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C++ RTMP_Init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_Init函数的典型用法代码示例。如果您正苦于以下问题：C++ RTMP_Init函数的具体用法？C++ RTMP_Init怎么用？C++ RTMP_Init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_Init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: rtmp_probe

static int
rtmp_probe(const char *url0, char *errbuf, size_t errlen, int timeout_ms)
{
  RTMP *r;
  char *url = mystrdupa(url0);

  r = RTMP_Alloc();
  RTMP_Init(r, NULL);

  if(!RTMP_SetupURL(r, url)) {
    snprintf(errbuf, errlen, "Unable to setup RTMP-session");
    RTMP_Free(r);
    return BACKEND_PROBE_FAIL;
  }

  if(!RTMP_Connect(r, NULL, errbuf, errlen, timeout_ms)) {
    RTMP_Close(r);
    RTMP_Free(r);
    return BACKEND_PROBE_FAIL;
  }

  RTMP_SetReadTimeout(r, timeout_ms);

  if(!RTMP_ConnectStream(r, 0)) {
    snprintf(errbuf, errlen, "Unable to connect RTMP-stream");
    RTMP_Close(r);
    RTMP_Free(r);
    return BACKEND_PROBE_FAIL;
  }

  RTMP_Close(r);
  RTMP_Free(r);

  return BACKEND_PROBE_OK;
}
开发者ID:lprot，项目名称:showtime，代码行数:35，代码来源:rtmp.c


示例2: m_rtmp

QRtmp::QRtmp() :
    m_rtmp(new RTMP_private),
    m_swfSize(0),
    m_nSkipKeyFrames(0),
    m_bufferTime(10 * 60 * 60 * 1000), /* 10 hours default */
    m_bOverrideBufferTime(false),
    m_bLiveStream(true),
    m_port(-1),
    m_proto(Undefined),
    m_timeout(30),
    dStartOffset(0), dStopOffset(0), dSeek(0),
    m_bResume(false),
    m_stop(false),
    m_percent(0),
    m_duration(0),
    m_streamIsRunning(false)
{
    if(!m_socketsInitialized)
    {
#ifdef WIN32
      WORD version;
      WSADATA wsaData;

      version = MAKEWORD(1, 1);
      m_socketsInitialized = (WSAStartup(version, &wsaData) == 0);
#else
      m_socketsInitialized = true;
#endif
    }

    RTMP_Init(m_rtmp);
}
开发者ID:theappgeek，项目名称:Media-Stream-Downloader，代码行数:32，代码来源:qrtmp.cpp


示例3: gst_rtmp_sink_start

static gboolean
gst_rtmp_sink_start (GstBaseSink * basesink)
{
  GstRTMPSink *sink = GST_RTMP_SINK (basesink);

  if (!sink->uri) {
    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
        ("Please set URI for RTMP output"), ("No URI set before starting"));
    return FALSE;
  }

  sink->rtmp_uri = g_strdup (sink->uri);
  sink->rtmp = RTMP_Alloc ();
  RTMP_Init (sink->rtmp);
  if (!RTMP_SetupURL (sink->rtmp, sink->rtmp_uri)) {
    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
        ("Failed to setup URL '%s'", sink->uri));
    RTMP_Free (sink->rtmp);
    sink->rtmp = NULL;
    g_free (sink->rtmp_uri);
    sink->rtmp_uri = NULL;
    return FALSE;
  }

  GST_DEBUG_OBJECT (sink, "Created RTMP object");

  /* Mark this as an output connection */
  RTMP_EnableWrite (sink->rtmp);

  sink->first = TRUE;

  return TRUE;
}
开发者ID:dylansong77，项目名称:gstreamer，代码行数:33，代码来源:gstrtmpsink.c


示例4: RTMP_Alloc

/*
 * Class:     net_butterflytv_rtmp_client_RtmpClient
 * Method:    open
 * Signature: (Ljava/lang/String;)I
 */
JNIEXPORT jint JNICALL Java_net_ossrs_sea_RtmpClient_open
        (JNIEnv * env, jobject thiz, jstring url_, jboolean isPublishMode) {

    const char *url = (*env)->GetStringUTFChars(env, url_, 0);
    rtmp = RTMP_Alloc();
    if (rtmp == NULL) {
        return -1;
    }

