streamToAdd.getPosition()) &&
            audioZones[zoneSettings[i].listener].contains(listeningNodeStream.getPosition())) {
            attenuationPerDoublingInDistance = zoneSettings[i].coefficient;
            break;
        }
    }

    // distance attenuation
    const float ATTENUATION_START_DISTANCE = 1.0f;
    float distance = glm::length(relativePosition);
    assert(ATTENUATION_START_DISTANCE > EPSILON);
    if (distance >= ATTENUATION_START_DISTANCE) {

        // translate the zone setting to gain per log2(distance)
        float g = 1.0f - attenuationPerDoublingInDistance;
        g = glm::clamp(g, EPSILON, 1.0f);

        // calculate the distance coefficient using the distance to this node
        float distanceCoefficient = fastExp2f(fastLog2f(g) * fastLog2f(distance/ATTENUATION_START_DISTANCE));

        // multiply the current attenuation coefficient by the distance coefficient
        gain *= distanceCoefficient;
    }

    return gain;
}

float AudioMixer::gainForSource(const PositionalAudioStream& streamToAdd,
                                const AvatarAudioStream& listeningNodeStream, const glm::vec3& relativePosition, bool isEcho) {
    float gain = 1.0f;

    float distanceBetween = glm::length(relativePosition);

    if (distanceBetween < EPSILON) {
        distanceBetween = EPSILON;
    }

    if (streamToAdd.getType() == PositionalAudioStream::Injector) {
        gain *= reinterpret_cast<const InjectedAudioStream*>(&streamToAdd)->getAttenuationRatio();
    }

    if (!isEcho && (streamToAdd.getType() == PositionalAudioStream::Microphone)) {
        //  source is another avatar, apply fixed off-axis attenuation to make them quieter as they turn away from listener
        glm::vec3 rotatedListenerPosition = glm::inverse(streamToAdd.getOrientation()) * relativePosition;

        float angleOfDelivery = glm::angle(glm::vec3(0.0f, 0.0f, -1.0f),
                                           glm::normalize(rotatedListenerPosition));

        const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
        const float OFF_AXIS_ATTENUATION_FORMULA_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;

        float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION +
        (OFF_AXIS_ATTENUATION_FORMULA_STEP * (angleOfDelivery / PI_OVER_TWO));

        // multiply the current attenuation coefficient by the calculated off axis coefficient
        gain *= offAxisCoefficient;
    }

    float attenuationPerDoublingInDistance = _attenuationPerDoublingInDistance;
    for (int i = 0; i < _zonesSettings.length(); ++i) {
        if (_audioZones[_zonesSettings[i].source].contains(streamToAdd.getPosition()) &&
            _audioZones[_zonesSettings[i].listener].contains(listeningNodeStream.getPosition())) {
            attenuationPerDoublingInDistance = _zonesSettings[i].coefficient;
            break;
        }
    }

    if (distanceBetween >= ATTENUATION_BEGINS_AT_DISTANCE) {

        // translate the zone setting to gain per log2(distance)
        float g = 1.0f - attenuationPerDoublingInDistance;
        g = (g < EPSILON) ? EPSILON : g;
        g = (g > 1.0f) ? 1.0f : g;

        // calculate the distance coefficient using the distance to this node
        float distanceCoefficient = exp2f(log2f(g) * log2f(distanceBetween/ATTENUATION_BEGINS_AT_DISTANCE));

        // multiply the current attenuation coefficient by the distance coefficient
        gain *= distanceCoefficient;
    }

    return gain;
}

void AudioMixerClientData::checkBuffersBeforeFrameSend() {
    QHash<QUuid, PositionalAudioStream*>::ConstIterator i;
    for (i = _audioStreams.constBegin(); i != _audioStreams.constEnd(); i++) {
        PositionalAudioStream* stream = i.value();

        if (stream->popFrames(1, true) > 0) {
            stream->updateLastPopOutputLoudnessAndTrailingLoudness();
        }
    }
}

void AudioMixerClientData::sendAudioStreamStatsPackets(const SharedNodePointer& destinationNode) {

    auto nodeList = DependencyManager::get<NodeList>();

