• 设为首页
  • 点击收藏
  • 手机版
    手机扫一扫访问
    迪恩网络手机版
  • 关注官方公众号
    微信扫一扫关注
    公众号

C++ RTMP_LogPrintf函数代码示例

原作者: [db:作者] 来自: [db:来源] 收藏 邀请

本文整理汇总了C++中RTMP_LogPrintf函数的典型用法代码示例。如果您正苦于以下问题:C++ RTMP_LogPrintf函数的具体用法?C++ RTMP_LogPrintf怎么用?C++ RTMP_LogPrintf使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。



在下文中一共展示了RTMP_LogPrintf函数的20个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于我们的系统推荐出更棒的C++代码示例。

示例1: sigIntHandler

void
sigIntHandler(int sig)
{
  RTMP_ctrlC = TRUE;
  RTMP_LogPrintf("Caught signal: %d, cleaning up, just a second...\n", sig);
  if (rtmpServer)
    stopStreaming(rtmpServer);
  signal(SIGINT, SIG_DFL);
}
开发者ID:firemyst,项目名称:rtmpdump,代码行数:9,代码来源:rtmpsrv.c


示例2: rtmp_rvod_stop_notify

static void  rtmp_rvod_stop_notify(void* ctx)
{
    if (!ctx)
    {
        return;
    }
    RTMP_LogPrintf("rtmp stopeed\n");
    t_rtmp_vod_ctx *c = ctx;
    c->stopeed = 1;

}
开发者ID:taolinbg,项目名称:video_sdk,代码行数:11,代码来源:ont_video_rvod.c


示例3: sigIntHandler

void sigIntHandler(int sig)
{
	RTMP_ctrlC = TRUE;
	RTMP_LogPrintf("Caught signal: %d, cleaning up, just a second...\n", sig);
	// ignore all these signals now and let the connection close
	signal(SIGINT, SIG_IGN);
	signal(SIGTERM, SIG_IGN);
	signal(SIGHUP, SIG_IGN);
	signal(SIGPIPE, SIG_IGN);
	signal(SIGQUIT, SIG_IGN);
}
开发者ID:kulv2012,项目名称:rtmpdump-librtmp,代码行数:11,代码来源:simplertmpdump.c


示例4: controlServerThread

TFTYPE
controlServerThread(void *unused)
{
  char ich;
  while (1)
    {
      ich = getchar();
      switch (ich)
	{
	case 'q':
	  RTMP_LogPrintf("Exiting\n");
	  stopStreaming(rtmpServer);
	  exit(0);
	  break;
	default:
	  RTMP_LogPrintf("Unknown command \'%c\', ignoring\n", ich);
	}
    }
  TFRET();
}
开发者ID:firemyst,项目名称:rtmpdump,代码行数:20,代码来源:rtmpsrv.c


示例5: sigIntHandler

void
sigIntHandler(int sig)
{
    //xjzhang, 屏蔽这些signal,设置RTMP_ctrlC为TRUE从而进入正常退出流程;
    RTMP_ctrlC = TRUE;
    RTMP_LogPrintf("Caught signal: %d, cleaning up, just a second...\n", sig);
    // ignore all these signals now and let the connection close
    signal(SIGINT, SIG_IGN);
    signal(SIGTERM, SIG_IGN);
#ifndef WIN32
    signal(SIGHUP, SIG_IGN);
    signal(SIGPIPE, SIG_IGN);
    signal(SIGQUIT, SIG_IGN);
#endif
}
开发者ID:minico,项目名称:rtmpdump,代码行数:15,代码来源:rtmpdump.cpp


示例6: rtmp_rvod_pause_notify

static void  rtmp_rvod_pause_notify(void* ctx, int paused, double ts)
{
    RTMP_LogPrintf("rtmp pause %d ts %lf\n", paused, ts);
    if (!ctx)
    {
        return;
    }
    t_rtmp_vod_ctx *c = ctx;
    c->paused = paused;

    if (paused)
    {
        c->start_timestamp = ts;
    }
    else
    {
		c->start_timestamp = ts;
        c->epoch = RTMP_GetTime();
    }
}
开发者ID:taolinbg,项目名称:video_sdk,代码行数:20,代码来源:ont_video_rvod.c


示例7: main

int
main(int argc, char **argv)
{
  int nStatus = RD_SUCCESS;

  // http streaming server
  char DEFAULT_HTTP_STREAMING_DEVICE[] = "0.0.0.0";	// 0.0.0.0 is any device

  char *rtmpStreamingDevice = DEFAULT_HTTP_STREAMING_DEVICE;	// streaming device, default 0.0.0.0
  int nRtmpStreamingPort = 1935;	// port

  RTMP_LogPrintf("RTMP Server %s\n", RTMPDUMP_VERSION);
  RTMP_LogPrintf("(c) 2010 Andrej Stepanchuk, Howard Chu; license: GPL\n\n");

  RTMP_debuglevel = RTMP_LOGINFO;

  if (argc > 1 && !strcmp(argv[1], "-z"))
    RTMP_debuglevel = RTMP_LOGALL;