	RTMP_Init(rtmp);
	int ret = RTMP_SetupURL(rtmp, url);

    if (!ret) {
        RTMP_Free(rtmp);
        return -2;
    }
    if (isPublishMode) {
        RTMP_EnableWrite(rtmp);
    }

	ret = RTMP_Connect(rtmp, NULL);
    if (!ret) {
        RTMP_Free(rtmp);
        return -3;
    }
	ret = RTMP_ConnectStream(rtmp, 0);

    if (!ret) {
        return -4;
    }
    (*env)->ReleaseStringUTFChars(env, url_, url);
    return 1;
}
开发者ID:haifengdeng，项目名称:Android_Caputure_push，代码行数:38，代码来源:librtmp-jni.c


示例5: rtmp_open

/**
 * Open RTMP connection and verify that the stream can be played.
 *
 * URL syntax: rtmp://server[:port][/app][/playpath][ keyword=value]...
 *             where 'app' is first one or two directories in the path
 *             (e.g. /ondemand/, /flash/live/, etc.)
 *             and 'playpath' is a file name (the rest of the path,
 *             may be prefixed with "mp4:")
 *
 *             Additional RTMP library options may be appended as
 *             space-separated key-value pairs.
 */
static int rtmp_open(URLContext *s, const char *uri, int flags)
{
    LibRTMPContext *ctx = s->priv_data;
    RTMP *r = &ctx->rtmp;
    int rc = 0, level;
    char *filename = s->filename;

    switch (av_log_get_level()) {
    default:
    case AV_LOG_FATAL:   level = RTMP_LOGCRIT;    break;
    case AV_LOG_ERROR:   level = RTMP_LOGERROR;   break;
    case AV_LOG_WARNING: level = RTMP_LOGWARNING; break;
    case AV_LOG_INFO:    level = RTMP_LOGINFO;    break;
    case AV_LOG_VERBOSE: level = RTMP_LOGDEBUG;   break;
    case AV_LOG_DEBUG:   level = RTMP_LOGDEBUG2;  break;
    }
    RTMP_LogSetLevel(level);
    RTMP_LogSetCallback(rtmp_log);

    if (ctx->app || ctx->playpath) {
        int len = strlen(s->filename) + 1;
        if (ctx->app)      len += strlen(ctx->app)      + sizeof(" app=");
        if (ctx->playpath) len += strlen(ctx->playpath) + sizeof(" playpath=");

        if (!(filename = av_malloc(len)))
            return AVERROR(ENOMEM);

        av_strlcpy(filename, s->filename, len);
        if (ctx->app) {
            av_strlcat(filename, " app=", len);
            av_strlcat(filename, ctx->app, len);
        }
        if (ctx->playpath) {
            av_strlcat(filename, " playpath=", len);
            av_strlcat(filename, ctx->playpath, len);
        }
    }

    RTMP_Init(r);
    if (!RTMP_SetupURL(r, filename)) {
        rc = AVERROR_UNKNOWN;
        goto fail;
    }

    if (flags & AVIO_FLAG_WRITE)
        RTMP_EnableWrite(r);

    if (!RTMP_Connect(r, NULL) || !RTMP_ConnectStream(r, 0)) {
        rc = AVERROR_UNKNOWN;
        goto fail;
    }

    s->is_streamed = 1;
    rc = 0;
fail:
    if (filename != s->filename)
        av_freep(&filename);
    return rc;
}
开发者ID:AVbin，项目名称:libav，代码行数:71，代码来源:librtmp.c


