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C++ Groupsock::removeAllDestinations方法代码示例

本文整理汇总了C++中Groupsock::removeAllDestinations方法的典型用法代码示例。如果您正苦于以下问题:C++ Groupsock::removeAllDestinations方法的具体用法?C++ Groupsock::removeAllDestinations怎么用?C++ Groupsock::removeAllDestinations使用的例子?那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在Groupsock的用法示例。


在下文中一共展示了Groupsock::removeAllDestinations方法的1个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C++代码示例。

示例1: dummy

void OnDemandServerMediaSubsession
::getStreamParameters(unsigned clientSessionId,
		      netAddressBits clientAddress,
		      Port const& clientRTPPort,
		      Port const& clientRTCPPort,
		      int tcpSocketNum,
		      unsigned char rtpChannelId,
		      unsigned char rtcpChannelId,
		      netAddressBits& destinationAddress,
		      u_int8_t& /*destinationTTL*/,
		      Boolean& isMulticast,
		      Port& serverRTPPort,
		      Port& serverRTCPPort,
		      void*& streamToken) {
  if (destinationAddress == 0) destinationAddress = clientAddress;
  struct in_addr destinationAddr; destinationAddr.s_addr = destinationAddress;
  isMulticast = False;

  if (fLastStreamToken != NULL && fReuseFirstSource) {
    // Special case: Rather than creating a new 'StreamState',
    // we reuse the one that we've already created:
    serverRTPPort = ((StreamState*)fLastStreamToken)->serverRTPPort();
    serverRTCPPort = ((StreamState*)fLastStreamToken)->serverRTCPPort();
    ++((StreamState*)fLastStreamToken)->referenceCount();
    streamToken = fLastStreamToken;
  } else {
    // Normal case: Create a new media source:
    unsigned streamBitrate;
    FramedSource* mediaSource
      = createNewStreamSource(clientSessionId, streamBitrate);

    // Create 'groupsock' and 'sink' objects for the destination,
    // using previously unused server port numbers:
    RTPSink* rtpSink;
    BasicUDPSink* udpSink;
    Groupsock* rtpGroupsock;
    Groupsock* rtcpGroupsock;
    portNumBits serverPortNum;
    if (clientRTCPPort.num() == 0) {
      // We're streaming raw UDP (not RTP). Create a single groupsock:
      NoReuse dummy(envir()); // ensures that we skip over ports that are already in use
      for (serverPortNum = fInitialPortNum; ; ++serverPortNum) {
	struct in_addr dummyAddr; dummyAddr.s_addr = 0;

	serverRTPPort = serverPortNum;
	rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
	if (rtpGroupsock->socketNum() >= 0) break; // success
      }

      rtcpGroupsock = NULL;
      rtpSink = NULL;
      udpSink = BasicUDPSink::createNew(envir(), rtpGroupsock);
    } else {
      // Normal case: We're streaming RTP (over UDP or TCP).  Create a pair of
      // groupsocks (RTP and RTCP), with adjacent port numbers (RTP port number even):
      NoReuse dummy(envir()); // ensures that we skip over ports that are already in use
      for (portNumBits serverPortNum = fInitialPortNum; ; serverPortNum += 2) {
	struct in_addr dummyAddr; dummyAddr.s_addr = 0;

	serverRTPPort = serverPortNum;
	rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
	if (rtpGroupsock->socketNum() < 0) {
	  delete rtpGroupsock;
	  continue; // try again
	}

	serverRTCPPort = serverPortNum+1;
	rtcpGroupsock = new Groupsock(envir(), dummyAddr, serverRTCPPort, 255);
	if (rtcpGroupsock->socketNum() < 0) {
	  delete rtpGroupsock;
	  delete rtcpGroupsock;
	  continue; // try again
	}

	break; // success
      }

      unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
      rtpSink = createNewRTPSink(rtpGroupsock, rtpPayloadType, mediaSource);
      udpSink = NULL;
    }

    // Turn off the destinations for each groupsock.  They'll get set later
    // (unless TCP is used instead):
    if (rtpGroupsock != NULL) rtpGroupsock->removeAllDestinations();
    if (rtcpGroupsock != NULL) rtcpGroupsock->removeAllDestinations();

    if (rtpGroupsock != NULL) {
      // Try to use a big send buffer for RTP -  at least 0.1 second of
      // specified bandwidth and at least 50 KB
      unsigned rtpBufSize = streamBitrate * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
      if (rtpBufSize < 50 * 1024) rtpBufSize = 50 * 1024;
      increaseSendBufferTo(envir(), rtpGroupsock->socketNum(), rtpBufSize);
    }

    // Set up the state of the stream.  The stream will get started later:
    streamToken = fLastStreamToken
      = new StreamState(*this, serverRTPPort, serverRTCPPort, rtpSink, udpSink,
			streamBitrate, mediaSource,
			rtpGroupsock, rtcpGroupsock);
//.........这里部分代码省略.........
开发者ID:DayDayUpCQ,项目名称:live555,代码行数:101,代码来源:OnDemandServerMediaSubsession.cpp


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