  // init request
  memset(&defaultRTMPRequest, 0, sizeof(RTMP_REQUEST));

  defaultRTMPRequest.rtmpport = -1;
  defaultRTMPRequest.protocol = RTMP_PROTOCOL_UNDEFINED;
  defaultRTMPRequest.bLiveStream = FALSE;	// is it a live stream? then we can't seek/resume

  defaultRTMPRequest.timeout = 300;	// timeout connection afte 300 seconds
  defaultRTMPRequest.bufferTime = 20 * 1000;


  signal(SIGINT, sigIntHandler);
#ifndef WIN32
  signal(SIGPIPE, SIG_IGN);
#endif

#ifdef _DEBUG
  netstackdump = fopen("netstackdump", "wb");
  netstackdump_read = fopen("netstackdump_read", "wb");
#endif

  InitSockets();

  // start text UI
  ThreadCreate(controlServerThread, 0);

  // start http streaming
  if ((rtmpServer =
       startStreaming(rtmpStreamingDevice, nRtmpStreamingPort)) == 0)
    {
      RTMP_Log(RTMP_LOGERROR, "Failed to start RTMP server, exiting!");
      return RD_FAILED;
    }
  RTMP_LogPrintf("Streaming on rtmp://%s:%d\n", rtmpStreamingDevice,
	    nRtmpStreamingPort);

  while (rtmpServer->state != STREAMING_STOPPED)
    {
      sleep(1);
    }
  RTMP_Log(RTMP_LOGDEBUG, "Done, exiting...");

  CleanupSockets();

#ifdef _DEBUG
  if (netstackdump != 0)
    fclose(netstackdump);
  if (netstackdump_read != 0)
    fclose(netstackdump_read);
#endif
  return nStatus;
}
开发者ID:firemyst,项目名称:rtmpdump,代码行数:71,代码来源:rtmpsrv.c


示例8: main

int
main(int argc, char **argv)
{
  int nStatus = RD_SUCCESS;

  // http streaming server
  char DEFAULT_HTTP_STREAMING_DEVICE[] = "0.0.0.0";	// 0.0.0.0 is any device

  char *httpStreamingDevice = DEFAULT_HTTP_STREAMING_DEVICE;	// streaming device, default 0.0.0.0
  int nHttpStreamingPort = 80;	// port
  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'},
    {"rtmp", 1, NULL, 'r'},
    {"swfUrl", 1, NULL, 's'},
    {"tcUrl", 1, NULL, 't'},
    {"pageUrl", 1, NULL, 'p'},
    {"app", 1, NULL, 'a'},
#ifdef CRYPTO
    {"swfhash", 1, NULL, 'w'},
    {"swfsize", 1, NULL, 'x'},
    {"swfVfy", 1, NULL, 'W'},
    {"swfAge", 1, NULL, 'X'},
#endif
    {"auth", 1, NULL, 'u'},
    {"conn", 1, NULL, 'C'},
    {"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'},
    {"device", 1, NULL, 'D'},
    {"sport", 1, NULL, 'g'},
    {"subscribe", 1, NULL, 'd'},
    {"start", 1, NULL, 'A'},
    {"stop", 1, NULL, 'B'},
    {"token", 1, NULL, 'T'},
    {"debug", 0, NULL, 'z'},
    {"quiet", 0, NULL, 'q'},
    {"verbose", 0, NULL, 'V'},
    {0, 0, 0, 0}
  };
  RTMP_LogPrintf("HTTP-RTMP Stream Gateway %s\n", RTMPDUMP_VERSION);
  RTMP_LogPrintf("(c) 2010 Andrej Stepanchuk, Howard Chu; license: GPL\n\n");

  // init request
  memset(&defaultRTMPRequest, 0, sizeof(RTMP_REQUEST));

  defaultRTMPRequest.rtmpport = -1;
  defaultRTMPRequest.protocol = RTMP_PROTOCOL_UNDEFINED;
  defaultRTMPRequest.bLiveStream = FALSE;	// is it a live stream? then we can't seek/resume

  defaultRTMPRequest.timeout = 120;	// timeout connection after 120 seconds
  defaultRTMPRequest.bufferTime = 20 * 1000;

  defaultRTMPRequest.swfAge = 30;



  signal(SIGINT, sigIntHandler);
#ifndef WIN32
  signal(SIGPIPE, SIG_IGN);
#endif

  InitSockets();

  while ((opt =
	  getopt_long(argc, argv,
		      "hvqVzr:s:t:p:a:f:u:n:c:l:y:m:d:D:A:B:T:g:w:x:W:X:S:", longopts,
		      NULL)) != -1)
    {
      switch (opt)
	{
	case 'h':
	  RTMP_LogPrintf
	    ("\nThis program serves media content streamed from RTMP onto HTTP.\n\n");
	  RTMP_LogPrintf("--help|-h               Prints this help screen.\n");
	  RTMP_LogPrintf
	    ("--rtmp|-r url           URL (e.g. rtmp://host[:port]/path)\n");
	  RTMP_LogPrintf
	    ("--host|-n hostname      Overrides the hostname in the rtmp url\n");
	  RTMP_LogPrintf
	    ("--port|-c port          Overrides the port in the rtmp url\n");
	  RTMP_LogPrintf
	    ("--socks|-S host:port    Use the specified SOCKS proxy\n");
	  RTMP_LogPrintf
	    ("--protocol|-l           Overrides the protocol in the rtmp url (0 - RTMP, 2 - RTMPE)\n");
	  RTMP_LogPrintf
	    ("--playpath|-y           Overrides the playpath parsed from rtmp url\n");
	  RTMP_LogPrintf("--swfUrl|-s url         URL to player swf file\n");
	  RTMP_LogPrintf
	    ("--tcUrl|-t url          URL to played stream (default: \"rtmp://host[:port]/app\")\n");
	  RTMP_LogPrintf("--pageUrl|-p url        Web URL of played programme\n");
//.........这里部分代码省略.........
开发者ID:bohrasd,项目名称:windowsrtdev,代码行数:101,代码来源:rtmpgw.c


