RTSP服务运作
基础基本搞明白了,那么RTSP,RTP等这些协议又是如何利用这些基础机制运作的呢?
首先来看RTSP.
RTSP首先需建立TCP侦听socket。可见于此函数:
要帧听客户端的连接,就需要利用任务调度机制了,所以需添加一个socket handler。可见于此函数:
下面重点讲一下下RTSPClientSession响应DESCRIBE请求的过程:
fOurServer.lookupServerMediaSession(urlTotalSuffix)中会在找不到同名ServerMediaSession时新建一个,代表一个RTP流的ServerMediaSession们是被RTSPServer管理的,而不是被RTSPClientSession拥有。为什么呢?因为ServerMediaSession代表的是一个静态的流,也就是可以从它里面获取一个流的各种信息,但不能获取传输状态。不同客户可能连接到同一个流,所以ServerMediaSession应被RTSPServer所拥有。创建一个ServerMediaSession过程值得一观:
可以看到NEW_SMS("AMR Audio")会创建新的ServerMediaSession,之后马上调用sms->addSubsession()为这个ServerMediaSession添加一个 ServerMediaSubSession 。看起来ServerMediaSession应该可以添加多个ServerMediaSubSession,但这里并没有这样做。如果可以添加多个 ServerMediaSubsession 那么ServerMediaSession与流名字所指定与文件是没有关系的,也就是说它不会操作文件,而文件的操作是放在 ServerMediaSubsession中的。具体应改是在ServerMediaSubsession的sdpLines()函数中打开。
基础基本搞明白了,那么RTSP,RTP等这些协议又是如何利用这些基础机制运作的呢?
首先来看RTSP.
RTSP首先需建立TCP侦听socket。可见于此函数:
- DynamicRTSPServer* DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,
- UserAuthenticationDatabase* authDatabase,
- unsigned reclamationTestSeconds) {
- int ourSocket = setUpOurSocket(env, ourPort); //建立TCP socket
- if (ourSocket == -1)
- return NULL;
- return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase,
- reclamationTestSeconds);
- }
- DynamicRTSPServer* DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,
- UserAuthenticationDatabase* authDatabase,
- unsigned reclamationTestSeconds) {
- int ourSocket = setUpOurSocket(env, ourPort); //建立TCP socket
- if (ourSocket == -1)
- return NULL;
- return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase,
- reclamationTestSeconds);
- }
要帧听客户端的连接,就需要利用任务调度机制了,所以需添加一个socket handler。可见于此函数:
- RTSPServer::RTSPServer(UsageEnvironment& env,
- int ourSocket,
- Port ourPort,
- UserAuthenticationDatabase* authDatabase,
- unsigned reclamationTestSeconds) :
- Medium(env),
- fRTSPServerSocket(ourSocket),
- fRTSPServerPort(ourPort),
- fHTTPServerSocket(-1),
- fHTTPServerPort(0),
- fClientSessionsForHTTPTunneling(NULL),
- fAuthDB(authDatabase),
- fReclamationTestSeconds(reclamationTestSeconds),
- fServerMediaSessions(HashTable::create(STRING_HASH_KEYS))
- {
- #ifdef USE_SIGNALS
- // Ignore the SIGPIPE signal, so that clients on the same host that are killed
- // don't also kill us:
- signal(SIGPIPE, SIG_IGN);
- #endif
- // Arrange to handle connections from others:
- env.taskScheduler().turnOnBackgroundReadHandling(
- fRTSPServerSocket,
- (TaskScheduler::BackgroundHandlerProc*) &incomingConnectionHandlerRTSP,
- this);
- }
- RTSPServer::RTSPServer(UsageEnvironment& env,
- int ourSocket,
- Port ourPort,
- UserAuthenticationDatabase* authDatabase,
- unsigned reclamationTestSeconds) :
- Medium(env),
- fRTSPServerSocket(ourSocket),
- fRTSPServerPort(ourPort),
- fHTTPServerSocket(-1),
- fHTTPServerPort(0),
- fClientSessionsForHTTPTunneling(NULL),
- fAuthDB(authDatabase),
- fReclamationTestSeconds(reclamationTestSeconds),
- fServerMediaSessions(HashTable::create(STRING_HASH_KEYS))
- {
- #ifdef USE_SIGNALS
- // Ignore the SIGPIPE signal, so that clients on the same host that are killed
