live555音视频同步

live555音视频同步

通过testOnDemandRTSPServer.cpp实现，仅需修改部分代码即可，

具体思路：

流式传输音频+视频很简单：只需调用“addSubsession()”两次 - 一次用于视频源，另一次用于音频源。（这样你的“ServerMediaSession”对象包含两个“ServerMediaSubsession”对象 - 一个用于视频，一个用于音频。）

但是，为了音频/视频同步正常工作，每个源（视频和音频）必须生成正确的“每个帧的“fPresentationTime”值，并且这些“必须”与“挂钟”时间对齐 - 即，与调用“gettimeofday()”时获得的时间相同。

注意

测试用的h264和mp3尽量从同一个mp4上进行提取。以验证同步问题。

提取具体操作：

Mp4文件提取详细H.264和MP3文件_泷fyk的博客-CSDN博客

现成的我已提取好的文件：

音视频同步h264-live555(水龙吟)-直播技术文档类资源-CSDN文库

音视频同步测试mp3-live555(水龙吟)资源-CSDN文库

具体代码（对testOnDemandRTSPServer.cpp进行修改）

#include "liveMedia.hh"

#include "BasicUsageEnvironment.hh"
#include "announceURL.hh"

UsageEnvironment* env;

// To make the second and subsequent client for each stream reuse the same
// input stream as the first client (rather than playing the file from the
// start for each client), change the following "False" to "True":
//单播、组播开关，false为单播，true为组播，单播每个链接接入时都会重新播放，组播的话会随着上一个接入流的位置进行播放
//经过实测，组播模式各链接同步流畅，延迟很小。
Boolean reuseFirstSource = false;

// To stream *only* MPEG-1 or 2 video "I" frames
// (e.g., to reduce network bandwidth),
// change the following "False" to "True":
Boolean iFramesOnly = False;

static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
			   char const* streamName, char const* inputFileName); // forward

int main(int argc, char** argv) {
  // Begin by setting up our usage environment:
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();
  env = BasicUsageEnvironment::createNew(*scheduler);

  UserAuthenticationDatabase* authDB = NULL;
#ifdef ACCESS_CONTROL
  // To implement client access control to the RTSP server, do the following:
  authDB = new UserAuthenticationDatabase;
  authDB->addUserRecord("username1", "password1"); // replace these with real strings
  // Repeat the above with each ,  that you wish to allow
  // access to the server.
#endif

  // Create the RTSP server:
#ifdef SERVER_USE_TLS
  // Serve RTSPS: RTSP over a TLS connection:
  RTSPServer* rtspServer = RTSPServer::createNew(*env, 322, authDB);
#else
  // Serve regular RTSP (over a TCP connection):
  RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
#endif
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }
#ifdef SERVER_USE_TLS
#ifndef STREAM_USING_SRTP
#define STREAM_USING_SRTP True
#endif
  rtspServer->setTLSState(PATHNAME_TO_CERTIFICATE_FILE, PATHNAME_TO_PRIVATE_KEY_FILE,
			  STREAM_USING_SRTP);
#endif

  char const* descriptionString
    = "Session streamed by \"testOnDemandRTSPServer\"";

  // Set up each of the possible streams that can be served by the
  // RTSP server.  Each such stream is implemented using a
  // "ServerMediaSession" object, plus one or more
  // "ServerMediaSubsession" objects for each audio/video substream.

  // A H.264 video elementary stream:
  //在此处进行修改
  {
    //输出文件名称，按需求修改
    char const* streamName = "h264ESVideoTest";
    //h264输入文件所在位置，按实际修改
    char const* inputFileName = "./test/output_video.264";
    //mp3输入文件名称所在位置，按实际修改
    char const* inputFileNameAdus = "./test/output_audio.mp3";
    //创建ServerMediaSession
    //下面这几行定义了两个变量，useADUs 和 interleaving，用于配置MP3音频流的传输方式。useADUs 设置为False 表示不使用ADUs（Audio Data Units），即不进行分组音频数据的传输。interleaving 初始化为NULL，表示不进行数据交错。
    ServerMediaSession* sms
      = ServerMediaSession::createNew(*env, streamName, streamName,
				      descriptionString);
              Boolean useADUs = False;
    Interleaving* interleaving = NULL;
#ifdef STREAM_USING_ADUS
    useADUs = True;
#ifdef INTERLEAVE_ADUS
    unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
    unsigned const interleaveCycleSize
      = (sizeof interleaveCycle)/(sizeof (unsigned char));
    interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
#endif
#endif
    //添加音频子会话
    sms->addSubsession(MP3AudioFileServerMediaSubsession
		       ::createNew(*env, inputFileNameAdus, reuseFirstSource,
				   useADUs, interleaving));
    //添加视频子会话
    sms->addSubsession(H264VideoFileServerMediaSubsession
		       ::createNew(*env, inputFileName, reuseFirstSource));
    //将创建的ServerMediaSession* sms添加到rtsp服务器。
    rtspServer->addServerMediaSession(sms);
	//
    announceStream(rtspServer, sms, streamName, inputFileName);
  }
	
  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}

static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
			   char const* streamName, char const* inputFileName) {
  UsageEnvironment& env = rtspServer->envir();

  env << "\n\"" << streamName << "\" stream, from the file \""
      << inputFileName << "\"\n";
  announceURL(rtspServer, sms);
}

修改之后成功在所在目录下直接分步执行以下命令

make
./testOnDemandRTSPServer

windowsVLC接到rtsp流

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