overte-HifiExperiments/assignment-client/src/audio/AudioMixer.h

//
//  AudioMixer.h
//  assignment-client/src/audio
//
//  Created by Stephen Birarda on 8/22/13.
//  Copyright 2013 High Fidelity, Inc.
//
//  Distributed under the Apache License, Version 2.0.
//  See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
//

#ifndef hifi_AudioMixer_h
#define hifi_AudioMixer_h

#include <AABox.h>
#include <AudioRingBuffer.h>
#include <ThreadedAssignment.h>

class PositionalAudioStream;
class AvatarAudioStream;
class AudioMixerClientData;

const int SAMPLE_PHASE_DELAY_AT_90 = 20;

const int READ_DATAGRAMS_STATS_WINDOW_SECONDS = 30;

/// Handles assignments of type AudioMixer - mixing streams of audio and re-distributing to various clients.
class AudioMixer : public ThreadedAssignment {
    Q_OBJECT
public:
    AudioMixer(const QByteArray& packet);
    ~AudioMixer();
public slots:
    /// threaded run of assignment
    void run();
    
    void readPendingDatagrams() { }; // this will not be called since our datagram processing thread will handle
    void readPendingDatagram(const QByteArray& receivedPacket, const HifiSockAddr& senderSockAddr);
    
    void sendStatsPacket();

    static const InboundAudioStream::Settings& getStreamSettings() { return _streamSettings; }
    
private:
    /// adds one stream to the mix for a listening node
    int addStreamToMixForListeningNodeWithStream(AudioMixerClientData* listenerNodeData,
                                                    const QUuid& streamUUID,
                                                    PositionalAudioStream* streamToAdd,
                                                    AvatarAudioStream* listeningNodeStream);
    
    /// prepares and sends a mix to one Node
    int prepareMixForListeningNode(Node* node);

    // used on a per stream basis to run the filter on before mixing, large enough to handle the historical
    // data from a phase delay as well as an entire network buffer
    int16_t _preMixSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)];
    
    // client samples capacity is larger than what will be sent to optimize mixing
    // we are MMX adding 4 samples at a time so we need client samples to have an extra 4
    int16_t _mixSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)];

    void perSecondActions();

    QString getReadPendingDatagramsCallsPerSecondsStatsString() const;
    QString getReadPendingDatagramsPacketsPerCallStatsString() const;
    QString getReadPendingDatagramsTimeStatsString() const;
    QString getReadPendingDatagramsHashMatchTimeStatsString() const;
    
    float _trailingSleepRatio;
    float _minAudibilityThreshold;
    float _performanceThrottlingRatio;
    int _numStatFrames;
    int _sumListeners;
    int _sumMixes;
    AABox* _sourceUnattenuatedZone;
    AABox* _listenerUnattenuatedZone;

    static InboundAudioStream::Settings _streamSettings;

    static bool _printStreamStats;
    static bool _enableFilter;
    
    quint64 _lastPerSecondCallbackTime;

    bool _sendAudioStreamStats;

    // stats
    MovingMinMaxAvg<int> _datagramsReadPerCallStats;     // update with # of datagrams read for each readPendingDatagrams call
    MovingMinMaxAvg<quint64> _timeSpentPerCallStats;     // update with usecs spent inside each readPendingDatagrams call
    MovingMinMaxAvg<quint64> _timeSpentPerHashMatchCallStats; // update with usecs spent inside each packetVersionAndHashMatch call

    MovingMinMaxAvg<int> _readPendingCallsPerSecondStats;     // update with # of readPendingDatagrams calls in the last second
};

#endif // hifi_AudioMixer_h