基于LIVE555的RTSP QoS实现

如何从OnDemandServerMediaSubsession类以及继承类对象中获取RTCP信息（句柄）

OnDemandServerMediaSubsession.cpp void StreamState::startPlaying函数中添加：

fRTCPInstance->setRRHandler(fMaster.fRRHandlerTask, fMaster.fRRHandlerClientData);

OnDemandServerMediaSubsession.hh 中OnDemandServerMediaSubsession添加两个成员：

 TaskFunc* fRRHandlerTask;
 void* fRRHandlerClientData;

以及成员函数

setRTCPRRPacketHandler(TaskFunc* handler, void* clientData) {
  fRRHandlerTask = handler;
  fRRHandlerClientData = clientData;
}

创建CamServerMediaSubsession 对象时，设置回调。（CamServerMediaSubsession 是继承于 OnDemandServerMediaSubsession，重写createNewStreamSource和createNewRTPSink即可）

CamServerMediaSubsession   *sub = CamServerMediaSubsession::createNew(*env, inputDevice, &device);
...
sub->setRTCPRRPacketHandler(RTCPRRHandler, (void *)sub);
...

函调函数中获取RTCP RR信息

void RTCPRRHandler(void* clientData)
{
    using namespace CamStream;
    CamServerMediaSubsession *sub = (CamServerMediaSubsession *)clientData;
    RTPSink *sink = sub->get_rtp_sink();
    if (!sink) {
        std::cout<<"unable to get sink obj, not ready"<<std::endl;
        return;
    }
    bool ignore_firstRR = true;

    RTPTransmissionStatsDB& transmissionStats = sink->transmissionStatsDB();
    RTPTransmissionStatsDB::Iterator iter(transmissionStats);
    RTPTransmissionStats* substat;

    while ((substat = iter.next()) != NULL) {
        auto cam = sub->get_cam_instance();
        auto jitter = substat->jitter();
        auto loss_ratio = ((float)substat->packetLossRatio()/)*;   // %
        auto rtt = (int)(((float)substat->roundTripDelay()/)*);  //ms
        auto last_bitrate = cam->get_bitrate();

        std::cout<<"SSRC "<<substat->SSRC()
            <<" RTT "<<rtt<<" ms"
            <<" jitter "<<jitter
            <<" loss "<<(int)loss_ratio<<"%"<<std::endl;
　　}
}

最后，根据丢帧率以及RTTD等信息，我们可以调整视频源的码率，实现QoS。

关于CamServerMediaSubsession的实现（实现下面两个函数，就可以蒋H264视频流转为RTP传输流，从而实现RTSP服务器）　

　　/*source */
　　FramedSource* CamServerMediaSubsession::createNewStreamSource(unsigned clientSessionId, unsigned& estBitrate)
    {
        estBitrate = static_cast<unsigned int>(this->bit_rate_);
        FramedSource *source = replicator_->createStreamReplica();
        //H264VideoStreamDiscreteFramer的输入是离散的NALU
        //H264VideoStreamFramer的输入是stream bit流
        FramedSource *h264_source = H264VideoStreamDiscreteFramer::createNew(envir(), source);
        return h264_source;
    }

    /*sink */
    RTPSink* CamServerMediaSubsession::createNewRTPSink(Groupsock* rtpGroupsock,
            unsigned char rtpPayloadTypeIfDynamic,
            FramedSource* inputSource)
    {
        auto sink = H264VideoRTPSink::createNew(envir(),
                                                rtpGroupsock,
                                                rtpPayloadTypeIfDynamic,
                                                sps_nal_,
                                                sps_nal_size_,
                                                pps_nal_,
                                                pps_nal_size_);
        return sink;
    }

总结，上面的实现修改了live555源码，官方推荐的方式是通过继承现有类重写方法来实现，不过代码看了半天没头绪，有知道怎么弄的告诉我声（vslinux@qq.com）