程序的main()在live555MediaServer.cpp中，在main()中调用了DynamicRTSPServer中的类

 

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 3 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
**********/
// Copyright (c) 1996-2020, Live Networks, Inc.  All rights reserved
// LIVE555 Media Server
// main program

#include <BasicUsageEnvironment.hh>
#include "DynamicRTSPServer.hh"
#include "version.hh"

int main(int argc, char** argv) {
  // Begin by setting up our usage environment:
  /************************************************************************
  TaskScheduler:
	实现时间的异步处理，时间处理函数的注册等
	它通过维护一个异步读取源对诸如通信消息到达等事件的处理。通过DelayQueue实现其他
	注册函数的延时调度。
  *************************************************************************/
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();

  
 /************************************************************************
 UsageEnvironment：用于消息的输入，输出以及用户的交互功能
 BasicUsageEnvironment：该模块是 UsageEnvironment的一个控制台应用的实现，针对
 输入输出的信号，相应的进行具体实现。
 *************************************************************************/
  UsageEnvironment* 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 <username>, <password> that you wish to allow
  // access to the server.
#endif

  //建立RTSP server,默认使用端口（554）
  // Create the RTSP server.  Try first with the default port number (554),
  // and then with the alternative port number (8554):
  RTSPServer* rtspServer;
  portNumBits rtspServerPortNum = 554;

  //创建rtspServer实例
  rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
  if (rtspServer == NULL) {
    rtspServerPortNum = 8554;
    rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
  }
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }

  //运算符重载
  *env << "LIVE555 Media Server\n";
  *env << "\tversion " << MEDIA_SERVER_VERSION_STRING
       << " (LIVE555 Streaming Media library version "
       << LIVEMEDIA_LIBRARY_VERSION_STRING << ").\n";

  char* urlPrefix = rtspServer->rtspURLPrefix();
  *env << "Play streams from this server using the URL\n\t"
       << urlPrefix << "<filename>\nwhere <filename> is a file present in the current directory.\n";
  *env << "Each file's type is inferred from its name suffix:\n";
  *env << "\t\".264\" => a H.264 Video Elementary Stream file\n";
  *env << "\t\".265\" => a H.265 Video Elementary Stream file\n";
  *env << "\t\".aac\" => an AAC Audio (ADTS format) file\n";
  *env << "\t\".ac3\" => an AC-3 Audio file\n";
  *env << "\t\".amr\" => an AMR Audio file\n";
  *env << "\t\".dv\" => a DV Video file\n";
  *env << "\t\".m4e\" => a MPEG-4 Video Elementary Stream file\n";
  *env << "\t\".mkv\" => a Matroska audio+video+(optional)subtitles file\n";
  *env << "\t\".mp3\" => a MPEG-1 or 2 Audio file\n";
  *env << "\t\".mpg\" => a MPEG-1 or 2 Program Stream (audio+video) file\n";
  *env << "\t\".ogg\" or \".ogv\" or \".opus\" => an Ogg audio and/or video file\n";
  *env << "\t\".ts\" => a MPEG Transport Stream file\n";
  *env << "\t\t(a \".tsx\" index file - if present - provides server 'trick play' support)\n";
  *env << "\t\".vob\" => a VOB (MPEG-2 video with AC-3 audio) file\n";
  *env << "\t\".wav\" => a WAV Audio file\n";
  *env << "\t\".webm\" => a WebM audio(Vorbis)+video(VP8) file\n";
  *env << "See http://www.live555.com/mediaServer/ for additional documentation.\n";

  // Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling.
  // Try first with the default HTTP port (80), and then with the alternative HTTP
  // port numbers (8000 and 8080).

  if (rtspServer->setUpTunnelingOverHTTP(80) || rtspServer->setUpTunnelingOverHTTP(8000) || rtspServer->setUpTunnelingOverHTTP(8080)) {
    *env << "(We use port " << rtspServer->httpServerPortNum() << " for optional RTSP-over-HTTP tunneling, or for HTTP live streaming (for indexed Transport Stream files only).)\n";
  } else {
    *env << "(RTSP-over-HTTP tunneling is not available.)\n";
  }
  
  //进入一个永久的循环
  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 3 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
**********/
// Copyright (c) 1996-2020, Live Networks, Inc.  All rights reserved
// A subclass of "RTSPServer" that creates "ServerMediaSession"s on demand,
// based on whether or not the specified stream name exists as a file
// Implementation

#include "DynamicRTSPServer.hh"
#include <liveMedia.hh>
#include <string.h>

DynamicRTSPServer*
DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,
			     UserAuthenticationDatabase* authDatabase,
			     unsigned reclamationTestSeconds) {
			     