- // don't also kill us:
- signal(SIGPIPE, SIG_IGN);
- #endif
- // Arrange to handle connections from others:
- env.taskScheduler().turnOnBackgroundReadHandling(
- fRTSPServerSocket,
- (TaskScheduler::BackgroundHandlerProc*) &incomingConnectionHandlerRTSP,
- this);
- }
当收到客户的连接时需保存下代表客户端的新socket,以后用这个socket与这个客户通讯。每个客户将来会对应一个rtp会话,而且各客户的RTSP请求只控制自己的rtp会话,那么最好建立一个会话类,代表各客户的rtsp会话。于是类RTSPServer::RTSPClientSession产生,它保存的代表客户的socket。下为RTSPClientSession的创建过程
- void RTSPServer::incomingConnectionHandler(int serverSocket)
- {
- struct sockaddr_in clientAddr;
- SOCKLEN_T clientAddrLen = sizeof clientAddr;
- //接受连接
- int clientSocket = accept(serverSocket,
- (struct sockaddr*) &clientAddr,
- &clientAddrLen);
- if (clientSocket < 0) {
- int err = envir().getErrno();
- if (err != EWOULDBLOCK) {
- envir().setResultErrMsg("accept() failed: ");
- }
- return;
- }
- //设置socket的参数
- makeSocketNonBlocking(clientSocket);
- increaseSendBufferTo(envir(), clientSocket, 50 * 1024);
- #ifdef DEBUG
- envir() << "accept()ed connection from " << our_inet_ntoa(clientAddr.sin_addr) << "\n";
- #endif
- //产生一个sesson id
- // Create a new object for this RTSP session.
- // (Choose a random 32-bit integer for the session id (it will be encoded as a 8-digit hex number). We don't bother checking for
- // a collision; the probability of two concurrent sessions getting the same session id is very low.)
- // (We do, however, avoid choosing session id 0, because that has a special use (by "OnDemandServerMediaSubsession").)
- unsigned sessionId;
- do {
- sessionId = (unsigned) our_random();
- } while (sessionId == 0);
- //创建RTSPClientSession,注意传入的参数
- (void) createNewClientSession(sessionId, clientSocket, clientAddr);
- }
- void RTSPServer::incomingConnectionHandler(int serverSocket)
- {
- struct sockaddr_in clientAddr;
- SOCKLEN_T clientAddrLen = sizeof clientAddr;
- //接受连接
- int clientSocket = accept(serverSocket,
- (struct sockaddr*) &clientAddr,
- &clientAddrLen);
- if (clientSocket < 0) {
- int err = envir().getErrno();
- if (err != EWOULDBLOCK) {
- envir().setResultErrMsg("accept() failed: ");
- }
- return;
- }
- //设置socket的参数
- makeSocketNonBlocking(clientSocket);
- increaseSendBufferTo(envir(), clientSocket, 50 * 1024);
- #ifdef DEBUG
- envir() << "accept()ed connection from " << our_inet_ntoa(clientAddr.sin_addr) << "\n";
- #endif
- //产生一个sesson id
- // Create a new object for this RTSP session.
- // (Choose a random 32-bit integer for the session id (it will be encoded as a 8-digit hex number). We don't bother checking for
- // a collision; the probability of two concurrent sessions getting the same session id is very low.)
- // (We do, however, avoid choosing session id 0, because that has a special use (by "OnDemandServerMediaSubsession").)
- unsigned sessionId;
- do {
- sessionId = (unsigned) our_random();
- } while (sessionId == 0);
- //创建RTSPClientSession,注意传入的参数
- (void) createNewClientSession(sessionId, clientSocket, clientAddr);
- }
RTSPClientSession要提供什么功能呢?可以想象:需要监听客户端的rtsp请求并回应它,需要在DESCRIBE请求中返回所请求的流的信息,需要在SETUP请求中建立起RTP会话,需要在TEARDOWN请求中关闭RTP会话,等等...
