当前位置: 首页>>代码示例>>C++>>正文


C++ HashTable::RemoveNext方法代码示例

本文整理汇总了C++中HashTable::RemoveNext方法的典型用法代码示例。如果您正苦于以下问题:C++ HashTable::RemoveNext方法的具体用法?C++ HashTable::RemoveNext怎么用?C++ HashTable::RemoveNext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HashTable的用法示例。


在下文中一共展示了HashTable::RemoveNext方法的2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: initiate

Boolean MediaSubsession::initiate(int useSpecialRTPoffset) {
  if (fReadSource != NULL) return True; // has already been initiated

  do {
    if (fCodecName == NULL) {
      env().setResultMsg("Codec is unspecified");
      break;
    }

    // Create RTP and RTCP 'Groupsocks' on which to receive incoming data.
    // (Groupsocks will work even for unicast addresses)
    struct in_addr tempAddr;
    tempAddr.s_addr = connectionEndpointAddress();
        // This could get changed later, as a result of a RTSP "SETUP"

    if (fClientPortNum != 0) {
      // The sockets' port numbers were specified for us.  Use these:
      fClientPortNum = fClientPortNum&~1; // even
      if (isSSM()) {
	fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, fClientPortNum);
      } else {
	fRTPSocket = new Groupsock(env(), tempAddr, fClientPortNum, 255);
      }
      if (fRTPSocket == NULL) {
	env().setResultMsg("Failed to create RTP socket");
	break;
      }
      
      // Set our RTCP port to be the RTP port +1
      portNumBits const rtcpPortNum = fClientPortNum|1;
      if (isSSM()) {
	fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
      } else {
	fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
      }
      if (fRTCPSocket == NULL) {
	char tmpBuf[100];
	sprintf(tmpBuf, "Failed to create RTCP socket (port %d)", rtcpPortNum);
	env().setResultMsg(tmpBuf);
	break;
      }
    } else {
      // Port numbers were not specified in advance, so we use ephemeral port numbers.
      // Create sockets until we get a port-number pair (even: RTP; even+1: RTCP).
      // We need to make sure that we don't keep trying to use the same bad port numbers over and over again.
      // so we store bad sockets in a table, and delete them all when we're done.
      HashTable* socketHashTable = HashTable::create(ONE_WORD_HASH_KEYS);
      if (socketHashTable == NULL) break;
      Boolean success = False;

      while (1) {
	// Create a new socket:
	if (isSSM()) {
	  fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, 0);
	} else {
	  fRTPSocket = new Groupsock(env(), tempAddr, 0, 255);
	}
	if (fRTPSocket == NULL) {
	  env().setResultMsg("MediaSession::initiate(): unable to create RTP and RTCP sockets");
	  break;
	}

	// Get the client port number, and check whether it's even (for RTP):
	Port clientPort(0);
	if (!getSourcePort(env(), fRTPSocket->socketNum(), clientPort)) {
	  break;
	}
	fClientPortNum = ntohs(clientPort.num()); 
	if ((fClientPortNum&1) != 0) { // it's odd
	  // Record this socket in our table, and keep trying:
	  unsigned key = (unsigned)fClientPortNum;
	  socketHashTable->Add((char const*)key, fRTPSocket);
	  continue;
	}

	// Make sure we can use the next (i.e., odd) port number, for RTCP:
	portNumBits rtcpPortNum = fClientPortNum|1;
	if (isSSM()) {
	  fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
	} else {
	  fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
	}
	if (fRTCPSocket != NULL) {
	  // Success! Use these two sockets (and delete any others that we've created):
	  Groupsock* oldGS;
	  while ((oldGS = (Groupsock*)socketHashTable->RemoveNext()) != NULL) {
	    delete oldGS;
	  }
	  delete socketHashTable;
	  success = True;
	  break;
	} else {
	  // We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).
	  // Record the first socket in our table, and keep trying:
	  unsigned key = (unsigned)fClientPortNum;
	  socketHashTable->Add((char const*)key, fRTPSocket);
	  continue;
	}
      }
      if (!success) break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue
//.........这里部分代码省略.........
开发者ID:dalinhuang,项目名称:ffmpeg-port,代码行数:101,代码来源:MediaSession.cpp