示例6: rtmp_open

/**
 * Open RTMP connection and verify that the stream can be played.
 *
 * URL syntax: rtmp://server[:port][/app][/playpath][ keyword=value]...
 *             where 'app' is first one or two directories in the path
 *             (e.g. /ondemand/, /flash/live/, etc.)
 *             and 'playpath' is a file name (the rest of the path,
 *             may be prefixed with "mp4:")
 *
 *             Additional RTMP library options may be appended as
 *             space-separated key-value pairs.
 */
static int rtmp_open(URLContext *s, const char *uri, int flags)
{
	RTMP *r;
	int rc;

	r = av_mallocz(sizeof(RTMP));
	if (!r)
		return AVERROR(ENOMEM);

	switch (av_log_get_level())
	{
	default:
	case AV_LOG_FATAL:
		rc = RTMP_LOGCRIT;
		break;
	case AV_LOG_ERROR:
		rc = RTMP_LOGERROR;
		break;
	case AV_LOG_WARNING:
		rc = RTMP_LOGWARNING;
		break;
	case AV_LOG_INFO:
		rc = RTMP_LOGINFO;
		break;
	case AV_LOG_VERBOSE:
		rc = RTMP_LOGDEBUG;
		break;
	case AV_LOG_DEBUG:
		rc = RTMP_LOGDEBUG2;
		break;
	}
	RTMP_LogSetLevel(rc);
	RTMP_LogSetCallback(rtmp_log);

	RTMP_Init(r);
	if (!RTMP_SetupURL(r, s->filename))
	{
		rc = -1;
		goto fail;
	}

	if (flags & AVIO_WRONLY)
		RTMP_EnableWrite(r);

	if (!RTMP_Connect(r, NULL) || !RTMP_ConnectStream(r, 0))
	{
		rc = -1;
		goto fail;
	}

	s->priv_data   = r;
	s->is_streamed = 1;
	return 0;
fail:
	av_free(r);
	return rc;
}
开发者ID:hicks0074，项目名称:freescale_omx_framework，代码行数:69，代码来源:librtmp.c


示例7: rtmp_playvideo

static event_t *
rtmp_playvideo(const char *url0, media_pipe_t *mp,
	       int flags, int priority,
	       char *errbuf, size_t errlen,
	       const char *mimetype)
{
  rtmp_t r = {0};
  event_t *e;
  char *url = mystrdupa(url0);

  prop_set_string(mp->mp_prop_type, "video");

  RTMP_LogSetLevel(RTMP_LOGINFO);

  r.r = RTMP_Alloc();
  RTMP_Init(r.r);

  if(!RTMP_SetupURL(r.r, url)) {
    snprintf(errbuf, errlen, "Unable to setup RTMP-session");
    rtmp_free(&r);
    return NULL;
  }

  if(!RTMP_Connect(r.r, NULL)) {
    snprintf(errbuf, errlen, "Unable to connect RTMP-session");
    rtmp_free(&r);
    return NULL;
  }

  if(!RTMP_ConnectStream(r.r, 0)) {
    snprintf(errbuf, errlen, "Unable to connect RTMP-stream");
    rtmp_free(&r);
    return NULL;
  }

  mp->mp_audio.mq_stream = 0;
  mp->mp_video.mq_stream = 0;

  mp_configure(mp, MP_PLAY_CAPS_PAUSE, MP_BUFFER_DEEP);
  mp->mp_max_realtime_delay = (r.r->Link.timeout - 1) * 1000000;

  mp_become_primary(mp);

  e = rtmp_loop(&r, mp, url, errbuf, errlen);

  mp_flush(mp, 0);
  mp_shutdown(mp);

  TRACE(TRACE_DEBUG, "RTMP", "End of stream");

  rtmp_free(&r);
  return e;
}
开发者ID:bielorkut，项目名称:showtime，代码行数:53，代码来源:rtmp.c


示例8: main

int main(int argc, char **argv)
{
    RTMP *rtmp=RTMP_Alloc();

    if(!rtmp)
        return 1;

