Live555主要有四个类库:
libUsageEnvironment.lib;libliveMedia.lib;libgroupsock.lib;libBasicUsageEnvironment.lib
将这四个类库以及相关的头文件导入VC++2010之后,可以轻松实现网络直播系统。
在这里直接贴上完整代码,粘贴到VC里面就可以运行。
注:程序运行后,使用播放器软件(VLC Media Player,FFplay等),打开URL:rtp://239.255.42.42:1234,即可收看直播的视频。
// 网络直播系统.cpp : 定义控制台应用程序的入口点。 // 雷霄骅 // 中国传媒大学/数字电视技术 // 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) { // 首先建立使用环境: TaskScheduler* scheduler = BasicTaskScheduler::createNew(); env = BasicUsageEnvironment::createNew(*scheduler); // 创建 'groupsocks' for RTP and RTCP: char const* destinationAddressStr #ifdef USE_SSM = "232.255.42.42"; #else = "239.255.42.42"; // Note: 这是一个多播地址。如果你希望流使用单播地址,然后替换这个字符串与单播地址 #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 // 创建一个适当的“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] = ' '; #ifdef IMPLEMENT_RTSP_SERVER RTCPInstance* rtcp = #endif RTCPInstance::createNew(*env, &rtcpGroupsock, estimatedSessionBandwidth, CNAME, videoSink, NULL /* we're a server */, isSSM); // 开始自动运行的媒体 #ifdef IMPLEMENT_RTSP_SERVER RTSPServer* rtspServer = RTSPServer::createNew(*env); if (rtspServer == NULL) { *env << "Failed to create RTSP server: " << env->getResultMsg() << " "; 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 << "" "; delete[] url; #endif *env << "开始发送流媒体... "; play(); env->taskScheduler().doEventLoop(); return 0; // 只是为了防止编译器警告 } void afterPlaying(void* /*clientData*/) { *env << "...从文件中读取完毕 "; Medium::close(videoSource); // 将关闭从源读取的输入文件 play(); } void play() { unsigned const inputDataChunkSize = TRANSPORT_PACKETS_PER_NETWORK_PACKET*TRANSPORT_PACKET_SIZE; // 打开输入文件作为一个“ByteStreamFileSource": ByteStreamFileSource* fileSource = ByteStreamFileSource::createNew(*env, inputFileName, inputDataChunkSize); if (fileSource == NULL) { *env << "无法打开文件 "" << inputFileName << "" 作为 file source "; exit(1); } videoSource = MPEG2TransportStreamFramer::createNew(*env, fileSource); *env << "Beginning to read from file... "; videoSink->startPlaying(*videoSource, afterPlaying, videoSink); }
完整工程下载地址:http://download.csdn.net/detail/leixiaohua1020/6272839