live555 源代码简单分析1 主程序

               

live555是使用十分广泛的开源流媒体服务器，之前也看过其他人写的live555的学习笔记，在这里自己简单总结下。

live555源代码有以下几个明显的特点：

1.头文件是.hh后缀的，但没觉得和.h后缀的有什么不同

2.采用了面向对象的程序设计思路，里面各种对象

 

好了，不罗嗦，使用vc2010打开live555的vc工程，看到live555源代码结构如下：

源代码由5个工程构成（4个库和一个主程序）：

libUsageEnvironment.lib；libliveMedia.lib；libgroupsock.lib；libBasicUsageEnvironment.lib；以及live555MediaServer

这里我们只分析live555MediaServer这个主程序，其实代码量并不大，主要有两个CPP：DynamicRTSPServer.cpp和live555MediaServer.cpp

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

 

不废话，直接贴上有注释的源码

live555MediaServer.cpp：

#include #include "DynamicRTSPServer.hh"#include "version.hh"int main(int argc, char** argv) {  // Begin by setting up our usage environment:  // TaskScheduler用于任务计划  TaskScheduler* scheduler = BasicTaskScheduler::createNew();  // UsageEnvironment用于输出  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 ,  that you wish to allow  // access to the server.#endif  //建立 RTSP server.  使用默认端口 (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);  }  //用到了运算符重载  *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 << "\nwhere  is a file present in the current directory.\n";  *env << "Each file's type is inferred from its name suffix:\n";  *env << "\t\".aac\" => an AAC Audio (ADTS format) file\n";  *env << "\t\".amr\" => an AMR Audio file\n";  *env << "\t\".m4e\" => a MPEG-4 Video Elementary Stream file\n";  *env << "\t\".dv\" => a DV Video 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\".ts\" => a MPEG Transport Stream file\n";  *env << "\t\t(a \".tsx\" index file - if present - provides server 'trick play' support)\n";  *env << "\t\".wav\" => a WAV Audio 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.)\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}

DynamicRTSPServer.cpp：

#include "DynamicRTSPServer.hh"#include #include DynamicRTSPServer*DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,        UserAuthenticationDatabase* authDatabase,        unsigned reclamationTestSeconds) {  int ourSocket = -1;  do { //建立TCP socket(socket(),bind(),listen()...)    int ourSocket = setUpOurSocket(env, ourPort);    if (ourSocket == -1) break;    return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds);  } while (0);  if (ourSocket != -1) ::closeSocket(ourSocket);  return NULL;}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（对应服务器上一个媒体文件，，或设备），如果没有的话就创建一个//streamName例:A.aviServerMediaSession*DynamicRTSPServer::lookupServerMediaSession(char const* streamName) {  // First, check whether the specified "streamName" exists as a local file:  FILE* fid = fopen(streamName, "rb");  //如果返回文件指针不为空，则文件存在  Boolean fileExists = fid != NULL;  // Next, check whether we already have a "ServerMediaSession" for this file:  //看看是否有这个ServerMediaSession  ServerMediaSession* sms = RTSPServer::lookupServerMediaSession(streamName);  Boolean smsExists = sms != NULL;  // Handle the four possibilities for "fileExists" and "smsExists":  //文件没了，ServerMediaSession有，删之  if (!fileExists) {    if (smsExists) {      // "sms" was created for a file that no longer exists. Remove it:      removeServerMediaSession(sms);    }    return NULL;  } else { //文件有，ServerMediaSession无，加之    if (!smsExists) {      // Create a new "ServerMediaSession" object for streaming from the named file.      sms = createNewSMS(envir(), streamName, fid);      addServerMediaSession(sms);    }    fclose(fid);    return sms;  }}#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ServerMediaSessionstatic 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。不严密，万一文件名有多个点？  char const* extension = strrchr(fileName, '.');  if (extension == NULL) return NULL;  ServerMediaSession* sms = NULL;  Boolean const reuseSource = False;  if (strcmp(extension, ".aac") == 0) {    // Assumed to be an AAC Audio (ADTS format) file: // 调用ServerMediaSession::createNew（） //还会调用MediaSubsession    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, ".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, ".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 )    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, ".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));  }  return sms;}

 live555源代码（VC6）：http://download.csdn.net/detail/leixiaohua1020/6374387

           

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