    // The append flag is a boolean value that will be packed right after the header.
    // This flag allows the client to know when it has received all stats packets, so it can group any downstream effects,
    // and clear its cache of injector stream stats; it helps to prevent buildup of dead audio stream stats in the client.
    quint8 appendFlag = AudioStreamStats::START;

    auto streamsCopy = getAudioStreams();

    // pack and send stream stats packets until all audio streams' stats are sent
    int numStreamStatsRemaining = int(streamsCopy.size());
    auto it = streamsCopy.cbegin();

    while (numStreamStatsRemaining > 0) {
        auto statsPacket = NLPacket::create(PacketType::AudioStreamStats);

        int numStreamStatsRoomFor = (int)(statsPacket->size() - sizeof(quint8) - sizeof(quint16)) / sizeof(AudioStreamStats);

        // calculate the number of stream stats to follow
        quint16 numStreamStatsToPack = std::min(numStreamStatsRemaining, numStreamStatsRoomFor);

        // is this the terminal packet?
        if (numStreamStatsRemaining <= numStreamStatsToPack) {
            appendFlag |= AudioStreamStats::END;
        }

        // pack the append flag in this packet
        statsPacket->writePrimitive(appendFlag);
        appendFlag = 0;

        // pack the number of stream stats to follow
        statsPacket->writePrimitive(numStreamStatsToPack);

        // pack the calculated number of stream stats
        for (int i = 0; i < numStreamStatsToPack; i++) {
            PositionalAudioStream* stream = it->second.get();

            stream->perSecondCallbackForUpdatingStats();

            AudioStreamStats streamStats = stream->getAudioStreamStats();
            statsPacket->writePrimitive(streamStats);

            ++it;
        }

        numStreamStatsRemaining -= numStreamStatsToPack;

        // send the current packet
        nodeList->sendPacket(std::move(statsPacket), *destinationNode);
    }
}

void AudioMixerClientData::sendAudioStreamStatsPackets(const SharedNodePointer& destinationNode) {

    // since audio stream stats packets are sent periodically, this is a good place to remove our dead injected streams.
    removeDeadInjectedStreams();

    auto nodeList = DependencyManager::get<NodeList>();

    // The append flag is a boolean value that will be packed right after the header.  The first packet sent
    // inside this method will have 0 for this flag, while every subsequent packet will have 1 for this flag.
    // The sole purpose of this flag is so the client can clear its map of injected audio stream stats when
    // it receives a packet with an appendFlag of 0. This prevents the buildup of dead audio stream stats in the client.
    quint8 appendFlag = 0;

    // pack and send stream stats packets until all audio streams' stats are sent
    int numStreamStatsRemaining = _audioStreams.size();
    QHash<QUuid, PositionalAudioStream*>::ConstIterator audioStreamsIterator = _audioStreams.constBegin();

    while (numStreamStatsRemaining > 0) {
        auto statsPacket = NLPacket::create(PacketType::AudioStreamStats);

        // pack the append flag in this packet
        statsPacket->writePrimitive(appendFlag);
        appendFlag = 1;

        int numStreamStatsRoomFor = (statsPacket->size() - sizeof(quint8) - sizeof(quint16)) / sizeof(AudioStreamStats);

        // calculate and pack the number of stream stats to follow
        quint16 numStreamStatsToPack = std::min(numStreamStatsRemaining, numStreamStatsRoomFor);
        statsPacket->writePrimitive(numStreamStatsToPack);

        // pack the calculated number of stream stats
        for (int i = 0; i < numStreamStatsToPack; i++) {
            PositionalAudioStream* stream = audioStreamsIterator.value();

            stream->perSecondCallbackForUpdatingStats();

            AudioStreamStats streamStats = stream->getAudioStreamStats();
            statsPacket->writePrimitive(streamStats);

            audioStreamsIterator++;
        }

        numStreamStatsRemaining -= numStreamStatsToPack;

        // send the current packet
        nodeList->sendPacket(std::move(statsPacket), *destinationNode);
    }
}

void AudioMixerSlave::mixStream(AudioMixerClientData& listenerNodeData, const QUuid& sourceNodeID,
        const AvatarAudioStream& listeningNodeStream, const PositionalAudioStream& streamToAdd) {
    // only add the stream to the mix if it has a valid position, we won't know how to mix it otherwise
    if (streamToAdd.hasValidPosition()) {
        addStream(listenerNodeData, sourceNodeID, listeningNodeStream, streamToAdd, false);
    }
}

float approximateGain(const AvatarAudioStream& listeningNodeStream, const PositionalAudioStream& streamToAdd) {
    float gain = 1.0f;