    RTMP_Init(rtmp);
    RTMP_Free(rtmp);

    return 0;
}
开发者ID:Cyberunner23，项目名称:hunter，代码行数:12，代码来源:main.cpp


示例9: gst_rtmp_src_start

/* open the file, do stuff necessary to go to PAUSED state */
static gboolean
gst_rtmp_src_start (GstBaseSrc * basesrc)
{
  GstRTMPSrc *src;

  src = GST_RTMP_SRC (basesrc);

  if (!src->uri) {
    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given"));
    return FALSE;
  }

  src->cur_offset = 0;
  src->last_timestamp = 0;
  src->discont = TRUE;

  src->rtmp = RTMP_Alloc ();

  if (!src->rtmp) {
    GST_ERROR_OBJECT (src, "Could not allocate librtmp's RTMP context");
    goto error;
  }

  RTMP_Init (src->rtmp);
  if (!RTMP_SetupURL (src->rtmp, src->uri)) {
    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
        ("Failed to setup URL '%s'", src->uri));
    goto error;
  }
  src->seekable = !(src->rtmp->Link.lFlags & RTMP_LF_LIVE);
  GST_INFO_OBJECT (src, "seekable %d", src->seekable);

  /* open if required */
  if (!RTMP_IsConnected (src->rtmp)) {
    if (!RTMP_Connect (src->rtmp, NULL)) {
      GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
          ("Could not connect to RTMP stream \"%s\" for reading", src->uri));
      goto error;
    }
  }

  return TRUE;

error:
  if (src->rtmp) {
    RTMP_Free (src->rtmp);
    src->rtmp = NULL;
  }
  return FALSE;
}
开发者ID:asrashley，项目名称:gst-plugins-bad，代码行数:51，代码来源:gstrtmpsrc.c


示例10: rtmp_setup_connection

static CURLcode rtmp_setup_connection(struct connectdata *conn)
{
  RTMP *r = RTMP_Alloc();
  if(!r)
    return CURLE_OUT_OF_MEMORY;

  RTMP_Init(r);
  RTMP_SetBufferMS(r, DEF_BUFTIME);
  if(!RTMP_SetupURL(r, conn->data->change.url)) {
    RTMP_Free(r);
    return CURLE_URL_MALFORMAT;
  }
  conn->proto.generic = r;
  return CURLE_OK;
}
开发者ID:601040605，项目名称:WNetLicensor，代码行数:15，代码来源:curl_rtmp.c


示例11: ll

int CRtmpdSession::Init( sqbind::CSqSocket *pSocket )
{
	oexAutoLock ll( _g_rtmpd_lock );
	if ( !ll.IsLocked() ) return 0;

	// Out with the old
	Destroy();

#if _DEBUG

	// Unfortunately, this is a must for the debug version
	if ( !netstackdump || !netstackdump_read )
	{	setLastErrorStr( "You must call StartDebugLog() in debug versions" );
		return 0;
	} // end if

#endif

	// Sanity check
	if ( !pSocket || !pSocket->Ptr() )
	{	setLastErrorStr( "Invalid socket" );
		return 0;
	} // end if

	// Initialize the session object
	RTMP_Init( &m_session );

	// Mark stream as live
//	m_session.Link.lFlags |= RTMP_LF_LIVE;

	// Set short timeout
//	m_session.Link.timeout = 15;

	// Give the rtmpd object control of the socket handle
	m_session.m_sb.sb_socket = oexPtrToInt( pSocket->Ptr()->Detach() );

	// Disable Nagle's algorithm
	int on = 1;
	setsockopt( m_session.m_sb.sb_socket, IPPROTO_TCP, TCP_NODELAY, (char*)&on, sizeof( on ) );

