void shutdownStream(RTSPClient* rtspClient, int exitCode) {
(void)exitCode;
// UsageEnvironment& env = rtspClient->envir(); // alias
StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
int channel = ((ourRTSPClient*)rtspClient)->_channel;
while(pthread_mutex_lock(&g_context[channel].mutex) != 0 && errno == EINTR){};
// First, check whether any subsessions have still to be closed:
if (scs.session != NULL) {
Boolean someSubsessionsWereActive = False;
MediaSubsessionIterator iter(*scs.session);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
if (subsession->sink != NULL) {
Medium::close(subsession->sink);
subsession->sink = NULL;
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
}
someSubsessionsWereActive = True;
}
}
if (someSubsessionsWereActive) {
// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
// Don't bother handling the response to the "TEARDOWN".
rtspClient->sendTeardownCommand(*scs.session, NULL);
}
}
// env << *rtspClient << "Closing the stream.\n";
Medium::close(rtspClient);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
//if (--rtspClientCount == 0)
{
// The final stream has ended, so exit the application now.
// (Of course, if you're embedding this code into your own application, you might want to comment this out,
// and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
// exit(exitCode);
g_context[channel].shutdown = true;
g_context[channel].eventLoopVar = 1;
}
while(pthread_mutex_unlock(&g_context[channel].mutex) != 0 && errno == EINTR){};
}
AVIFileSink::AVIFileSink(UsageEnvironment& env,
MediaSession& inputSession,
char const* outputFileName,
unsigned bufferSize,
unsigned short movieWidth, unsigned short movieHeight,
unsigned movieFPS, Boolean packetLossCompensate)
: Medium(env), fInputSession(inputSession),
fIndexRecordsHead(NULL), fIndexRecordsTail(NULL), fNumIndexRecords(0),
fBufferSize(bufferSize), fPacketLossCompensate(packetLossCompensate),
fAreCurrentlyBeingPlayed(False), fNumSubsessions(0), fNumBytesWritten(0),
fHaveCompletedOutputFile(False),
fMovieWidth(movieWidth), fMovieHeight(movieHeight), fMovieFPS(movieFPS) {
fOutFid = OpenOutputFile(env, outputFileName);
if (fOutFid == NULL) return;
// Set up I/O state for each input subsession:
MediaSubsessionIterator iter(fInputSession);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
// Ignore subsessions without a data source:
FramedSource* subsessionSource = subsession->readSource();
if (subsessionSource == NULL) continue;
// If "subsession's" SDP description specified screen dimension
// or frame rate parameters, then use these.
if (subsession->videoWidth() != 0) {
fMovieWidth = subsession->videoWidth();
}
if (subsession->videoHeight() != 0) {
fMovieHeight = subsession->videoHeight();
}
if (subsession->videoFPS() != 0) {
fMovieFPS = subsession->videoFPS();
}
AVISubsessionIOState* ioState
= new AVISubsessionIOState(*this, *subsession);
subsession->miscPtr = (void*)ioState;
// Also set a 'BYE' handler for this subsession's RTCP instance:
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(onRTCPBye, ioState);
}
++fNumSubsessions;
}
// Begin by writing an AVI header:
addFileHeader_AVI();
}
void CAimer39RTSPClient::shutdownStream( RTSPClient* rtspClient )
{
UsageEnvironment& env = rtspClient->envir(); // alias
StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
CAimer39RTSPClient * arc = findClient( (ourRTSPClient*)rtspClient );
if ( NULL == arc ) {
env << "some how the system in to a dangerous situation!" << "\n";
return;
}
// First, check whether any subsessions have still to be closed:
if (scs.session != NULL) {
Boolean someSubsessionsWereActive = False;
MediaSubsessionIterator iter(*scs.session);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
if (subsession->sink != NULL) {
Medium::close(subsession->sink);
subsession->sink = NULL;
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
}
someSubsessionsWereActive = True;
}
}
if (someSubsessionsWereActive) {
// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
// Don't bother handling the response to the "TEARDOWN".
rtspClient->sendTeardownCommand(*scs.session, NULL);
}
}
env << *rtspClient << "Closing the stream.\n";
arc->m_bIsShutDown = true;
if (arc->m_pFinishCallback) arc->m_pFinishCallback(arc->m_pCallBackParam, arc); //call back inform stream over
Medium::close(rtspClient);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
}
// Used to shut down and close a stream (including its "RTSPClient" object):
void StreamClient::shutdownStream()
{
// First, check whether any subsessions have still to be closed:
if (_state.session != nullptr) {
Boolean someSubsessionsWereActive = False;
MediaSubsessionIterator iter(*_state.session);
MediaSubsession * subsession = nullptr;
while ((subsession = iter.next()) != nullptr) {
if (subsession->sink != nullptr) {
Medium::close(subsession->sink);
subsession->sink = nullptr;
if (subsession->rtcpInstance() != nullptr) {
subsession->rtcpInstance()->setByeHandler(nullptr, nullptr); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
}
someSubsessionsWereActive = True;
}
}
if (someSubsessionsWereActive) {
sendTeardown(*_state.session);
}
}
crLogN("StreamClient::shutdownStream() Closing the stream.");
Medium::close(this);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
// if (--rtspClientCount == 0) {
// // The final stream has ended, so exit the application now.
// // (Of course, if you're embedding this code into your own application, you might want to comment this out,
// // and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
// exit(exitCode);
// }
exit(0);
}
int CMediaNet::MediaNet_Thread( void * pThisVoid )
{
CMediaNet *pThis = ( CMediaNet* )pThisVoid;
do
{
// 开始初始化.
pThis->SetRtspStatus( RTSPStatus_Init );
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
progName = "M_CU";
string strUrl = pThis->m_strRTSPUrlA;
gettimeofday(&startTime, NULL);
unsigned short desiredPortNum = 0;
// unfortunately we can't use getopt() here, as Windoze doesn't have it
// Create our client object:
ourClient = createClient(*env, verbosityLevel, progName);
if (ourClient == NULL)
{
*env << "Failed to create " << clientProtocolName
<< " client: " << env->getResultMsg() << "\n";
pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
break;
}
// 开始获取Opition.
pThis->SetRtspStatus( RTSPStatus_Opitiion );
// Begin by sending an "OPTIONS" command:
char* optionsResponse
= getOptionsResponse(ourClient, pThis->m_strRTSPUrlA.c_str(), username, password);
if (optionsResponse == NULL)
{
*env << clientProtocolName << " \"OPTIONS\" request failed: "
<< env->getResultMsg() << "\n";
pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
break;
}
else
{
*env << clientProtocolName << " \"OPTIONS\" request returned: "
<< optionsResponse << "\n";
}
if( optionsResponse )
{
delete[] optionsResponse;
}
// 开始获取Description.