示例9: usage

void usage(char *prog)
{
	RTMP_LogPrintf
		("\n%s: This program dumps the media content streamed over RTMP.\n\n", prog);
	RTMP_LogPrintf("--help|-h               Prints this help screen.\n");
	RTMP_LogPrintf
		("--url|-i url            URL with options included (e.g. rtmp://host[:port]/path swfUrl=url tcUrl=url)\n");
	RTMP_LogPrintf
		("--rtmp|-r url           URL (e.g. rtmp://host[:port]/path)\n");
	RTMP_LogPrintf
		("--live|-v               Save a live stream, no --resume (seeking) of live streams possible\n");
	RTMP_LogPrintf
		("--flv|-o string         FLV output file name, if the file name is - print stream to stdout\n");
	RTMP_LogPrintf
		("--timeout|-m num        Timeout connection num seconds (default: %u)\n", DEF_TIMEOUT);
	RTMP_LogPrintf
		("--quiet|-q              Suppresses all command output.\n");
	RTMP_LogPrintf("--verbose|-V            Verbose command output.\n");
	RTMP_LogPrintf
		("If you don't pass parameters for swfUrl, pageUrl, or auth these properties will not be included in the connect ");
	RTMP_LogPrintf("packet.\n\n");
}
开发者ID:kulv2012,项目名称:rtmpdump-librtmp,代码行数:22,代码来源:simplertmpdump.c


示例10: main

int main(int argc, char* argv[])
{
    InitSockets();

    double duration = -1;
    int nRead;
    //is live stream ?
    bool bLiveStream = TRUE;

    int bufsize = 1024 * 1024 * 10;
    char *buf = (char*)malloc(bufsize);
    memset(buf, 0, bufsize);
    long countbufsize = 0;

    FILE *fp = fopen("receive.flv", "wb");
    if (!fp)
    {
        RTMP_LogPrintf("Open File Error.\n");
        CleanupSockets();
        return -1;
    }

    /* set log level */
    RTMP_LogLevel loglvl = RTMP_LOGDEBUG;
    RTMP_LogSetLevel(loglvl);

    RTMP *rtmp = RTMP_Alloc();
    RTMP_Init(rtmp);
    //set connection timeout,default 30s
    rtmp->Link.timeout = 10;

	char* rtmpFilePath = "d:\\rtmp.raw";
	rtmp->m_pRTMPFile = fopen(rtmpFilePath,"rb");
	if (!rtmp->m_pRTMPFile)
	{
		RTMP_LogPrintf("Failed to open File :%s\n", rtmpFilePath);
		return FALSE;
	}

	// HKS's live URL
	if(!RTMP_SetupURL(rtmp, "rtmp://live.hkstv.hk.lxdns.com/live/hks"))
    {
        RTMP_Log(RTMP_LOGERROR, "SetupURL Err\n");
        RTMP_Free(rtmp);
        CleanupSockets();
        return -1;
    }
    if (bLiveStream)
    {
        rtmp->Link.lFlags |= RTMP_LF_LIVE;
    }

    //1hour
    RTMP_SetBufferMS(rtmp, 3600 * 1000);

    if(!RTMP_Connect(rtmp, NULL))
    {
        RTMP_Log(RTMP_LOGERROR, "Connect Err\n");
        RTMP_Free(rtmp);
        CleanupSockets();
        return -1;
    }

    if(!RTMP_ConnectStream(rtmp, 0))
    {
        RTMP_Log(RTMP_LOGERROR, "ConnectStream Err\n");
        RTMP_Close(rtmp);
        RTMP_Free(rtmp);
        CleanupSockets();
        return -1;
    }

    while(nRead = RTMP_Read(rtmp, buf, bufsize))
    {
        fwrite(buf, 1, nRead, fp);

        countbufsize += nRead;
        RTMP_LogPrintf("Receive: %5dByte, Total: %5.2fkB\n", nRead, countbufsize * 1.0 / 1024);
    }

    if(fp)
        fclose(fp);

    if(fpPcap)
        fclose(fpPcap);

	if (rtmp->m_pRTMPFile)
	{
		fclose(rtmp->m_pRTMPFile);
	}

    if(buf)
    {
        free(buf);
    }

    if(rtmp)
    {
        RTMP_Close(rtmp);
        RTMP_Free(rtmp);
//.........这里部分代码省略.........
开发者ID:minico,项目名称:rtmpdump,代码行数:101,代码来源:simplest_librtmp_receive.cpp