	// Attempt handshake
	if ( !RTMP_Serve( &m_session ) )
	{	setLastErrorStr( "RTMP handshake failed" );
		return 0;
	} // end if

	return 1;
}
开发者ID:MangoCats，项目名称:winglib，代码行数:48，代码来源:rtmpd_session.cpp


示例12: bzalloc

static void *rtmp_stream_create(obs_data_t *settings, obs_output_t *output)
{
	struct rtmp_stream *stream = bzalloc(sizeof(struct rtmp_stream));
	stream->output = output;
	pthread_mutex_init_value(&stream->packets_mutex);

	RTMP_Init(&stream->rtmp);
	RTMP_LogSetCallback(log_rtmp);
	RTMP_LogSetLevel(RTMP_LOGWARNING);

	if (pthread_mutex_init(&stream->packets_mutex, NULL) != 0)
		goto fail;
	if (os_event_init(&stream->stop_event, OS_EVENT_TYPE_MANUAL) != 0)
		goto fail;

	if (pthread_mutex_init(&stream->write_buf_mutex, NULL) != 0) {
		warn("Failed to initialize write buffer mutex");
		goto fail;
	}

	if (os_event_init(&stream->buffer_space_available_event,
		OS_EVENT_TYPE_AUTO) != 0) {
		warn("Failed to initialize write buffer event");
		goto fail;
	}
	if (os_event_init(&stream->buffer_has_data_event,
		OS_EVENT_TYPE_AUTO) != 0) {
		warn("Failed to initialize data buffer event");
		goto fail;
	}
	if (os_event_init(&stream->socket_available_event,
		OS_EVENT_TYPE_AUTO) != 0) {
		warn("Failed to initialize socket buffer event");
		goto fail;
	}
	if (os_event_init(&stream->send_thread_signaled_exit,
		OS_EVENT_TYPE_MANUAL) != 0) {
		warn("Failed to initialize socket exit event");
		goto fail;
	}

	UNUSED_PARAMETER(settings);
	return stream;

fail:
	rtmp_stream_destroy(stream);
	return NULL;
}
开发者ID:chaturbatecom，项目名称:obs-studio，代码行数:48，代码来源:rtmp-stream.c


示例13: QObject

Rtmp::Rtmp(QUrl url, QObject *parent)
: QObject(parent)
{
    m_rtmp = RTMP_Alloc();
    RTMP_Init(m_rtmp);
    qDebug() << "Connecting to" << url;
    RTMP_SetupURL(m_rtmp, MY_URL );
    RTMP_EnableWrite(m_rtmp);

    RTMP_Connect(m_rtmp, NULL);
    RTMP_ConnectStream(m_rtmp, 0);
    memset(&m_rtmpPacket, 0, sizeof(RTMPPacket));
    qDebug() << RTMP_IsConnected(m_rtmp);



}
开发者ID:AlexSnet，项目名称:RtmpBroadcaster，代码行数:17，代码来源:rtmp.cpp


示例14: gst_rtmp_src_start

/* open the file, do stuff necessary to go to PAUSED state */
static gboolean
gst_rtmp_src_start (GstBaseSrc * basesrc)
{
  GstRTMPSrc *src;
  gchar *uri_copy;

  src = GST_RTMP_SRC (basesrc);

  if (!src->uri) {
    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given"));
    return FALSE;
  }

  src->cur_offset = 0;
  src->last_timestamp = 0;
  src->seekable = TRUE;
  src->discont = TRUE;

  uri_copy = g_strdup (src->uri);
  src->rtmp = RTMP_Alloc ();
  RTMP_Init (src->rtmp);
  if (!RTMP_SetupURL (src->rtmp, uri_copy)) {
    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
        ("Failed to setup URL '%s'", src->uri));
    g_free (uri_copy);
    RTMP_Free (src->rtmp);
    src->rtmp = NULL;
    return FALSE;
  }

