- 浏览: 4398815 次
最新评论
RTMPdump 源代码分析 1: main()函数
注:此前写了一些列的分析RTMPdump(libRTMP)源代码的文章,在此列一个列表:
RTMPdump 源代码分析 1: main()函数
RTMPDump(libRTMP)源代码分析 2:解析RTMP地址——RTMP_ParseURL()
RTMPdump(libRTMP) 源代码分析 3: AMF编码
RTMPdump(libRTMP)源代码分析 4: 连接第一步——握手(Hand Shake)
RTMPdump(libRTMP) 源代码分析 5: 建立一个流媒体连接 (NetConnection部分)
RTMPdump(libRTMP) 源代码分析 6: 建立一个流媒体连接 (NetStream部分 1)
RTMPdump(libRTMP) 源代码分析 7: 建立一个流媒体连接 (NetStream部分 2)
RTMPdump(libRTMP) 源代码分析 8: 发送消息(Message)
RTMPdump(libRTMP) 源代码分析 9: 接收消息(Message)(接收视音频数据)
RTMPdump(libRTMP) 源代码分析 10: 处理各种消息(Message)
其中InitSocket()代码很简单,初始化了Socket,如下:
CleanupSockets()则更简单:
Download()函数则比较复杂:
RTMPdump 源代码分析 1: main()函数
RTMPDump(libRTMP)源代码分析 2:解析RTMP地址——RTMP_ParseURL()
RTMPdump(libRTMP) 源代码分析 3: AMF编码
RTMPdump(libRTMP)源代码分析 4: 连接第一步——握手(Hand Shake)
RTMPdump(libRTMP) 源代码分析 5: 建立一个流媒体连接 (NetConnection部分)
RTMPdump(libRTMP) 源代码分析 6: 建立一个流媒体连接 (NetStream部分 1)
RTMPdump(libRTMP) 源代码分析 7: 建立一个流媒体连接 (NetStream部分 2)
RTMPdump(libRTMP) 源代码分析 8: 发送消息(Message)
RTMPdump(libRTMP) 源代码分析 9: 接收消息(Message)(接收视音频数据)
RTMPdump(libRTMP) 源代码分析 10: 处理各种消息(Message)
===============================
rtmpdump 是一个用来处理 RTMP 流媒体的工具包,支持 rtmp://, rtmpt://, rtmpe://, rtmpte://, and rtmps:// 等。之前在学习RTMP协议的时候,发现没有讲它源代码的,只好自己分析,现在打算把自己学习的成果写出来,可能结果不一定都对,先暂且记录一下。
使用RTMPdump下载一个流媒体的大致流程是这样的:
RTMP_Init();//初始化结构体 InitSockets();//初始化Socket RTMP_ParseURL();//解析输入URL RTMP_SetupStream();//一些设置 fopen();//打开文件,准备写入 RTMP_Connect();//建立NetConnection RTMP_ConnectStream()//建立NetStream Download();//下载函数 RTMP_Close();//关闭连接 fclose();//关闭文件 CleanupSockets();//清理Socket
其中Download()主要是使用RTMP_Read()进行下载的。
注:可以参考:RTMP流媒体播放过程
下面贴上自己注释的RTMPDump源代码。注意以下几点:
1.此RTMPDump已经被移植进VC 2010 的 MFC的工程,所以main()函数已经被改名为rtmpdump(),而且参数也改了,传进来一个MFC窗口的句柄。不过功能没怎么改(控制台程序移植到MFC以后,main()就不是程序的入口了,所以main()名字改成什么是无所谓的)
2.里面有很多提取信息的代码形如:rtmp.dlg->AppendCInfo("开始初始化Socket...");这些代码是我为了获取RTMP信息而自己加的,并不影响程序的执行。
int rtmpdump(LPVOID lpParam,int argc,char **argv) { extern char *optarg; //一定要设置,否则只能运行一次 extern int optind; optind=0; int nStatus = RD_SUCCESS; double percent = 0; double duration = 0.0; int nSkipKeyFrames = DEF_SKIPFRM; // skip this number of keyframes when resuming int bOverrideBufferTime = FALSE; // if the user specifies a buffer time override this is true int bStdoutMode = TRUE; // if true print the stream directly to stdout, messages go to stderr int bResume = FALSE; // true in resume mode uint32_t dSeek = 0; // seek position in resume mode, 0 otherwise uint32_t bufferTime = DEF_BUFTIME; // meta header and initial frame for the resume mode (they are read from the file and compared with // the stream we are trying to continue char *metaHeader = 0; uint32_t nMetaHeaderSize = 0; // video keyframe for matching char *initialFrame = 0; uint32_t nInitialFrameSize = 0; int initialFrameType = 0; // tye: audio or video AVal hostname = { 0, 0 }; AVal playpath = { 0, 0 }; AVal subscribepath = { 0, 0 }; int port = -1; int protocol = RTMP_PROTOCOL_UNDEFINED; int retries = 0; int bLiveStream = FALSE; // 是直播流吗? then we can't seek/resume int bHashes = FALSE; // display byte counters not hashes by default long int timeout = DEF_TIMEOUT; // timeout connection after 120 seconds uint32_t dStartOffset = 0; // 非直播流搜寻点seek position in non-live mode uint32_t dStopOffset = 0; RTMP rtmp = { 0 }; AVal swfUrl = { 0, 0 }; AVal tcUrl = { 0, 0 }; AVal pageUrl = { 0, 0 }; AVal app = { 0, 0 }; AVal auth = { 0, 0 }; AVal swfHash = { 0, 0 }; uint32_t swfSize = 0; AVal flashVer = { 0, 0 }; AVal sockshost = { 0, 0 }; #ifdef CRYPTO int swfAge = 30; /* 30 days for SWF cache by default */ int swfVfy = 0; unsigned char hash[RTMP_SWF_HASHLEN]; #endif char *flvFile = 0; signal(SIGINT, sigIntHandler); signal(SIGTERM, sigIntHandler); #ifndef WIN32 signal(SIGHUP, sigIntHandler); signal(SIGPIPE, sigIntHandler); signal(SIGQUIT, sigIntHandler); #endif RTMP_debuglevel = RTMP_LOGINFO; //首先搜寻“ --quiet”选项 int index = 0; while (index < argc) { if (strcmp(argv[index], "--quiet") == 0 || strcmp(argv[index], "-q") == 0) RTMP_debuglevel = RTMP_LOGCRIT; index++; } #define RTMPDUMP_VERSION "1.0" RTMP_LogPrintf("RTMP流媒体下载 %s\n", RTMPDUMP_VERSION); RTMP_LogPrintf ("2012 雷霄骅 中国传媒大学/信息工程学院/通信与信息系统/数字电视技术\n"); //RTMP_LogPrintf("输入 -h 获取命令选项\n"); RTMP_Init(&rtmp); //句柄----------------------------- rtmp.dlg=(CSpecialPRTMPDlg *)lpParam; //--------------------------------- //---------------------- rtmp.dlg->AppendCInfo("开始初始化Socket..."); //----------------------------- if (!InitSockets()) { //---------------------- rtmp.dlg->AppendCInfo("初始化Socket失败!"); //----------------------------- RTMP_Log(RTMP_LOGERROR, "Couldn't load sockets support on your platform, exiting!"); return RD_FAILED; } //---------------------- rtmp.dlg->AppendCInfo("成功初始化Socket"); //----------------------------- /* sleep(30); */ int opt; /* struct option longopts[] = { {"help", 0, NULL, 'h'}, {"host", 1, NULL, 'n'}, {"port", 1, NULL, 'c'}, {"socks", 1, NULL, 'S'}, {"protocol", 1, NULL, 'l'}, {"playpath", 1, NULL, 'y'}, {"playlist", 0, NULL, 'Y'}, {"rtmp", 1, NULL, 'r'}, {"swfUrl", 1, NULL, 's'}, {"tcUrl", 1, NULL, 't'}, {"pageUrl", 1, NULL, 'p'}, {"app", 1, NULL, 'a'}, {"auth", 1, NULL, 'u'}, {"conn", 1, NULL, 'C'}, #ifdef CRYPTO {"swfhash", 1, NULL, 'w'}, {"swfsize", 1, NULL, 'x'}, {"swfVfy", 1, NULL, 'W'}, {"swfAge", 1, NULL, 'X'}, #endif {"flashVer", 1, NULL, 'f'}, {"live", 0, NULL, 'v'}, {"flv", 1, NULL, 'o'}, {"resume", 0, NULL, 'e'}, {"timeout", 1, NULL, 'm'}, {"buffer", 1, NULL, 'b'}, {"skip", 1, NULL, 'k'}, {"subscribe", 1, NULL, 'd'}, {"start", 1, NULL, 'A'}, {"stop", 1, NULL, 'B'}, {"token", 1, NULL, 'T'}, {"hashes", 0, NULL, '#'}, {"debug", 0, NULL, 'z'}, {"quiet", 0, NULL, 'q'}, {"verbose", 0, NULL, 'V'}, {0, 0, 0, 0} };*/ //分析命令行参数,注意用法。 //选项都是一个字母,后面有冒号的代表该选项还有相关参数 //一直循环直到获取所有的opt while ((opt = getopt/*_long*/(argc, argv, "hVveqzr:s:t:p:a:b:f:o:u:C:n:c:l:y:Ym:k:d:A:B:T:w:x:W:X:S:#"/*, longopts, NULL*/)) != -1) { //不同的选项做不同的处理 switch (opt) { case 'h': usage(argv[0]); return RD_SUCCESS; #ifdef CRYPTO case 'w': { int res = hex2bin(optarg, &swfHash.av_val); if (res != RTMP_SWF_HASHLEN) { swfHash.av_val = NULL; RTMP_Log(RTMP_LOGWARNING, "Couldn't parse swf hash hex string, not hexstring or not %d bytes, ignoring!", RTMP_SWF_HASHLEN); } swfHash.av_len = RTMP_SWF_HASHLEN; break; } case 'x': { int size = atoi(optarg); if (size <= 0) { RTMP_Log(RTMP_LOGERROR, "SWF Size must be at least 1, ignoring\n"); } else { swfSize = size; } break; } case 'W': STR2AVAL(swfUrl, optarg); swfVfy = 1; break; case 'X': { int num = atoi(optarg); if (num < 0) { RTMP_Log(RTMP_LOGERROR, "SWF Age must be non-negative, ignoring\n"); } else { swfAge = num; } } break; #endif case 'k': nSkipKeyFrames = atoi(optarg); if (nSkipKeyFrames < 0) { RTMP_Log(RTMP_LOGERROR, "Number of keyframes skipped must be greater or equal zero, using zero!"); nSkipKeyFrames = 0; } else { RTMP_Log(RTMP_LOGDEBUG, "Number of skipped key frames for resume: %d", nSkipKeyFrames); } break; case 'b': { int32_t bt = atol(optarg); if (bt < 0) { RTMP_Log(RTMP_LOGERROR, "Buffer time must be greater than zero, ignoring the specified value %d!", bt); } else { bufferTime = bt; bOverrideBufferTime = TRUE; } break; } //直播流 case 'v': //---------------- rtmp.dlg->AppendCInfo("该RTMP的URL是一个直播流"); //---------------- bLiveStream = TRUE; // no seeking or resuming possible! break; case 'd': STR2AVAL(subscribepath, optarg); break; case 'n': STR2AVAL(hostname, optarg); break; case 'c': port = atoi(optarg); break; case 'l': protocol = atoi(optarg); if (protocol < RTMP_PROTOCOL_RTMP || protocol > RTMP_PROTOCOL_RTMPTS) { RTMP_Log(RTMP_LOGERROR, "Unknown protocol specified: %d", protocol); return RD_FAILED; } break; case 'y': STR2AVAL(playpath, optarg); break; case 'Y': RTMP_SetOpt(&rtmp, &av_playlist, (AVal *)&av_true); break; //路径参数-r case 'r': { AVal parsedHost, parsedApp, parsedPlaypath; unsigned int parsedPort = 0; int parsedProtocol = RTMP_PROTOCOL_UNDEFINED; //解析URL。注optarg指向参数(URL) RTMP_LogPrintf("RTMP URL : %s\n",optarg); //---------------- rtmp.dlg->AppendCInfo("解析RTMP的URL..."); //---------------- if (!RTMP_ParseURL (optarg, &parsedProtocol, &parsedHost, &parsedPort, &parsedPlaypath, &parsedApp)) { //---------------- rtmp.dlg->AppendCInfo("解析RTMP的URL失败!"); //---------------- RTMP_Log(RTMP_LOGWARNING, "无法解析 url (%s)!", optarg); } else { //---------------- rtmp.dlg->AppendCInfo("解析RTMP的URL成功"); //---------------- //把解析出来的数据赋值 if (!hostname.av_len) hostname = parsedHost; if (port == -1) port = parsedPort; if (playpath.av_len == 0 && parsedPlaypath.av_len) { playpath = parsedPlaypath; } if (protocol == RTMP_PROTOCOL_UNDEFINED) protocol = parsedProtocol; if (app.av_len == 0 && parsedApp.av_len) { app = parsedApp; } } break; } case 's': STR2AVAL(swfUrl, optarg); break; case 't': STR2AVAL(tcUrl, optarg); break; case 'p': STR2AVAL(pageUrl, optarg); break; case 'a': STR2AVAL(app, optarg); break; case 'f': STR2AVAL(flashVer, optarg); break; //指定输出文件 case 'o': flvFile = optarg; if (strcmp(flvFile, "-")) bStdoutMode = FALSE; break; case 'e': bResume = TRUE; break; case 'u': STR2AVAL(auth, optarg); break; case 'C': { AVal av; STR2AVAL(av, optarg); if (!RTMP_SetOpt(&rtmp, &av_conn, &av)) { RTMP_Log(RTMP_LOGERROR, "Invalid AMF parameter: %s", optarg); return RD_FAILED; } } break; case 'm': timeout = atoi(optarg); break; case 'A': dStartOffset = (int) (atof(optarg) * 1000.0); break; case 'B': dStopOffset = (int) (atof(optarg) * 1000.0); break; case 'T': { AVal token; STR2AVAL(token, optarg); RTMP_SetOpt(&rtmp, &av_token, &token); } break; case '#': bHashes = TRUE; break; case 'q': RTMP_debuglevel = RTMP_LOGCRIT; break; case 'V': RTMP_debuglevel = RTMP_LOGDEBUG; break; case 'z': RTMP_debuglevel = RTMP_LOGALL; break; case 'S': STR2AVAL(sockshost, optarg); break; default: RTMP_LogPrintf("unknown option: %c\n", opt); usage(argv[0]); return RD_FAILED; break; } } if (!hostname.av_len) { RTMP_Log(RTMP_LOGERROR, "您必须指定 主机名(hostname) (--host) 或 url (-r \"rtmp://host[:port]/playpath\") 包含 a hostname"); return RD_FAILED; } if (playpath.av_len == 0) { RTMP_Log(RTMP_LOGERROR, "您必须指定 播放路径(playpath) (--playpath) 或 url (-r \"rtmp://host[:port]/playpath\") 包含 a playpath"); return RD_FAILED; } if (protocol == RTMP_PROTOCOL_UNDEFINED) { RTMP_Log(RTMP_LOGWARNING, "您没有指定 协议(protocol) (--protocol) 或 rtmp url (-r), 默认协议 RTMP"); protocol = RTMP_PROTOCOL_RTMP; } if (port == -1) { RTMP_Log(RTMP_LOGWARNING, "您没有指定 端口(port) (--port) 或 rtmp url (-r), 默认端口 1935"); port = 0; } if (port == 0) { if (protocol & RTMP_FEATURE_SSL) port = 443; else if (protocol & RTMP_FEATURE_HTTP) port = 80; else port = 1935; } if (flvFile == 0) { RTMP_Log(RTMP_LOGWARNING, "请指定一个输出文件 (-o filename), using stdout"); bStdoutMode = TRUE; } if (bStdoutMode && bResume) { RTMP_Log(RTMP_LOGWARNING, "Can't resume in stdout mode, ignoring --resume option"); bResume = FALSE; } if (bLiveStream && bResume) { RTMP_Log(RTMP_LOGWARNING, "Can't resume live stream, ignoring --resume option"); bResume = FALSE; } #ifdef CRYPTO if (swfVfy) { if (RTMP_HashSWF(swfUrl.av_val, (unsigned int *)&swfSize, hash, swfAge) == 0) { swfHash.av_val = (char *)hash; swfHash.av_len = RTMP_SWF_HASHLEN; } } if (swfHash.av_len == 0 && swfSize > 0) { RTMP_Log(RTMP_LOGWARNING, "Ignoring SWF size, supply also the hash with --swfhash"); swfSize = 0; } if (swfHash.av_len != 0 && swfSize == 0) { RTMP_Log(RTMP_LOGWARNING, "Ignoring SWF hash, supply also the swf size with --swfsize"); swfHash.av_len = 0; swfHash.av_val = NULL; } #endif if (tcUrl.av_len == 0) { char str[512] = { 0 }; tcUrl.av_len = snprintf(str, 511, "%s://%.*s:%d/%.*s", RTMPProtocolStringsLower[protocol], hostname.av_len, hostname.av_val, port, app.av_len, app.av_val); tcUrl.av_val = (char *) malloc(tcUrl.av_len + 1); strcpy(tcUrl.av_val, str); } int first = 1; // User defined seek offset if (dStartOffset > 0) { //直播流 if (bLiveStream) { RTMP_Log(RTMP_LOGWARNING, "Can't seek in a live stream, ignoring --start option"); dStartOffset = 0; } } //---------------- rtmp.dlg->AppendCInfo("开始初始化RTMP连接的参数..."); //---------------- //设置 RTMP_SetupStream(&rtmp, protocol, &hostname, port, &sockshost, &playpath, &tcUrl, &swfUrl, &pageUrl, &app, &auth, &swfHash, swfSize, &flashVer, &subscribepath, dSeek, dStopOffset, bLiveStream, timeout); //此处设置参数----------------- rtmp.dlg->AppendCInfo("成功初始化RTMP连接的参数"); //----------------------------- char *temp=(char *)malloc(MAX_URL_LENGTH); memcpy(temp,rtmp.Link.hostname.av_val,rtmp.Link.hostname.av_len); temp[rtmp.Link.hostname.av_len]='\0'; rtmp.dlg->AppendB_R_L_Info("主机名",temp); itoa(rtmp.Link.port,temp,10); rtmp.dlg->AppendB_R_L_Info("端口号",temp); memcpy(temp,rtmp.Link.app.av_val,rtmp.Link.app.av_len); temp[rtmp.Link.app.av_len]='\0'; rtmp.dlg->AppendB_R_L_Info("应用程序",temp); memcpy(temp,rtmp.Link.playpath.av_val,rtmp.Link.playpath.av_len); temp[rtmp.Link.playpath.av_len]='\0'; rtmp.dlg->AppendB_R_L_Info("路径",temp); //----------------------------- /* Try to keep the stream moving if it pauses on us */ if (!bLiveStream && !(protocol & RTMP_FEATURE_HTTP)) rtmp.Link.lFlags |= RTMP_LF_BUFX; off_t size = 0; // ok,我们必须获得timestamp of the last keyframe (only keyframes are seekable) / last audio frame (audio only streams) if (bResume) { //打开文件,输出的文件(Resume) nStatus = OpenResumeFile(flvFile, &file, &size, &metaHeader, &nMetaHeaderSize, &duration); if (nStatus == RD_FAILED) goto clean; if (!file) { // file does not exist, so go back into normal mode bResume = FALSE; // we are back in fresh file mode (otherwise finalizing file won't be done) } else { //获取最后一个关键帧 nStatus = GetLastKeyframe(file, nSkipKeyFrames, &dSeek, &initialFrame, &initialFrameType, &nInitialFrameSize); if (nStatus == RD_FAILED) { RTMP_Log(RTMP_LOGDEBUG, "Failed to get last keyframe."); goto clean; } if (dSeek == 0) { RTMP_Log(RTMP_LOGDEBUG, "Last keyframe is first frame in stream, switching from resume to normal mode!"); bResume = FALSE; } } } //如果输出文件不存在 if (!file) { if (bStdoutMode) { //直接输出到stdout file = stdout; SET_BINMODE(file); } else { //打开一个文件 //w+b 读写打开或建立一个二进制文件,允许读和写。 //----------------- rtmp.dlg->AppendCInfo("创建输出文件..."); //----------------------------- file = fopen(flvFile, "w+b"); if (file == 0) { //----------------- rtmp.dlg->AppendCInfo("创建输出文件失败!"); //----------------------------- RTMP_LogPrintf("Failed to open file! %s\n", flvFile); return RD_FAILED; } rtmp.dlg->AppendCInfo("成功创建输出文件"); } } #ifdef _DEBUG netstackdump = fopen("netstackdump", "wb"); netstackdump_read = fopen("netstackdump_read", "wb"); #endif while (!RTMP_ctrlC) { RTMP_Log(RTMP_LOGDEBUG, "Setting buffer time to: %dms", bufferTime); //设置Buffer时间 //----------------- rtmp.dlg->AppendCInfo("设置缓冲(Buffer)的时间"); //----------------------------- RTMP_SetBufferMS(&rtmp, bufferTime); //第一次执行 if (first) { first = 0; RTMP_LogPrintf("开始建立连接!\n"); //----------------- rtmp.dlg->AppendCInfo("开始建立连接(NetConnection)..."); //----------------------------- //建立连接(Connect) if (!RTMP_Connect(&rtmp, NULL)) { //----------------- rtmp.dlg->AppendCInfo("建立连接(NetConnection)失败!"); //----------------------------- nStatus = RD_FAILED; break; } //----------------- rtmp.dlg->AppendCInfo("成功建立连接(NetConnection)"); //----------------------------- //RTMP_Log(RTMP_LOGINFO, "已链接..."); // User defined seek offset if (dStartOffset > 0) { // Don't need the start offset if resuming an existing file if (bResume) { RTMP_Log(RTMP_LOGWARNING, "Can't seek a resumed stream, ignoring --start option"); dStartOffset = 0; } else { dSeek = dStartOffset; } } // Calculate the length of the stream to still play if (dStopOffset > 0) { // Quit if start seek is past required stop offset if (dStopOffset <= dSeek) { RTMP_LogPrintf("Already Completed\n"); nStatus = RD_SUCCESS; break; } } //创建流(Stream)(发送connect命令消息后处理传来的数据) itoa(rtmp.m_inChunkSize,temp,10); rtmp.dlg->AppendB_R_Info("输入Chunk大小",temp); itoa(rtmp.m_outChunkSize,temp,10); rtmp.dlg->AppendB_R_Info("输出Chunk大小",temp); itoa(rtmp.m_stream_id,temp,10); rtmp.dlg->AppendB_R_Info("Stream ID",temp); itoa(rtmp.m_nBufferMS,temp,10); rtmp.dlg->AppendB_R_Info("Buffer时长(ms)",temp); itoa(rtmp.m_nServerBW,temp,10); rtmp.dlg->AppendB_R_Info("ServerBW",temp); itoa(rtmp.m_nClientBW,temp,10); rtmp.dlg->AppendB_R_Info("ClientBW",temp); itoa((int)rtmp.m_fEncoding,temp,10); rtmp.dlg->AppendB_R_Info("命令消息编码方法",temp); itoa((int)rtmp.m_fDuration,temp,10); rtmp.dlg->AppendB_R_Info("时长(s)",temp); rtmp.dlg->ShowBInfo(); free(temp); //----------------- rtmp.dlg->AppendCInfo("开始建立网络流(NetStream)"); //----------------------------- if (!RTMP_ConnectStream(&rtmp, dSeek)) { //----------------- rtmp.dlg->AppendCInfo("建立网络流(NetStream)失败!"); //----------------- nStatus = RD_FAILED; break; } //----------------- rtmp.dlg->AppendCInfo("成功建立网络流(NetStream)!"); //----------------- } else { nInitialFrameSize = 0; if (retries) { RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n"); if (!RTMP_IsTimedout(&rtmp)) nStatus = RD_FAILED; else nStatus = RD_INCOMPLETE; break; } RTMP_Log(RTMP_LOGINFO, "Connection timed out, trying to resume.\n\n"); /* Did we already try pausing, and it still didn't work? */ if (rtmp.m_pausing == 3) { /* Only one try at reconnecting... */ retries = 1; dSeek = rtmp.m_pauseStamp; if (dStopOffset > 0) { if (dStopOffset <= dSeek) { RTMP_LogPrintf("Already Completed\n"); nStatus = RD_SUCCESS; break; } } if (!RTMP_ReconnectStream(&rtmp, dSeek)) { RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n"); if (!RTMP_IsTimedout(&rtmp)) nStatus = RD_FAILED; else nStatus = RD_INCOMPLETE; break; } } else if (!RTMP_ToggleStream(&rtmp)) { RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n"); if (!RTMP_IsTimedout(&rtmp)) nStatus = RD_FAILED; else nStatus = RD_INCOMPLETE; break; } bResume = TRUE; } //----------------- //----------------- rtmp.dlg->AppendCInfo("开始将媒体数据写入文件"); //----------------- //下载,写入文件 nStatus = Download(&rtmp, file, dSeek, dStopOffset, duration, bResume, metaHeader, nMetaHeaderSize, initialFrame, initialFrameType, nInitialFrameSize, nSkipKeyFrames, bStdoutMode, bLiveStream, bHashes, bOverrideBufferTime, bufferTime, &percent); free(initialFrame); initialFrame = NULL; /* If we succeeded, we're done. */ if (nStatus != RD_INCOMPLETE || !RTMP_IsTimedout(&rtmp) || bLiveStream) break; } //当下载完的时候 if (nStatus == RD_SUCCESS) { //----------------- rtmp.dlg->AppendCInfo("写入文件完成"); //----------------- RTMP_LogPrintf("Download complete\n"); } //没下载完的时候 else if (nStatus == RD_INCOMPLETE) { //----------------- rtmp.dlg->AppendCInfo("写入文件可能不完整"); //----------------- RTMP_LogPrintf ("Download may be incomplete (downloaded about %.2f%%), try resuming\n", percent); } //后续清理工作 clean: //----------------- rtmp.dlg->AppendCInfo("关闭连接"); //----------------- RTMP_Log(RTMP_LOGDEBUG, "Closing connection.\n"); RTMP_Close(&rtmp); rtmp.dlg->AppendCInfo("关闭文件"); if (file != 0) fclose(file); rtmp.dlg->AppendCInfo("关闭Socket"); CleanupSockets(); #ifdef _DEBUG if (netstackdump != 0) fclose(netstackdump); if (netstackdump_read != 0) fclose(netstackdump_read); #endif return nStatus; }
其中InitSocket()代码很简单,初始化了Socket,如下:
// 初始化 sockets int InitSockets() { #ifdef WIN32 WORD version; WSADATA wsaData; version = MAKEWORD(1, 1); return (WSAStartup(version, &wsaData) == 0); #else return TRUE; #endif }
CleanupSockets()则更简单:
inline void CleanupSockets() { #ifdef WIN32 WSACleanup(); #endif }
Download()函数则比较复杂:
int Download(RTMP * rtmp, // connected RTMP object FILE * file, uint32_t dSeek, uint32_t dStopOffset, double duration, int bResume, char *metaHeader, uint32_t nMetaHeaderSize, char *initialFrame, int initialFrameType, uint32_t nInitialFrameSize, int nSkipKeyFrames, int bStdoutMode, int bLiveStream, int bHashes, int bOverrideBufferTime, uint32_t bufferTime, double *percent) // percentage downloaded [out] { int32_t now, lastUpdate; int bufferSize = 64 * 1024; char *buffer = (char *) malloc(bufferSize); int nRead = 0; //long ftell(FILE *stream); //返回当前文件指针 RTMP_LogPrintf("开始下载!\n"); off_t size = ftello(file); unsigned long lastPercent = 0; //时间戳 rtmp->m_read.timestamp = dSeek; *percent = 0.0; if (rtmp->m_read.timestamp) { RTMP_Log(RTMP_LOGDEBUG, "Continuing at TS: %d ms\n", rtmp->m_read.timestamp); } //是直播 if (bLiveStream) { RTMP_LogPrintf("直播流\n"); } else { // print initial status // Workaround to exit with 0 if the file is fully (> 99.9%) downloaded if (duration > 0) { if ((double) rtmp->m_read.timestamp >= (double) duration * 999.0) { RTMP_LogPrintf("Already Completed at: %.3f sec Duration=%.3f sec\n", (double) rtmp->m_read.timestamp / 1000.0, (double) duration / 1000.0); return RD_SUCCESS; } else { *percent = ((double) rtmp->m_read.timestamp) / (duration * 1000.0) * 100.0; *percent = ((double) (int) (*percent * 10.0)) / 10.0; RTMP_LogPrintf("%s download at: %.3f kB / %.3f sec (%.1f%%)\n", bResume ? "Resuming" : "Starting", (double) size / 1024.0, (double) rtmp->m_read.timestamp / 1000.0, *percent); } } else { RTMP_LogPrintf("%s download at: %.3f kB\n", bResume ? "Resuming" : "Starting", (double) size / 1024.0); } } if (dStopOffset > 0) RTMP_LogPrintf("For duration: %.3f sec\n", (double) (dStopOffset - dSeek) / 1000.0); //各种设置参数到rtmp连接 if (bResume && nInitialFrameSize > 0) rtmp->m_read.flags |= RTMP_READ_RESUME; rtmp->m_read.initialFrameType = initialFrameType; rtmp->m_read.nResumeTS = dSeek; rtmp->m_read.metaHeader = metaHeader; rtmp->m_read.initialFrame = initialFrame; rtmp->m_read.nMetaHeaderSize = nMetaHeaderSize; rtmp->m_read.nInitialFrameSize = nInitialFrameSize; now = RTMP_GetTime(); lastUpdate = now - 1000; do { //从rtmp中把bufferSize(64k)个数据读入buffer nRead = RTMP_Read(rtmp, buffer, bufferSize); //RTMP_LogPrintf("nRead: %d\n", nRead); if (nRead > 0) { //函数:size_t fwrite(const void* buffer,size_t size,size_t count,FILE* stream); //向文件读入写入一个数据块。