  //建立TCP socket(socket(),bind(),listen()...)		     
  int ourSocket = setUpOurSocket(env, ourPort);
  if (ourSocket == -1) return NULL;

  return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds);
}

DynamicRTSPServer::DynamicRTSPServer(UsageEnvironment& env, int ourSocket,
				     Port ourPort,
				     UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds)
  : RTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds) {
}

DynamicRTSPServer::~DynamicRTSPServer() {
}

static ServerMediaSession* createNewSMS(UsageEnvironment& env,
					char const* fileName, FILE* fid); // forward


//查找ServerMediaSession (对应媒体服务器上的一个媒体文件或者设备)，如果没有的话就创建一个streamName
//例如：1080test.h264
ServerMediaSession* DynamicRTSPServer
::lookupServerMediaSession(char const* streamName, Boolean isFirstLookupInSession) {
  // First, check whether the specified "streamName" exists as a local file:
  //VLC播放:rtsp://192.168.35.249:8554/1080test.264,首先先判断一下指定的文件名1080test.264
  //是否存在一个本地的文件
  FILE* fid = fopen(streamName, "rb");

  //如果返回的文件指针不为空，则文件存在
  Boolean const fileExists = fid != NULL;

  // Next, check whether we already have a "ServerMediaSession" for this file:
  //下一步看看是否有这个ServerMediaSession
  ServerMediaSession* sms = RTSPServer::lookupServerMediaSession(streamName);
  Boolean const smsExists = sms != NULL;

  // Handle the four possibilities for "fileExists" and "smsExists":
  //如果文件没了但是ServerMediaSession还存在，就删除流媒体会话sms
  if (!fileExists) {
    if (smsExists) {
      // "sms" was created for a file that no longer exists. Remove it:
      removeServerMediaSession(sms);
      sms = NULL;
    }

    return NULL;
  } else {
  	//如果sms存在并且是第一次查找，删除流媒体会话sms
    if (smsExists && isFirstLookupInSession) { 
      // Remove the existing "ServerMediaSession" and create a new one, in case the underlying
      // file has changed in some way:
      removeServerMediaSession(sms); 
      sms = NULL;
    } 

	//流媒体会话sms不存在，则创建流媒体会话sms
    if (sms == NULL) {
      sms = createNewSMS(envir(), streamName, fid); 
      addServerMediaSession(sms);
    }

    fclose(fid);
    return sms;
  }
}

// Special code for handling Matroska files:
struct MatroskaDemuxCreationState {
  MatroskaFileServerDemux* demux;
  char watchVariable;
};
static void onMatroskaDemuxCreation(MatroskaFileServerDemux* newDemux, void* clientData) {
  MatroskaDemuxCreationState* creationState = (MatroskaDemuxCreationState*)clientData;
  creationState->demux = newDemux;
  creationState->watchVariable = 1;
}
// END Special code for handling Matroska files:

// Special code for handling Ogg files:
struct OggDemuxCreationState {
  OggFileServerDemux* demux;
  char watchVariable;
};
static void onOggDemuxCreation(OggFileServerDemux* newDemux, void* clientData) {
  OggDemuxCreationState* creationState = (OggDemuxCreationState*)clientData;
  creationState->demux = newDemux;
  creationState->watchVariable = 1;
}
// END Special code for handling Ogg files:

#define NEW_SMS(description) do {\
char const* descStr = description\
    ", streamed by the LIVE555 Media Server";\
sms = ServerMediaSession::createNew(env, fileName, fileName, descStr);\
} while(0)

	
//创建一个流媒体会话ServerMediaSession
static ServerMediaSession* createNewSMS(UsageEnvironment& env,
					char const* fileName, FILE* /*fid*/) {
  // Use the file name extension to determine the type of "ServerMediaSession":
  //获取扩展名，以“.”开始
  char const* extension = strrchr(fileName, '.');
  if (extension == NULL) return NULL;

  //定义变量
  ServerMediaSession* sms = NULL;
  Boolean const reuseSource = False;
  //判断视频类型：1080ptest.h264
  if (strcmp(extension, ".aac") == 0) {
    // Assumed to be an AAC Audio (ADTS format) file:
    NEW_SMS("AAC Audio");
    sms->addSubsession(ADTSAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".amr") == 0) {
    // Assumed to be an AMR Audio file:
    NEW_SMS("AMR Audio");
    sms->addSubsession(AMRAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".ac3") == 0) {
    // Assumed to be an AC-3 Audio file:
    NEW_SMS("AC-3 Audio");
    sms->addSubsession(AC3AudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".m4e") == 0) {
    // Assumed to be a MPEG-4 Video Elementary Stream file:
    NEW_SMS("MPEG-4 Video");
    sms->addSubsession(MPEG4VideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
	