  /* open if required */
  if (!RTMP_IsConnected (src->rtmp)) {
    if (!RTMP_Connect (src->rtmp, NULL)) {
      GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
          ("Could not connect to RTMP stream \"%s\" for reading", src->uri));
      RTMP_Free (src->rtmp);
      src->rtmp = NULL;
      return FALSE;
    }
  }

  return TRUE;
}
开发者ID:ylatuya，项目名称:gst-plugins-bad，代码行数:44，代码来源:gstrtmpsrc.c


示例15: fopen

LibRtmp::LibRtmp(bool isNeedLog, bool isNeedRecord)
{
    if (isNeedLog)
    {
        flog_ = fopen("librtmp.log", "w");
        RTMP_LogSetLevel(RTMP_LOGDEBUG2);
        RTMP_LogSetOutput(flog_);
    }
    else
    {
        flog_ = NULL;
    }

    rtmp_ = RTMP_Alloc();
    RTMP_Init(rtmp_);
    RTMP_SetBufferMS(rtmp_, 300);

    streming_url_ = NULL;
    is_need_record_ = isNeedRecord;
}
开发者ID:clzhan，项目名称:RtmpLive_RABDetection，代码行数:20，代码来源:LibRtmp.cpp


示例16: RtmpSessionHandshake

int RtmpSessionHandshake(RTMP_SESSION *pSession)
{
	int iRet = 0;
	RTMPPacket *packet = (RTMPPacket *)malloc(sizeof(RTMPPacket));
	if(NULL == packet)
	{
		
		return ERROR_FAILED;
	}
	RTMPPacket_Init(packet);
	pSession->pPkt = packet;
	
	RTMP *rtmp = RTMP_Alloc(); 
	if(NULL == rtmp )
	{
		return ERROR_FAILED;
	}

	RTMP_Init(rtmp);
	rtmp->m_sb.sb_socket =  pSession->socket;
	pSession->prtmp = rtmp;

	pSession->state = RTMPSERVER_STATE_INIT;
	
	/* 进行握手处理 */
	if(RTMP_Serve(rtmp))
	{
		RTMPPacket_Free(pSession->pPkt);
		RTMPPacket_Init(pSession->pPkt);
		pSession->handshake = 1;
		pSession->arglen = 0;
		SetNonBlocking(pSession->socket);
	}
	else
	{
		iRet = -1;
	}
	
	return iRet;
}
开发者ID:cnkedao，项目名称:rtmpserver_demo，代码行数:40，代码来源:rtmpepollsrv.c


示例17: bzalloc

static void *rtmp_stream_create(obs_data_t *settings, obs_output_t *output)
{
	struct rtmp_stream *stream = bzalloc(sizeof(struct rtmp_stream));
	stream->output = output;
	pthread_mutex_init_value(&stream->packets_mutex);

	RTMP_Init(&stream->rtmp);
	RTMP_LogSetCallback(log_rtmp);
	RTMP_LogSetLevel(RTMP_LOGWARNING);

	if (pthread_mutex_init(&stream->packets_mutex, NULL) != 0)
		goto fail;
	if (os_event_init(&stream->stop_event, OS_EVENT_TYPE_MANUAL) != 0)
		goto fail;

	UNUSED_PARAMETER(settings);
	return stream;

fail:
	rtmp_stream_destroy(stream);
	return NULL;
}
开发者ID:AmoghSubhedar，项目名称:obs-studio，代码行数:22，代码来源:rtmp-stream.c


示例18: TEXT

DWORD WINAPI RTMPPublisher::CreateConnectionThread(RTMPPublisher *publisher)
{
    //------------------------------------------------------
    // set up URL

    bool bRetry = false;
    bool bSuccess = false;
    bool bCanRetry = false;

    String failReason;
    String strBindIP;

    int    serviceID    = AppConfig->GetInt   (TEXT("Publish"), TEXT("Service"));
    String strURL       = AppConfig->GetString(TEXT("Publish"), TEXT("URL"));
    String strPlayPath  = AppConfig->GetString(TEXT("Publish"), TEXT("PlayPath"));

    strURL.KillSpaces();
    strPlayPath.KillSpaces();

