[35]_live.2018.04.25.tar.gz分析之LIVE555MediaServer.cpp和DynamicRTSPServer.cpp

简介：RTSP 在体系结构上位于 RTP 和 RTCP 之上， 其使用 TCP 或 UDP 完成数据的传输； HTTP 与 RTSP 相比， HTTP 请求由客户机发出， 服务器作出响应， 使用 RTSP 时， 客户机和服务器都可以发出请求， 即RTSP 可以是双向的； RTSP 是用来控制声音或影像多媒体串流协议， 并允许同时多个串流需求控制， 传输时所用的网络通信协定并不在其定义范围内。 RTSP 协议默认端口： 554， 默认承载协议为 TCP。

1.使用source insight建立工程 (这里就略过了...)

2.直接上代码：// LIVE555MediaServer.cpp

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

int main(int argc, char** argv) {
  //一：(准备)开始设置使用前环境
  // Begin by setting up our usage environment:
  //1.1 TaskScheduler用于任务      
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();
  //1.2 UsageEnvironment用于输出
  UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);

  UserAuthenticationDatabase* authDB = NULL;
#ifdef ACCESS_CONTROL
  //1.3 实现客户端访问RTSP服务器权限控制(即客户端通过密码才能访问服务器)
  // 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

  // Create the RTSP server.  Try first with the default port number (554),
  // and then with the alternative port number (8554):
  //二：建立 RTSPServer、使用默认端口(554)
  RTSPServer* rtspServer;
  portNumBits rtspServerPortNum = 554;
  //三：创建rtspServer实例
  rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
  if (rtspServer == NULL) { 
  	//3.1 重新设置端口号8554、再次创建rtspServer实例
    rtspServerPortNum = 8554;
    rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
  }
  if (rtspServer == NULL) { 
  	//3.2 如果再次创建rtspServer实例失败，打印输出信息，并退出程序
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }
  //四：rtspServer创建成功，打印输出，譬如Server名称、版本号、版本更新日期
  *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" //客户端连接服务器的使用方法
       << urlPrefix << "<filename>\nwhere <filename> is a file present in the current directory.\n"; //下面是访问支持的文件类型
  *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";
  }
  //五：进入一个循环
  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}

追进去看createNew，这部分后续补充......

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

DynamicRTSPServer*
DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,
			     UserAuthenticationDatabase* authDatabase,
			     unsigned reclamationTestSeconds) {
  //一：创建socket网络文件描述符ourSocket (底层还应该包括bind(...)、listen(...))
  int ourSocket = setUpOurSocket(env, ourPort); 
  if (ourSocket == -1) return NULL; //检错机制

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

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

DynamicRTSPServer::~DynamicRTSPServer() {
}

static ServerMediaSession* createNewSMS(UsageEnvironment& env,
					char const* fileName, FILE* fid); // forward
					
//二：查找服务器上一个媒体文件并打开等一系列的处理机制
ServerMediaSession* DynamicRTSPServer
::lookupServerMediaSession(char const* streamName, Boolean isFirstLookupInSession) {
  // First, check whether the specified "streamName" exists as a local file:
  FILE* fid = fopen(streamName, "rb");
  //2.1 如果返回文件指针不为空，则被打开的streamName存在
  Boolean fileExists = fid != NULL;  

  // Next, check whether we already have a "ServerMediaSession" for this file:
  //2.2 检查是否存在ServerMediaSession
  ServerMediaSession* sms = RTSPServer::lookupServerMediaSession(streamName);
  Boolean smsExists = sms != NULL;

  // Handle the four possibilities for "fileExists" and "smsExists":
  //2.3 fileExists不存在，ServerMediaSession存在，则删之
  if (!fileExists) {
    if (smsExists) {
      // "sms" was created for a file that no longer exists. Remove it:
      removeServerMediaSession(sms);
      sms = NULL;
    }

    return NULL;
  } 
  else {
  	//2.4 两者同时存在，则删除ServerMediaSession，防止通过其它方式改变存在的文件
    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;
    } 

    if (sms == NULL) {
	  //2.5 添加ServerMediaSession
      sms = createNewSMS(envir(), streamName, fid); 
      addServerMediaSession(sms);
    }
	//关闭网络文件描述符fid
    fclose(fid);
    return sms;
  }
}

//下面是处理Matroska、ogg文件的专用代码，见识有限，不知道什么是Matroska、ogg
// 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)

static ServerMediaSession* createNewSMS(UsageEnvironment& env,
					char const* fileName, FILE* /*fid*/) {
  // Use the file name extension to determine the type of "ServerMediaSession":
  //三：获取文件扩展名，以"."开始
  char const* extension = strrchr(fileName, '.');
  if (extension == NULL) return NULL;

  ServerMediaSession* sms = NULL;
  Boolean const reuseSource = False;
  //3.1 由文件后缀名确定

流媒体文件  的类型，然后createNew去处理，后面流程基本上是这样一个过程

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: 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 = 100000; // 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;}总结：参考博客上的一些文章辅助分析，可能是因为版本差异，代码上还是有很多的地方明显不一样，代码经过了多次的迭代...还好，分析起来不算难，但是细节地方挺多的....