LIVE555再学习 -- testH264VideoStreamer 源码分析

上一篇文章我们已经讲了一部分：

testH264VideoStreamer 重复从 H.264 基本流视频文件（名为“test.264”）中读取，并使用 RTP 多播进行流式传输。 该程序还具有内置的 RTSP 服务器。
Apple 的“QuickTime 播放器”可用于接收和播放此音频流。 要使用它，让玩家打开会话的“rtsp：//”URL（程序在开始流式传输时打印出来）。
开源“VLC”和“MPlayer”媒体播放器也可以使用。

因为都有英文注释的，所以源码分析也简单

一、源码分析

首先我要说明一下，testH264VideoStreamer 我执行了完，然后一直不出现视频

尝试了好几个版本的live555，但是都不成功，很影响心情啊。然后看到一篇文章。

参看：live555编译、播放示例

这里的源码我试了一下，人家的是可以的，好伤心。为什么呢？先看一下它的源码，讲解的很明白了。

/** 
本程序同时提供单播、组播功能。基于testH264VideoStreamer程序修改，另参考testOnDemandRTSPServer。 
注： 
单播：重开VLC连接，会重新读文件。无马赛克 
组播：重开VLC连接，会继续上一次的位置往下读文件。每次连接时，出现马赛克，VLC出现： 
      main error: pictures leaked, trying to workaround 
 
*/  
#include <liveMedia.hh>  
#include <BasicUsageEnvironment.hh>  
#include <GroupsockHelper.hh>  
  
UsageEnvironment* env;  
char inputFileName[128] = {0};  // 输入的视频文件  
H264VideoStreamFramer* videoSource;  
RTPSink* videoSink;  
  
Boolean reuseFirstSource = False;  
  
void play(); // forward  
  
void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,  
               char const* streamName, char const* inputFileName);  
  
int main(int argc, char** argv) {  
  strcpy(inputFileName, "test.264"); // 默认值  
  if (argc == 2) {  
    strcpy(inputFileName, argv[1]);  
  }  
  printf("Using file: %s\n", inputFileName);  
    
  // Begin by setting up our usage environment:  
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();  
  env = BasicUsageEnvironment::createNew(*scheduler);  
  
  // 描述信息  
  char const* descriptionString  
    = "Session streamed by \"testH264VideoStreamer\"";  
  
  // RTSP服务器，端口为8554  
  RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);  
  if (rtspServer == NULL) {  
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";  
    exit(1);  
  }  
  
  // 组播  
  // Create 'groupsocks' for RTP and RTCP:  
  struct in_addr destinationAddress;  
  destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);  
  
  const unsigned short rtpPortNum = 18888;  
  const unsigned short rtcpPortNum = rtpPortNum+1;  
  const unsigned char ttl = 255;  
  
  const Port rtpPort(rtpPortNum);  
  const Port rtcpPort(rtcpPortNum);  
  
  Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);  
  rtpGroupsock.multicastSendOnly(); // we're a SSM source  
  Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);  
  rtcpGroupsock.multicastSendOnly(); // we're a SSM source  
  
  // Create a 'H264 Video RTP' sink from the RTP 'groupsock':  
  OutPacketBuffer::maxSize = 200000;  
  videoSink = H264VideoRTPSink::createNew(*env, &rtpGroupsock, 96);  
  
  // Create (and start) a 'RTCP instance' for this RTP sink:  
  const unsigned estimatedSessionBandwidth = 500; // in kbps; for RTCP b/w share  
  const unsigned maxCNAMElen = 100;  
  unsigned char CNAME[maxCNAMElen+1];  
  gethostname((char*)CNAME, maxCNAMElen);  
  CNAME[maxCNAMElen] = '\0'; // just in case  
  RTCPInstance* rtcp  
  = RTCPInstance::createNew(*env, &rtcpGroupsock,  
                estimatedSessionBandwidth, CNAME,  
                videoSink, NULL /* we're a server */,  
                True /* we're a SSM source */);  
  // Note: This starts RTCP running automatically  
  
  char const* streamName = "h264ESVideoMulticast";  
  ServerMediaSession* sms  
    = ServerMediaSession::createNew(*env, streamName, inputFileName,  
           descriptionString, True /*SSM*/);  
  sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));  
  rtspServer->addServerMediaSession(sms);  
  
  announceStream(rtspServer, sms, streamName, inputFileName);  
  
  // Start the streaming:  
  *env << "Beginning streaming...\n";  
  play(); // 播放  
  
    
  
  // 单播  
  {  
    char const* streamName = "h264ESVideo";  
    ServerMediaSession* sms  
      = ServerMediaSession::createNew(*env, streamName, streamName,  
                      descriptionString);  
    sms->addSubsession(H264VideoFileServerMediaSubsession  
               ::createNew(*env, inputFileName, reuseFirstSource));  
    rtspServer->addServerMediaSession(sms);  
  
    announceStream(rtspServer, sms, streamName, inputFileName);  
  }  
  
  env->taskScheduler().doEventLoop(); // does not return  
  
  return 0; // only to prevent compiler warning  
}  
  
// 继续读取文件  
void afterPlaying(void* /*clientData*/) {  
  *env << "...done reading from file\n";  
  videoSink->stopPlaying();  
  Medium::close(videoSource);  
  // Note that this also closes the input file that this source read from.  
  
  // Start playing once again:  
  play();  
}  
  
void play() {  
  // Open the input file as a 'byte-stream file source':  
  ByteStreamFileSource* fileSource  
    = ByteStreamFileSource::createNew(*env, inputFileName);  
  if (fileSource == NULL) {  
    *env << "Unable to open file \"" << inputFileName  
         << "\" as a byte-stream file source\n";  
    exit(1);  
  }  
  
  FramedSource* videoES = fileSource;  
  
  // Create a framer for the Video Elementary Stream:  
  videoSource = H264VideoStreamFramer::createNew(*env, videoES);  
  
  // Finally, start playing:  
  *env << "Beginning to read from file...\n";  
  videoSink->startPlaying(*videoSource, afterPlaying, videoSink);  
}  
  
void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,  
               char const* streamName, char const* inputFileName) {  
  char* url = rtspServer->rtspURL(sms);  
  UsageEnvironment& env = rtspServer->envir();  
  env << "\n\"" << streamName << "\" stream, from the file \""  
      << inputFileName << "\"\n";  
  env << "Play this stream using the URL \"" << url << "\"\n";  
  delete[] url;  
}  

二、概念介绍

然后，源码里有这样一段说明：

// Note: This is a multicast address.  If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.

翻译一下：

注意：这是多播地址。 如果你想改为流
使用单播，那么你应该使用“testOnDemandRTSPServer”
测试程序 - 不是这个测试程序 - 只是一个模型。

他这里出现了单播和多播的概念？我之前还查考过还有直播和点播，这些都要什么区别？

参看：LIVE555再学习 -- 单播、多播、广播、直播、点播 都是个啥？

三、实例

参看：使用Live555类库实现的网络直播系统

该代码中，指定了多播IP地址和端口号。