Rtmp 流通过 gstreamer-1.0 appsrc 到 rtmpsink

Question

我正在尝试通过 rtmp 流式传输我的网络摄像头。我尝试通过以下管道流式传输数据：

gst-launch-1.0 -v v4l2src ! 'video/x-raw, width=640, height=480, framerate=30/1' ! queue ! videoconvert ! omxh264enc ! h264parse ! flvmux ! rtmpsink location='rtmp://{MY_IP}/rtmp/live'

它就像一个魅力。我可以在我的网站上看到视频。

那我想先抓帧，再做一些处理。我像以前一样通过将数据推送到 appsrc 并通过管道流式传输处理过的数据，但是出现了一些问题。

我在我的网站上看不到任何流媒体。服务器端和客户端都不会引发任何错误或警告。尽管如此，我仍然可以通过以下方式获取流媒体：

gst-launch-1.0 rtmpsrc location='rtmp://{MY_IP}/rtmp/live' ! filesink location='rtmpsrca.flv'

有人知道吗？

这是我的网站部分和 gstreamer 管道的片段

gstreamer 管道：

void threadgst(){

    App * app = &s_app; 
    GstCaps *srccap;
    GstCaps * filtercap;
    GstFlowReturn ret;
    GstBus *bus;
    GstElement *pipeline;

    gst_init (NULL,NULL);

    loop = g_main_loop_new (NULL, TRUE);

    //creazione della pipeline:
    pipeline = gst_pipeline_new ("gstreamer-encoder");
    if( ! pipeline ) {
        g_print("Error creating Pipeline, exiting...");
    }

    //creazione elemento appsrc:
    app-> videosrc = gst_element_factory_make ("appsrc", "videosrc");
    if( !  app->videosrc ) {
            g_print( "Error creating source element, exiting...");
    }

    //creazione elemento queue:
    app-> queue = gst_element_factory_make ("queue", "queue");
    if( !  app->queue ) {
            g_print( "Error creating queue element, exiting...");
    }

    app->videocoverter = gst_element_factory_make ("videoconvert", "videocoverter");
    if( ! app->videocoverter ) {
            g_print( "Error creating videocoverter, exiting...");
    }

    //creazione elemento filter:
    app->filter = gst_element_factory_make ("capsfilter", "filter");
    if( ! app->filter ) {
            g_print( "Error creating filter, exiting...");
    }

    app->h264enc = gst_element_factory_make ("omxh264enc", "h264enc");
    if( ! app->h264enc ) {
            g_print( "Error creating omxh264enc, exiting...");
    }

 app->h264parse = gst_element_factory_make ("h264parse", "h264parse");
    if( ! app->h264parse ) {
            g_print( "Error creating h264parse, exiting...");
    }
    app->flvmux = gst_element_factory_make ("flvmux", "flvmux");
    if( ! app->flvmux ) {
            g_print( "Error creating flvmux, exiting...");
    }
    app->rtmpsink = gst_element_factory_make ("rtmpsink", "rtmpsink");
    if( ! app->rtmpsink ) {
            g_print( "Error rtmpsink flvmux, exiting...");
    }



    g_print ("Elements are created\n");
    g_object_set (G_OBJECT (app->rtmpsink), "location" , "rtmp://192.168.3.107/rtmp/live live=1" ,  NULL);
    


    g_print ("end of settings\n");

    srccap = gst_caps_new_simple("video/x-raw",
            "format", G_TYPE_STRING, "RGB",
            "width", G_TYPE_INT, 640,
            "height", G_TYPE_INT, 480,
            //"width", G_TYPE_INT, 320,
            //"height", G_TYPE_INT, 240,
            "framerate", GST_TYPE_FRACTION, 30, 1,
            //"pixel-aspect-ratio", GST_TYPE_FRACTION, 1, 1,
        NULL);

    filtercap = gst_caps_new_simple("video/x-raw",
            "format", G_TYPE_STRING, "I420",
            "width", G_TYPE_INT, 640,
            "height", G_TYPE_INT, 480,
            //"width", G_TYPE_INT, 320,
            //"height", G_TYPE_INT, 240,
            "framerate", GST_TYPE_FRACTION, 30, 1,
        NULL);

    gst_app_src_set_caps(GST_APP_SRC( app->videosrc), srccap);
    g_object_set (G_OBJECT (app->filter), "caps", filtercap, NULL);
    bus = gst_pipeline_get_bus (GST_PIPELINE ( pipeline));
    g_assert(bus);
    gst_bus_add_watch ( bus, (GstBusFunc) bus_call, app);