// Open the URL, to get a SDP description:
pThis->SetRtspStatus( RTSPStatus_Description );
char* sdpDescription
= getSDPDescriptionFromURL(ourClient, pThis->m_strRTSPUrlA.c_str(), username, password,
proxyServerName, proxyServerPortNum,
desiredPortNum);
if (sdpDescription == NULL)
{
*env << "Failed to get a SDP description from URL \"" << pThis->m_strRTSPUrlA.c_str()
<< "\": " << env->getResultMsg() << "\n";
pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
break;
}
*env << "Opened URL \"" << pThis->m_strRTSPUrlA.c_str()
<< "\", returning a SDP description:\n" << sdpDescription << "\n";
// Create a media session object from this SDP description:
session = MediaSession::createNew(*env, sdpDescription);
delete[] sdpDescription;
if (session == NULL)
{
*env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << "\n";
pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
break;
}
else if (!session->hasSubsessions())
{
*env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
pThis->SetRtspStatus( RTSPStatus_Error_Connect_Srv );
break;
}
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*session);
MediaSubsession *subsession;
Boolean madeProgress = False;
char const* singleMediumToTest = singleMedium;
while ((subsession = iter.next()) != NULL)
{
// If we've asked to receive only a single medium, then check this now:
if (singleMediumToTest != NULL)
{
if (strcmp(subsession->mediumName(), singleMediumToTest) != 0)
{
*env << "Ignoring \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession, because we've asked to receive a single " << singleMedium
<< " session only\n";
continue;
}
else
{
// Receive this subsession only
singleMediumToTest = "xxxxx";
// this hack ensures that we get only 1 subsession of this type
}
}
desiredPortNum = 0;
if (desiredPortNum != 0)
{
subsession->setClientPortNum(desiredPortNum);
desiredPortNum += 2;
}
if (true)
{
if (!subsession->initiate(simpleRTPoffsetArg))
{
*env << "Unable to create receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession: " << env->getResultMsg() << "\n";
}
else
{
*env << "Created receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession (client ports " << subsession->clientPortNum()
<< "-" << subsession->clientPortNum()+1 << ")\n";
madeProgress = True;
if (subsession->rtpSource() != NULL)
{
// Because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
if (socketInputBufferSize > 0)
{
// Set the RTP source's input buffer size as specified:
int socketNum
= subsession->rtpSource()->RTPgs()->socketNum();
unsigned curBufferSize
= getReceiveBufferSize(*env, socketNum);
unsigned newBufferSize
= setReceiveBufferTo(*env, socketNum, socketInputBufferSize);
*env << "Changed socket receive buffer size for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession from "
<< curBufferSize << " to "
<< newBufferSize << " bytes\n";
}
}
}
}
else
{
mcu::tlog << _T( "Use port： " ) << (int)subsession->clientPortNum() << endl;
if (subsession->clientPortNum() == 0)
{
*env << "No client port was specified for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession. (Try adding the \"-p <portNum>\" option.)\n";
}
else
{
madeProgress = True;
}
}
}
if (!madeProgress)
break;
// Perform additional 'setup' on each subsession, before playing them:
pThis->SetRtspStatus( RTSPStatus_Setup );
unsigned nResponseCode = NULL;
BOOL bSetupSuccess = setupStreams( &nResponseCode );
if ( !bSetupSuccess )
{
// setup失败！
if ( RTSPResp_Error_Server_Full == nResponseCode )
{
pThis->SetRtspStatus( RTSPStatus_Error_Server_Full );
}
else
{
pThis->SetRtspStatus( RTSPStatus_Idle );
}
break;
}
// Create output files:
if ( true )
{
// Create and start "FileSink"s for each subsession:
madeProgress = False;
iter.reset();
while ((subsession = iter.next()) != NULL)
{
if (subsession->readSource() == NULL) continue; // was not initiated
MediaSink *pDecodeSink = 0;
if (strcmp(subsession->mediumName(), "video") == 0 )
{
int nBandWidth = subsession->GetBandWidth();
if ( strcmp(subsession->codecName(), "MP4V-ES") == 0 )
{
CMpeg4StreamDecodeSink *pMsds = CMpeg4StreamDecodeSink::CreateNew( *env, 20000, nBandWidth );
pDecodeSink = pMsds;
}
else if ( strcmp( subsession->codecName(), "H264" ) == 0 )
{
CH264StreamDecodeSink *pHsds = CH264StreamDecodeSink::CreateNew( *env, 20000, nBandWidth );
pDecodeSink = pHsds;
}
else
{
continue;
}
}
subsession->sink = pDecodeSink;
if (subsession->sink == NULL)
{
*env << "Failed to create CH264StreamDecodeSink \"" << "\n";
}
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL)
{
subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
subsession);
}
// 发送NAT探测包。
unsigned char temp[112] = {0};
temp[0] = 0x80;
subsession->rtpSource()->RTPgs()->output( *env, 0,temp, 112 );
madeProgress = True;
}
}
// Finally, start playing each subsession, to start the data flow:
pThis->SetRtspStatus( RTSPStatus_Play );
startPlayingStreams();
pThis->SetRtspStatus( RTSPStatus_Running );
// 传入结束标志指针。
env->taskScheduler().doEventLoop( &pThis->m_runFlag );
pThis->SetRtspStatus( RTSPStatus_Idle );
} while(0);
return 0;
}
bool MtkRTSPClient::handSetup(char* resultString)
{
CHECK_NULL_COND(session, false);
CHECK_NULL_COND(rtsp::env, false);
bool bSuccess = false;
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*(session));
MediaSubsession *subsession = NULL;
while ((subsession = iter.next()) != NULL)
{
if (subsession->readSource() == NULL)
{
LOG_ERR("warning");
continue; // was not initiated
}
if (subsession->sink != NULL)/*already be set*/
{
continue;
}
unsigned int type = getBufType(subsession);
if (type == 0)