RTSPClientSession要侦听客户端的请求,就需把自己的socket handler加入计划任务。证据如下:- RTSPServer::RTSPClientSession::RTSPClientSession(
- RTSPServer& ourServer,
- unsigned sessionId,
- int clientSocket,
- struct sockaddr_in clientAddr) :
- fOurServer(ourServer),
- fOurSessionId(sessionId),
- fOurServerMediaSession(NULL),
- fClientInputSocket(clientSocket),
- fClientOutputSocket(clientSocket),
- fClientAddr(clientAddr),
- fSessionCookie(NULL),
- fLivenessCheckTask(NULL),
- fIsMulticast(False),
- fSessionIsActive(True),
- fStreamAfterSETUP(False),
- fTCPStreamIdCount(0),
- fNumStreamStates(0),
- fStreamStates(NULL),
- fRecursionCount(0)
- {
- // Arrange to handle incoming requests:
- resetRequestBuffer();
- envir().taskScheduler().turnOnBackgroundReadHandling(fClientInputSocket,
- (TaskScheduler::BackgroundHandlerProc*) &incomingRequestHandler,
- this);
- noteLiveness();
- }
- RTSPServer::RTSPClientSession::RTSPClientSession(
- RTSPServer& ourServer,
- unsigned sessionId,
- int clientSocket,
- struct sockaddr_in clientAddr) :
- fOurServer(ourServer),
- fOurSessionId(sessionId),
- fOurServerMediaSession(NULL),
- fClientInputSocket(clientSocket),
- fClientOutputSocket(clientSocket),
- fClientAddr(clientAddr),
- fSessionCookie(NULL),
- fLivenessCheckTask(NULL),
- fIsMulticast(False),
- fSessionIsActive(True),
- fStreamAfterSETUP(False),
- fTCPStreamIdCount(0),
- fNumStreamStates(0),
- fStreamStates(NULL),
- fRecursionCount(0)
- {
- // Arrange to handle incoming requests:
- resetRequestBuffer();
- envir().taskScheduler().turnOnBackgroundReadHandling(fClientInputSocket,
- (TaskScheduler::BackgroundHandlerProc*) &incomingRequestHandler,
- this);
- noteLiveness();
- }
下面重点讲一下下RTSPClientSession响应DESCRIBE请求的过程:
- void RTSPServer::RTSPClientSession::handleCmd_DESCRIBE(
- char const* cseq,
- char const* urlPreSuffix,
- char const* urlSuffix,
- char const* fullRequestStr)
- {
- char* sdpDescription = NULL;
- char* rtspURL = NULL;
- do {
- //整理一下下RTSP地址
- char urlTotalSuffix[RTSP_PARAM_STRING_MAX];
- if (strlen(urlPreSuffix) + strlen(urlSuffix) + 2
- > sizeof urlTotalSuffix) {
- handleCmd_bad(cseq);
- break;
- }
- urlTotalSuffix[0] = '\0';
- if (urlPreSuffix[0] != '\0') {
- strcat(urlTotalSuffix, urlPreSuffix);
- strcat(urlTotalSuffix, "/");
- }
- strcat(urlTotalSuffix, urlSuffix);
- //验证帐户和密码
- if (!authenticationOK("DESCRIBE", cseq, urlTotalSuffix, fullRequestStr))
- break;
- // We should really check that the request contains an "Accept:" #####
- // for "application/sdp", because that's what we're sending back #####
- // Begin by looking up the "ServerMediaSession" object for the specified "urlTotalSuffix":
- //跟据流的名字查找ServerMediaSession,如果找不到,会创建一个。每个ServerMediaSession中至少要包含一个
- //ServerMediaSubsession。一个ServerMediaSession对应一个媒体,可以认为是Server上的一个文件,或一个实时获取设备。其包含的每个ServerMediaSubSession代表媒体中的一个Track。所以一个ServerMediaSession对应一个媒体,如果客户请求的媒体名相同,就使用已存在的ServerMediaSession,如果不同,就创建一个新的。一个流对应一个StreamState,StreamState与ServerMediaSubsession相关,但代表的是动态的,而ServerMediaSubsession代表静态的。
- ServerMediaSession* session = fOurServer.lookupServerMediaSession(urlTotalSuffix);
- if (session == NULL) {
- handleCmd_notFound(cseq);
- break;
- }
- // Then, assemble a SDP description for this session:
- //获取SDP字符串,在函数内会依次获取每个ServerMediaSubSession的字符串然连接起来。
- sdpDescription = session->generateSDPDescription();
- if (sdpDescription == NULL) {
- // This usually means that a file name that was specified for a
- // "ServerMediaSubsession" does not exist.
