Live555还提供了录像的示例程序，在testProgs目录下的playCommon.cpp中，Live555录像的基本原理就是创建一个RTSPClient去请求指定rtsp地址的视频，然后保存到文件里。
playCommon.cpp打开一看就发现首先是各种全局函数的声明，然后是各种全局变量的声明，然后是main函数和各个函数的实现。main函数中首先还是创建TaskScheduler对象和UsageEnvironment对象，然后根据各种输入参数设置各种全局变量，最后就是创建一个RTSPClient对象请求指定rtsp地址的视频。
1 int main(int argc, char** argv) 2 { 3 // Begin by setting up our usage environment: 4 TaskScheduler* scheduler = BasicTaskScheduler::createNew(); 5 env = BasicUsageEnvironment::createNew(*scheduler); 6 7 /* 8 处理各种输入参数，在此省略 9 */ 10 streamURL = argv[1]; 11 12 // Create (or arrange to create) our client object: 13 if (createHandlerServerForREGISTERCommand) { 14 handlerServerForREGISTERCommand 15 = HandlerServerForREGISTERCommand::createNew(*env, continueAfterClientCreation0, 16 handlerServerForREGISTERCommandPortNum, authDBForREGISTER, 17 verbosityLevel, progName); 18 if (handlerServerForREGISTERCommand == NULL) { 19 *env << "Failed to create a server for handling incoming "REGISTER" commands: " << env->getResultMsg() << " "; 20 } else { 21 *env << "Awaiting an incoming "REGISTER" command on port " << handlerServerForREGISTERCommand->serverPortNum() << " "; 22 } 23 } else { 24 ourClient = createClient(*env, streamURL, verbosityLevel, progName); 25 if (ourClient == NULL) { 26 *env << "Failed to create " << clientProtocolName << " client: " << env->getResultMsg() << " "; 27 shutdown(); 28 } 29 continueAfterClientCreation1(); 30 } 31 32 // All subsequent activity takes place within the event loop: 33 env->taskScheduler().doEventLoop(); // does not return 34 35 return 0; // only to prevent compiler warning 36 } 37
createClient函数在openRTSP.cpp文件中定义，内容很简单，就是调用了RTSPClient::createNew函数创建了一个RTSPClient对象。我们来看continueAfterClientCreation1函数
1 void continueAfterClientCreation1() { 2 setUserAgentString(userAgent); 3 4 if (sendOptionsRequest) { 5 // Begin by sending an "OPTIONS" command: 6 getOptions(continueAfterOPTIONS); // 发送OPTIONS命令，我们也可以跳过这一步直接发送DESCRIBE命令 7 } else { 8 continueAfterOPTIONS(NULL, 0, NULL); 9 } 10 } 11 12 void getOptions(RTSPClient::responseHandler* afterFunc) { 13 ourRTSPClient->sendOptionsCommand(afterFunc, ourAuthenticator); 14 } 15 16 void continueAfterOPTIONS(RTSPClient*, int resultCode, char* resultString) { 17 if (sendOptionsRequestOnly) { 18 if (resultCode != 0) { 19 *env << clientProtocolName << " "OPTIONS" request failed: " << resultString << " "; 20 } else { 21 *env << clientProtocolName << " "OPTIONS" request returned: " << resultString << " "; 22 } 23 shutdown(); 24 } 25 delete[] resultString; 26 27 // Next, get a SDP description for the stream: 28 getSDPDescription(continueAfterDESCRIBE); // 发送DESCRIBE命令 29 } 30 31 void getSDPDescription(RTSPClient::responseHandler* afterFunc) { 32 ourRTSPClient->sendDescribeCommand(afterFunc, ourAuthenticator); 33 } 34 35 void continueAfterDESCRIBE(RTSPClient*, int resultCode, char* resultString) { 36 if (resultCode != 0) { 37 *env << "Failed to get a SDP description for the URL "" << streamURL << "": " << resultString << " "; 38 delete[] resultString; 39 shutdown(); 40 } 41 42 char* sdpDescription = resultString; 43 *env << "Opened URL "" << streamURL << "", returning a SDP description: " << sdpDescription << " "; 44 45 // Create a media session object from this SDP description: 46 session = MediaSession::createNew(*env, sdpDescription); //创建MediaSession 47 delete[] sdpDescription; 48 if (session == NULL) { 49 *env << "Failed to create a MediaSession object from the SDP description: " << env->getResultMsg() << " "; 50 shutdown(); 51 } else if (!session->hasSubsessions()) { 52 *env << "This session has no media subsessions (i.e., no "m=" lines) "; 53 shutdown(); 54 } 55 56 // Then, setup the "RTPSource"s for the session: 57 MediaSubsessionIterator iter(*session); 58 MediaSubsession *subsession; 59 Boolean madeProgress = False; 60 char const* singleMediumToTest = singleMedium; 61 while ((subsession = iter.next()) != NULL) { 62 // If we've asked to receive only a single medium, then check this now: 63 if (singleMediumToTest != NULL) { 64 if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) { 65 *env << "Ignoring "" << subsession->mediumName() 66 << "/" << subsession->codecName() 67 << "" subsession, because we've asked to receive a single " << singleMedium 68 << " session only "; 69 continue; 70 } else { 71 // Receive this subsession only 72 singleMediumToTest = "xxxxx"; 73 // this hack ensures that we get only 1 subsession of this type 74 } 75 } 76 77 if (desiredPortNum != 0) { 78 subsession->setClientPortNum(desiredPortNum); //创建相关的RTPSource、Groupsock等资源 79 desiredPortNum += 2; 80 } 81 82 if (createReceivers) { //我们接收数据然后保存在文件中，createReceivers为true 83 if (!subsession->initiate(simpleRTPoffsetArg)) { //初始化MediaSubsession 84 *env << "Unable to create receiver for "" << subsession->mediumName() 85 << "/" << subsession->codecName() 86 << "" subsession: " << env->getResultMsg() << " "; 87 } else { 88 *env << "Created receiver for "" << subsession->mediumName() 89 << "/" << subsession->codecName() << "" subsession ("; 90 if (subsession->rtcpIsMuxed()) { 91 *env << "client port " << subsession->clientPortNum(); 92 } else { 93 *env << "client ports " << subsession->clientPortNum() 94 << "-" << subsession->clientPortNum()+1; 95 } 96 *env << ") "; 97 madeProgress = True; 98 99 if (subsession->rtpSource() != NULL) { 100 // Because we're saving the incoming data, rather than playing 101 // it in real time, allow an especially large time threshold 102 // (1 second) for reordering misordered incoming packets: 103 unsigned const thresh = 1000000; // 1 second 104 subsession->rtpSource()->setPacketReorderingThresholdTime(thresh); 105 106 // Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B), 107 // or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size. 108 // (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size, 109 // then the input data rate may be large enough to justify increasing the OS socket buffer size also.) 110 int socketNum = subsession->rtpSource()->RTPgs()->socketNum(); 111 unsigned curBufferSize = getReceiveBufferSize(*env, socketNum); 112 if (socketInputBufferSize > 0 || fileSinkBufferSize > curBufferSize) { 113 unsigned newBufferSize = socketInputBufferSize > 0 ? socketInputBufferSize : fileSinkBufferSize; 114 newBufferSize = setReceiveBufferTo(*env, socketNum, newBufferSize); 115 if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it: 116 *env << "Changed socket receive buffer size for the "" 117 << subsession->mediumName() 118 << "/" << subsession->codecName() 119 << "" subsession from " 120 << curBufferSize << " to " 121 << newBufferSize << " bytes "; 122 } 123 } 124 } 125 } 126 } else { 127 if (subsession->clientPortNum() == 0) { 128 *env << "No client port was specified for the "" 129 << subsession->mediumName() 130 << "/" << subsession->codecName() 131 << "" subsession. (Try adding the "-p <portNum>" option.) "; 132 } else { 133 madeProgress = True; 134 } 135 } 136 } 137 if (!madeProgress) shutdown(); 138 139 // Perform additional 'setup' on each subsession, before playing them: 140 setupStreams(); //对每个ServerMediaSubsession发送SETUP命令 141 }