    // injector: apply attenuation
    if (streamToAdd.getType() == PositionalAudioStream::Injector) {
        gain *= reinterpret_cast<const InjectedAudioStream*>(&streamToAdd)->getAttenuationRatio();
    }

    // avatar: skip attenuation - it is too costly to approximate

    // distance attenuation: approximate, ignore zone-specific attenuations
    glm::vec3 relativePosition = streamToAdd.getPosition() - listeningNodeStream.getPosition();
    float distance = glm::length(relativePosition);
    return gain / distance;

    // avatar: skip master gain - it is constant for all streams
}

void AudioMixerClientData::removeDeadInjectedStreams() {

    const int INJECTOR_CONSECUTIVE_NOT_MIXED_AFTER_STARTED_THRESHOLD = 100;

    // we have this second threshold in case the injected audio is so short that the injected stream
    // never even reaches its desired size, which means it will never start.
    const int INJECTOR_CONSECUTIVE_NOT_MIXED_THRESHOLD = 1000;

    QHash<QUuid, PositionalAudioStream*>::Iterator i = _audioStreams.begin(), end = _audioStreams.end();
    while (i != end) {
        PositionalAudioStream* audioStream = i.value();
        if (audioStream->getType() == PositionalAudioStream::Injector && audioStream->isStarved()) {
            int notMixedThreshold = audioStream->hasStarted() ? INJECTOR_CONSECUTIVE_NOT_MIXED_AFTER_STARTED_THRESHOLD
                                                              : INJECTOR_CONSECUTIVE_NOT_MIXED_THRESHOLD;
            if (audioStream->getConsecutiveNotMixedCount() >= notMixedThreshold) {
                delete audioStream;
                i = _audioStreams.erase(i);
                continue;
            }
        }
        ++i;
    }
}

int AudioMixer::prepareMixForListeningNode(Node* node) {
    AvatarAudioStream* nodeAudioStream = static_cast<AudioMixerClientData*>(node->getLinkedData())->getAvatarAudioStream();
    AudioMixerClientData* listenerNodeData = static_cast<AudioMixerClientData*>(node->getLinkedData());

    // zero out the client mix for this node
    memset(_preMixSamples, 0, sizeof(_preMixSamples));
    memset(_mixSamples, 0, sizeof(_mixSamples));

    // loop through all other nodes that have sufficient audio to mix
    int streamsMixed = 0;

    DependencyManager::get<NodeList>()->eachNode([&](const SharedNodePointer& otherNode){
        if (otherNode->getLinkedData()) {
            AudioMixerClientData* otherNodeClientData = (AudioMixerClientData*) otherNode->getLinkedData();

            // enumerate the ARBs attached to the otherNode and add all that should be added to mix

            const QHash<QUuid, PositionalAudioStream*>& otherNodeAudioStreams = otherNodeClientData->getAudioStreams();
            QHash<QUuid, PositionalAudioStream*>::ConstIterator i;
            for (i = otherNodeAudioStreams.constBegin(); i != otherNodeAudioStreams.constEnd(); i++) {
                PositionalAudioStream* otherNodeStream = i.value();
                QUuid streamUUID = i.key();

                if (otherNodeStream->getType() == PositionalAudioStream::Microphone) {
                    streamUUID = otherNode->getUUID();
                }

                if (*otherNode != *node || otherNodeStream->shouldLoopbackForNode()) {
                    streamsMixed += addStreamToMixForListeningNodeWithStream(listenerNodeData, streamUUID,
                                                                             otherNodeStream, nodeAudioStream);
                }
            }
        }
    });

    return streamsMixed;
}

float approximateGain(const AvatarAudioStream& listeningNodeStream, const PositionalAudioStream& streamToAdd,
        const glm::vec3& relativePosition) {
    float gain = 1.0f;

    // injector: apply attenuation
    if (streamToAdd.getType() == PositionalAudioStream::Injector) {
        gain *= reinterpret_cast<const InjectedAudioStream*>(&streamToAdd)->getAttenuationRatio();
    }