{
LOG_ERR("error type=%d", type);
continue;
}
{
iSetupCount--;
/*set mediay info*/
setMediaInfo(subsession, type);
}
CmpbSink *sink = NULL;
if ((type != mediatype_audio) && (strcmp(subsession->codecName(), "H264") == 0))
{
sink = CmpbH264Sink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
else if ((type == mediatype_audio) &&
((stMediaInfo.audioCodec == MEDIACODEC_AC3) ||
(stMediaInfo.audioCodec == MEDIACODEC_EAC3) ||
(stMediaInfo.audioCodec == MEDIACODEC_MPEG4_GENERIC)))
{
sink = CmpbAACSink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
else if ((type == mediatype_audio) && (stMediaInfo.audioCodec == MEDIACODEC_MP4A_LATM))
{
sink = CmpbLATMSink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
else
{
sink = CmpbSink::createNew(*env, *subsession, type, fileSinkBufferSize);
}
subsession->sink = sink;
if (subsession->sink == NULL)
{
LOG_ERR("error!");
}
else
{
#if 0 /*this should be remove to cmpb sink*/
if ((type != mediatype_audio) && (strcmp(subsession->codecName(), "MP4V-ES") == 0)
&& (subsession->fmtp_config() != NULL))
{
// For MPEG-4 video RTP streams, the 'config' information
// from the SDP description contains useful VOL etc. headers.
// Insert this data at the front of the output file:
unsigned configLen;
unsigned char* configData
= parseGeneralConfigStr(subsession->fmtp_config(), configLen);
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
sink->sendData(configData, configLen, timeNow);
delete[] configData;
}
#endif
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL)
{
subsession->rtcpInstance()->setByeHandler(subsessionAfterPlaying, subsession);
}
bSuccess = true;
}
break;
}
if (iSetupCount == 0)
{
mediaInfoReady();
}
return bSuccess ;
}
void shutdownStream(RTSPClient* rtspClient, int exitCode) {
OUTPUT_DEBUG_STRING("%s \n", __FUNCTION__);
OUTPUT_DEBUG_STRING("open times = %d close times = %d\n", open_times, ++close_times);
UsageEnvironment& env = rtspClient->envir(); // alias
StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
// First, check whether any subsessions have still to be closed:
if (scs.session != NULL) {
Boolean someSubsessionsWereActive = False;
MediaSubsessionIterator iter(*scs.session);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
if (subsession->sink != NULL) {
//subsession->sink->stopPlaying();
Medium::close(subsession->sink);
subsession->sink = NULL;
OUTPUT_DEBUG_STRING("22 add = %d, del = %d\n", sum_add, ++sum_del);
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
}
someSubsessionsWereActive = True;
}
}
if (someSubsessionsWereActive) {
// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
// Don't bother handling the response to the "TEARDOWN".
//gEnv->taskScheduler().unscheduleDelayedTask(scs.streamTimerTask);
rtspClient->sendTeardownCommand(*scs.session, NULL);
}
//Medium::close(scs.session);
//gEnv->taskScheduler().unscheduleDelayedTask(scs.streamTimerTask);
Medium::close(rtspClient);
rtspClient = NULL;
//gEnv->reclaim(); //del by huguohu
//gEnv = NULL;
if(gScheduler != NULL){
delete gScheduler;
gScheduler = NULL;
}
}
//env << *rtspClient << "Closing the stream.\n";
//Medium::close(scs.session);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
//
// if (--rtspClientCount == 0) {
// // The final stream has ended, so exit the application now.
// // (Of course, if you're embedding this code into your own application, you might want to comment this out,
// // and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
// //exit(exitCode);
// }
}
bool CRTSPClient::OpenStream(char* url)
{
XBMC->Log(LOG_DEBUG, "CRTSPClient::OpenStream()");
m_session=NULL;
strcpy(m_url,url);
// Open the URL, to get a SDP description:
char* sdpDescription= getSDPDescriptionFromURL(m_ourClient, url, ""/*username*/, ""/*password*/,""/*proxyServerName*/, 0/*proxyServerPortNum*/,1234/*desiredPortNum*/);
if (sdpDescription == NULL)
{
XBMC->Log(LOG_DEBUG, "Failed to get a SDP description from URL %s %s",url ,m_env->getResultMsg() );
shutdown();
return false;
}
XBMC->Log(LOG_DEBUG, "Opened URL %s %s",url,sdpDescription);
char* range=strstr(sdpDescription,"a=range:npt=");
if (range!=NULL)
{
char *pStart = range+strlen("a=range:npt=");
char *pEnd = strstr(range,"-") ;
if (pEnd!=NULL)
{
pEnd++ ;
double Start=atof(pStart) ;
double End=atof(pEnd) ;
XBMC->Log(LOG_DEBUG, "rangestart:%f rangeend:%f", Start,End);
m_duration=(long) ((End-Start)*1000.0);
}
}
// Create a media session object from this SDP description:
m_session = MediaSession::createNew(*m_env, sdpDescription);
delete[] sdpDescription;
if (m_session == NULL)
{
XBMC->Log(LOG_DEBUG, "Failed to create a MediaSession object from the SDP description:%s ",m_env->getResultMsg());
shutdown();
return false;
}
else if (!m_session->hasSubsessions())
{
XBMC->Log(LOG_DEBUG, "This session has no media subsessions");
shutdown();
return false;
}
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*m_session);
MediaSubsession *subsession;
Boolean madeProgress = False;
char const* singleMediumToTest = singleMedium;
while ((subsession = iter.next()) != NULL)
{
// If we've asked to receive only a single medium, then check this now:
if (singleMediumToTest != NULL)
{
if (strcmp(subsession->mediumName(), singleMediumToTest) != 0)
{
XBMC->Log(LOG_DEBUG, "Ignoring %s %s %s" , subsession->mediumName(),subsession->codecName(),singleMedium);
continue;
}
else
{
// Receive this subsession only
singleMediumToTest = "xxxxx";
// this hack ensures that we get only 1 subsession of this type
}
}
if (desiredPortNum != 0)
{
subsession->setClientPortNum(desiredPortNum);
