video interrupted,timestamping issue?

[Katerina Voulgary]

Hello,

the only evidence I have is the interrupted video on the webrtc client 
and it is random.

The processing of the frames is not consistent, so appsrc is not 
producing frames at a stable framerate.

I changed the pipeline to "appsrc 
caps=video/x-raw,format=BGR,format=GST_FORMAT_TIME,do-timestamp=true,stream-type=0,is-live=true", 
I suppose that sets the pts? I also added a videorate element. Still it 
happens.

Would anyone know what could cause these interrupts and how to debug?

Sender pipeline:


         self._videoRGBString = "v4l2src device=/dev/video9 ! 
video/x-raw,format=YUY2, width=1920, height=1080,framerate=15/1 ! 
videoconvert ! queue ! video/x-raw,format=BGR ! appsink "


         self._writerRGBString = "appsrc 
caps=video/x-raw,format=BGR,do-timestamp=true,stream-type=0 \
                                 ! queue ! videoconvert !  videorate ! 
video/x-raw,format=I420,width=1920,height=1080,framerate=15/1  \
                                 ! queue ! omxh264enc profile=baseline 
control-rate=1 qp-range=35,45:35,45:35,45 peak-bitrate=1500000 
preset-level=0 \
                                 ! video/x-h264, 
stream-format=byte-stream, alignment=au, 
width=1920,height=1080,framerate=15/1\
                                 ! queue ! h264parse ! video/x-h264, 
alignment=au, stream-format=byte-stream, 
width=1920,height=1080,framerate=15/1\
                                 ! queue ! rtph264pay config-interval=-1 
! queue ! udpsink "+ self._config['janus']['url_RGB'] +"  \
                                  async=false"

cv2.VideoWriter(self._writerRGBString,0,15.0,(1920,1080))


Receiver pipeline:

     gst-launch-1.0 udpsrc port=8006 ! 
application/x-rtp,encoding-name=H264,payload=96 ! rtpjitterbuffer mode=1 
! rtph264depay ! decodebin ! videoconvert ! fakesink dump=1

Στις 26/1/21 5:31 μ.μ., o/η gotsring έγραψε:
> Just guessing here, but did you try setting buffer timestamps before pushing
> the buffers through the appsrc? Probably just pts is needed, see docs here:
> https://gstreamer.freedesktop.org/documentation/gstreamer/gstbuffer.html?gi-language=python#GstBuffer
>
>
>
>
> --
> Sent from:http://gstreamer-devel.966125.n4.nabble.com/
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