示例11: ParseOption


//.........这里部分代码省略.........
      {
	int protocol = atoi(arg);
	if (protocol < RTMP_PROTOCOL_RTMP || protocol > RTMP_PROTOCOL_RTMPTS)
	  {
	    RTMP_Log(RTMP_LOGERROR, "Unknown protocol specified: %d, using default",
		protocol);
	    return FALSE;
	  }
	else
	  {
	    req->protocol = protocol;
	  }
	break;
      }
    case 'y':
      STR2AVAL(req->playpath, arg);
      break;
    case 'r':
      {
	req->rtmpurl = arg;



	if (!RTMP_ParseURL
	    (req->rtmpurl, &parsedProtocol, &parsedHost, &parsedPort,
	     &parsedPlaypath, &parsedApp))
	  {
	    RTMP_Log(RTMP_LOGWARNING, "Couldn't parse the specified url (%s)!", arg);
	  }
	else
	  {
	    if (!req->hostname.av_len)
	      req->hostname = parsedHost;
	    if (req->rtmpport == -1)
	      req->rtmpport = parsedPort;
	    if (req->playpath.av_len == 0 && parsedPlaypath.av_len)
	      {
		    req->playpath = parsedPlaypath;
	      }
	    if (req->protocol == RTMP_PROTOCOL_UNDEFINED)
	      req->protocol = parsedProtocol;
	    if (req->app.av_len == 0 && parsedApp.av_len)
	      {
		    req->app = parsedApp;
	      }
	  }
	break;
      }
    case 's':
      STR2AVAL(req->swfUrl, arg);
      break;
    case 't':
      STR2AVAL(req->tcUrl, arg);
      break;
    case 'p':
      STR2AVAL(req->pageUrl, arg);
      break;
    case 'a':
      STR2AVAL(req->app, arg);
      break;
    case 'f':
      STR2AVAL(req->flashVer, arg);
      break;
    case 'u':
      STR2AVAL(req->auth, arg);
      break;
    case 'C':
      parseAMF(&req->extras, optarg, &req->edepth);
      break;
    case 'm':
      req->timeout = atoi(arg);
      break;
    case 'A':
      req->dStartOffset = (int)(atof(arg) * 1000.0);
      //printf("dStartOffset = %d\n", dStartOffset);
      break;
    case 'B':
      req->dStopOffset = (int)(atof(arg) * 1000.0);
      //printf("dStartOffset = %d\n", dStartOffset);
      break;
    case 'T':
      STR2AVAL(req->token, arg);
      break;
    case 'S':
      STR2AVAL(req->sockshost, arg);
    case 'q':
      RTMP_debuglevel = RTMP_LOGCRIT;
      break;
    case 'V':
      RTMP_debuglevel = RTMP_LOGDEBUG;
      break;
    case 'z':
      RTMP_debuglevel = RTMP_LOGALL;
      break;
    default:
      RTMP_LogPrintf("unknown option: %c, arg: %s\n", opt, arg);
      return FALSE;
    }
  return TRUE;
}
开发者ID:bohrasd,项目名称:windowsrtdev,代码行数:101,代码来源:rtmpgw.c


示例12: usage

void usage(char *prog)
{
	RTMP_LogPrintf
	("\n%s: This program dumps the media content streamed over RTMP.\n\n", prog);
	RTMP_LogPrintf("--help|-h               Prints this help screen.\n");
	RTMP_LogPrintf
	("--url|-i url            URL with options included (e.g. rtmp://host[:port]/path swfUrl=url tcUrl=url)\n");
	RTMP_LogPrintf
	("--rtmp|-r url           URL (e.g. rtmp://host[:port]/path)\n");
	RTMP_LogPrintf
	("--host|-n hostname      Overrides the hostname in the rtmp url\n");
	RTMP_LogPrintf
	("--port|-c port          Overrides the port in the rtmp url\n");
	RTMP_LogPrintf
	("--socks|-S host:port    Use the specified SOCKS proxy\n");
	RTMP_LogPrintf
	("--protocol|-l num       Overrides the protocol in the rtmp url (0 - RTMP, 2 - RTMPE)\n");
	RTMP_LogPrintf
	("--playpath|-y path      Overrides the playpath parsed from rtmp url\n");
	RTMP_LogPrintf
	("--playlist|-Y           Set playlist before playing\n");
	RTMP_LogPrintf("--swfUrl|-s url         URL to player swf file\n");
	RTMP_LogPrintf
	("--tcUrl|-t url          URL to played stream (default: \"rtmp://host[:port]/app\")\n");
	RTMP_LogPrintf("--pageUrl|-p url        Web URL of played programme\n");
	RTMP_LogPrintf("--app|-a app            Name of target app on server\n");
#ifdef CRYPTO
	RTMP_LogPrintf
	("--swfhash|-w hexstring  SHA256 hash of the decompressed SWF file (32 bytes)\n");
	RTMP_LogPrintf
	("--swfsize|-x num        Size of the decompressed SWF file, required for SWFVerification\n");
	RTMP_LogPrintf
	("--swfVfy|-W url         URL to player swf file, compute hash/size automatically\n");
	RTMP_LogPrintf
	("--swfAge|-X days        Number of days to use cached SWF hash before refreshing\n");
#endif
	RTMP_LogPrintf
	("--auth|-u string        Authentication string to be appended to the connect string\n");
	RTMP_LogPrintf
	("--conn|-C type:data     Arbitrary AMF data to be appended to the connect string\n");
	RTMP_LogPrintf
	("                        B:boolean(0|1), S:string, N:number, O:object-flag(0|1),\n");
	RTMP_LogPrintf
	("                        Z:(null), NB:name:boolean, NS:name:string, NN:name:number\n");
	RTMP_LogPrintf
	("--flashVer|-f string    Flash version string (default: \"%s\")\n",
	 RTMP_DefaultFlashVer.av_val);
	RTMP_LogPrintf
	("--live|-v               Save a live stream, no --resume (seeking) of live streams possible\n");
	RTMP_LogPrintf
	("--subscribe|-d string   Stream name to subscribe to (otherwise defaults to playpath if live is specifed)\n");
	RTMP_LogPrintf
	("--realtime|-R           Don't attempt to speed up download via the Pause/Unpause BUFX hack\n");
	RTMP_LogPrintf
	("--flv|-o string         FLV output file name, if the file name is - print stream to stdout\n");
	RTMP_LogPrintf
	("--resume|-e             Resume a partial RTMP download\n");
	RTMP_LogPrintf
	("--timeout|-m num        Timeout connection num seconds (default: %u)\n",
	 DEF_TIMEOUT);
	RTMP_LogPrintf
	("--start|-A num          Start at num seconds into stream (not valid when using --live)\n");
	RTMP_LogPrintf
	("--stop|-B num           Stop at num seconds into stream\n");
	RTMP_LogPrintf
	("--token|-T key          Key for SecureToken response\n");
	RTMP_LogPrintf
	("--jtv|-j JSON           Authentication token for Justin.tv legacy servers\n");
	RTMP_LogPrintf
	("--hashes|-#             Display progress with hashes, not with the byte counter\n");
	RTMP_LogPrintf
	("--buffer|-b             Buffer time in milliseconds (default: %u)\n",
	 DEF_BUFTIME);
	RTMP_LogPrintf
	("--skip|-k num           Skip num keyframes when looking for last keyframe to resume from. Useful if resume fails (default: %d)\n\n",
	 DEF_SKIPFRM);
	RTMP_LogPrintf
	("--quiet|-q              Suppresses all command output.\n");
	RTMP_LogPrintf("--verbose|-V            Verbose command output.\n");
	RTMP_LogPrintf("--debug|-z              Debug level command output.\n");
	RTMP_LogPrintf
	("If you don't pass parameters for swfUrl, pageUrl, or auth these properties will not be included in the connect ");
	RTMP_LogPrintf("packet.\n\n");
}
开发者ID:odol0503,项目名称:rtmpdump_vs2005,代码行数:84,代码来源:rtmpdump.cpp