desiredPortNum += 2;
}
if (createReceivers)
{
if (!subsession->initiate(simpleRTPoffsetArg))
{
XBMC->Log(LOG_DEBUG, "Unable to create receiver for %s %s %s" ,subsession->mediumName(),subsession->codecName(),m_env->getResultMsg());
}
else
{
XBMC->Log(LOG_DEBUG, "Created receiver for type=%s codec=%s ports: %d %d " ,subsession->mediumName(),subsession->codecName(),subsession->clientPortNum(),subsession->clientPortNum()+1 );
madeProgress = True;
if (subsession->rtpSource() != NULL)
{
// Because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
int socketNum= subsession->rtpSource()->RTPgs()->socketNum();
XBMC->Log(LOG_DEBUG, "rtsp:increaseReceiveBufferTo to 2000000 for s:%d",socketNum);
increaseReceiveBufferTo( *m_env, socketNum, 2000000 );
unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
if (socketInputBufferSize > 0)
{
// Set the RTP source's input buffer size as specified:
int socketNum= subsession->rtpSource()->RTPgs()->socketNum();
unsigned curBufferSize= getReceiveBufferSize(*m_env, socketNum);
unsigned newBufferSize= setReceiveBufferTo(*m_env, socketNum, socketInputBufferSize);
XBMC->Log(LOG_DEBUG, "Changed socket receive buffer size for the %s %s %d %d",
subsession->mediumName(),subsession->codecName(),curBufferSize,newBufferSize);
}
}
}
}
else
{
if (subsession->clientPortNum() == 0)
{
XBMC->Log(LOG_DEBUG, "No client port was specified for the %s %s",subsession->mediumName(),subsession->codecName());
}
else
{
madeProgress = True;
}
}
}
if (!madeProgress)
{
shutdown();
return false;
}
// Perform additional 'setup' on each subsession, before playing them:
if (!setupStreams())
{
return false;
}
// Create output files:
// Create and start "FileSink"s for each subsession:
madeProgress = False;
iter.reset();
while ((subsession = iter.next()) != NULL)
{
if (subsession->readSource() == NULL) continue; // was not initiated
// Mediaportal:
CMemorySink* fileSink= CMemorySink::createNew(*m_env, *m_buffer, fileSinkBufferSize);
// XBMC test via file:
//FileSink* fileSink = FileSink::createNew(*m_env, m_outFileName, fileSinkBufferSize, false); //oneFilePerFrame
subsession->sink = fileSink;
if (subsession->sink == NULL)
{
XBMC->Log(LOG_DEBUG, "Failed to create FileSink %s",m_env->getResultMsg());
shutdown();
return false;
}
XBMC->Log(LOG_DEBUG, "Created output sink: %s", m_outFileName);
subsession->sink->startPlaying(*(subsession->readSource()),my_subsessionAfterPlaying,subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL)
{
subsession->rtcpInstance()->setByeHandler(my_subsessionByeHandler,subsession);
}
madeProgress = True;
}
return true;
}
int CMPIPTV_RTSP::OpenConnection(void)
{
this->logger.Log(LOGGER_INFO, METHOD_START_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
int result = STATUS_OK;
this->rtspClient = RTSPClient::createNew(*this->rtspEnvironment);
result |= (this->rtspClient == NULL);
if (result == STATUS_OK)
{
// RTSPClient works with char, not with TCHAR
char *tempRtspUrl = ConvertToMultiByte(this->rtspUrl);
result |= (tempRtspUrl == NULL);
if (result == STATUS_OK)
{
char* optionsResult = this->rtspClient->sendOptionsCmd(tempRtspUrl, NULL, NULL, NULL, this->receiveDataTimeout / 2000);
result |= (optionsResult == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while sending OPTIONS command"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("OPTIONS result"));
this->LogFullRtspMessage(LOGGER_VERBOSE, message, optionsResult);
FREE_MEM(message);
char *describeResult = this->rtspClient->describeURL(tempRtspUrl, NULL, FALSE, this->receiveDataTimeout / 2000);
result |= (describeResult == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while sending DESCRIBE command"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("DESCRIBE result"));
this->LogFullRtspMessage(LOGGER_VERBOSE, message, describeResult);
FREE_MEM(message);
this->rtspSession = MediaSession::createNew(*this->rtspEnvironment, describeResult);
result |= (this->rtspSession == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while creating new session"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
result |= (!this->rtspSession->hasSubsessions());
if (result != STATUS_OK)
{
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("session doesn't have subsessions"));
}
else
{
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*this->rtspSession);
MediaSubsession *subsession = NULL;
while ((result == STATUS_OK) && ((subsession = iter.next()) != NULL))
{
char *tempSubSessionName = (char *)subsession->mediumName();
char *tempSubSessionCodecName = (char *)subsession->codecName();
#ifdef _MBCS
TCHAR *subSessionName = ConvertToMultiByteA(tempSubSessionName);
TCHAR *subSessionCodecName = ConvertToMultiByteA(tempSubSessionCodecName);
#else
TCHAR *subSessionName = ConvertToUnicodeA(tempSubSessionName);
TCHAR *subSessionCodecName = ConvertToUnicodeA(tempSubSessionCodecName);
#endif
if (!subsession->initiate())
{
result = STATUS_ERROR;
TCHAR *message = FormatString(_T("%s: %s: unable to create receiver for subsession '%s', codec '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: created receiver for subsession '%s', codec '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);
// set session ID, doesn't matter what
subsession->sessionId = tempSubSessionName;
if (subsession->rtpSource() != NULL)
{
// because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
// set the RTP source's OS socket buffer size as appropriate
int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
unsigned int currentBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
