WebRTC学习之十：最简单的视频聊天（使用WebRtcVideoEngine2）

来源：互联网发布：手机网络创业编辑：程序博客网时间：2024/06/06 17:38

这篇在上篇WebRTC学习之九：摄像头的捕捉和显示的基础上修改而来。上篇中主要使用了WebRtcVideoEngine2中的WebRtcVideoCapturer类，而本篇中主要使用了WebRtcVideoEngine2中的WebRtcVideoChannel2类。

一.环境

参考：WebRTC学习之三：录音和播放

二.实现

在WebRTC学习之四：最简单的语音聊天中我们是通过实现Transport类的两个纯虚函数SendRtp和SendRtcp，并在它们的实现中调用通信协议的发送函数将数据发送出去。然而WebRtcVideoChannel2类继承自Transport，实现了SendRtp和SendRtcp，并在实现中分别调用了MediaChannel类的SendPacket和SendRtcp函数。MediaChannel类也是WebRtcVideoChannel2的基类。SendPacket和SendRtcp函数最终调用的是MediaChannel类中NetworkInterface接口的同名函数SendPacket和SendRtcp。因此我们只需要去继承NetworkInterface接口，实现其中的函数SendPacket和SendRtcp，并在它们的实现中调用通信协议的发送函数将数据发送出去。

mynetworkinterface.h

#ifndef MYNETWORKINTERFACE_H#define MYNETWORKINTERFACE_H#include <QUdpSocket>#include "webrtc/media/base/mediachannel.h"#include "webrtc/modules/rtp_rtcp/source/byte_io.h"class MyNetworkInterface:public QObject,public cricket::MediaChannel::NetworkInterface{    Q_OBJECTpublic:    MyNetworkInterface();    ~MyNetworkInterface();    void setLocalReceiver(int port);    void stopRecieve();    void setSendDestination(QString ip, int port);    void stopSend();    // NetworkInterface functions override    bool SendPacket(rtc::CopyOnWriteBuffer* packet,const rtc::PacketOptions& options) override;    bool SendRtcp(rtc::CopyOnWriteBuffer* packet,const rtc::PacketOptions& options) override;    int SetOption(SocketType type, rtc::Socket::Option opt,int option) override    {        return 0;    }private:    QUdpSocket * udpsocketSendRTP;    QUdpSocket * udpsocketSendRTCP;    QUdpSocket * udpSocketRecvRTP;    QUdpSocket * udpSocketRecvRTCP;    QString destIP;    int destPort;    bool sendFlag;    bool recvFlag;signals:    void signalRecvRTPData(char *data,int length);    void signalRecvRTCPData(char *data,int length);    void signalSendRTPData(char *data,int length);    void signalSendRTCPData(char *data,int length);private slots:    void slotRTPReadPendingDatagrams();    void slotRTCPReadPendingDatagrams();    void slotSendRTPData(char *data,int length);    void slotSendRTCPData(char *data,int length);};#endif // MYNETWORKINTERFACE_H

