[Bug 797280] webrtcbin: video pixelated

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=797280

--- Comment #2 from Dmitriy <degtyaryov at gmail.com> ---
(In reply to Mathieu Duponchelle from comment #1)
> Do you observe high CPU usage with the unmodified version of the webrtc demo?
Intel Core i5-2300 not support vp8 and vp9. Without vaapi used vp8dec/vp8dec
and CPU usage 150% - 250%. Pixelated always.
Intel Core i7-7500U support vp8 and vp9. Without vaapi used vp8dec/vp8dec and
CPU usage 150% - 250%. With vaapi ~10%. Pixelated sometimes.

Intel Core i5-2300 my working computer, everything works quickly, it suits me
for many years and to this day. Watch YouTube, Skype without problems, webrtc
browser to browser no problem.

> Have you tried modifying the media constraints in webrtc.js to a lower resolution?
{ "video": true, "audio": true } - no problem
{ "video": { "width": { "exact": 1280 }, "height": { "exact": 800 } }, "audio":
true } - pixelated

> Pixelated is not a very good description, do you mean chrome lowers the quality it sends or the decoded video is actually corrupted?
The image is degradated and is not restored. I can not see anything.

> If the video is corrupted, that means gstreamer hasn't received a keyframe and should be requesting a new one through RTCP.
According to my observations, this is most likely what happens. In the case of
packet loss, the image is pixelated and immediately restored. Often not
pixelated, I just see in the statistics a packet loss, but the image is good on
CPU Intel Core i7-7500U with vaapi.


> If the quality is lowered, it means chrome has probably detected packet loss (based on the receiver reports sent by gstreamer through RTCP), maybe the statistics are affected when the pipeline struggles to perform decoding in real time.
In chrome video good, no problem. Video pixelated in gstreamer from camera on
chrome. Pixelated when gstreamer <-> gstreamer. I think the problem is between
webrtcbin ! decodebin .

I also described this problem in issee on github
https://github.com/centricular/gstwebrtc-demos/issues/31

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