if (this->defaultBufferSize > currentBufferSize)
{
setReceiveBufferTo(*this->rtspEnvironment, socketNum, this->defaultBufferSize);
unsigned setBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
if (setBufferSize == this->defaultBufferSize)
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: set buffer size for subsession '%s' successful, previous size: %i, requested size: %i, current size: %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, currentBufferSize, this->defaultBufferSize, setBufferSize);
}
else
{
result = STATUS_ERROR;
this->logger.Log(LOGGER_ERROR, _T("%s: %s: set buffer size for subsession '%s' failed, previous size: %i, requested size: %i, current size: %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, currentBufferSize, this->defaultBufferSize, setBufferSize);
}
}
if (_tcsncicmp(subSessionName, _T("audio"), 5) == 0)
{
// audio
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: audio subsession '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
result |= (!rtspClient->setupMediaSubsession(*subsession));
if (result != STATUS_OK)
{
// error occured
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while setup subsession"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
this->logger.Log(LOGGER_WARNING, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("subsession audio codec not supported"));
}
}
else if (_tcsncicmp(subSessionName, _T("video"), 5) == 0)
{
// video
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: video subsession '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
result |= (!rtspClient->setupMediaSubsession(*subsession));
if (result != STATUS_OK)
{
// error occured
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while setup subsession"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
if (_tcsncicmp(subSessionCodecName, _T("MP2T"), 4) == 0)
{
// MPEG2 Transport Stream
// set new RTSP source
this->rtspSource = subsession->rtpSource();
if (subsession->rtcpInstance() != NULL)
{
this->logger.Log(LOGGER_VERBOSE, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("set subsession 'Bye' handler"));
subsession->rtcpInstance()->setByeHandler(SubsessionByeHandler, this);
}
}
else if (_tcsncicmp(subSessionCodecName, _T("H264"), 4) == 0)
{
// H264 codec, HD TV
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("H264 not supported"));
result = STATUS_ERROR;
}
else
{
// SD TV
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("other subsession video codec than MP2T not supported"));
result = STATUS_ERROR;
}
}
}
else
{
this->logger.Log(LOGGER_WARNING, _T("%s: %s: unknown subsession '%s', ignored"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
}
}
else
{
this->logger.Log(LOGGER_WARNING, _T("%s: %s: subsession '%s' doesn't have RTP source"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
}
}
// free subsession name and codec name
FREE_MEM(subSessionName);
FREE_MEM(subSessionCodecName);
}
// we should have some RTSP source
result |= (this->rtspSource == NULL);
if (result == STATUS_OK)
{
result |= (!this->rtspClient->playMediaSession(*this->rtspSession));
if (result != STATUS_OK)
{
// error occured
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while playing session"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
// create UDP socket and start playing
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr("127.0.0.1");
unsigned int port = this->rtspUdpPortRangeStart;
do
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: UDP port %u"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
// special construction force not reuse same UDP port
{
NoReuse noReuse;
this->rtspUdpGroupsock = new Groupsock(*this->rtspEnvironment, destinationAddress, port, 1);
}
if (this->rtspUdpGroupsock->socketNum() == (-1))
{
this->logger.Log(LOGGER_ERROR, _T("%s: %s: UDP port %u occupied, trying another port"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
port++;
delete this->rtspUdpGroupsock;
this->rtspUdpGroupsock = NULL;
}
}
while ((this->rtspUdpGroupsock == NULL) && (port <= this->rtspUdpPortRangeEnd));
result |= (this->rtspUdpGroupsock == NULL);
if (result != STATUS_OK)
{
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("cannot create UDP sink, no free port"));
}
else
{
this->rtspUdpSink = BasicUDPSink::createNew(*this->rtspEnvironment, this->rtspUdpGroupsock, this->rtspUdpSinkMaxPayloadSize);
result |= (this->rtspUdpSink == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("cannot create UDP sink"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
if (this->rtspUdpSink->startPlaying(*this->rtspSource, NULL, NULL))
{
this->logger.Log(LOGGER_INFO, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("playing started"));
// now create UDP connection
TCHAR *url = FormatString(_T("udp://@127.0.0.1:%u"), port);
result |= (url == NULL);
if (result == STATUS_OK)
{
// parse UDP url
// ParseURL calls ClearSession and IsConnected must return FALSE
// in another case will be current RTSP connection closed
result = this->CMPIPTV_UDP::ParseUrl(url, NULL);
if (result == STATUS_OK)
{
// connect to UDP url
result = this->CMPIPTV_UDP::OpenConnection();
}
}
FREE_MEM(url);
}
else
{
result = STATUS_ERROR;
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while starting playing"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
}
}
}
}
}
}
}
}
if (optionsResult != NULL)
{
delete[] optionsResult;
optionsResult = NULL;
}
}
FREE_MEM(tempRtspUrl);
}
if (result == STATUS_OK)
{
// start winsock worker thread
this->rtspSchedulerThreadHandle = CreateThread(