    // avatar: skip attenuation - it is too costly to approximate

    // distance attenuation: approximate, ignore zone-specific attenuations
    // this is a good approximation for streams further than ATTENUATION_START_DISTANCE
    // those streams closer will be amplified; amplifying close streams is acceptable
    // when throttling, as close streams are expected to be heard by a user
    float distance = glm::length(relativePosition);
    return gain / distance;

    // avatar: skip master gain - it is constant for all streams
}

void AudioMixerSlave::addStream(AudioMixerClientData& listenerNodeData, const QUuid& sourceNodeID,
        const AvatarAudioStream& listeningNodeStream, const PositionalAudioStream& streamToAdd,
        bool throttle) {
    ++stats.totalMixes;

    // to reduce artifacts we call the HRTF functor for every source, even if throttled or silent
    // this ensures the correct tail from last mixed block and the correct spatialization of next first block

    // check if this is a server echo of a source back to itself
    bool isEcho = (&streamToAdd == &listeningNodeStream);

    glm::vec3 relativePosition = streamToAdd.getPosition() - listeningNodeStream.getPosition();

    float distance = glm::max(glm::length(relativePosition), EPSILON);
    float gain = computeGain(listenerNodeData, listeningNodeStream, streamToAdd, relativePosition, isEcho);
    float azimuth = isEcho ? 0.0f : computeAzimuth(listeningNodeStream, listeningNodeStream, relativePosition);
    const int HRTF_DATASET_INDEX = 1;

    if (!streamToAdd.lastPopSucceeded()) {
        bool forceSilentBlock = true;

        if (!streamToAdd.getLastPopOutput().isNull()) {
            bool isInjector = dynamic_cast<const InjectedAudioStream*>(&streamToAdd);

            // in an injector, just go silent - the injector has likely ended
            // in other inputs (microphone, &c.), repeat with fade to avoid the harsh jump to silence
            if (!isInjector) {
                // calculate its fade factor, which depends on how many times it's already been repeated.
                float fadeFactor = calculateRepeatedFrameFadeFactor(streamToAdd.getConsecutiveNotMixedCount() - 1);
                if (fadeFactor > 0.0f) {
                    // apply the fadeFactor to the gain
                    gain *= fadeFactor;
                    forceSilentBlock = false;
                }
            }
        }

        if (forceSilentBlock) {
            // call renderSilent with a forced silent block to reduce artifacts
            // (this is not done for stereo streams since they do not go through the HRTF)
            if (!streamToAdd.isStereo() && !isEcho) {
                // get the existing listener-source HRTF object, or create a new one
                auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());

                static int16_t silentMonoBlock[AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL] = {};
                hrtf.renderSilent(silentMonoBlock, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
                                  AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

                ++stats.hrtfSilentRenders;
            }

            return;
        }
    }

    // grab the stream from the ring buffer
    AudioRingBuffer::ConstIterator streamPopOutput = streamToAdd.getLastPopOutput();

    // stereo sources are not passed through HRTF
    if (streamToAdd.isStereo()) {
        for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; ++i) {
            _mixSamples[i] += float(streamPopOutput[i] * gain / AudioConstants::MAX_SAMPLE_VALUE);
        }

        ++stats.manualStereoMixes;
        return;
    }

    // echo sources are not passed through HRTF
    if (isEcho) {
        for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i += 2) {
            auto monoSample = float(streamPopOutput[i / 2] * gain / AudioConstants::MAX_SAMPLE_VALUE);
            _mixSamples[i] += monoSample;
            _mixSamples[i + 1] += monoSample;
        }

        ++stats.manualEchoMixes;
        return;
    }

    // get the existing listener-source HRTF object, or create a new one
    auto& hrtf = listenerNodeData.hrtfForStream(sourceNodeID, streamToAdd.getStreamIdentifier());

    streamPopOutput.readSamples(_bufferSamples, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

    if (streamToAdd.getLastPopOutputLoudness() == 0.0f) {
        // call renderSilent to reduce artifacts
        hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, gain,
                          AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

        ++stats.hrtfSilentRenders;
        return;
    }

    if (throttle) {
        // call renderSilent with actual frame data and a gain of 0.0f to reduce artifacts
        hrtf.renderSilent(_bufferSamples, _mixSamples, HRTF_DATASET_INDEX, azimuth, distance, 0.0f,
                          AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);