返回值:返回实际写入的数据块数目 //(1)buffer:是一个指针,对fwrite来说,是要输出数据的地址。 //(2)size:要写入内容的单字节数; //(3)count:要进行写入size字节的数据项的个数; //(4)stream:目标文件指针。 //(5)返回实际写入的数据项个数count。 //关键。把buffer里面的数据写成文件 if (fwrite(buffer, sizeof(unsigned char), nRead, file) != (size_t) nRead) { RTMP_Log(RTMP_LOGERROR, "%s: Failed writing, exiting!", __FUNCTION__); free(buffer); return RD_FAILED; } //记录已经写入的字节数 size += nRead; //RTMP_LogPrintf("write %dbytes (%.1f kB)\n", nRead, nRead/1024.0); if (duration <= 0) // if duration unknown try to get it from the stream (onMetaData) duration = RTMP_GetDuration(rtmp); if (duration > 0) { // make sure we claim to have enough buffer time! if (!bOverrideBufferTime && bufferTime < (duration * 1000.0)) { bufferTime = (uint32_t) (duration * 1000.0) + 5000; // 再加5s以确保buffertime足够长 RTMP_Log(RTMP_LOGDEBUG, "Detected that buffer time is less than duration, resetting to: %dms", bufferTime); //重设Buffer长度 RTMP_SetBufferMS(rtmp, bufferTime); //给服务器发送UserControl消息通知Buffer改变 RTMP_UpdateBufferMS(rtmp); } //计算百分比 *percent = ((double) rtmp->m_read.timestamp) / (duration * 1000.0) * 100.0; *percent = ((double) (int) (*percent * 10.0)) / 10.0; if (bHashes) { if (lastPercent + 1 <= *percent) { RTMP_LogStatus("#"); lastPercent = (unsigned long) *percent; } } else { //设置显示数据的更新间隔200ms now = RTMP_GetTime(); if (abs(now - lastUpdate) > 200) { RTMP_LogStatus("\r%.3f kB / %.2f sec (%.1f%%)", (double) size / 1024.0, (double) (rtmp->m_read.timestamp) / 1000.0, *percent); lastUpdate = now; } } } else { //现在距离开机的毫秒数 now = RTMP_GetTime(); //每间隔200ms刷新一次数据 if (abs(now - lastUpdate) > 200) { if (bHashes) RTMP_LogStatus("#"); else //size为已写入文件的字节数 RTMP_LogStatus("\r%.3f kB / %.2f sec", (double) size / 1024.0, (double) (rtmp->m_read.timestamp) / 1000.0); lastUpdate = now; } } } #ifdef _DEBUG else { RTMP_Log(RTMP_LOGDEBUG, "zero read!"); } #endif } while (!RTMP_ctrlC && nRead > -1 && RTMP_IsConnected(rtmp) && !RTMP_IsTimedout(rtmp)); free(buffer); if (nRead < 0) //nRead是读取情况 nRead = rtmp->m_read.status; /* Final status update */ if (!bHashes) { if (duration > 0) { *percent = ((double) rtmp->m_read.timestamp) / (duration * 1000.0) * 100.0; *percent = ((double) (int) (*percent * 10.0)) / 10.0; //输出 RTMP_LogStatus("\r%.3f kB / %.2f sec (%.1f%%)", (double) size / 1024.0, (double) (rtmp->m_read.timestamp) / 1000.0, *percent); } else { RTMP_LogStatus("\r%.3f kB / %.2f sec", (double) size / 1024.0, (double) (rtmp->m_read.timestamp) / 1000.0); } } RTMP_Log(RTMP_LOGDEBUG, "RTMP_Read returned: %d", nRead); //读取错误 if (bResume && nRead == -2) { RTMP_LogPrintf("Couldn't resume FLV file, try --skip %d\n\n", nSkipKeyFrames + 1); return RD_FAILED; } //读取正确 if (nRead == -3) return RD_SUCCESS; //没读完... if ((duration > 0 && *percent < 99.9) || RTMP_ctrlC || nRead < 0 || RTMP_IsTimedout(rtmp)) { return RD_INCOMPLETE; } return RD_SUCCESS; }
以上内容是我能理解到的rtmpdump.c里面的内容。
rtmpdump源代码(Linux):http://download.csdn.net/detail/leixiaohua1020/6376561
rtmpdump源代码(VC 2005 工程):http://download.csdn.net/detail/leixiaohua1020/6563163
相关推荐
这个压缩包“rtmpdump源代码.rar”包含rtmpdump的源代码,使得开发者可以深入理解其工作原理,进行二次开发或者调试。 **RTMP协议** RTMP是一种用于实时数据传输的协议,常用于在线视频直播和点播服务。它由Adobe ...
1. **获取源代码**: 解压rtmpdump-2.4压缩包。 2. **依赖安装**: 安装必要的编译依赖,例如`libssl-dev`和`librtmp-dev`。 3. **编译源码**: 使用`./configure`进行配置,`make`进行编译,`sudo make install`进行...