示例13: setError

void QRtmp::run()
{
    bool first = true;
    int retries = 0;

    if(!m_destFile.fileName().isEmpty())
        if(!m_destFile.open(QIODevice::WriteOnly))
        {
            setError(QString("Can't open %1 for writing").arg(m_destFile.fileName()));
            return;
        }

    while(!m_stop)
    {
        RTMP_Log(RTMP_LOGDEBUG, "Setting buffer time to: %dms", m_bufferTime);
        RTMP_SetBufferMS(m_rtmp, m_bufferTime);

        if(first)
        {
            first = false;
            RTMP_LogPrintf("Connecting ...\n");

            if(!RTMP_Connect(m_rtmp, NULL))
            {
                setError("RTMP_Connect failed");
                break;
            }

            RTMP_Log(RTMP_LOGINFO, "Connected...");

            // User defined seek offset
            if(dStartOffset > 0)
            {
                // Don't need the start offset if resuming an existing file
                if(m_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");
                    break;
                }
            }

            if(!RTMP_ConnectStream(m_rtmp, dSeek))
            {
                setError("RTMP_ConnectStream failed");
                break;
            }
        }
        else
        {
            if(retries)
            {
                RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n");
                if(!RTMP_IsTimedout(m_rtmp))
                    setError("RTMP_IsTimedout failed");
                else
                    setError("RTMP_IsTimedout 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(m_rtmp->m_pausing == 3)
            {
                /* Only one try at reconnecting... */
                retries = 1;
                dSeek = m_rtmp->m_pauseStamp;
                if(dStopOffset > 0)
                {
                    if(dStopOffset <= dSeek)
                    {
                        RTMP_LogPrintf("Already Completed\n");
                        break;
                    }
                }
                if(!RTMP_ReconnectStream(m_rtmp, dSeek))
                {
                    RTMP_Log(RTMP_LOGERROR, "Failed to resume the stream\n\n");
                    if(!RTMP_IsTimedout(m_rtmp))
                        setError("RTMP_IsTimedout failed");
                    else
                        setError("RTMP_IsTimedout RD_INCOMPLETE");
                    break;
                }
            }
            else if(!RTMP_ToggleStream(m_rtmp))
            {
//.........这里部分代码省略.........
开发者ID:theappgeek,项目名称:Media-Stream-Downloader,代码行数:101,代码来源:qrtmp.cpp


示例14: doServe

void doServe(STREAMING_SERVER * server,	// server socket and state (our listening socket)
  int sockfd	// client connection socket
  )
{
  server->state = STREAMING_IN_PROGRESS;

  RTMP rtmp = { 0 };		/* our session with the real client */
  RTMPPacket packet = { 0 };

  // timeout for http requests
  fd_set fds;
  struct timeval tv;

  memset(&tv, 0, sizeof(struct timeval));
  tv.tv_sec = 5;