NULL, // default security attributes
0, // use default stack size
&CMPIPTV_RTSP::RtspSchedulerWorker, // thread function name
this, // argument to thread function
0, // use default creation flags
&this->rtspSchedulerThreadId); // returns the thread identifier
if (this->rtspSchedulerThreadHandle == NULL)
{
// thread not created
result = STATUS_ERROR;
this->logger.Log(LOGGER_ERROR, _T("%s: %s: cannot create RTSP scheduler thread, error: %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, GetLastError());
}
}
if (result != STATUS_OK)
{
// if failed opening connection, than close connection
this->CloseConnection();
}
this->logger.Log(LOGGER_INFO, (result == STATUS_OK) ? METHOD_END_FORMAT : METHOD_END_FAIL_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
return (result == STATUS_OK) ? STATUS_OK : STATUS_ERROR;
}
Boolean MediaSession
::initiateByMediaType(char const* mimeType,
MediaSubsession*& resultSubsession,
PrioritizedRTPStreamSelector*& resultMultiSource,
int& resultMultiSourceSessionId,
int useSpecialRTPoffset) {
// Look through this session's subsessions for media that match "mimeType"
resultSubsession = NULL;
resultMultiSource = NULL;
resultMultiSourceSessionId = 0;
unsigned maxStaggerSeconds = 0;
MediaSubsessionIterator iter(*this);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
if (resultMultiSourceSessionId != 0
&& subsession->mctSLAPSessionId() != resultMultiSourceSessionId) {
// We're using a multi-source SLAP session, but this subsession
// isn't part of it
continue;
}
Boolean wasAlreadyInitiated = subsession->readSource() != NULL;
if (!wasAlreadyInitiated) {
// Try to create a source for this subsession:
if (!subsession->initiate(useSpecialRTPoffset)) return False;
}
// Make sure the source's MIME type is one that we handle:
if (strcmp(subsession->readSource()->MIMEtype(), mimeType) != 0) {
if (!wasAlreadyInitiated) subsession->deInitiate();
continue;
}
if (subsession->mctSLAPSessionId() == 0) {
// Normal case: a single session
resultSubsession = subsession;
break; // use this
} else {
// Special case: a multi-source SLAP session
resultMultiSourceSessionId = subsession->mctSLAPSessionId();
unsigned subsessionStaggerSeconds = subsession->mctSLAPStagger();
if (subsessionStaggerSeconds > maxStaggerSeconds) {
maxStaggerSeconds = subsessionStaggerSeconds;
}
}
}
if (resultSubsession == NULL && resultMultiSourceSessionId == 0) {
envir().setResultMsg("Session has no usable media subsession");
return False;
}
if (resultMultiSourceSessionId != 0) {
// We have a multi-source MCT SLAP session; create a selector for it:
unsigned seqNumStagger = computeSeqNumStagger(maxStaggerSeconds);
resultMultiSource
= PrioritizedRTPStreamSelector::createNew(envir(), seqNumStagger);
if (resultMultiSource == NULL) return False;
// Note: each subsession has its own RTCP instance; we don't return them
// Then run through the subsessions again, adding each of the sources:
iter.reset();
while ((subsession = iter.next()) != NULL) {
if (subsession->mctSLAPSessionId() == resultMultiSourceSessionId) {
resultMultiSource->addInputRTPStream(subsession->rtpSource(),
subsession->rtcpInstance());
}
}
}
return True;
}
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
progName = argv[0];
gettimeofday(&startTime, NULL);
#ifdef USE_SIGNALS
// Allow ourselves to be shut down gracefully by a SIGHUP or a SIGUSR1:
signal(SIGHUP, signalHandlerShutdown);
signal(SIGUSR1, signalHandlerShutdown);
#endif
unsigned short desiredPortNum = 0;
// unfortunately we can't use getopt() here, as Windoze doesn't have it
while (argc > 2) {
char* const opt = argv[1];
if (opt[0] != '-') usage();
switch (opt[1]) {
case 'p': { // specify start port number
int portArg;
if (sscanf(argv[2], "%d", &portArg) != 1) {
usage();
}
if (portArg <= 0 || portArg >= 65536 || portArg&1) {
*env << "bad port number: " << portArg
<< " (must be even, and in the range (0,65536))\n";
usage();
}
desiredPortNum = (unsigned short)portArg;
++argv; --argc;
break;
}
case 'r': { // do not receive data (instead, just 'play' the stream(s))
createReceivers = False;
break;
}
case 'q': { // output a QuickTime file (to stdout)
outputQuickTimeFile = True;
break;
}
case '4': { // output a 'mp4'-format file (to stdout)
outputQuickTimeFile = True;
generateMP4Format = True;
break;
}
case 'i': { // output an AVI file (to stdout)
outputAVIFile = True;
break;
}
case 'I': { // specify input interface...
NetAddressList addresses(argv[2]);
if (addresses.numAddresses() == 0) {
*env << "Failed to find network address for \"" << argv[2] << "\"";
break;
}
ReceivingInterfaceAddr = *(unsigned*)(addresses.firstAddress()->data());
++argv; --argc;
break;
}
case 'a': { // receive/record an audio stream only
audioOnly = True;
singleMedium = "audio";
break;
}
case 'v': { // receive/record a video stream only
videoOnly = True;
singleMedium = "video";
break;
}
case 'V': { // disable verbose output
verbosityLevel = 0;
break;
}
case 'd': { // specify duration, or how much to delay after end time
float arg;
if (sscanf(argv[2], "%g", &arg) != 1) {
usage();
}
if (argv[2][0] == '-') { // not "arg<0", in case argv[2] was "-0"
// a 'negative' argument was specified; use this for "durationSlop":
duration = 0; // use whatever's in the SDP
durationSlop = -arg;
} else {
duration = arg;
durationSlop = 0;
}
++argv; --argc;
break;
}
case 'D': { // specify maximum number of seconds to wait for packets:
if (sscanf(argv[2], "%u", &interPacketGapMaxTime) != 1) {
usage();
}
++argv; --argc;
break;
}
case 'c': { // play continuously