        ++stats.hrtfThrottleRenders;
//.........这里部分代码省略.........

float computeGain(float masterListenerGain, const AvatarAudioStream& listeningNodeStream,
        const PositionalAudioStream& streamToAdd, const glm::vec3& relativePosition, float distance, bool isEcho) {
    float gain = 1.0f;

    // injector: apply attenuation
    if (streamToAdd.getType() == PositionalAudioStream::Injector) {
        gain *= reinterpret_cast<const InjectedAudioStream*>(&streamToAdd)->getAttenuationRatio();

    // avatar: apply fixed off-axis attenuation to make them quieter as they turn away
    } else if (!isEcho && (streamToAdd.getType() == PositionalAudioStream::Microphone)) {
        glm::vec3 rotatedListenerPosition = glm::inverse(streamToAdd.getOrientation()) * relativePosition;

        // source directivity is based on angle of emission, in local coordinates
        glm::vec3 direction = glm::normalize(rotatedListenerPosition);
        float angleOfDelivery = fastAcosf(glm::clamp(-direction.z, -1.0f, 1.0f));   // UNIT_NEG_Z is "forward"

        const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
        const float OFF_AXIS_ATTENUATION_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
        float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION + (angleOfDelivery * (OFF_AXIS_ATTENUATION_STEP / PI_OVER_TWO));

        gain *= offAxisCoefficient;

        // apply master gain, only to avatars
        gain *= masterListenerGain;
    }

    auto& audioZones = AudioMixer::getAudioZones();
    auto& zoneSettings = AudioMixer::getZoneSettings();

    // find distance attenuation coefficient
    float attenuationPerDoublingInDistance = AudioMixer::getAttenuationPerDoublingInDistance();
    for (const auto& settings : zoneSettings) {
        if (audioZones[settings.source].area.contains(streamToAdd.getPosition()) &&
            audioZones[settings.listener].area.contains(listeningNodeStream.getPosition())) {
            attenuationPerDoublingInDistance = settings.coefficient;
            break;
        }
    }

    if (attenuationPerDoublingInDistance < 0.0f) {
        // translate a negative zone setting to distance limit
        const float MIN_DISTANCE_LIMIT = ATTN_DISTANCE_REF + 1.0f;  // silent after 1m
        float distanceLimit = std::max(-attenuationPerDoublingInDistance, MIN_DISTANCE_LIMIT);

        // calculate the LINEAR attenuation using the distance to this node
        // reference attenuation of 0dB at distance = ATTN_DISTANCE_REF
        float d = distance - ATTN_DISTANCE_REF;
        gain *= std::max(1.0f - d / (distanceLimit - ATTN_DISTANCE_REF), 0.0f);
        gain = std::min(gain, ATTN_GAIN_MAX);

    } else {
        // translate a positive zone setting to gain per log2(distance)
        const float MIN_ATTENUATION_COEFFICIENT = 0.001f;   // -60dB per log2(distance)
        float g = glm::clamp(1.0f - attenuationPerDoublingInDistance, MIN_ATTENUATION_COEFFICIENT, 1.0f);

        // calculate the LOGARITHMIC attenuation using the distance to this node
        // reference attenuation of 0dB at distance = ATTN_DISTANCE_REF
        float d = (1.0f / ATTN_DISTANCE_REF) * std::max(distance, HRTF_NEARFIELD_MIN);
        gain *= fastExp2f(fastLog2f(g) * fastLog2f(d));
        gain = std::min(gain, ATTN_GAIN_MAX);
    }

    return gain;
}

int AudioMixerClientData::parseData(NLPacket& packet) {
    PacketType packetType = packet.getType();
    
    if (packetType == PacketType::AudioStreamStats) {

        // skip over header, appendFlag, and num stats packed
        packet.seek(sizeof(quint8) + sizeof(quint16));

        // read the downstream audio stream stats
        packet.readPrimitive(&_downstreamAudioStreamStats);

        return packet.pos();

    } else {
        PositionalAudioStream* matchingStream = NULL;

        if (packetType == PacketType::MicrophoneAudioWithEcho
            || packetType == PacketType::MicrophoneAudioNoEcho
            || packetType == PacketType::SilentAudioFrame) {