    LPSTR lpAnsiURL = NULL, lpAnsiPlaypath = NULL;
    RTMP *rtmp = NULL;

    //--------------------------------
    // unbelievably disgusting hack for elgato devices

    String strOldDirectory;
    UINT dirSize = GetCurrentDirectory(0, 0);
    strOldDirectory.SetLength(dirSize);
    GetCurrentDirectory(dirSize, strOldDirectory.Array());

    OSSetCurrentDirectory(API->GetAppPath());

    //--------------------------------

    if(!strURL.IsValid())
    {
        failReason = TEXT("No server specified to connect to");
        goto end;
    }

    if(serviceID != 0)
    {
        XConfig serverData;
        if(!serverData.Open(TEXT("services.xconfig")))
        {
            failReason = TEXT("Could not open services.xconfig");
            goto end;
        }

        XElement *services = serverData.GetElement(TEXT("services"));
        if(!services)
        {
            failReason = TEXT("Could not find any services in services.xconfig");
            goto end;
        }

        XElement *service = NULL;
        DWORD numServices = services->NumElements();
        for(UINT i=0; i<numServices; i++)
        {
            XElement *curService = services->GetElementByID(i);
            if(curService->GetInt(TEXT("id")) == serviceID)
            {
                service = curService;
                break;
            }
        }

        if(!service)
        {
            failReason = TEXT("Could not find the service specified in services.xconfig");
            goto end;
        }

        XElement *servers = service->GetElement(TEXT("servers"));
        if(!servers)
        {
            failReason = TEXT("Could not find any servers for the service specified in services.xconfig");
            goto end;
        }

        XDataItem *item = servers->GetDataItem(strURL);
        if(!item)
            item = servers->GetDataItemByID(0);

        strURL = item->GetData();

        Log(TEXT("Using RTMP service: %s"), service->GetName());
        Log(TEXT("  Server selection: %s"), strURL.Array());
    }

    //------------------------------------------------------
    // now back to the elgato directory if it needs the directory changed still to function *sigh*

    OSSetCurrentDirectory(strOldDirectory);

    //------------------------------------------------------

    rtmp = RTMP_Alloc();
    RTMP_Init(rtmp);
//.........这里部分代码省略.........
开发者ID:Crimson13，项目名称:OBS，代码行数:101，代码来源:RTMPPublisher.cpp


示例19: main

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

	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 };
	AVal usherToken = { 0, 0 };	//Justin.tv auth token
	int port = -1;
	int protocol = RTMP_PROTOCOL_UNDEFINED;
	int retries = 0;
	int bLiveStream = FALSE;  // is it a live stream? then we can't seek/resume
	int bRealtimeStream = FALSE;  // If true, disable the BUFX hack (be patient)
	int bHashes = FALSE;	  // display byte counters not hashes by default

	long int timeout = DEF_TIMEOUT;	  // timeout connection after 120 seconds
	uint32_t dStartOffset = 0;	  // seek position in non-live mode
	uint32_t dStopOffset = 0;
	RTMP rtmp = { 0 };
	FILE *pLogFile;

	AVal fullUrl = { 0, 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_LOGALL;

	//pLogFile = fopen("log.txt", "w");
	//RTMP_LogSetOutput(pLogFile);

	// Check for --quiet option before printing any output
	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 "2.4"
	RTMP_LogPrintf("RTMPDump %s\n", RTMPDUMP_VERSION);
	RTMP_LogPrintf
	("(c) 2010 Andrej Stepanchuk, Howard Chu, The Flvstreamer Team; license: GPL\n");

	if (!InitSockets())
	{
		RTMP_Log(RTMP_LOGERROR,
				 "Couldn't load sockets support on your platform, exiting!");
		return RD_FAILED;
	}