示例2: initiate


//.........这里部分代码省略.........
	if (fRTPSocket == NULL) {
	  env().setResultMsg("MediaSession::initiate(): unable to create RTP and RTCP sockets");
	  break;
	}

	// Get the client port number, and check whether it's even (for RTP):
	Port clientPort(0);
	if (!getSourcePort(env(), fRTPSocket->socketNum(), clientPort)) {
	  break;
	}
	fClientPortNum = ntohs(clientPort.num()); 
	if ((fClientPortNum&1) != 0) { // it's odd
	  // Record this socket in our table, and keep trying:
	  unsigned key = (unsigned)fClientPortNum;
	  Groupsock* existing = (Groupsock*)socketHashTable->Add((char const*)key, fRTPSocket);
	  delete existing; // in case it wasn't NULL
	  continue;
	}

	// Make sure we can use the next (i.e., odd) port number, for RTCP:
	portNumBits rtcpPortNum = fClientPortNum|1;
	if (isSSM()) {
	  fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
	} else {
	  fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
	}
	if (fRTCPSocket != NULL && fRTCPSocket->socketNum() >= 0) {
	  // Success! Use these two sockets.
	  success = True;
	  break;
	} else {
	  // We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).
	  delete fRTCPSocket;

	  // Record the first socket in our table, and keep trying:
	  unsigned key = (unsigned)fClientPortNum;
	  Groupsock* existing = (Groupsock*)socketHashTable->Add((char const*)key, fRTPSocket);
	  delete existing; // in case it wasn't NULL
	  continue;
	}
      }

      // Clean up the socket hash table (and contents):
      Groupsock* oldGS;
      while ((oldGS = (Groupsock*)socketHashTable->RemoveNext()) != NULL) {
	delete oldGS;
      }
      delete socketHashTable;

      if (!success) break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue
    }

    // Try to use a big receive buffer for RTP - at least 0.1 second of
    // specified bandwidth and at least 50 KB
    unsigned rtpBufSize = fBandwidth * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
    if (rtpBufSize < 50 * 1024)
      rtpBufSize = 50 * 1024;
    increaseReceiveBufferTo(env(), fRTPSocket->socketNum(), rtpBufSize);

    if (isSSM() && fRTCPSocket != NULL) {
      // Special case for RTCP SSM: Send RTCP packets back to the source via unicast:
      fRTCPSocket->changeDestinationParameters(fSourceFilterAddr,0,~0);
    }

    // Create "fRTPSource" and "fReadSource":
    if (!createSourceObjects(useSpecialRTPoffset)) break;

    if (fReadSource == NULL) {
      env().setResultMsg("Failed to create read source");
      break;
    }

    // Finally, create our RTCP instance. (It starts running automatically)
    if (fRTPSource != NULL && fRTCPSocket != NULL) {
      // If bandwidth is specified, use it and add 5% for RTCP overhead.
      // Otherwise make a guess at 500 kbps.
      unsigned totSessionBandwidth
	= fBandwidth ? fBandwidth + fBandwidth / 20 : 500;
      fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,
					      totSessionBandwidth,
					      (unsigned char const*)
					      fParent.CNAME(),
					      NULL /* we're a client */,
					      fRTPSource);
      if (fRTCPInstance == NULL) {
	env().setResultMsg("Failed to create RTCP instance");
	break;
      }
    }

    return True;
  } while (0);

  delete fRTPSocket; fRTPSocket = NULL;
  delete fRTCPSocket; fRTCPSocket = NULL;
  Medium::close(fRTCPInstance); fRTCPInstance = NULL;
  Medium::close(fReadSource); fReadSource = fRTPSource = NULL;
  fClientPortNum = 0;
  return False;
}
开发者ID:hatboyzero,项目名称:live456,代码行数:101,代码来源:MediaSession.cpp


注:本文中的HashTable::RemoveNext方法示例由纯净天空整理自Github/MSDocs等开源代码及文档管理平台,相关代码片段筛选自各路编程大神贡献的开源项目,源码版权归原作者所有,传播和使用请参考对应项目的License;未经允许,请勿转载。