ondemandservermediasubsession.cpp

H.264 RTSP 串流(live 555)視窗版本

						void* serverRequestAlternativeByteHandlerClientData) {
  StreamState* streamState = (StreamState*)streamToken;
  Destinations* destinations
    = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
  if (streamState != NULL) {
    streamState->startPlaying(destinations,
			      rtcpRRHandler, rtcpRRHandlerClientData,
			      serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
    if (streamState->rtpSink() != NULL) {
      rtpSeqNum = streamState->rtpSink()->currentSeqNo();
      rtpTimestamp = streamState->rtpSink()->presetNextTimestamp();
    }
  }
}

void OnDemandServerMediaSubsession::pauseStream(unsigned /*clientSessionId*/,
						void* streamToken) {
  // Pausing isn't allowed if multiple clients are receiving data from
  // the same source:
  if (fReuseFirstSource) return;

  StreamState* streamState = (StreamState*)streamToken;
  if (streamState != NULL) streamState->pause();
}

void OnDemandServerMediaSubsession::seekStream(unsigned /*clientSessionId*/,
					       void* streamToken, double seekNPT) {
  // Seeking isn't allowed if multiple clients are receiving data from
  // the same source:
  if (fReuseFirstSource) return;

  StreamState* streamState = (StreamState*)streamToken;
  if (streamState != NULL && streamState->mediaSource() != NULL) {
    seekStreamSource(streamState->mediaSource(), seekNPT);
  }
}

void OnDemandServerMediaSubsession::setStreamScale(unsigned /*clientSessionId*/,
						   void* streamToken, float scale) {
  // Changing the scale factor isn't allowed if multiple clients are receiving data
  // from the same source:
  if (fReuseFirstSource) return;

  StreamState* streamState = (StreamState*)streamToken;
  if (streamState != NULL && streamState->mediaSource() != NULL) {
    setStreamSourceScale(streamState->mediaSource(), scale);
  }
}

void OnDemandServerMediaSubsession::deleteStream(unsigned clientSessionId,
						 void*& streamToken) {
  StreamState* streamState = (StreamState*)streamToken;

  // Look up (and remove) the destinations for this client session:
  Destinations* destinations
    = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
  if (destinations != NULL) {
    fDestinationsHashTable->Remove((char const*)clientSessionId);

    // Stop streaming to these destinations:
    if (streamState != NULL) streamState->endPlaying(destinations);
  }

  // Delete the "StreamState" structure if it's no longer being used:
  if (streamState != NULL) {
    if (streamState->referenceCount() > 0) --streamState->referenceCount();
    if (streamState->referenceCount() == 0) {
      delete streamState;
      if (fLastStreamToken == streamToken) fLastStreamToken = NULL;
      streamToken = NULL;
    }
  }

  // Finally, delete the destinations themselves:
  delete destinations;
}

char const* OnDemandServerMediaSubsession
::getAuxSDPLine(RTPSink* rtpSink, FramedSource* /*inputSource*/) {
  // Default implementation:
  return rtpSink == NULL ? NULL : rtpSink->auxSDPLine();
}

void OnDemandServerMediaSubsession::seekStreamSource(FramedSource* /*inputSource*/,
						     double /*seekNPT*/) {
  // Default implementation: Do nothing
}

void OnDemandServerMediaSubsession
::setStreamSourceScale(FramedSource* /*inputSource*/, float /*scale*/) {
  // Default implementation: Do nothing
}

void OnDemandServerMediaSubsession::closeStreamSource(FramedSource *inputSource) {
  Medium::close(inputSource);
}

void OnDemandServerMediaSubsession
::setSDPLinesFromRTPSink(RTPSink* rtpSink, FramedSource* inputSource, unsigned estBitrate) {
  if (rtpSink == NULL) return;

  char const* mediaType = rtpSink->sdpMediaType();
  unsigned char rtpPayloadType = rtpSink->rtpPayloadType();
  struct in_addr serverAddrForSDP; serverAddrForSDP.s_addr = fServerAddressForSDP;
  char* const ipAddressStr = strDup(our_inet_ntoa(serverAddrForSDP));
  char* rtpmapLine = rtpSink->rtpmapLine();
  char const* rangeLine = rangeSDPLine();
  char const* auxSDPLine = getAuxSDPLine(rtpSink, inputSource);
  if (auxSDPLine == NULL) auxSDPLine = "";

  char const* const sdpFmt =
    "m=%s %u RTP/AVP %d\r\n"
    "c=IN IP4 %s\r\n"
    "b=AS:%u\r\n"
    "%s"
    "%s"
    "%s"
    "a=control:%s\r\n";
  unsigned sdpFmtSize = strlen(sdpFmt)
    + strlen(mediaType) + 5 /* max short len */ + 3 /* max char len */
    + strlen(ipAddressStr)
    + 20 /* max int len */
    + strlen(rtpmapLine)
    + strlen(rangeLine)
    + strlen(auxSDPLine)
    + strlen(trackId());
  char* sdpLines = new char[sdpFmtSize];
  sprintf(sdpLines, sdpFmt,
	  mediaType, // m= <media>
	  fPortNumForSDP, // m= <port>
	  rtpPayloadType, // m= <fmt list>
	  ipAddressStr, // c= address
	  estBitrate, // b=AS:<bandwidth>
	  rtpmapLine, // a=rtpmap:... (if present)
	  rangeLine, // a=range:... (if present)
	  auxSDPLine, // optional extra SDP line
	  trackId()); // a=control:<track-id>
  delete[] (char*)rangeLine; delete[] rtpmapLine; delete[] ipAddressStr;

