vlc源码分析之调用live555接收RTSP数据

Boolean RTPInterface::handleRead(unsigned char* buffer, unsigned bufferMaxSize,
                 unsigned& bytesRead, struct sockaddr_in& fromAddress, Boolean& packetReadWasIncomplete)

Boolean BufferedPacket::fillInData(RTPInterface& rtpInterface, Boolean& packetReadWasIncomplete)；

static H264VideoRTPSource* createNew(UsageEnvironment& env, Groupsock* RTPgs,
　　unsigned char rtpPayloadFormat,
　　unsigned rtpTimestampFrequency = 90000);

p_sys->scheduler->doEventLoop( &p_sys->event_data );

// 做了两件事，一件是分析RTP包，放入ReorderingPacketBuffer管理的BufferedPacket中；
// 另一件是读取的BufferedPacket，进行一系列拆包操作后，将数据放入数据fifo中
void MultiFramedRTPSource::networkReadHandler1() {
  BufferedPacket* bPacket = fPacketReadInProgress;
  if (bPacket == NULL) {
    // Normal case: Get a free BufferedPacket descriptor to hold the new network packet:
    bPacket = fReorderingBuffer->getFreePacket(this);
  }

  // Read the network packet, and perform sanity checks on the RTP header:
  Boolean readSuccess = False;
  // do-while(0)结构，出现错误直接break
  do {
    Boolean packetReadWasIncomplete = fPacketReadInProgress != NULL;
    if (!bPacket->fillInData(fRTPInterface, packetReadWasIncomplete)) break;
    if (packetReadWasIncomplete) {
      // We need additional read(s) before we can process the incoming packet:
      fPacketReadInProgress = bPacket;
      return;
    } else {
      fPacketReadInProgress = NULL;
    }
#ifdef TEST_LOSS
    setPacketReorderingThresholdTime(0);
       // don‘t wait for ‘lost‘ packets to arrive out-of-order later
    if ((our_random()%10) == 0) break; // simulate 10% packet loss
#endif

    // Check for the 12-byte RTP header:
    if (bPacket->dataSize() < 12) break;
    // 读取RTP头，向前移4个字节
    unsigned rtpHdr = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);
    // 读取RTP头中的标记位
    Boolean rtpMarkerBit = (rtpHdr&0x00800000) != 0;
    // 读取时间戳，向前移4个字节
    unsigned rtpTimestamp = ntohl(*(u_int32_t*)(bPacket->data()));ADVANCE(4);
    // 读取SSRC，向前移4个字节
    unsigned rtpSSRC = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);

    // Check the RTP version number (it should be 2):
    // 检查RTP头版本，不是2的话，break
    if ((rtpHdr&0xC0000000) != 0x80000000) break;

    // Skip over any CSRC identifiers in the header:
    // 跳过CSRC计数字节
    unsigned cc = (rtpHdr>>24)&0xF;
    if (bPacket->dataSize() < cc) break;
    ADVANCE(cc*4);

    // Check for (& ignore) any RTP header extension
    // 如果扩展头标志被置位
    if (rtpHdr&0x10000000) {
      if (bPacket->dataSize() < 4) break;
      // 获取扩展头
      unsigned extHdr = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);
      // 获取扩展字节数
      unsigned remExtSize = 4*(extHdr&0xFFFF);
      if (bPacket->dataSize() < remExtSize) break;
      // 直接跳过扩展字节？？？
      ADVANCE(remExtSize);
    }

    // Discard any padding bytes:
    // 如果填充标志被置位，直接丢弃不处理
    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.
    // 检查载荷类型，如果源数据H264类型，则其值为96
    // 如果与我们生成的source类型不同，则break
    if ((unsigned char)((rtpHdr&0x007F0000)>>16)
    != rtpPayloadFormat()) {
      break;
    }

    // The rest of the packet is the usable data.  Record and save it:
    if (rtpSSRC != fLastReceivedSSRC) {
      // The SSRC of incoming packets has changed.  Unfortunately we don‘t yet handle streams that contain multiple SSRCs,
      // but we can handle a single-SSRC stream where the SSRC changes occasionally:
      fLastReceivedSSRC = rtpSSRC;
      fReorderingBuffer->resetHaveSeenFirstPacket();
    }
    // RTP包序号，随RTP数据包而自增，由接收者用来探测包损失
    unsigned short rtpSeqNo = (unsigned short)(rtpHdr&0xFFFF);
    Boolean usableInJitterCalculation
      = packetIsUsableInJitterCalculation((bPacket->data()),
                          bPacket->dataSize());
    struct timeval presentationTime; // computed by:
    Boolean hasBeenSyncedUsingRTCP; // computed by:
    // 根据数据包的一些信息，进行一些计算和记录
    receptionStatsDB()
      .noteIncomingPacket(rtpSSRC, rtpSeqNo, rtpTimestamp,
              timestampFrequency(),
              usableInJitterCalculation, presentationTime,
              hasBeenSyncedUsingRTCP, bPacket->dataSize());

    // Fill in the rest of the packet descriptor, and store it:
    struct timeval timeNow;
    gettimeofday(&timeNow, NULL);
    // 将计算所得的一些参数再赋值到包中
    bPacket->assignMiscParams(rtpSeqNo, rtpTimestamp, presentationTime,
                  hasBeenSyncedUsingRTCP, rtpMarkerBit,
                  timeNow);
    // 经过以上判断和检查，没有发现问题，则由管理类fReorderingBuffer存储包
    if (!fReorderingBuffer->storePacket(bPacket)) break;

    readSuccess = True;// 读取成功
  } while (0);
  if (!readSuccess) fReorderingBuffer->freePacket(bPacket);// 如果读取不成功，则释放内存

  // 将读取到的数据包送至数据fifo中，等待解码线程解码
  doGetNextFrame1();
  // If we didn‘t get proper data this time, we‘ll get another chance
}