GStreaner rtpbin problem

Sat Jan 26 18:52:02 UTC 2019

Hi

New on this list, hope that this question fits here. I am working to
implement a SCReAM (https://tools.ietf.org/html/rfc8298 ) plugin for
GStreamer. The end goal is to enable congestion control especially for
video. 

Currently I try to make applications for the sender and receiver side. I
have a problem with the code further down below. The error occurs at the
line

"lres = gst_pad_link (srcpad, sinkpad);"

I have tried to double check agaist other examples and I don't find anything
wrong but obviously there must be something ?

Regards
Ingemar

=========

(gscream_app_rx:24376): GLib-GObject-CRITICAL **: 18:40:46.833:
g_object_unref: assertion 'G_IS_OBJECT (object)' failed

**

ERROR:gscream_app_rx.cpp:166:int main(int, char**): assertion failed: (lres
== GST_PAD_LINK_OK)

Aborted (core dumped)

=========

int main (int argc, char *argv[])

{

  std::printf("going for video from IP: %s\n  on port 5000", argv[1]);

  GMainLoop *loop;

  GstElement *rtpbin, *pipeline, *capsfilter, *rtpsrc, *rtcpsrc, *rtcpsink,
*rtpdepay,

    *decodebin, *videoconvert, *autovideosink;

  GstBus *bus;

  guint bus_watch_id;

  GstCaps *udpcaps;

  GstPadLinkReturn lres;

  GstPad *srcpad, *sinkpad;

  /* Initialisation */

  gst_init (&argc, &argv);

  loop = g_main_loop_new (NULL, FALSE);

  /* Create gstreamer elements */

  pipeline        = gst_pipeline_new ("video-pipeline");

  rtpsrc          = gst_element_factory_make("udpsrc",          "rtpsrc");

  udpcaps = gst_caps_new_simple ("application/x-rtp",

    "media", G_TYPE_STRING,"video",

    "clock-rate", G_TYPE_INT, 90000,

    "encoding-name",G_TYPE_STRING,"H264",

    NULL);

  g_object_set(G_OBJECT(rtpsrc), "port", 5000, "caps", udpcaps, NULL);

  gst_object_unref(udpcaps);

  rtcpsrc         = gst_element_factory_make("udpsrc",           "rtcpsrc");

  g_object_set(G_OBJECT(rtcpsrc), "port", 5001, NULL);

  g_assert (rtcpsrc);

  rtcpsink        = gst_element_factory_make("udpsink",
"rtcpsink");

  g_object_set (rtcpsink, "port", 5003, "host", argv[1], NULL);

  /* no need for synchronisation or preroll on the RTCP sink */

  g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);

  g_assert (rtcpsink);

  rtpdepay        = gst_element_factory_make("rtph264depay",
"rtph264depay");

  decodebin       = gst_element_factory_make ("avdec_h264",
"avdec_h264");

  videoconvert    = gst_element_factory_make ("videoconvert",
"videoconvert");

  autovideosink   = gst_element_factory_make ("autovideosink",
"videosink");

  if (!pipeline || !rtpdepay || !decodebin || !videoconvert ||
!autovideosink) {

    g_printerr ("One element could not be created. Exiting.\n");

    return -1;

  }

  /* Set up the pipeline */

  /* we add a message handler */

  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));

  bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);

  gst_object_unref (bus);

  /* we add all elements into the pipeline */

  gst_bin_add_many (GST_BIN (pipeline),

                    rtpsrc, rtpdepay, decodebin, videoconvert,
autovideosink, NULL);

  /* we link the elements together */

  /* videotestsrcm -> autovideosink */

  if(!gst_element_link_many (rtpsrc, rtpdepay, decodebin, videoconvert,
autovideosink, NULL)){

    g_error("Could not link on ore more of elements udpsrc, rtpdepay
decodebin");

    return -1;

  }

  /* the rtpbin element */

  rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");

  g_assert (rtpbin);

  gst_bin_add(GST_BIN (pipeline), rtpbin);

  /* now link all to the rtpbin, start by getting an RTP sinkpad for session
0 */

  srcpad = gst_element_get_static_pad (rtpsrc, "src");

  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");

  lres = gst_pad_link (srcpad, sinkpad);

  g_assert (lres == GST_PAD_LINK_OK);

  gst_object_unref (srcpad);

  /* get an RTCP sinkpad in session 0 */

  srcpad = gst_element_get_static_pad (rtcpsrc, "src");

  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0");

  lres = gst_pad_link (srcpad, sinkpad);

  g_assert (lres == GST_PAD_LINK_OK);

  gst_object_unref (srcpad);

  gst_object_unref (sinkpad);

  /* get an RTCP srcpad for sending RTCP back to the sender */

  srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");

  sinkpad = gst_element_get_static_pad (rtcpsink, "sink");

  lres = gst_pad_link (srcpad, sinkpad);

  g_assert (lres == GST_PAD_LINK_OK);

  gst_object_unref (sinkpad);

  /* the RTP pad that we have to connect to the depayloader will be created

   * dynamically so we connect to the pad-added signal, pass the depayloader
as

   * user_data so that we can link to it. */

  g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb),
rtpdepay);

  /* give some stats when we receive RTCP */

  g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK
(on_ssrc_active_cb),

      rtpdepay);

  //GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL,
"recv-pipeline-bf-playing");

  /* Set the pipeline to "playing" state*/

  g_print ("Now set pipeline in state playing\n");

  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL,
"recv-pipeline-af-playing-bf-running");

  /* Iterate */

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

  g_main_loop_run (loop);

  GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL,
"recv-pipeline-af-running");

  /* Out of the main loop, clean up nicely */

  g_print ("Returned, stopping playback\n");

  gst_element_set_state (pipeline, GST_STATE_NULL);

  GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL,
"recv-pipeline-af-stop");

  g_print ("Deleting pipeline\n");

  gst_object_unref (GST_OBJECT (pipeline));

  g_source_remove (bus_watch_id);

  g_main_loop_unref (loop);

  return 0;

}

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