  } else if (strcmp(extension, ".264") == 0) {
    // Assumed to be a H.264 Video Elementary Stream file:
	//调用ServerMediaSession::createNew（）还会调用MediaSubsession
    NEW_SMS("H.264 Video");
    OutPacketBuffer::maxSize = 100000; // allow for some possibly large H.264 frames
    sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".265") == 0) {
    // Assumed to be a H.265 Video Elementary Stream file:
    NEW_SMS("H.265 Video");
    OutPacketBuffer::maxSize = 100000; // allow for some possibly large H.265 frames
    sms->addSubsession(H265VideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".mp3") == 0) {
    // Assumed to be a MPEG-1 or 2 Audio file:
    NEW_SMS("MPEG-1 or 2 Audio");
    // To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
//#define STREAM_USING_ADUS 1
    // To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
    // (For more information about ADUs and interleaving,
    //  see <http://www.live555.com/rtp-mp3/>)
    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
    sms->addSubsession(MP3AudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, useADUs, interleaving));
  } else if (strcmp(extension, ".mpg") == 0) {
    // Assumed to be a MPEG-1 or 2 Program Stream (audio+video) file:
    NEW_SMS("MPEG-1 or 2 Program Stream");
    MPEG1or2FileServerDemux* demux
      = MPEG1or2FileServerDemux::createNew(env, fileName, reuseSource);
    sms->addSubsession(demux->newVideoServerMediaSubsession());
    sms->addSubsession(demux->newAudioServerMediaSubsession());
  } else if (strcmp(extension, ".vob") == 0) {
    // Assumed to be a VOB (MPEG-2 Program Stream, with AC-3 audio) file:
    NEW_SMS("VOB (MPEG-2 video with AC-3 audio)");
    MPEG1or2FileServerDemux* demux
      = MPEG1or2FileServerDemux::createNew(env, fileName, reuseSource);
    sms->addSubsession(demux->newVideoServerMediaSubsession());
    sms->addSubsession(demux->newAC3AudioServerMediaSubsession());
  } else if (strcmp(extension, ".ts") == 0) {
    // Assumed to be a MPEG Transport Stream file:
    // Use an index file name that's the same as the TS file name, except with ".tsx":
    unsigned indexFileNameLen = strlen(fileName) + 2; // allow for trailing "x\0"
    char* indexFileName = new char[indexFileNameLen];
    sprintf(indexFileName, "%sx", fileName);
    NEW_SMS("MPEG Transport Stream");
    sms->addSubsession(MPEG2TransportFileServerMediaSubsession::createNew(env, fileName, indexFileName, reuseSource));
    delete[] indexFileName;
  } else if (strcmp(extension, ".wav") == 0) {
    // Assumed to be a WAV Audio file:
    NEW_SMS("WAV Audio Stream");
    // To convert 16-bit PCM data to 8-bit u-law, prior to streaming,
    // change the following to True:
    Boolean convertToULaw = False;
    sms->addSubsession(WAVAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, convertToULaw));
  } else if (strcmp(extension, ".dv") == 0) {
    // Assumed to be a DV Video file
    // First, make sure that the RTPSinks' buffers will be large enough to handle the huge size of DV frames (as big as 288000).
    OutPacketBuffer::maxSize = 300000;

    NEW_SMS("DV Video");
    sms->addSubsession(DVVideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".mkv") == 0 || strcmp(extension, ".webm") == 0) {
    // Assumed to be a Matroska file (note that WebM ('.webm') files are also Matroska files)
    OutPacketBuffer::maxSize = 300000; // allow for some possibly large VP8 or VP9 frames
    NEW_SMS("Matroska video+audio+(optional)subtitles");

    // Create a Matroska file server demultiplexor for the specified file.
    // (We enter the event loop to wait for this to complete.)
    MatroskaDemuxCreationState creationState;
    creationState.watchVariable = 0;
    MatroskaFileServerDemux::createNew(env, fileName, onMatroskaDemuxCreation, &creationState);
    env.taskScheduler().doEventLoop(&creationState.watchVariable);

    ServerMediaSubsession* smss;
    while ((smss = creationState.demux->newServerMediaSubsession()) != NULL) {
      sms->addSubsession(smss);
    }
  } else if (strcmp(extension, ".ogg") == 0 || strcmp(extension, ".ogv") == 0 || strcmp(extension, ".opus") == 0) {
    // Assumed to be an Ogg file
    NEW_SMS("Ogg video and/or audio");

    // Create a Ogg file server demultiplexor for the specified file.
    // (We enter the event loop to wait for this to complete.)
    OggDemuxCreationState creationState;
    creationState.watchVariable = 0;
    OggFileServerDemux::createNew(env, fileName, onOggDemuxCreation, &creationState);
    env.taskScheduler().doEventLoop(&creationState.watchVariable);

    ServerMediaSubsession* smss;
    while ((smss = creationState.demux->newServerMediaSubsession()) != NULL) {
      sms->addSubsession(smss);
    }
  }

  return sms;
}