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