- snprintf((char*) fResponseBuffer, sizeof fResponseBuffer,
- "RTSP/1.0 404 File Not Found, Or In Incorrect Format\r\n"
- "CSeq: %s\r\n"
- "%s\r\n", cseq, dateHeader());
- break;
- }
- unsigned sdpDescriptionSize = strlen(sdpDescription);
- // Also, generate our RTSP URL, for the "Content-Base:" header
- // (which is necessary to ensure that the correct URL gets used in
- // subsequent "SETUP" requests).
- rtspURL = fOurServer.rtspURL(session, fClientInputSocket);
- //形成响应DESCRIBE请求的RTSP字符串。
- snprintf((char*) fResponseBuffer, sizeof fResponseBuffer,
- "RTSP/1.0 200 OK\r\nCSeq: %s\r\n"
- "%s"
- "Content-Base: %s/\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: %d\r\n\r\n"
- "%s", cseq, dateHeader(), rtspURL, sdpDescriptionSize,
- sdpDescription);
- } while (0);
- delete[] sdpDescription;
- delete[] rtspURL;
- //返回后会被立即发送(没有把socket write操作放入计划任务中)。
- }
- void RTSPServer::RTSPClientSession::handleCmd_DESCRIBE(
- char const* cseq,
- char const* urlPreSuffix,
- char const* urlSuffix,
- char const* fullRequestStr)
- {
- char* sdpDescription = NULL;
- char* rtspURL = NULL;
- do {
- //整理一下下RTSP地址
- char urlTotalSuffix[RTSP_PARAM_STRING_MAX];
- if (strlen(urlPreSuffix) + strlen(urlSuffix) + 2
- > sizeof urlTotalSuffix) {
- handleCmd_bad(cseq);
- break;
- }
- urlTotalSuffix[0] = '\0';
- if (urlPreSuffix[0] != '\0') {
- strcat(urlTotalSuffix, urlPreSuffix);
- strcat(urlTotalSuffix, "/");
- }
- strcat(urlTotalSuffix, urlSuffix);
- //验证帐户和密码
- if (!authenticationOK("DESCRIBE", cseq, urlTotalSuffix, fullRequestStr))
- break;
- // We should really check that the request contains an "Accept:" #####
- // for "application/sdp", because that's what we're sending back #####
- // Begin by looking up the "ServerMediaSession" object for the specified "urlTotalSuffix":
- //跟据流的名字查找ServerMediaSession,如果找不到,会创建一个。每个ServerMediaSession中至少要包含一个
- //ServerMediaSubsession。一个ServerMediaSession对应一个媒体,可以认为是Server上的一个文件,或一个实时获取设备。其包含的每个ServerMediaSubSession代表媒体中的一个Track。所以一个ServerMediaSession对应一个媒体,如果客户请求的媒体名相同,就使用已存在的ServerMediaSession,如果不同,就创建一个新的。一个流对应一个StreamState,StreamState与ServerMediaSubsession相关,但代表的是动态的,而ServerMediaSubsession代表静态的。
- ServerMediaSession* session = fOurServer.lookupServerMediaSession(urlTotalSuffix);
- if (session == NULL) {
- handleCmd_notFound(cseq);
- break;
- }
- // Then, assemble a SDP description for this session:
- //获取SDP字符串,在函数内会依次获取每个ServerMediaSubSession的字符串然连接起来。
- sdpDescription = session->generateSDPDescription();
- if (sdpDescription == NULL) {
- // This usually means that a file name that was specified for a
- // "ServerMediaSubsession" does not exist.