            QUuid nullUUID = QUuid();
            if (!_audioStreams.contains(nullUUID)) {
                // we don't have a mic stream yet, so add it

                // read the channel flag to see if our stream is stereo or not
                packet.seek(sizeof(quint16));

                quint8 channelFlag;
                packet.readPrimitive(&channelFlag);

                bool isStereo = channelFlag == 1;

                _audioStreams.insert(nullUUID, matchingStream = new AvatarAudioStream(isStereo, AudioMixer::getStreamSettings()));
            } else {
                matchingStream = _audioStreams.value(nullUUID);
            }
        } else if (packetType == PacketType::InjectAudio) {
            // this is injected audio

            // grab the stream identifier for this injected audio
            packet.seek(sizeof(quint16));
            QUuid streamIdentifier = QUuid::fromRfc4122(packet.readWithoutCopy(NUM_BYTES_RFC4122_UUID));

            bool isStereo;
            packet.readPrimitive(&isStereo);

            if (!_audioStreams.contains(streamIdentifier)) {
                // we don't have this injected stream yet, so add it
                _audioStreams.insert(streamIdentifier,
                        matchingStream = new InjectedAudioStream(streamIdentifier, isStereo, AudioMixer::getStreamSettings()));
            } else {
                matchingStream = _audioStreams.value(streamIdentifier);
            }
        }

        // seek to the beginning of the packet so that the next reader is in the right spot
        packet.seek(0);

        return matchingStream->parseData(packet);
    }
    return 0;
}

int AudioMixerClientData::parseData(ReceivedMessage& message) {
    PacketType packetType = message.getType();
    
    if (packetType == PacketType::AudioStreamStats) {

        // skip over header, appendFlag, and num stats packed
        message.seek(sizeof(quint8) + sizeof(quint16));

        // read the downstream audio stream stats
        message.readPrimitive(&_downstreamAudioStreamStats);

        return message.getPosition();

    } else {
        PositionalAudioStream* matchingStream = NULL;

        bool isMicStream = false;

        if (packetType == PacketType::MicrophoneAudioWithEcho
            || packetType == PacketType::MicrophoneAudioNoEcho
            || packetType == PacketType::SilentAudioFrame) {

            QUuid nullUUID = QUuid();
            if (!_audioStreams.contains(nullUUID)) {
                // we don't have a mic stream yet, so add it

                // read the channel flag to see if our stream is stereo or not
                message.seek(sizeof(quint16));

                quint8 channelFlag;
                message.readPrimitive(&channelFlag);

                bool isStereo = channelFlag == 1;

                _audioStreams.insert(nullUUID, matchingStream = new AvatarAudioStream(isStereo, AudioMixer::getStreamSettings()));
            } else {
                matchingStream = _audioStreams.value(nullUUID);
            }

            isMicStream = true;
        } else if (packetType == PacketType::InjectAudio) {
            // this is injected audio

            // grab the stream identifier for this injected audio
            message.seek(sizeof(quint16));
            QUuid streamIdentifier = QUuid::fromRfc4122(message.readWithoutCopy(NUM_BYTES_RFC4122_UUID));

            bool isStereo;
            message.readPrimitive(&isStereo);

            if (!_audioStreams.contains(streamIdentifier)) {
                // we don't have this injected stream yet, so add it
                _audioStreams.insert(streamIdentifier,
                        matchingStream = new InjectedAudioStream(streamIdentifier, isStereo, AudioMixer::getStreamSettings()));
            } else {
                matchingStream = _audioStreams.value(streamIdentifier);
            }
        }

        // seek to the beginning of the packet so that the next reader is in the right spot
        message.seek(0);

        // check the overflow count before we parse data
        auto overflowBefore = matchingStream->getOverflowCount();
        auto parseResult = matchingStream->parseData(message);

        if (matchingStream->getOverflowCount() > overflowBefore) {
            qDebug() << "Just overflowed on stream from" << message.getSourceID() << "at" << message.getSenderSockAddr();
            qDebug() << "This stream is for" << (isMicStream ? "microphone audio" : "injected audio");
        }

        return parseResult;
    }
    return 0;
}