这个压缩包“rtmpdump.zip”包含了在Visual Studio环境中编译rtmpdump项目所需的所有关键文件,包括头文件和源代码文件。RTMP是一种广泛用于在线直播和视频分发的协议,由Adobe Systems开发。 首先,让我们深入了解...
这个压缩包“rtmpdump-2.2e.tar.gz”包含了rtmpdump的源代码,允许用户深入理解其工作原理并可能进行自定义修改。下面将详细讨论RTMP协议、rtmpdump工具、以及它在IT领域的应用。 RTMP(Real Time Messaging ...
2. **获取源代码**:从rtmpdump项目的官方仓库或第三方源获取源代码。 3. **配置项目**:在VC6.0中创建新项目,将源代码文件添加到项目中,配置相应的编译选项和链接器设置。 4. **解决依赖**:如果项目依赖于特定库...
"bin"目录通常存放可执行文件,而"librtmp_src"则表示RTMP库的源代码,这是rtmpdump的核心部分,用于处理与RTMP服务器的通信。 编译rtmpdump时,你需要确保你有正确的开发环境,例如Visual Studio 2010,因为它是一...
在压缩包中的"rtmpdump-2.3"可能是rtmpdump的源代码或者编译好的二进制文件,用户可以进一步查看源码理解其工作原理,或者直接在Android设备上使用编译后的二进制文件进行流媒体操作。对于开发者来说,这为自定义...
1. rtmpdump源代码:这是工具的源代码,程序员可以查看并理解其工作原理,或者根据需求进行定制。 2. librtmp源代码:提供了库的源代码,开发者可以修改、优化或扩展功能。 3. 静态库(.lib文件):适用于那些希望将...
1. **源代码**:这是rtmpdump-2.3的核心部分,由C或C++编写,包含了实现RTMP协议解析和交互的所有函数和结构。开发者可以查看源码,理解其工作原理,甚至进行自定义修改。 2. **编译好的动态库**:预编译的动态库...
2. **下载源代码**:从官方网站或者GitHub仓库获取rtmpdump-2.3的源代码包。 3. **解压与配置**:使用`tar`命令解压缩源代码包,然后进入目录并使用`./configure`进行配置,根据提示检查依赖是否齐全。 4. **编译与...
- 添加源文件:将rtmpdump和librtmp的源代码添加到项目中。 - 适应平台:修改源代码中的编译器特定宏和包含路径,以适应VS2008环境。 - 链接库设置:确保正确配置了所有依赖库,如zlib等。 - 编译和调试:编译...
1. **流媒体下载**:rtmpdump支持从RTMP服务器上抓取视频流并将其保存到本地文件,这对于离线观看或者备份在线视频非常有用。 2. **支持多种平台**:rtmpdump不仅可以在常见的桌面操作系统(如Windows、Linux和macOS...
**rtmpdump包详解** RTMP(Real Time Messaging ...请注意,rtmpdump是一个开源工具,其源代码可供开发者参考和扩展。然而,不合法的使用可能会引起法律问题,因此在使用rtmpdump时,务必尊重内容提供商的版权规定。
rtmpdump is a toolkit for RTMP streams. All forms of RTMP are supported, including rtmp://, rtmpt://, rtmpe://, rtmpte://, and rtmps://. License: GPLv2 Copyright (C) 2009 Andrej Stepanchuk ...
源代码可以让你深入理解RTMP协议的工作原理,以及rtmpdump如何实现数据的捕获和处理。这对于想要自定义或扩展rtmpdump功能的人来说尤其重要。 rtmpdump.exe是rtmpdump的可执行文件,你可以用它来从RTMP服务器上下载...
1. 源代码文件:包含了实现RTMP协议解析和操作的C++源代码。 2. 头文件:定义了类、函数和常量等接口,供其他源文件引用。 3. 编译脚本或配置文件:用于设置编译选项和链接库,确保在VS2005环境下成功编译。 4. ...
rtmpdump-v1.6.tar.gz是rtmpdump的一个特定版本,通过解压这个tar.gz压缩包,我们可以获取到rtmpdump的源代码和其他相关文件。在这个压缩包中,唯一列出的子文件名是"rtmpdump",这通常指的是rtmpdump的可执行程序...
1. rtmpdump源码:这是rtmpdump项目的完整源代码,包含了C或C++语言编写的函数和结构,开发者可以通过阅读源码了解其工作原理,或者根据需求进行修改和定制。 2. openssl源码和编译结果:openssl库的源代码和编译好...
在编译rtmpdump时,需要将librtmp_src文件夹中的源代码与其他依赖库合并,并配置合适的编译选项。对于Windows环境,这通常涉及设置项目属性、链接库路径和包含目录。完成编译后,将生成rtmpdump可执行文件,可用于与...
同时,开发者也可以利用rtmpdump的源代码,学习如何实现RTMP协议的相关功能,为自己的项目提供支持。 然而,随着HLS (HTTP Live Streaming) 和DASH (Dynamic Adaptive Streaming over HTTP) 等适应性更强的流媒体...