multiframedrtpsink.cpp

H.264 RTSP 串流(live 555)視窗版本

    fOutBuf->skipBytes(fCurFrameSpecificHeaderSize);
    fTotalFrameSpecificHeaderSizes += fCurFrameSpecificHeaderSize;

    fSource->getNextFrame(fOutBuf->curPtr(), fOutBuf->totalBytesAvailable(),
			  afterGettingFrame, this, ourHandleClosure, this);
  }
}

void MultiFramedRTPSink
::afterGettingFrame(void* clientData, unsigned numBytesRead,
		    unsigned numTruncatedBytes,
		    struct timeval presentationTime,
		    unsigned durationInMicroseconds) {
  MultiFramedRTPSink* sink = (MultiFramedRTPSink*)clientData;
  sink->afterGettingFrame1(numBytesRead, numTruncatedBytes,
			   presentationTime, durationInMicroseconds);
}

void MultiFramedRTPSink
::afterGettingFrame1(unsigned frameSize, unsigned numTruncatedBytes,
		     struct timeval presentationTime,
		     unsigned durationInMicroseconds) {
  if (fIsFirstPacket) {
    // Record the fact that we're starting to play now:
    gettimeofday(&fNextSendTime, NULL);
  }

  if (numTruncatedBytes > 0) {
    unsigned const bufferSize = fOutBuf->totalBytesAvailable();
    unsigned newMaxSize = frameSize + numTruncatedBytes;
    envir() << "MultiFramedRTPSink::afterGettingFrame1(): The input frame data was too large for our buffer size ("
	    << bufferSize << ").  "
	    << numTruncatedBytes << " bytes of trailing data was dropped!  Correct this by increasing \"OutPacketBuffer::maxSize\" to at least "
	    << newMaxSize << ", *before* creating this 'RTPSink'.  (Current value is "
	    << OutPacketBuffer::maxSize << ".)\n";
  }
  unsigned curFragmentationOffset = fCurFragmentationOffset;
  unsigned numFrameBytesToUse = frameSize;
  unsigned overflowBytes = 0;

  // If we have already packed one or more frames into this packet,
  // check whether this new frame is eligible to be packed after them.
  // (This is independent of whether the packet has enough room for this
  // new frame; that check comes later.)
  if (fNumFramesUsedSoFar > 0) {
    if ((fPreviousFrameEndedFragmentation
	 && !allowOtherFramesAfterLastFragment())
	|| !frameCanAppearAfterPacketStart(fOutBuf->curPtr(), frameSize)) {
      // Save away this frame for next time:
      numFrameBytesToUse = 0;
      fOutBuf->setOverflowData(fOutBuf->curPacketSize(), frameSize,
			       presentationTime, durationInMicroseconds);
    }
  }
  fPreviousFrameEndedFragmentation = False;

  if (numFrameBytesToUse > 0) {
    // Check whether this frame overflows the packet
    if (fOutBuf->wouldOverflow(frameSize)) {
      // Don't use this frame now; instead, save it as overflow data, and
      // send it in the next packet instead.  However, if the frame is too
      // big to fit in a packet by itself, then we need to fragment it (and
      // use some of it in this packet, if the payload format permits this.)
      if (isTooBigForAPacket(frameSize)
          && (fNumFramesUsedSoFar == 0 || allowFragmentationAfterStart())) {
        // We need to fragment this frame, and use some of it now:
        overflowBytes = computeOverflowForNewFrame(frameSize);
        numFrameBytesToUse -= overflowBytes;
        fCurFragmentationOffset += numFrameBytesToUse;
      } else {
        // We don't use any of this frame now:
        overflowBytes = frameSize;
        numFrameBytesToUse = 0;
      }
      fOutBuf->setOverflowData(fOutBuf->curPacketSize() + numFrameBytesToUse,
			       overflowBytes, presentationTime, durationInMicroseconds);
    } else if (fCurFragmentationOffset > 0) {
      // This is the last fragment of a frame that was fragmented over
      // more than one packet.  Do any special handling for this case:
      fCurFragmentationOffset = 0;
      fPreviousFrameEndedFragmentation = True;
    }
  }

  if (numFrameBytesToUse == 0) {
    // Send our packet now, because we have filled it up:
    sendPacketIfNecessary();
  } else {
    // Use this frame in our outgoing packet:
    unsigned char* frameStart = fOutBuf->curPtr();
    fOutBuf->increment(numFrameBytesToUse);
        // do this now, in case "doSpecialFrameHandling()" calls "setFramePadding()" to append padding bytes

    // Here's where any payload format specific processing gets done:
    doSpecialFrameHandling(curFragmentationOffset, frameStart,
			   numFrameBytesToUse, presentationTime,
			   overflowBytes);

