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 <BasicUsageEnvironment.hh>
#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 <username>, <password> 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 << "<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\".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 returnreturn 0; // only to prevent compiler warning
}

DynamicRTSPServer.cpp：

#include "DynamicRTSPServer.hh"
#include <liveMedia.hh>
#include <string.h>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.avi
ServerMediaSession*
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ServerMediaSessionServerMediaSession* 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。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MediaSubsessionNEW_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 <http://www.live555.com/rtp-mp3/>)Boolean useADUs = False;Interleaving* interleaving = NULL;
#ifdef STREAM_USING_ADUSuseADUs = True;
#ifdef INTERLEAVE_ADUSunsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...unsigned const interleaveCycleSize= (sizeof interleaveCycle)/(sizeof (unsigned char));interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
#endif
#endifsms->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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