上面的流程和RTSPClient端与服务器建立连接的过程基本类似，先发送OPTIONS命令，然后发送DESCRIBE命令，然后发送SETUP命令。我们在此也可以忽略发送OPTIONS命令，直接从发送DESCRIBE命令开始。在setupStreams函数中分别对每个ServerMediaSubsession发送SETUP命令，我们来看一下setupStreams函数
1 void setupStreams() { 2 static MediaSubsessionIterator* setupIter = NULL; 3 if (setupIter == NULL) setupIter = new MediaSubsessionIterator(*session); 4 while ((subsession = setupIter->next()) != NULL) { 5 // We have another subsession left to set up: 6 if (subsession->clientPortNum() == 0) continue; // port # was not set 7 8 setupSubsession(subsession, streamUsingTCP, forceMulticastOnUnspecified, continueAfterSETUP); //发送SETUP命令，建立与ServerMediaSubsession的连接 9 return; 10 } 11 12 // We're done setting up subsessions. //与所有的ServerMediaSubsession建立连接成功 13 delete setupIter; 14 if (!madeProgress) shutdown(); 15 16 // Create output files: 17 if (createReceivers) { 18 if (fileOutputInterval > 0) { 19 createPeriodicOutputFiles(); //创建周期性的输出文件，例如：我们可以设置每一个小时输出一个录像文件 20 } else { 21 createOutputFiles(""); 22 } 23 } 24 25 // Finally, start playing each subsession, to start the data flow: 26 if (duration == 0) { 27 if (scale > 0) duration = session->playEndTime() - initialSeekTime; // use SDP end time 28 else if (scale < 0) duration = initialSeekTime; 29 } 30 if (duration < 0) duration = 0.0; 31 32 endTime = initialSeekTime; 33 if (scale > 0) { 34 if (duration <= 0) endTime = -1.0f; 35 else endTime = initialSeekTime + duration; 36 } else { 37 endTime = initialSeekTime - duration; 38 if (endTime < 0) endTime = 0.0f; 39 } 40 // 发送PLAY命令请求开始播放视频 41 char const* absStartTime = initialAbsoluteSeekTime != NULL ? initialAbsoluteSeekTime : session->absStartTime(); 42 if (absStartTime != NULL) { 43 // Either we or the server have specified that seeking should be done by 'absolute' time: 44 startPlayingSession(session, absStartTime, session->absEndTime(), scale, continueAfterPLAY); 45 } else { 46 // Normal case: Seek by relative time (NPT): 47 startPlayingSession(session, initialSeekTime, endTime, scale, continueAfterPLAY); 48 } 49 } 50 51 void setupSubsession(MediaSubsession* subsession, Boolean streamUsingTCP, Boolean forceMulticastOnUnspecified, RTSPClient::responseHandler* afterFunc) { 52 53 ourRTSPClient->sendSetupCommand(*subsession, afterFunc, False, streamUsingTCP, forceMulticastOnUnspecified, ourAuthenticator); 54 } 55 56 void startPlayingSession(MediaSession* session, double start, double end, float scale, RTSPClient::responseHandler* afterFunc) { 57 printf(" %f - %f ",start,end); 58 ourRTSPClient->sendPlayCommand(*session, afterFunc, start, end, scale, ourAuthenticator); 59 }
在setupStreams函数中依次与每个ServerMediaSubsession建立连接，都建立连接成功后，然后就调用createPeriodicOutputFiles函数周期性地创建输出录像的文件，然后开始发送PLAY命令请求播放视频。接下来我们看看createPeriodicOutputFiles这个函数
1 void createPeriodicOutputFiles() { 2 // Create a filename suffix that notes the time interval that's being recorded: 3 char periodicFileNameSuffix[100]; 4 snprintf(periodicFileNameSuffix, sizeof periodicFileNameSuffix, "-%05d-%05d", 5 fileOutputSecondsSoFar, fileOutputSecondsSoFar + fileOutputInterval); 6 createOutputFiles(periodicFileNameSuffix); //创建输出文件 7 8 // Schedule an event for writing the next output file: //添加一个停止当前录像，创建新录像文件的任务 9 periodicFileOutputTask 10 = env->taskScheduler().scheduleDelayedTask(fileOutputInterval*1000000, 11 (TaskFunc*)periodicFileOutputTimerHandler, 12 (void*)NULL); 13 } 14 15 void createOutputFiles(char const* periodicFilenameSuffix) { 16 char outFileName[1000]; 17
