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 }