GStreaner rtpbin problem
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GStreaner rtpbin problem
 
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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 ========= (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; } _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
smime.p7s (8K)
Re: GStreaner rtpbin problem
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Re: GStreaner rtpbin problem
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Re: GStreaner rtpbin problem
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