  FD_ZERO(&fds);
  FD_SET(sockfd, &fds);

  if (select(sockfd + 1, &fds, NULL, NULL, &tv) <= 0)
    {
      RTMP_Log(RTMP_LOGERROR, "Request timeout/select failed, ignoring request");
      goto quit;
    }
  else
    {
      RTMP_Init(&rtmp);
      rtmp.m_sb.sb_socket = sockfd;
      if (!RTMP_Serve(&rtmp))
	{
	  RTMP_Log(RTMP_LOGERROR, "Handshake failed");
	  goto cleanup;
	}
    }
  server->arglen = 0;
  while (RTMP_IsConnected(&rtmp) && RTMP_ReadPacket(&rtmp, &packet))
    {
      if (!RTMPPacket_IsReady(&packet))
	continue;
      ServePacket(server, &rtmp, &packet);
      RTMPPacket_Free(&packet);
    }

cleanup:
  RTMP_LogPrintf("Closing connection... ");
  RTMP_Close(&rtmp);
  /* Should probably be done by RTMP_Close() ... */
  rtmp.Link.playpath.av_val = NULL;
  rtmp.Link.tcUrl.av_val = NULL;
  rtmp.Link.swfUrl.av_val = NULL;
  rtmp.Link.pageUrl.av_val = NULL;
  rtmp.Link.app.av_val = NULL;
  rtmp.Link.flashVer.av_val = NULL;
  RTMP_LogPrintf("done!\n\n");

quit:
  if (server->state == STREAMING_IN_PROGRESS)
    server->state = STREAMING_ACCEPTING;

  return;
}
开发者ID:firemyst,项目名称:rtmpdump,代码行数:61,代码来源:rtmpsrv.c


示例15: main

int main(int argc, char **argv)
{
	extern char *optarg;

	int nStatus = RD_SUCCESS;
	int bStdoutMode = TRUE;	// if true print the stream directly to stdout, messages go to stderr
	int bLiveStream = TRUE;	// is it a live stream? then we can't seek/resume

	long int timeout = DEF_TIMEOUT;	// timeout connection after 120 seconds
	RTMP rtmp = { 0 };
	AVal fullUrl = { 0, 0 };
	RTMP_debuglevel = RTMP_LOGINFO;
	char *flvFile = 0;

	signal(SIGINT, sigIntHandler);
	signal(SIGTERM, sigIntHandler);
	signal(SIGHUP, sigIntHandler);
	signal(SIGPIPE, sigIntHandler);
	signal(SIGQUIT, sigIntHandler);


	RTMP_LogPrintf("RTMPDump %s\n", RTMPDUMP_VERSION);
	RTMP_LogPrintf("(c) 2010 Andrej Stepanchuk, Howard Chu, The Flvstreamer Team; license: GPL\n");


	int opt;
	struct option longopts[] = {
		{"help", 0, NULL, 'h'},
		{"url", 1, NULL, 'i'},
		{"rtmp", 1, NULL, 'r'},
		{"live", 0, NULL, 'v'},
		{"timeout", 1, NULL, 'm'},
		{"hashes", 0, NULL, '#'},
		{"quiet", 0, NULL, 'q'},
		{"verbose", 0, NULL, 'V'},
		{0, 0, 0, 0}
	};

	while ((opt = getopt_long(argc, argv, "hVvqzRr:s:t:i:p:a:b:f:o:u:C:n:c:l:y:Ym:k:d:A:B:T:w:x:W:X:S:#j:", longopts, NULL)) != -1)
	{
		switch (opt)
		{
			case 'h':
				usage(argv[0]);
				return RD_SUCCESS;

			case 'v':
				bLiveStream = TRUE;	// no seeking or resuming possible!
				break;
			case 'i':
				STR2AVAL(fullUrl, optarg);
				break;
			case 'o':
				flvFile = optarg;
				if (strcmp(flvFile, "-"))
					bStdoutMode = FALSE;

				break;
			case 'm':
					  timeout = atoi(optarg);
					  break;
			case 'q':
					  RTMP_debuglevel = RTMP_LOGCRIT;
					  break;
			case 'V':
					  RTMP_debuglevel = RTMP_LOGDEBUG;
					  break;
			case 'z':
					  RTMP_debuglevel = RTMP_LOGALL;
					  break;
			default:
					  RTMP_LogPrintf("unknown option: %c\n", opt);
					  usage(argv[0]);
					  return RD_FAILED;
					  break;
		}
	}

	if (flvFile == 0) {
		RTMP_Log(RTMP_LOGWARNING, "You haven't specified an output file (-o filename), using stdout");
		bStdoutMode = TRUE;
	}
	if (!file)	{
		if (bStdoutMode) {
			file = stdout;
			SET_BINMODE(file);
		}
		else {
			file = fopen(flvFile, "w+b");
			if (file == 0)	{
				RTMP_LogPrintf("Failed to open file! %s\n", flvFile);
				return RD_FAILED;
			}
		}
	}

	////////////////////////////////////////////////////////////////////////////////////////
	RTMP_Init(&rtmp);//初始化RTMP参数
	//指定了-i 参数,直接设置URL
	if (RTMP_SetupURL(&rtmp, fullUrl.av_val) == FALSE) {
//.........这里部分代码省略.........
开发者ID:kulv2012,项目名称:rtmpdump-librtmp,代码行数:101,代码来源:simplertmpdump.c


示例16: start_sample_rtmp_server

int
start_sample_rtmp_server(int argc, char **argv)
{
	int nStatus = RD_SUCCESS;
	int i;

	// http streaming server
	char DEFAULT_HTTP_STREAMING_DEVICE[] = "0.0.0.0";	// 0.0.0.0 is any device

	char *rtmpStreamingDevice = DEFAULT_HTTP_STREAMING_DEVICE;	// streaming device, default 0.0.0.0
	int nRtmpStreamingPort = 1935;	// port
	char *cert = NULL, *key = NULL;