playContinuously = True;
break;
}
case 'S': { // specify an offset to use with "SimpleRTPSource"s
if (sscanf(argv[2], "%d", &simpleRTPoffsetArg) != 1) {
usage();
}
if (simpleRTPoffsetArg < 0) {
*env << "offset argument to \"-S\" must be >= 0\n";
usage();
}
++argv; --argc;
break;
}
case 'O': { // Don't send an "OPTIONS" request before "DESCRIBE"
sendOptionsRequest = False;
break;
}
case 'o': { // Send only the "OPTIONS" request to the server
sendOptionsRequestOnly = True;
break;
}
case 'm': { // output multiple files - one for each frame
oneFilePerFrame = True;
break;
}
case 'n': { // notify the user when the first data packet arrives
notifyOnPacketArrival = True;
break;
}
case 't': {
// stream RTP and RTCP over the TCP 'control' connection
if (controlConnectionUsesTCP) {
streamUsingTCP = True;
} else {
usage();
}
break;
}
case 'T': {
// stream RTP and RTCP over a HTTP connection
if (controlConnectionUsesTCP) {
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the HTTP server port number:
if (sscanf(argv[2], "%hu", &tunnelOverHTTPPortNum) == 1
&& tunnelOverHTTPPortNum > 0) {
++argv; --argc;
break;
}
}
}
// If we get here, the option was specified incorrectly:
usage();
break;
}
case 'u': { // specify a username and password
username = argv[2];
password = argv[3];
argv+=2; argc-=2;
if (allowProxyServers && argc > 3 && argv[2][0] != '-') {
// The next argument is the name of a proxy server:
proxyServerName = argv[2];
++argv; --argc;
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the proxy server port number:
if (sscanf(argv[2], "%hu", &proxyServerPortNum) != 1) {
usage();
}
++argv; --argc;
}
}
break;
}
case 'A': { // specify a desired audio RTP payload format
unsigned formatArg;
if (sscanf(argv[2], "%u", &formatArg) != 1
|| formatArg >= 96) {
usage();
}
desiredAudioRTPPayloadFormat = (unsigned char)formatArg;
++argv; --argc;
break;
}
case 'M': { // specify a MIME subtype for a dynamic RTP payload type
mimeSubtype = argv[2];
if (desiredAudioRTPPayloadFormat==0) desiredAudioRTPPayloadFormat =96;
++argv; --argc;
break;
}
case 'w': { // specify a width (pixels) for an output QuickTime or AVI movie
if (sscanf(argv[2], "%hu", &movieWidth) != 1) {
usage();
}
movieWidthOptionSet = True;
++argv; --argc;
break;
}
case 'h': { // specify a height (pixels) for an output QuickTime or AVI movie
if (sscanf(argv[2], "%hu", &movieHeight) != 1) {
usage();
}
movieHeightOptionSet = True;
++argv; --argc;
break;
}
case 'f': { // specify a frame rate (per second) for an output QT or AVI movie
if (sscanf(argv[2], "%u", &movieFPS) != 1) {
usage();
}
movieFPSOptionSet = True;
++argv; --argc;
break;
}
case 'F': { // specify a prefix for the audio and video output files
fileNamePrefix = argv[2];
++argv; --argc;
break;
}
case 'b': { // specify the size of buffers for "FileSink"s
if (sscanf(argv[2], "%u", &fileSinkBufferSize) != 1) {
usage();
}
++argv; --argc;
break;
}
case 'B': { // specify the size of input socket buffers
if (sscanf(argv[2], "%u", &socketInputBufferSize) != 1) {
usage();
}
++argv; --argc;
break;
}
// Note: The following option is deprecated, and may someday be removed:
case 'l': { // try to compensate for packet loss by repeating frames
packetLossCompensate = True;
break;
}
case 'y': { // synchronize audio and video streams
syncStreams = True;
break;
}
case 'H': { // generate hint tracks (as well as the regular data tracks)
generateHintTracks = True;
break;
}
case 'Q': { // output QOS measurements
qosMeasurementIntervalMS = 1000; // default: 1 second
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the measurement interval,
// in multiples of 100 ms
if (sscanf(argv[2], "%u", &qosMeasurementIntervalMS) != 1) {
usage();
}
qosMeasurementIntervalMS *= 100;
++argv; --argc;
}
break;
}
case 's': { // specify initial seek time (trick play)
double arg;
if (sscanf(argv[2], "%lg", &arg) != 1 || arg < 0) {
usage();
}
initialSeekTime = arg;
++argv; --argc;
break;
}
case 'z': { // scale (trick play)
float arg;
if (sscanf(argv[2], "%g", &arg) != 1 || arg == 0.0f) {
usage();
}
scale = arg;
++argv; --argc;
break;
}
default: {
usage();
break;
}
}
++argv; --argc;
}
if (argc != 2) usage();
if (outputQuickTimeFile && outputAVIFile) {
*env << "The -i and -q (or -4) flags cannot both be used!\n";
usage();
}
Boolean outputCompositeFile = outputQuickTimeFile || outputAVIFile;
if (!createReceivers && outputCompositeFile) {
*env << "The -r and -q (or -4 or -i) flags cannot both be used!\n";
usage();
}
if (outputCompositeFile && !movieWidthOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -w. Assuming a video width of "
<< movieWidth << " pixels\n";
}
if (outputCompositeFile && !movieHeightOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -h. Assuming a video height of "
<< movieHeight << " pixels\n";
}
if (outputCompositeFile && !movieFPSOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -f. Assuming a video frame rate of "
<< movieFPS << " frames-per-second\n";
}
if (audioOnly && videoOnly) {
*env << "The -a and -v flags cannot both be used!\n";
usage();
}
if (sendOptionsRequestOnly && !sendOptionsRequest) {
*env << "The -o and -O flags cannot both be used!\n";
usage();
}
if (tunnelOverHTTPPortNum > 0) {
if (streamUsingTCP) {
*env << "The -t and -T flags cannot both be used!\n";
usage();
} else {
streamUsingTCP = True;
}
}
if (!createReceivers && notifyOnPacketArrival) {
*env << "Warning: Because we're not receiving stream data, the -n flag has no effect\n";
}
if (durationSlop < 0) {
// This parameter wasn't set, so use a default value.
