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 return return 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 <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, ".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;
}