  fSDPLines = strDup(sdpLines);
  delete[] sdpLines;
}


////////// StreamState implementation //////////

static void afterPlayingStreamState(void* clientData) {
  StreamState* streamState = (StreamState*)clientData;
  if (streamState->streamDuration() == 0.0) {
    // When the input stream ends, tear it down.  This will cause a RTCP "BYE"
    // to be sent to each client, teling it that the stream has ended.
    // (Because the stream didn't have a known duration, there was no other
    //  way for clients to know when the stream ended.)
    streamState->reclaim();
  }
  // Otherwise, keep the stream alive, in case a client wants to
  // subsequently re-play the stream starting from somewhere other than the end.
  // (This can be done only on streams that have a known duration.)
}

StreamState::StreamState(OnDemandServerMediaSubsession& master,
                         Port const& serverRTPPort, Port const& serverRTCPPort,
			 RTPSink* rtpSink, BasicUDPSink* udpSink,
			 unsigned totalBW, FramedSource* mediaSource,
			 Groupsock* rtpGS, Groupsock* rtcpGS)
  : fMaster(master), fAreCurrentlyPlaying(False), fReferenceCount(1),
    fServerRTPPort(serverRTPPort), fServerRTCPPort(serverRTCPPort),
    fRTPSink(rtpSink), fUDPSink(udpSink), fStreamDuration(master.duration()),
    fTotalBW(totalBW), fRTCPInstance(NULL) /* created later */,
    fMediaSource(mediaSource), fRTPgs(rtpGS), fRTCPgs(rtcpGS) {
}

StreamState::~StreamState() {
  reclaim();
}

void StreamState
::startPlaying(Destinations* dests,
	       TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
	       ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
	       void* serverRequestAlternativeByteHandlerClientData) {
  if (dests == NULL) return;
  if (!fAreCurrentlyPlaying && fMediaSource != NULL) {
    if (fRTPSink != NULL) {
      fRTPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
      fAreCurrentlyPlaying = True;
    } else if (fUDPSink != NULL) {
      fUDPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
      fAreCurrentlyPlaying = True;
    }
  }

  if (fRTCPInstance == NULL && fRTPSink != NULL) {
    // Create (and start) a 'RTCP instance' for this RTP sink:
    fRTCPInstance
      = RTCPInstance::createNew(fRTPSink->envir(), fRTCPgs,
				fTotalBW, (unsigned char*)fMaster.fCNAME,
				fRTPSink, NULL /* we're a server */);
        // Note: This starts RTCP running automatically
  }

  if (dests->isTCP) {
    // Change RTP and RTCP to use the TCP socket instead of UDP:
    if (fRTPSink != NULL) {
      fRTPSink->addStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);
      fRTPSink->setServerRequestAlternativeByteHandler(serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
    }
    if (fRTCPInstance != NULL) {
      fRTCPInstance->addStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);
      fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,
					  rtcpRRHandler, rtcpRRHandlerClientData);
    }
  } else {
    // Tell the RTP and RTCP 'groupsocks' about this destination
    // (in case they don't already have it):
    if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort);
    if (fRTCPgs != NULL) fRTCPgs->addDestination(dests->addr, dests->rtcpPort);
    if (fRTCPInstance != NULL) {
      fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
					  rtcpRRHandler, rtcpRRHandlerClientData);
    }
  }
}

void StreamState::pause() {
  if (fRTPSink != NULL) fRTPSink->stopPlaying();
  if (fUDPSink != NULL) fUDPSink->stopPlaying();
  fAreCurrentlyPlaying = False;
}

void StreamState::endPlaying(Destinations* dests) {
  if (dests->isTCP) {
    if (fRTPSink != NULL) {
      fRTPSink->removeStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);
    }
    if (fRTCPInstance != NULL) {
      fRTCPInstance->removeStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);
      fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,
					  NULL, NULL);
    }
  } else {
    // Tell the RTP and RTCP 'groupsocks' to stop using these destinations:
    if (fRTPgs != NULL) fRTPgs->removeDestination(dests->addr, dests->rtpPort);
    if (fRTCPgs != NULL) fRTCPgs->removeDestination(dests->addr, dests->rtcpPort);
    if (fRTCPInstance != NULL) {
      fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
					  NULL, NULL);
    }
  }
}

void StreamState::reclaim() {
  // Delete allocated media objects
  Medium::close(fRTCPInstance) /* will send a RTCP BYE */; fRTCPInstance = NULL;
  Medium::close(fRTPSink); fRTPSink = NULL;
  Medium::close(fUDPSink); fUDPSink = NULL;

  fMaster.closeStreamSource(fMediaSource); fMediaSource = NULL;

  delete fRTPgs; fRTPgs = NULL;
  delete fRTCPgs; fRTCPgs = NULL;
}