live555中关于rtp包接收后的处理

// Check for the 12-byte RTP header:

    if (bPacket->dataSize() < 12) break;

    unsigned rtpHdr = ntohl(*(unsigned*)(bPacket->data())); ADVANCE(4);

    Boolean rtpMarkerBit = (rtpHdr&0x00800000) >> 23;

    unsigned rtpTimestamp = ntohl(*(unsigned*)(bPacket->data()));ADVANCE(4);

    unsigned rtpSSRC = ntohl(*(unsigned*)(bPacket->data())); ADVANCE(4);

 

    // Check the RTP version number (it should be 2):

    if ((rtpHdr&0xC0000000) != 0x80000000) break;

 

    // Skip over any CSRC identifiers in the header:

    unsigned cc = (rtpHdr>>24)&0xF;

    if (bPacket->dataSize() < cc) break;

    ADVANCE(cc*4);

 

    // Check for (& ignore) any RTP header extension

    if (rtpHdr&0x10000000) {

      if (bPacket->dataSize() < 4) break;

      unsigned extHdr = ntohl(*(unsigned*)(bPacket->data())); ADVANCE(4);

      unsigned remExtSize = 4*(extHdr&0xFFFF);

      if (bPacket->dataSize() < remExtSize) break;

      ADVANCE(remExtSize);

    }

 

    // Discard any padding bytes:

    if (rtpHdr&0x20000000) {

      if (bPacket->dataSize() == 0) break;

      unsigned numPaddingBytes

       = (unsigned)(bPacket->data())[bPacket->dataSize()-1];

      if (bPacket->dataSize() < numPaddingBytes) break;

      bPacket->removePadding(numPaddingBytes);

    }

    // Check the Payload Type.

    if ((unsigned char)((rtpHdr&0x007F0000)>>16)

       != source->rtpPayloadFormat()) {

      break;

    }

 

    // The rest of the packet is the usable data.  Record and save it:

    source->fLastReceivedSSRC = rtpSSRC;

    unsigned short rtpSeqNo = (unsigned short)(rtpHdr&0xFFFF);

    Boolean usableInJitterCalculation

      = source->packetIsUsableInJitterCalculation((bPacket->data()),

                                            bPacket->dataSize());

    struct timeval presentationTime; // computed by:

    Boolean hasBeenSyncedUsingRTCP; // computed by:

    source->receptionStatsDB()

      .noteIncomingPacket(rtpSSRC, rtpSeqNo, rtpTimestamp,

                       source->timestampFrequency(),

                       usableInJitterCalculation, presentationTime,

                       hasBeenSyncedUsingRTCP, bPacket->dataSize());

 

    // Fill in the rest of the packet descriptor, and store it:

    struct timeval timeNow;

    gettimeofday(&timeNow, NULL);

    bPacket->assignMiscParams(rtpSeqNo, rtpTimestamp, presentationTime,

                           hasBeenSyncedUsingRTCP, rtpMarkerBit,

                           timeNow);

    if (!source->fReorderingBuffer->storePacket(bPacket)) break;

 

readSuccess = True;

BufferedPacket* ReorderingPacketBuffer
::getNextCompletedPacket(Boolean& packetLossPreceded) {
  if (fHeadPacket == NULL) return NULL;

关于丢包的处理：

  // Check whether the next packet we want is already at the head
  // of the queue:检查接受包中的序列号是否是所希望的序列号
    if (fHeadPacket->rtpSeqNo() == fNextExpectedSeqNo) {
    packetLossPreceded = fHeadPacket->isFirstPacket();
        // (The very first packet is treated as if there was packet loss beforehand.)
    return fHeadPacket;
  }

  // We're still waiting for our desired packet to arrive.  However, if
  // our time threshold has been exceeded, then forget it, and return
  // the head packet instead:等待所希望的包的到来，一段时间后丢弃
  struct timeval timeNow;
  gettimeofday(&timeNow, NULL);
  unsigned uSecondsSinceReceived
    = (timeNow.tv_sec - fHeadPacket->timeReceived().tv_sec)*1000000
    + (timeNow.tv_usec - fHeadPacket->timeReceived().tv_usec);
  if (uSecondsSinceReceived > fThresholdTime) {
    fNextExpectedSeqNo = fHeadPacket->rtpSeqNo();
        // we've given up on earlier packets now
    packetLossPreceded = True;
    return fHeadPacket;
  }

  // Otherwise, keep waiting for our desired packet to arrive:
  return NULL;
}

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