	/* sleep(30); */

	RTMP_Init(&rtmp);
//.........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: rtmp_playvideo

static event_t *
rtmp_playvideo(const char *url0, media_pipe_t *mp,
	       char *errbuf, size_t errlen,
	       video_queue_t *vq, struct vsource_list *vsl,
	       const video_args_t *va0)
{
  video_args_t va = *va0;
  rtmp_t r = {0};
  event_t *e;
  char *url = mystrdupa(url0);

  mp_set_url(mp, va0->canonical_url, va0->parent_url, va0->parent_title);

  usage_event("Play video", 1, USAGE_SEG("format", "RTMP"));

  prop_set(mp->mp_prop_metadata, "format", PROP_SET_STRING, "RTMP");
  prop_set(mp->mp_prop_root, "loading", PROP_SET_INT, 1);

  va.flags |= BACKEND_VIDEO_NO_FS_SCAN;

  rtmp_log_level = RTMP_LOGINFO;
  RTMP_LogSetLevel(rtmp_log_level);

  r.r = RTMP_Alloc();
  RTMP_Init(r.r, mp->mp_cancellable);

  int64_t start = playinfo_get_restartpos(va.canonical_url, va.title, va.resume_mode);

  if(!RTMP_SetupURL(r.r, url)) {
    snprintf(errbuf, errlen, "Unable to setup RTMP-session");
    rtmp_free(&r);
    return NULL;
  }

  r.r->Link.lFlags |= RTMP_LF_SWFV;

  if(!RTMP_Connect(r.r, NULL, errbuf, errlen, 5000)) {
    rtmp_free(&r);
    return NULL;
  }

  if(!RTMP_ConnectStream(r.r, 0)) {
    snprintf(errbuf, errlen, "Unable to connect RTMP-stream");
    rtmp_free(&r);
    return NULL;
  }

  if(start)
    RTMP_SendSeek(r.r, start);
    
  r.mp = mp;
  
  mp->mp_audio.mq_stream = 0;
  mp->mp_video.mq_stream = 0;

  if(start > 0) {
    r.seekpos_video = start * 1000;
    r.seekpos_audio = start * 1000;
    mp->mp_seek_base = r.seekpos_video;
    mp->mp_video.mq_seektarget = r.seekpos_video;
    mp->mp_audio.mq_seektarget = r.seekpos_video;
  } else {
    mp->mp_video.mq_seektarget = AV_NOPTS_VALUE;
    mp->mp_audio.mq_seektarget = AV_NOPTS_VALUE;
    mp->mp_seek_base = 0;
    r.seekpos_audio = AV_NOPTS_VALUE;
    r.seekpos_video = AV_NOPTS_VALUE;
  }

  mp_configure(mp, MP_CAN_PAUSE, MP_BUFFER_DEEP, 0, "video");
  mp->mp_max_realtime_delay = (r.r->Link.timeout - 1) * 1000000;

  mp_become_primary(mp);

  playinfo_register_play(va.canonical_url, 0);

  r.canonical_url = va.canonical_url;
  r.restartpos_last = -1;

  r.url = url;
  r.va = &va;
  r.is_loading = 1;
  e = rtmp_loop(&r, mp, url, errbuf, errlen);

  if(r.ss)
    sub_scanner_destroy(r.ss);

  if(r.total_duration) {
    int p = mp->mp_seek_base / (r.total_duration * 10);
    if(p >= video_settings.played_threshold) {
      TRACE(TRACE_DEBUG, "RTMP", "Playback reached %d%%, counting as played",
	    p);
      playinfo_register_play(va.canonical_url, 1);
      playinfo_set_restartpos(va.canonical_url, -1, 0);
    } else {
      playinfo_set_restartpos(va.canonical_url, mp->mp_seek_base / 1000, 0);
    }
  }

  mp_shutdown(mp);
//.........这里部分代码省略.........
开发者ID:lprot，项目名称:showtime，代码行数:101，代码来源:rtmp.c



注：本文中的RTMP_Init函数示例由纯净天空整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。

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