<html> <head> <meta http-equiv="Content-Type" content="text/html; charset=UTF-8"> </head> <body text="#000000" bgcolor="#FFFFFF"> Generally when i have issues where only the first frame gets drawn, it is a time stamping issue. The pipeline will preroll and pull the first frame, but then wait for data to match the current time clock. The buffers you are supplying to the appsrc should be timestamped. The videorate plugin might also work (it works really well with sources that sometimes have timestamps). But when you tell appsrc format=time, it is expecting timestamps. Or at least that is my understanding. In my situation, I have a pipeline that I pass frames to after the main pipeline has started, so I have to adjust each timestamp by an offset (this is on the new-sample callback on my appsink which feeds my appsrc). GST_BUFFER_PTS(buffer) = GST_BUFFER_PTS(buffer) - offset; <div class="moz-cite-prefix">Hopefully this helps!</div> <div class="moz-cite-prefix"> </div> <div class="moz-cite-prefix">On 11/18/2019 7:10 AM, Shishigami wrote: </div> <blockquote type="cite" cite="mid:1574089803584-0.post@n4.nabble.com"> <pre class="moz-quote-pre" wrap="">I'm on Windows 10 using Gstreamer 1.16.1. I'm trying to run two pipelines in parallel. Sender gets data from appsrc (OpenCV) and writes them to UDP sink: appsrc is-live=true format=time ! videoconvert ! video/x-raw, format=I420 ! x264enc speed-preset=superfast tune=zerolatency byte-stream=true key-int-max=15 ! rtph264pay pt=96 config-interval=5 ! udpsink sync=false host=localhost port=8553 Receiver reads from UDP source and uses gst-rtsp-server to provide an RTSP stream: udpsrc address=localhost port=8553 caps="application/x-rtp, clock-rate=90000, payload=96" ! rtpjitterbuffer ! rtph264depay ! rtph264pay name=pay0 pt=96 Opening the stream in VLC media player, the resulting frame from OpenCV does show up, but it never changes. In Unreal Engine (target environment) the stream opens fine, but it only shows a gray image. I'm not sure if this is related, but I set GST_DEBUG and get the following output from gst-rstp-server: 0:00:03.329599300 6124 0000024F68D88700 FIXME rtspmedia rtsp-media.c:4168:gst_rtsp_media_suspend: suspend for dynamic pipelines needs fixing 0:00:03.329990500 6124 0000024F68D88700 WARN rtspmedia rtsp-media.c:4194:gst_rtsp_media_suspend: media 0000024F68DB3220 was not prepared 0:00:03.339047900 6124 0000024F68D88700 FIXME rtspclient rtsp-client.c:1818:handle_play_request:<GstRTSPClient@0000024F68D89850> Add support for seek style (null) 0:00:03.339776300 6124 0000024F68D88700 FIXME rtspmedia rtsp-media.c:2684:gst_rtsp_media_seek_full:<GstRTSPMedia@0000024F68DB3220> Handle going back to 0 for none live not seekable streams. I tried using fpsdisplaysink with the following pipeline (instead of receiver) and that works: gst-launch-1.0 -v udpsrc address=localhost port=8553 caps="application/x-rtp,clock-rate=90000,payload=96" ! rtpjitterbuffer ! rtph264depay ! decodebin ! videoconvert ! fpsdisplaysink But the FPS is pretty bad, averaging at 2.29, which is a far cry from the 30 FPS I'm looking to achieve: last-message = rendered: 240, dropped: 0, current: 2.25, average: 2.29 Why does it work with fpsdisplaysink but not RTSP? I'd highly highly appreciate any help! -- Sent from: <a class="moz-txt-link-freetext" href="http://gstreamer-devel.966125.n4.nabble.com/">http://gstreamer-devel.966125.n4.nabble.com/</a> _______________________________________________ gstreamer-devel mailing list <a class="moz-txt-link-abbreviated" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a class="moz-txt-link-freetext" href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a></pre> </blockquote> </body> </html>