使用Live555類庫實現的網絡直播系統

Live555主要有四個類庫：

libUsageEnvironment.lib；libliveMedia.lib；libgroupsock.lib；libBasicUsageEnvironment.lib

將這四個類庫以及相關的頭文件導入VC++2010之後，可以輕松實現網絡直播系統。

在這裡直接貼上完整代碼，粘貼到VC裡面就可以運行。

注：程序運行後，使用播放器軟件VLC Media Player，FFplay等），打開URL：rtp://239.255.42.42:1234，即可收看直播的視頻。

// 網絡直播系統.cpp : 定義控制台應用程序的入口點。
// 雷霄骅
// 中國傳媒大學/數字電視技術
// leixiaohua1020@126.com
#include "stdafx.h"
#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
//#define IMPLEMENT_RTSP_SERVER
//#define USE_SSM 1
#ifdef USE_SSM
Boolean const isSSM = True;
#else
Boolean const isSSM = False;
#endif
#define TRANSPORT_PACKET_SIZE 188
#define TRANSPORT_PACKETS_PER_NETWORK_PACKET 7
UsageEnvironment* env;
char const* inputFileName = "test.ts";
FramedSource* videoSource;
RTPSink* videoSink;
void play(); // forward
int main(int argc, char** argv) {
  // 首先建立使用環境：
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();
  env = BasicUsageEnvironment::createNew(*scheduler);
  // 創建 'groupsocks' for RTP and RTCP:
  char const* destinationAddressStr
#ifdef USE_SSM
    = "232.255.42.42";
#else
    = "239.255.42.42";
  // Note: 這是一個多播地址。如果你希望流使用單播地址,然後替換這個字符串與單播地址 
#endif
  const unsigned short rtpPortNum = 1234;
  const unsigned short rtcpPortNum = rtpPortNum+1;
  const unsigned char ttl = 7; //
  struct in_addr destinationAddress;
  destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
  const Port rtpPort(rtpPortNum);
  const Port rtcpPort(rtcpPortNum);
  Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
  Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
#ifdef USE_SSM
  rtpGroupsock.multicastSendOnly();
  rtcpGroupsock.multicastSendOnly();
#endif
  // 創建一個適當的“RTPSink”:
  videoSink =
    SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video", "mp2t",
                 1, True, False /*no 'M' bit*/);
  const unsigned estimatedSessionBandwidth = 5000; // in kbps; for RTCP b/w share
  const unsigned maxCNAMElen = 100;
  unsigned char CNAME[maxCNAMElen+1];
  gethostname((char*)CNAME, maxCNAMElen);
  CNAME[maxCNAMElen] = '\0';
#ifdef IMPLEMENT_RTSP_SERVER
  RTCPInstance* rtcp =
#endif
    RTCPInstance::createNew(*env, &rtcpGroupsock,
                estimatedSessionBandwidth, CNAME,
                videoSink, NULL /* we're a server */, isSSM);
  // 開始自動運行的媒體
#ifdef IMPLEMENT_RTSP_SERVER
  RTSPServer* rtspServer = RTSPServer::createNew(*env);
    
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }
  ServerMediaSession* sms
    = ServerMediaSession::createNew(*env, "testStream", inputFileName,
           "Session streamed by \"testMPEG2TransportStreamer\"",
                       isSSM);
  sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
  rtspServer->addServerMediaSession(sms);
  char* url = rtspServer->rtspURL(sms);
  *env << "Play this stream using the URL \"" << url << "\"\n";
  delete[] url;
#endif
   
  *env << "開始發送流媒體...\n";
  play();
  env->taskScheduler().doEventLoop();
  return 0; // 只是為了防止編譯器警告
}
void afterPlaying(void* /*clientData*/) {
  *env << "...從文件中讀取完畢\n";
  Medium::close(videoSource);
  // 將關閉從源讀取的輸入文件
  play();
}
void play() {
  unsigned const inputDataChunkSize
    = TRANSPORT_PACKETS_PER_NETWORK_PACKET*TRANSPORT_PACKET_SIZE;
  // 打開輸入文件作為一個“ByteStreamFileSource":
  ByteStreamFileSource* fileSource
    = ByteStreamFileSource::createNew(*env, inputFileName, inputDataChunkSize);
  if (fileSource == NULL) {
    *env << "無法打開文件 \"" << inputFileName
     << "\" 作為 file source\n";
    exit(1);
  }
    
  videoSource = MPEG2TransportStreamFramer::createNew(*env, fileSource);
    
  *env << "Beginning to read from file...\n";
  videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}

本文出自 “leixiaohua1020視音頻技術” 博客，請務必保留此出處http://leixiaohua1020.blog.51cto.com/3974648/1303876