	RTMP_LogPrintf("RTMP Server %s\n", "AVRemoteControl Build 1.0");
	RTMP_LogPrintf("(c) 2010 Andrej Stepanchuk, Howard Chu; license: GPL\n\n");

	RTMP_debuglevel = RTMP_LOGINFO;

	for (i = 1; i < argc; i++)
	{
		if (!strcmp(argv[i], "-z"))
			RTMP_debuglevel = RTMP_LOGALL;
		else if (!strcmp(argv[i], "-c") && i + 1 < argc)
			cert = argv[++i];
		else if (!strcmp(argv[i], "-k") && i + 1 < argc)
			key = argv[++i];
	}

	if (cert && key)
		sslCtx = RTMP_TLS_AllocServerContext(cert, key);

	// init request
	memset(&defaultRTMPRequest, 0, sizeof(RTMP_REQUEST));

	defaultRTMPRequest.rtmpport = -1;
	defaultRTMPRequest.protocol = RTMP_PROTOCOL_UNDEFINED;
	defaultRTMPRequest.bLiveStream = true;	// is it a live stream? then we can't seek/resume

	defaultRTMPRequest.timeout = 500;	// timeout connection after 300 seconds
	defaultRTMPRequest.bufferTime = 20 * 1000;


	signal(SIGINT, sigIntHandler);
#ifndef WIN32
	signal(SIGPIPE, SIG_IGN);
#endif

#ifdef _DEBUG
	netstackdump = fopen("netstackdump", "wb");
	netstackdump_read = fopen("netstackdump_read", "wb");
#endif

	InitSockets();

	// start text UI
	//ThreadCreate(controlServerThread, 0);
	std::thread theThread(controlServerThread);

	// start http streaming
	if ((rtmpServer =
		startStreaming(rtmpStreamingDevice, nRtmpStreamingPort)) == 0)
	{
		RTMP_Log(RTMP_LOGERROR, "Failed to start RTMP server, exiting!");
		return RD_FAILED;
	}
	RTMP_LogPrintf("Streaming on rtmp://%s:%d\n", rtmpStreamingDevice,
		nRtmpStreamingPort);

	while (rtmpServer->state != STREAMING_STOPPED)
	{
		sleep(1);
	}
	RTMP_Log(RTMP_LOGDEBUG, "Done, exiting...");

	if (sslCtx)
		RTMP_TLS_FreeServerContext(sslCtx);

	CleanupSockets();

#ifdef _DEBUG
	if (netstackdump != 0)
		fclose(netstackdump);
	if (netstackdump_read != 0)
		fclose(netstackdump_read);
#endif
	return nStatus;
}
开发者ID:373137461,项目名称:AVRemoteControl,代码行数:87,代码来源:rtmpsrv.cpp


示例17: do_rtmp

int
do_rtmp(int port,int protocol,char* playpath_arg,char* host,char* swfVfy_arg,char* tcUrl_arg,char* app_arg,char* pageUrl_arg,char* flashVer_arg,char* conn,char* outfile)
{
  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 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 };
  AVal usherToken = { 0, 0 }; //Justin.tv auth token
  int retries = 0;
  int bLiveStream = FALSE;	// is it a live stream? 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 dStopOffset = 0;
  RTMP rtmp;

  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);
  signal(SIGHUP, sigIntHandler);
  signal(SIGPIPE, sigIntHandler);
  signal(SIGQUIT, sigIntHandler);

  RTMP_debuglevel = RTMP_LOGINFO;

  RTMP_Init(&rtmp);

  STR2AVAL(&swfUrl,swfVfy_arg);
  swfVfy = 1;

  if (host) { STR2AVAL(&hostname, host); }
  if (playpath_arg) { STR2AVAL(&playpath, playpath_arg); }
  if (tcUrl_arg) { STR2AVAL(&tcUrl, tcUrl_arg); }
  if (pageUrl_arg) { STR2AVAL(&pageUrl, pageUrl_arg); }
  if (app_arg) { STR2AVAL(&app, app_arg); }
  if (flashVer_arg) { STR2AVAL(&flashVer, flashVer_arg); }

  if (conn) {
    AVal av;
    STR2AVAL(&av, conn);
    if (!RTMP_SetOpt(&rtmp, &av_conn, &av))
    {
      RTMP_Log(RTMP_LOGERROR, "Invalid AMF parameter: %s", conn);
      return RD_FAILED;
    }
  }

  flvFile = outfile;

  file = fopen(flvFile, "w+b");
  if (file == 0)
    {
      RTMP_LogPrintf("Failed to open file! %s\n", flvFile);
      return RD_FAILED;
    }

  if (port == 0)
    {
      if (protocol & RTMP_FEATURE_SSL)
	port = 443;
      else if (protocol & RTMP_FEATURE_HTTP)
	port = 80;
//.........这里部分代码省略.........
开发者ID:linuxstb,项目名称:catchup,代码行数:101,代码来源:rtmpdump.c