- snprintf((char*) fResponseBuffer, sizeof fResponseBuffer,
- "RTSP/1.0 404 File Not Found, Or In Incorrect Format\r\n"
- "CSeq: %s\r\n"
- "%s\r\n", cseq, dateHeader());
- break;
- }
- unsigned sdpDescriptionSize = strlen(sdpDescription);
- // Also, generate our RTSP URL, for the "Content-Base:" header
- // (which is necessary to ensure that the correct URL gets used in
- // subsequent "SETUP" requests).
- rtspURL = fOurServer.rtspURL(session, fClientInputSocket);
- //形成响应DESCRIBE请求的RTSP字符串。
- snprintf((char*) fResponseBuffer, sizeof fResponseBuffer,
- "RTSP/1.0 200 OK\r\nCSeq: %s\r\n"
- "%s"
- "Content-Base: %s/\r\n"
- "Content-Type: application/sdp\r\n"
- "Content-Length: %d\r\n\r\n"
- "%s", cseq, dateHeader(), rtspURL, sdpDescriptionSize,
- sdpDescription);
- } while (0);
- delete[] sdpDescription;
- delete[] rtspURL;
- //返回后会被立即发送(没有把socket write操作放入计划任务中)。
- }
fOurServer.lookupServerMediaSession(urlTotalSuffix)中会在找不到同名ServerMediaSession时新建一个,代表一个RTP流的ServerMediaSession们是被RTSPServer管理的,而不是被RTSPClientSession拥有。为什么呢?因为ServerMediaSession代表的是一个静态的流,也就是可以从它里面获取一个流的各种信息,但不能获取传输状态。不同客户可能连接到同一个流,所以ServerMediaSession应被RTSPServer所拥有。创建一个ServerMediaSession过程值得一观:
- static ServerMediaSession* createNewSMS(UsageEnvironment& env,char const* fileName, FILE* /*fid*/)
- {
- // Use the file name extension to determine the type of "ServerMediaSession":
- 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:
- 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, ".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));
- } else if (strcmp(extension, ".mkv") == 0) {
- // Assumed to be a Matroska file
- 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.)
- newMatroskaDemuxWatchVariable = 0;
- MatroskaFileServerDemux::createNew(env, fileName,
- onMatroskaDemuxCreation, NULL);
- env.taskScheduler().doEventLoop(&newMatroskaDemuxWatchVariable);
- ServerMediaSubsession* smss;
- while ((smss = demux->newServerMediaSubsession()) != NULL) {
- sms->addSubsession(smss);
- }
- }
- return sms;
- }
- static ServerMediaSession* createNewSMS(UsageEnvironment& env,char const* fileName, FILE* /*fid*/)
- {
- // Use the file name extension to determine the type of "ServerMediaSession":
- 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:
- 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, ".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));
- } else if (strcmp(extension, ".mkv") == 0) {
- // Assumed to be a Matroska file
- 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.)
- newMatroskaDemuxWatchVariable = 0;
- MatroskaFileServerDemux::createNew(env, fileName,
- onMatroskaDemuxCreation, NULL);
- env.taskScheduler().doEventLoop(&newMatroskaDemuxWatchVariable);
- ServerMediaSubsession* smss;
- while ((smss = demux->newServerMediaSubsession()) != NULL) {
- sms->addSubsession(smss);
- }
- }
- return sms;
- }
可以看到NEW_SMS("AMR Audio")会创建新的ServerMediaSession,之后马上调用sms->addSubsession()为这个ServerMediaSession添加一个 ServerMediaSubSession 。看起来ServerMediaSession应该可以添加多个ServerMediaSubSession,但这里并没有这样做。如果可以添加多个 ServerMediaSubsession 那么ServerMediaSession与流名字所指定与文件是没有关系的,也就是说它不会操作文件,而文件的操作是放在 ServerMediaSubsession中的。具体应改是在ServerMediaSubsession的sdpLines()函数中打开。