Rtmp streaming via gstreamer-1.0 appsrc to rtmpsin

I am trying to stream my webcam via rtmp. I tried to stream data via following pipe line:

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'

and it works like a charm. I can see the video on my website.

Then I would like to capture frames firstly, and do some process. I streamed processed data by pushing data into appsrc and streaming through pipeline just like before, but some problems occured.

I cannot see any streaming on my website. Both server side and client doesn't raise any error or warning. Nevertheless, I can still acquire streaming by using:

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

Does anyone have any idea about this?

Here are the snippets of my website part and gstreamer pipeline

gstreamer pipeline:

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");


}

Source of my webpage

<!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>

When using appsrc and appsink people usualy do some stuff with the buffers, sometimes they get data and process them in some way and then create new buffer but forget to timestamp it properly..

What is timestamping? Its appending time information to the audio/video buffers. Why? - its synchronisation mechanism for every application (vlc, web .. ) to display (present) video/audio to user in certain time in certain rate (this is the PTS)..

This has something to do with framerate (in video) or frequency (in audio - but timestamps work differently here - its not per every audio sample which has 4 bytes usually).

So what probably happend on your web side - it recieved buffers but without this timestamping information. So the app didnt know how/when to display the video so it silently failed and displayed nothing.

The GStreamer app worked because it apparently has some algorithms how to guess framerate etc..

As I said you have two options.

1, Calculating your PTS and duration yourselve with:

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, Or by turning do-timestamp on for appsrc which will automatically generate timestamps - now I am not sure how is it doing - it either pick framerate from caps or it generate the PTS according to how you push frames into it.