示例18: Download

int Download(RTMP * rtmp, FILE * file, int bStdoutMode, int bLiveStream)
{
	int32_t now, lastUpdate;
	int bufferSize = 64 * 1024;
	char *buffer;
	int nRead = 0;
	off_t size = ftello(file);

	rtmp->m_read.timestamp = 0;

	if (rtmp->m_read.timestamp)	{
		RTMP_Log(RTMP_LOGDEBUG, "Continuing at TS: %d ms\n", rtmp->m_read.timestamp);
	}

	if (bLiveStream) {
		RTMP_LogPrintf("Starting Live Stream\n");
	}
	else {
		RTMP_LogPrintf("Starting download at: %.3f kB\n", (double) size / 1024.0);
	}

	rtmp->m_read.initialFrameType = 0;
	rtmp->m_read.nResumeTS = 0;
	rtmp->m_read.metaHeader = 0;
	rtmp->m_read.initialFrame = 0;
	rtmp->m_read.nMetaHeaderSize = 0;
	rtmp->m_read.nInitialFrameSize = 0;

	buffer = (char *) malloc(bufferSize);

	unsigned long lasttimestamp = 0 ;	
	unsigned long streamlasttimestamp = 0 ;	
	now = RTMP_GetTime();
	lastUpdate = now - 1000;
	do
	{
		nRead = RTMP_Read(rtmp, buffer, bufferSize);
		//RTMP_LogPrintf("nRead: %d\n", nRead);
		if (nRead > 0) {
			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;
			}
			size += nRead;

			///////log 
			struct timeval tnow ;
			gettimeofday(&tnow, NULL);
			unsigned long t2_ms = tnow.tv_sec * 1000 + tnow.tv_usec/1000;
			if( lasttimestamp == 0 ) {
				streamlasttimestamp = rtmp->m_read.timestamp ;//这个是音频流里面的时间戳。这个算出来的就是音频的码率
				lasttimestamp = t2_ms ; //这个是我们收到数据的时间戳,这个算出来的是音频的实际接收码率
				continue ;
			}

			float tmprate = (double)8*nRead/1024/( (double)(t2_ms-lasttimestamp)/1000)  ;
			float streamtmprate = (double)8*nRead/1024/( (double)(rtmp->m_read.timestamp - streamlasttimestamp)/1000)  ;
			KULV_LOG("RTMP_Read,ms,nread:%d\trecv timediff:%dms,receive rate:%.2fkbps\tstream timediff:%dms,stream rate:%.1fkbps.", 
					nRead, t2_ms-lasttimestamp, tmprate, rtmp->m_read.timestamp - streamlasttimestamp, streamtmprate );
			lasttimestamp = t2_ms ;
			streamlasttimestamp = rtmp->m_read.timestamp ;

			now = RTMP_GetTime();
			if (abs(now - lastUpdate) > 200) {
				RTMP_LogStatus("\r%.3f kB / %.2f sec. recv speed:%.2f kbps. stream rate:%.2f kbps.", (double) size / 1024.0, (double) (rtmp->m_read.timestamp) / 1000.0, tmprate, streamtmprate);
				lastUpdate = now;
			}
		}
		else {
			RTMP_Log(RTMP_LOGDEBUG, "zero read!");
			if (rtmp->m_read.status == RTMP_READ_EOF)
				break;
		}

	}
	while (!RTMP_ctrlC && nRead > -1 && RTMP_IsConnected(rtmp) && !RTMP_IsTimedout(rtmp));

	free(buffer);
	if (nRead < 0)
		nRead = rtmp->m_read.status;

	RTMP_Log(RTMP_LOGDEBUG, "RTMP_Read returned: %d", nRead);

	if (nRead == -3)
		return RD_SUCCESS;

	if ( RTMP_ctrlC || nRead < 0 || RTMP_IsTimedout(rtmp)) {
		return RD_INCOMPLETE;
	}

	return RD_SUCCESS;
}
开发者ID:kulv2012,项目名称:rtmpdump-librtmp,代码行数:93,代码来源:simplertmpdump.c


示例19: Download

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 bRealtimeStream, 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;
	off_t size = ftello(file);
	unsigned long lastPercent = 0;

	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);
	}

	if (bLiveStream)
	{
		RTMP_LogPrintf("Starting Live Stream\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 (bRealtimeStream)
			RTMP_LogPrintf("  in approximately realtime (disabled BUFX speedup hack)\n");
	}

	if (dStopOffset > 0)
		RTMP_LogPrintf("For duration: %.3f sec\n", (double) (dStopOffset - dSeek) / 1000.0);

	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;

	buffer = (char *) malloc(bufferSize);

	now = RTMP_GetTime();
	lastUpdate = now - 1000;
	do
	{
		nRead = RTMP_Read(rtmp, buffer, bufferSize);
		//RTMP_LogPrintf("nRead: %d\n", nRead);
		if (nRead > 0)
		{
			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;
			}
			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;	  // extra 5sec to make sure we've got enough

					RTMP_Log(RTMP_LOGDEBUG,
//.........这里部分代码省略.........
开发者ID:odol0503,项目名称:rtmpdump_vs2005,代码行数:101,代码来源:rtmpdump.cpp


示例20: publish_using_write

该文章已有0人参与评论

请发表评论

全部评论

专题导读
上一篇:
C++ RTMP_MLME_HANDLER函数代码示例发布时间:2022-05-30
下一篇:
C++ RTMP_Log函数代码示例发布时间:2022-05-30
热门推荐
阅读排行榜

扫描微信二维码

查看手机版网站

随时了解更新最新资讯

139-2527-9053

在线客服(服务时间 9:00~18:00)

在线QQ客服
地址:深圳市南山区西丽大学城创智工业园
电邮:jeky_zhao#qq.com
移动电话:139-2527-9053

Powered by 互联科技 X3.4© 2001-2213 极客世界.|Sitemap