发现live555中一个小bug

今天忽然发现了live555中的openRTSP.exe中的一个小问题。

在openRTSP.exe连上服务器后，如果服务器退出而没有发送TEARDOWN，openRTSP.exe不会退出，即使收不到数据，它也会一直等待下去。当然，这也可以不认为是一个bug，因为服务器不辞而别也不对。究其原因，主要是openRTSP.exe中所使用的RTPSource类没有对网络出错进行处理，所以尽管网络出错，收不到数据，接收数据的循环会一直进行下去。从这个解度讲也算是个bug吧。

要改正很容易，下面用//---------------标注者即是改正处：

 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 {
         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;
         unsigned rtpHdr = ntohl(*(u_int32_t*) (bPacket->data()));
         ADVANCE(4);
         Boolean rtpMarkerBit = (rtpHdr & 0x00800000) >> 23;
         unsigned rtpTimestamp = ntohl(*(u_int32_t*) (bPacket->data()));
         ADVANCE(4);
         unsigned rtpSSRC = ntohl(*(u_int32_t*) (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(*(u_int32_t*) (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)
                 != 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();
         }
         unsigned short rtpSeqNo = (unsigned short) (rtpHdr & 0xFFFF);
         Boolean usableInJitterCalculation = packetIsUsableInJitterCalculation(
                 (bPacket->data()), bPacket->dataSize());
         struct timeval presentationTime; // computed by:
         Boolean hasBeenSyncedUsingRTCP; // computed by:
         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);
         bPacket->assignMiscParams(rtpSeqNo, rtpTimestamp, presentationTime,
                 hasBeenSyncedUsingRTCP, rtpMarkerBit, timeNow);
         if (!fReorderingBuffer->storePacket(bPacket))
             break;

         readSuccess = True;
     } while (0);
     if (!readSuccess){
         fReorderingBuffer->freePacket(bPacket);
         //------------通知调用者，我结束了！-----------------------------
         RTPSource::handleClosure(this);
         return;
         //--------------------------------------------------------------
     }

     doGetNextFrame1();
     // If we didn't get proper data this time, we'll get another chance
 }

转自http://blog.csdn.net/nkmnkm/article/details/7003956