// 创建对应的FileSink来获取和保存视频数据 18 if (outputQuickTimeFile || outputAVIFile) { 19 if (periodicFilenameSuffix[0] == ' ') { 20 // Normally (unless the '-P <interval-in-seconds>' option was given) we output to 'stdout': 21 sprintf(outFileName, "stdout"); 22 } else { 23 // Otherwise output to a type-specific file name, containing "periodicFilenameSuffix": 24 char const* prefix = fileNamePrefix[0] == ' ' ? "output" : fileNamePrefix; 25 snprintf(outFileName, sizeof outFileName, "%s%s.%s", prefix, periodicFilenameSuffix, 26 outputAVIFile ? "avi" : generateMP4Format ? "mp4" : "mov"); 27 } 28 29 if (outputQuickTimeFile) { 30 qtOut = QuickTimeFileSink::createNew(*env, *session, outFileName, 31 fileSinkBufferSize, 32 movieWidth, movieHeight, 33 movieFPS, 34 packetLossCompensate, 35 syncStreams, 36 generateHintTracks, 37 generateMP4Format); 38 if (qtOut == NULL) { 39 *env << "Failed to create a "QuickTimeFileSink" for outputting to "" 40 << outFileName << "": " << env->getResultMsg() << " "; 41 shutdown(); 42 } else { 43 *env << "Outputting to the file: "" << outFileName << "" "; 44 } 45 46 qtOut->startPlaying(sessionAfterPlaying, NULL); 47 } else { // outputAVIFile 48 aviOut = AVIFileSink::createNew(*env, *session, outFileName, 49 fileSinkBufferSize, 50 movieWidth, movieHeight, 51 movieFPS, 52 packetLossCompensate); 53 if (aviOut == NULL) { 54 *env << "Failed to create an "AVIFileSink" for outputting to "" 55 << outFileName << "": " << env->getResultMsg() << " "; 56 shutdown(); 57 } else { 58 *env << "Outputting to the file: "" << outFileName << "" "; 59 } 60 61 aviOut->startPlaying(sessionAfterPlaying, NULL); 62 } 63 } else { //我直接保存录像成.H264文件，所以执行此else分支 64 // Create and start "FileSink"s for each subsession: 65 madeProgress = False; 66 MediaSubsessionIterator iter(*session); 67 while ((subsession = iter.next()) != NULL) { 68 if (subsession->readSource() == NULL) continue; // was not initiated 69 70 // Create an output file for each desired stream: 71 if (singleMedium == NULL || periodicFilenameSuffix[0] != ' ') { 72 // Output file name is 73 // "<filename-prefix><medium_name>-<codec_name>-<counter><periodicFilenameSuffix>" 74 static unsigned streamCounter = 0; 75 snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d%s", 76 fileNamePrefix, subsession->mediumName(), 77 subsession->codecName(), ++streamCounter, periodicFilenameSuffix); 78 } else { 79 // When outputting a single medium only, we output to 'stdout 80 // (unless the '-P <interval-in-seconds>' option was given): 81 sprintf(outFileName, "stdout"); 82 } 83 84 FileSink* fileSink = NULL; 85 Boolean createOggFileSink = False; // by default 86 if (strcmp(subsession->mediumName(), "video") == 0) { 87 if (strcmp(subsession->codecName(), "H264") == 0) { // 创建H264VideoFileSink来获取和保存H264视频数据 88 // For H.264 video stream, we use a special sink that adds 'start codes', 89 // and (at the start) the SPS and PPS NAL units: 90 fileSink = H264VideoFileSink::createNew(*env, outFileName, 91 subsession->fmtp_spropparametersets(), 92 fileSinkBufferSize, oneFilePerFrame); 93 } else if (strcmp(subsession->codecName(), "H265") == 0) { 94 // For H.265 video stream, we use a special sink that adds 'start codes', 95 // and (at the start) the VPS, SPS, and PPS NAL units: 96 fileSink = H265VideoFileSink::createNew(*env, outFileName, 97 subsession->fmtp_spropvps(), 98 subsession->fmtp_spropsps(), 99 subsession->fmtp_sproppps(), 100 fileSinkBufferSize, oneFilePerFrame); 101 } else if (strcmp(subsession->codecName(), "THEORA") == 0) { 102 createOggFileSink = True; 103 } 104 } else if (strcmp(subsession->mediumName(), "audio") == 0) { 105 if (strcmp(subsession->codecName(), "AMR") == 0 || 106 strcmp(subsession->codecName(), "AMR-WB") == 0) { 107 // For AMR audio streams, we use a special sink that inserts AMR frame hdrs: 108 fileSink = AMRAudioFileSink::createNew(*env, outFileName, 109 fileSinkBufferSize, oneFilePerFrame); 110 } else if (strcmp(subsession->codecName(), "VORBIS") == 0 || 111 strcmp(subsession->codecName(), "OPUS") == 0) { 112 createOggFileSink = True; 113 } 114 } 115 if (createOggFileSink) { 116 fileSink = OggFileSink 117 ::createNew(*env, outFileName, 118 subsession->rtpTimestampFrequency(), subsession->fmtp_config()); 119 } else if (fileSink == NULL) { 120 // Normal case: //不属于上面的各种情形，创建一个普通FileSink获取和保存数据 121 fileSink = FileSink::createNew(*env, outFileName, 122 fileSinkBufferSize, oneFilePerFrame); 123 } 124 subsession->sink = fileSink; 125 126 if (subsession->sink == NULL) { 127 *env << "Failed to create FileSink for "" << outFileName 128 << "": " << env->getResultMsg() << " "; 129 } else { 130 if (singleMedium == NULL) { 131 *env << "Created output file: "" << outFileName << "" "; 132 } else { 133 *env << "Outputting data from the "" << subsession->mediumName() 134 << "/" << subsession->codecName() 135 << "" subsession to "" << outFileName << "" "; 136 } 137 138 if (strcmp(subsession->mediumName(), "video") == 0 && 139 strcmp(subsession->codecName(), "MP4V-ES") == 0 && 140 subsession->fmtp_config() != NULL) { 141 // For MPEG-4 video RTP streams, the 'config' information 142 // from the SDP description contains useful VOL etc. headers. 143 // Insert this data at the front of the output file: 144 unsigned configLen; 145 unsigned char* configData 146 = parseGeneralConfigStr(subsession->fmtp_config(), configLen); 147 struct timeval timeNow; 148 gettimeofday(&timeNow, NULL); 149 fileSink->addData(configData, configLen, timeNow); 150 delete[] configData; 151 } 152 153 subsession->sink->startPlaying(*(subsession->readSource()), // 开始获取数据并保存 154 subsessionAfterPlaying, 155 subsession); 156 157 // Also set a handler to be called if a RTCP "BYE" arrives 158 // for this subsession: 159 if (subsession->rtcpInstance() != NULL) { 160 subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, subsession); 161 } 162 163 madeProgress = True; 164 } 165 } 166 if (!madeProgress) shutdown(); 167 } 168 } 169 170 void periodicFileOutputTimerHandler(void* /*clientData*/) { //完成了一个录像文件，关闭相关资源，开始下一个录像文件 171 fileOutputSecondsSoFar += fileOutputInterval; 172 173 // First, close the existing output files: 174 closeMediaSinks(); 175 176 // Then, create new output files: 177 createPeriodicOutputFiles(); 178 } 179 180 void closeMediaSinks() { //关闭FileSink，释放资源 181 Medium::close(qtOut); qtOut = NULL; 182 Medium::close(aviOut); aviOut = NULL; 183 184 if (session == NULL) return; 185 MediaSubsessionIterator iter(*session); 186 MediaSubsession* subsession; 187 while ((subsession = iter.next()) != NULL) { 188 Medium::close(subsession->sink); 189 subsession->sink = NULL; 190 } 191 192 }
首先创建一个录像文件，然后创建FileSink从服务器获取和保存录像数据，当指定的录像时长到了，就终止当前录像，开始下一个录像文件。如此，就可以实现每隔指定的一段时间录像成一个文件的功能。我这里是获取的H264编码的视频，创建的是H264VideoFileSink来获取和保存数据，H264VideoFileSink是H264or5VideoFileSink的子类，H264or5VideoFileSink又是FileSink的子类。FileSink通过MediaSubsession的FramedSource获取数据，然后保存到文件中，我们来看一下保存文件的过程。