// If we're measuring QOS stats, then don't add any slop, to avoid
// having 'empty' measurement intervals at the end.
durationSlop = qosMeasurementIntervalMS > 0 ? 0.0 : 5.0;
}
char* url = argv[1];
// Create our client object:
ourClient = createClient(*env, verbosityLevel, progName);
if (ourClient == NULL) {
*env << "Failed to create " << clientProtocolName
<< " client: " << env->getResultMsg() << "\n";
shutdown();
}
if (sendOptionsRequest) {
// Begin by sending an "OPTIONS" command:
char* optionsResponse
= getOptionsResponse(ourClient, url, username, password);
if (sendOptionsRequestOnly) {
if (optionsResponse == NULL) {
*env << clientProtocolName << " \"OPTIONS\" request failed: "
<< env->getResultMsg() << "\n";
} else {
*env << clientProtocolName << " \"OPTIONS\" request returned: "
<< optionsResponse << "\n";
}
shutdown();
}
delete[] optionsResponse;
}
// Open the URL, to get a SDP description:
char* sdpDescription
= getSDPDescriptionFromURL(ourClient, url, username, password,
proxyServerName, proxyServerPortNum,
desiredPortNum);
if (sdpDescription == NULL) {
*env << "Failed to get a SDP description from URL \"" << url
<< "\": " << env->getResultMsg() << "\n";
shutdown();
}
*env << "Opened URL \"" << url
<< "\", returning a SDP description:\n" << sdpDescription << "\n";
// Create a media session object from this SDP description:
session = MediaSession::createNew(*env, sdpDescription);
delete[] sdpDescription;
if (session == NULL) {
*env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << "\n";
shutdown();
} else if (!session->hasSubsessions()) {
*env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
shutdown();
}
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*session);
MediaSubsession *subsession;
Boolean madeProgress = False;
char const* singleMediumToTest = singleMedium;
while ((subsession = iter.next()) != NULL) {
// If we've asked to receive only a single medium, then check this now:
if (singleMediumToTest != NULL) {
if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) {
*env << "Ignoring \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession, because we've asked to receive a single " << singleMedium
<< " session only\n";
continue;
} else {
// Receive this subsession only
singleMediumToTest = "xxxxx";
// this hack ensures that we get only 1 subsession of this type
}
}
if (desiredPortNum != 0) {
subsession->setClientPortNum(desiredPortNum);
desiredPortNum += 2;
}
if (createReceivers) {
if (!subsession->initiate(simpleRTPoffsetArg)) {
*env << "Unable to create receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession: " << env->getResultMsg() << "\n";
} else {
*env << "Created receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession (client ports " << subsession->clientPortNum()
<< "-" << subsession->clientPortNum()+1 << ")\n";
madeProgress = True;
if (subsession->rtpSource() != NULL) {
// Because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
// Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),
// or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.
// (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,
// then the input data rate may be large enough to justify increasing the OS socket buffer size also.)
int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
unsigned curBufferSize = getReceiveBufferSize(*env, socketNum);
if (socketInputBufferSize > 0 || fileSinkBufferSize > curBufferSize) {
unsigned newBufferSize = socketInputBufferSize > 0 ? socketInputBufferSize : fileSinkBufferSize;
newBufferSize = setReceiveBufferTo(*env, socketNum, newBufferSize);
if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it:
*env << "Changed socket receive buffer size for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession from "
<< curBufferSize << " to "
<< newBufferSize << " bytes\n";
}
}
}
}
} else {
if (subsession->clientPortNum() == 0) {
*env << "No client port was specified for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession. (Try adding the \"-p <portNum>\" option.)\n";
} else {
madeProgress = True;
}
}
}
if (!madeProgress) shutdown();
// Perform additional 'setup' on each subsession, before playing them:
setupStreams();
// Create output files:
if (createReceivers) {
if (outputQuickTimeFile) {
// Create a "QuickTimeFileSink", to write to 'stdout':
qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate,
syncStreams,
generateHintTracks,
generateMP4Format);
if (qtOut == NULL) {
*env << "Failed to create QuickTime file sink for stdout: " << env->getResultMsg();
shutdown();
}
qtOut->startPlaying(sessionAfterPlaying, NULL);
} else if (outputAVIFile) {
// Create an "AVIFileSink", to write to 'stdout':
aviOut = AVIFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate);
if (aviOut == NULL) {
*env << "Failed to create AVI file sink for stdout: " << env->getResultMsg();
shutdown();
}
aviOut->startPlaying(sessionAfterPlaying, NULL);
} else {
// Create and start "FileSink"s for each subsession:
madeProgress = False;
iter.reset();
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
// Create an output file for each desired stream:
char outFileName[1000];
if (singleMedium == NULL) {
// Output file name is
// "<filename-prefix><medium_name>-<codec_name>-<counter>"
static unsigned streamCounter = 0;
snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",
fileNamePrefix, subsession->mediumName(),
subsession->codecName(), ++streamCounter);
} else {
sprintf(outFileName, "stdout");
}
FileSink* fileSink;
if (strcmp(subsession->mediumName(), "audio") == 0 &&
(strcmp(subsession->codecName(), "AMR") == 0 ||
strcmp(subsession->codecName(), "AMR-WB") == 0)) {
// For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
fileSink = AMRAudioFileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
} else if (strcmp(subsession->mediumName(), "video") == 0 &&
(strcmp(subsession->codecName(), "H264") == 0)) {
// For H.264 video stream, we use a special sink that insert start_codes:
fileSink = H264VideoFileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
} else {
// Normal case:
fileSink = FileSink::createNew(*env, outFileName,
fileSinkBufferSize, oneFilePerFrame);
}
subsession->sink = fileSink;
if (subsession->sink == NULL) {
*env << "Failed to create FileSink for \"" << outFileName
<< "\": " << env->getResultMsg() << "\n";
} else {
if (singleMedium == NULL) {
*env << "Created output file: \"" << outFileName << "\"\n";
} else {
*env << "Outputting data from the \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession to 'stdout'\n";
}
if (strcmp(subsession->mediumName(), "video") == 0 &&
strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
subsession->fmtp_config() != NULL) {
// For MPEG-4 video RTP streams, the 'config' information
// from the SDP description contains useful VOL etc. headers.
// Insert this data at the front of the output file:
unsigned configLen;
unsigned char* configData
= parseGeneralConfigStr(subsession->fmtp_config(), configLen);
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
fileSink->addData(configData, configLen, timeNow);
delete[] configData;
}
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
subsession);
}
madeProgress = True;
}
}
if (!madeProgress) shutdown();
}
}
// Finally, start playing each subsession, to start the data flow:
startPlayingStreams();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