 gst_bin_add_many (GST_BIN ( pipeline), app-> videosrc, app->queue, app->videocoverter,app->filter, app->h264enc,  app->h264parse, app->flvmux, app->rtmpsink, NULL);

    g_print ("Added all the Elements into the pipeline\n");

    int ok = false;
    ok = gst_element_link_many ( app-> videosrc, app->queue, app->videocoverter, app->filter,app->h264enc,  app->h264parse, app->flvmux, app->rtmpsink, NULL);


    if(ok)g_print ("Linked all the Elements together\n");
    else g_print("*** Linking error ***\n");

    g_assert(app->videosrc);
    g_assert(GST_IS_APP_SRC(app->videosrc));

    g_signal_connect (app->videosrc, "need-data", G_CALLBACK (start_feed), app);
    g_signal_connect (app->videosrc, "enough-data", G_CALLBACK (stop_feed),app);


    g_print ("Playing the video\n");
    gst_element_set_state (pipeline, GST_STATE_PLAYING);

    g_print ("Running...\n");
        g_main_loop_run ( loop);

    g_print ("Returned, stopping playback\n");
    gst_element_set_state (pipeline, GST_STATE_NULL);
    gst_object_unref ( bus);
    g_main_loop_unref (loop);
    g_print ("Deleting pipeline\n");


}

我的网页来源

<!DOCTYPE html>
<meta content="text/html;charset=utf-8" http-equiv="Content-Type">
<meta content="utf-8" http-equiv="encoding">
<html>
<head>
<title>Live Streaming</title>

<!-- strobe -->
<script type="text/javascript" src="strobe/lib/swfobject.js"></script>
<script type="text/javascript">
  var parameters = {  
     src: "rtmp://192.168.3.107/rtmp/live",  
     autoPlay: true,  
     controlBarAutoHide: false,  
     playButtonOverlay: true,  
     showVideoInfoOverlayOnStartUp: true,  
     optimizeBuffering : false,  
     initialBufferTime : 0.1,  
     expandedBufferTime : 0.1,  
     minContinuousPlayback : 0.1,  
     //poster: "images/poster.png"  
  };  
  swfobject.embedSWF(
    "strobe/StrobeMediaPlayback.swf"
    , "StrobeMediaPlayback"
    , 1024
    , 768
    , "10.1.0"
    , "strobe/expressInstall.swf"
    , parameters
    , {
      allowFullScreen: "true"
    }
    , {
      name: "StrobeMediaPlayback"
    }
  );
</script>

</head>
<body>
<div id="StrobeMediaPlayback"></div>
</body>
</html>

Answer 1

当使用 appsrc 和 appsink 时，人们通常会用缓冲区做一些事情，有时他们会获取数据并以某种方式处理它们，然后创建新的缓冲区但忘记正确地为其添加时间戳..

什么是时间戳？它将时间信息附加到 audio/video 缓冲区。为什么？ - 每个应用程序（vlc，web ..）的同步机制在特定时间以特定速率向用户显示（呈现）video/audio（这是 PTS）..

这与帧速率（在视频中）或频率（在音频中 - 但时间戳在这里的工作方式不同 - 它不是每个通常有 4 个字节的音频样本）。

那么您的 Web 端可能发生了什么 - 它收到了缓冲区但没有此时间戳信息。所以应用程序不知道 how/when 显示视频，所以它默默地失败了，什么也不显示。

GStreamer 应用程序之所以有效，是因为它显然有一些算法可以猜测帧率等。

正如我所说，你有两个选择。

1，自己计算 PTS 和持续时间：

guint64 calculated_pts = some_cool_algorithm();
GstBuffer *buffer = gst_buffer_new(data);//your processed data
GST_BUFFER_PTS(buffer) = calculated_pts; // in nanoseconds
GST_BUFFER_DURATION(buffer) = 1234567890; // in nanoseconds
//push buffer to appsrc

2，或者为 appsrc 打开 do-timestamp，这将自动生成时间戳 - 现在我不确定它是怎么做的 - 它要么从上限中选择帧率，要么根据你的推送方式生成 PTS框入其中。

Rtmp 流通过 gstreamer-1.0 appsrc 到 rtmpsink

Rtmp streaming via gstreamer-1.0 appsrc to rtmpsink

c

rtmp

gstreamer