mynetworkinterface.cpp

#include "mynetworkinterface.h"#include "QDebug"#include <QTime>MyNetworkInterface::MyNetworkInterface()    :destIP(""),    destPort(0),    sendFlag(true),    recvFlag(true){    udpsocketSendRTP=new QUdpSocket();    udpSocketRecvRTP = new QUdpSocket();    udpsocketSendRTCP=new QUdpSocket();    udpSocketRecvRTCP = new QUdpSocket();    connect(udpSocketRecvRTP, SIGNAL(readyRead()), this, SLOT(slotRTPReadPendingDatagrams()));    connect(udpSocketRecvRTCP, SIGNAL(readyRead()), this, SLOT(slotRTCPReadPendingDatagrams()));    connect(this,SIGNAL(signalSendRTPData(char *,int)),this,SLOT(slotSendRTPData(char *,int)));    connect(this,SIGNAL(signalSendRTCPData(char *,int)),this,SLOT(slotSendRTCPData(char *,int)));}MyNetworkInterface::~MyNetworkInterface(){   udpsocketSendRTP->deleteLater();   udpSocketRecvRTP->deleteLater();   udpsocketSendRTCP->deleteLater();   udpSocketRecvRTCP->deleteLater();}void MyNetworkInterface::setLocalReceiver(int port){    udpSocketRecvRTP->bind(port, QUdpSocket::ShareAddress);    udpSocketRecvRTCP->bind(port+1, QUdpSocket::ShareAddress);    recvFlag=true;}void MyNetworkInterface::stopRecieve(){    udpSocketRecvRTP->abort();    udpSocketRecvRTCP->abort();    recvFlag=false;}void MyNetworkInterface::setSendDestination(QString ip, int port){    destIP=ip;    destPort=port;    sendFlag=true;}void MyNetworkInterface::stopSend(){    sendFlag=false;}//为何不直接调用udpsocketSendRTP->writeDatagram，而用信号，是因为SendPacket在另一个线程里bool MyNetworkInterface::SendPacket(rtc::CopyOnWriteBuffer* packet,const rtc::PacketOptions& options){    Q_UNUSED(options);    //测试发送的ssrc//    uint8_t* packet1=(uint8_t*)data;//    uint32_t ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(&packet1[8]);//    qDebug()<<"Send SSRC:"<<ssrc;    if(sendFlag)    emit signalSendRTPData((char*)packet->data<uint8_t>(),packet->size());    return true;}//为何不直接调用udpsocketSendRTCP->writeDatagram，而用信号，是因为SendRtcp在另一个线程里bool MyNetworkInterface::SendRtcp(rtc::CopyOnWriteBuffer* packet,const rtc::PacketOptions& options){   Q_UNUSED(options);   if(sendFlag)   emit signalSendRTCPData(packet->data<char>(),packet->size());   return true;}void MyNetworkInterface::slotSendRTPData(char *data,int length){    //测试发送的ssrc//    uint8_t* packet1=(uint8_t*)data;//    uint32_t ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(&packet1[8]);//    qDebug()<<"Send SSRC:"<<ssrc;    udpsocketSendRTP->writeDatagram(data, length,QHostAddress(destIP), destPort);}//RTCP端口为RTP端口+1void MyNetworkInterface::slotSendRTCPData(char *data,int length){    udpsocketSendRTCP->writeDatagram(data, length,QHostAddress(destIP), destPort+1);}void MyNetworkInterface::slotRTPReadPendingDatagrams(){   QByteArray datagram;   while (udpSocketRecvRTP->hasPendingDatagrams()&&recvFlag)   {          datagram.resize(udpSocketRecvRTP->pendingDatagramSize());          QHostAddress sender;          quint16 senderPort;          int size=udpSocketRecvRTP->readDatagram(          datagram.data(),          datagram.size(),          &sender,          &senderPort);          if(size>0)          {              //测试接收的ssrc          //    uint8_t* packet1=(uint8_t*)data;          //    uint32_t ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(&packet1[8]);          //    qDebug()<<"Receive SSRC:"<<ssrc;              emit signalRecvRTPData(datagram.data(),datagram.size());          }    }}void MyNetworkInterface::slotRTCPReadPendingDatagrams(){   QByteArray datagram;   while (udpSocketRecvRTCP->hasPendingDatagrams()&&recvFlag)   {          datagram.resize(udpSocketRecvRTCP->pendingDatagramSize());          QHostAddress sender;          quint16 senderPort;          int size=udpSocketRecvRTCP->readDatagram(          datagram.data(),          datagram.size(),          &sender,          &senderPort);          if(size>0)          {              emit signalRecvRTCPData(datagram.data(),datagram.size());          }    }}

WebRtcVideoChannel2中有个专门设置网络接口的函数SetInterface，其参数就是NetworkInterface，这样视频数据发送就与WebRtcVideoChannel2关联起来了。

至于频数据接收，需要调用通信协议的接收函数，将接收到的数据传递到WebRtcVideoChannel2的OnPacketReceived和OnRtcpReceived函数，这两个函数是其基类对应纯虚函数的实现。那么解码后的frame我们如何获取呢？在WebRtcVideoChannel2中有个函数SetSink，它的第二个参数是rtc::VideoSinkInterface<VideoFrame>*，因此我们只需要继承rtc::VideoSinkInterface<VideoFrame>，并实现其中的void OnFrame(const VideoFrameT& frame) =0;纯虚函数，就能得到解码后的frame，然后转换成RGB格式，就可以显示到Qt界面了。

void MainWindow::OnFrame(const cricket::VideoFrame& frame){    //复制一份    const cricket::VideoFrame* frameTemp = frame.GetCopyWithRotationApplied();    frameTemp->ConvertToRgbBuffer(cricket::FOURCC_ARGB,                             receivedVideoImageData.get(),                             frameTemp->width()*frameTemp->height()*32/8,                             frameTemp->width()*32/8);    QImage image(receivedVideoImageData.get(), frameTemp->width(), frameTemp->height(), QImage::Format_RGB32);    //因为OnFrame在另一个线程，所以不直接显示，用信号中转一下,将显示放在GUI线程（主线程）    if(!image.isNull())        emit signalReceivedFrame(image);}void MainWindow::slotReceivedFrame(const QImage &image){     ui->labelReceivedVideo->setPixmap(QPixmap::fromImage(image));}

三.效果

本人光荣上镜，长相一般，用个口罩挡一下。大窗口是摄像头捕捉和显示的图像，小窗口是通过网络接收到并显示的图像。

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