1 void H264or5VideoFileSink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime) { 2 unsigned char const start_code[4] = {0x00, 0x00, 0x00, 0x01}; //H264帧的开头是0x00000001 3 4 if (!fHaveWrittenFirstFrame) { 5 // If we have NAL units encoded in "sprop parameter strings", prepend these to the file: 6 for (unsigned j = 0; j < 3; ++j) { 7 unsigned numSPropRecords; 8 SPropRecord* sPropRecords 9 = parseSPropParameterSets(fSPropParameterSetsStr[j], numSPropRecords); 10 for (unsigned i = 0; i < numSPropRecords; ++i) { 11 addData(start_code, 4, presentationTime); 12 addData(sPropRecords[i].sPropBytes, sPropRecords[i].sPropLength, presentationTime); 13 } 14 delete[] sPropRecords; 15 } 16 fHaveWrittenFirstFrame = True; // for next time 17 } 18 19 // Write the input data to the file, with the start code in front: 20 addData(start_code, 4, presentationTime); //先把每一帧的头部写入文件 21 //调用FileSink类的afterGettingFrame函数写入一帧的图像数据到文件 22 // Call the parent class to complete the normal file write with the input data: 23 FileSink::afterGettingFrame(frameSize, numTruncatedBytes, presentationTime); 24 } 25 26 void FileSink::addData(unsigned char const* data, unsigned dataSize, 27 struct timeval presentationTime) { 28 if (fPerFrameFileNameBuffer != NULL && fOutFid == NULL) { 29 // Special case: Open a new file on-the-fly for this frame 30 if (presentationTime.tv_usec == fPrevPresentationTime.tv_usec && 31 presentationTime.tv_sec == fPrevPresentationTime.tv_sec) { 32 // The presentation time is unchanged from the previous frame, so we add a 'counter' 33 // suffix to the file name, to distinguish them: 34 sprintf(fPerFrameFileNameBuffer, "%s-%lu.%06lu-%u", fPerFrameFileNamePrefix, 35 presentationTime.tv_sec, presentationTime.tv_usec, ++fSamePresentationTimeCounter); 36 } else { 37 sprintf(fPerFrameFileNameBuffer, "%s-%lu.%06lu", fPerFrameFileNamePrefix, 38 presentationTime.tv_sec, presentationTime.tv_usec); 39 fPrevPresentationTime = presentationTime; // for next time 40 fSamePresentationTimeCounter = 0; // for next time 41 } 42 fOutFid = OpenOutputFile(envir(), fPerFrameFileNameBuffer); 43 } 44 45 // Write to our file: 46 #ifdef TEST_LOSS 47 static unsigned const framesPerPacket = 10; 48 static unsigned const frameCount = 0; 49 static Boolean const packetIsLost; 50 if ((frameCount++)%framesPerPacket == 0) { 51 packetIsLost = (our_random()%10 == 0); // simulate 10% packet loss ##### 52 } 53 54 if (!packetIsLost) 55 #endif 56 if (fOutFid != NULL && data != NULL) { 57 fwrite(data, 1, dataSize, fOutFid); // 写数据到文件 58 } 59 } 60 61 62 void FileSink::afterGettingFrame(unsigned frameSize, 63 unsigned numTruncatedBytes, 64 struct timeval presentationTime) { 65 if (numTruncatedBytes > 0) { 66 envir() << "FileSink::afterGettingFrame(): The input frame data was too large for our buffer size (" 67 << fBufferSize << "). " 68 << numTruncatedBytes << " bytes of trailing data was dropped! Correct this by increasing the "bufferSize" parameter in the "createNew()" call to at least " 69 << fBufferSize + numTruncatedBytes << " "; 70 } 71 addData(fBuffer, frameSize, presentationTime); // 写视频数据到文件 72 73 if (fOutFid == NULL || fflush(fOutFid) == EOF) { 74 // The output file has closed. Handle this the same way as if the input source had closed: 75 if (fSource != NULL) fSource->stopGettingFrames(); 76 onSourceClosure(); 77 return; 78 } 79 80 if (fPerFrameFileNameBuffer != NULL) { 81 if (fOutFid != NULL) { fclose(fOutFid); fOutFid = NULL; } 82 } 83 84 // Then try getting the next frame: 85 continuePlaying(); // 获取下一帧的数据 86 }
H264VideoFileSink从FramedSource获取一帧数据后，先写H264帧的头部，然后再写视频数据，然后再获取下一帧数据，如此不断地将获取的视频数据写到录像文件中去。