    ++fNumFramesUsedSoFar;

    // Update the time at which the next packet should be sent, based
    // on the duration of the frame that we just packed into it.
    // However, if this frame has overflow data remaining, then don't
    // count its duration yet.
    if (overflowBytes == 0) {
      fNextSendTime.tv_usec += durationInMicroseconds;
      fNextSendTime.tv_sec += fNextSendTime.tv_usec/1000000;
      fNextSendTime.tv_usec %= 1000000;
    }

    // Send our packet now if (i) it's already at our preferred size, or
    // (ii) (heuristic) another frame of the same size as the one we just
    //      read would overflow the packet, or
    // (iii) it contains the last fragment of a fragmented frame, and we
    //      don't allow anything else to follow this or
    // (iv) one frame per packet is allowed:
    if (fOutBuf->isPreferredSize()
        || fOutBuf->wouldOverflow(numFrameBytesToUse)
        || (fPreviousFrameEndedFragmentation &&
            !allowOtherFramesAfterLastFragment())
        || !frameCanAppearAfterPacketStart(fOutBuf->curPtr() - frameSize,
					   frameSize) ) {
      // The packet is ready to be sent now
      sendPacketIfNecessary();
    } else {
      // There's room for more frames; try getting another:
      packFrame();
    }
  }
}

static unsigned const rtpHeaderSize = 12;

Boolean MultiFramedRTPSink::isTooBigForAPacket(unsigned numBytes) const {
  // Check whether a 'numBytes'-byte frame - together with a RTP header and
  // (possible) special headers - would be too big for an output packet:
  // (Later allow for RTP extension header!) #####
  numBytes += rtpHeaderSize + specialHeaderSize() + frameSpecificHeaderSize();
  return fOutBuf->isTooBigForAPacket(numBytes);
}

void MultiFramedRTPSink::sendPacketIfNecessary() {
  if (fNumFramesUsedSoFar > 0) {
    // Send the packet:
#ifdef TEST_LOSS
    if ((our_random()%10) != 0) // simulate 10% packet loss #####
#endif
    fRTPInterface.sendPacket(fOutBuf->packet(), fOutBuf->curPacketSize());
    ++fPacketCount;
    fTotalOctetCount += fOutBuf->curPacketSize();
    fOctetCount += fOutBuf->curPacketSize()
      - rtpHeaderSize - fSpecialHeaderSize - fTotalFrameSpecificHeaderSizes;

    ++fSeqNo; // for next time
  }

  if (fOutBuf->haveOverflowData()
      && fOutBuf->totalBytesAvailable() > fOutBuf->totalBufferSize()/2) {
    // Efficiency hack: Reset the packet start pointer to just in front of
    // the overflow data (allowing for the RTP header and special headers),
    // so that we probably don't have to "memmove()" the overflow data
    // into place when building the next packet:
    unsigned newPacketStart = fOutBuf->curPacketSize()
      - (rtpHeaderSize + fSpecialHeaderSize + frameSpecificHeaderSize());
    fOutBuf->adjustPacketStart(newPacketStart);
  } else {
    // Normal case: Reset the packet start pointer back to the start:
    fOutBuf->resetPacketStart();
  }
  fOutBuf->resetOffset();
  fNumFramesUsedSoFar = 0;

  if (fNoFramesLeft) {
    // We're done:
    onSourceClosure(this);
  } else {
    // We have more frames left to send.  Figure out when the next frame
    // is due to start playing, then make sure that we wait this long before
    // sending the next packet.
    struct timeval timeNow;
    gettimeofday(&timeNow, NULL);
    int uSecondsToGo;
    if (fNextSendTime.tv_sec < timeNow.tv_sec
	|| (fNextSendTime.tv_sec == timeNow.tv_sec && fNextSendTime.tv_usec < timeNow.tv_usec)) {
      uSecondsToGo = 0; // prevents integer underflow if too far behind
    } else {
      uSecondsToGo = (fNextSendTime.tv_sec - timeNow.tv_sec)*1000000 + (fNextSendTime.tv_usec - timeNow.tv_usec);
    }

    // Delay this amount of time:
    nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecondsToGo,
						(TaskFunc*)sendNext, this);
  }
}

// The following is called after each delay between packet sends:
void MultiFramedRTPSink::sendNext(void* firstArg) {
  MultiFramedRTPSink* sink = (MultiFramedRTPSink*)firstArg;
  sink->buildAndSendPacket(False);
}

void MultiFramedRTPSink::ourHandleClosure(void* clientData) {
  MultiFramedRTPSink* sink = (MultiFramedRTPSink*)clientData;
  // There are no frames left, but we may have a partially built packet
  //  to send
  sink->fNoFramesLeft = True;
  sink->sendPacketIfNecessary();
}