Hi gstreamer team, I’ve noticed this commit from Havard Graff to support transport-wide congestion control in Gstreamer https://github.com/GStreamer/gst-plugins-good/commit/1df706448ca2f6116173f879f43d596e026e2dc5.
Is this feature now available in the latest Gstreamer release 1.16.2? I need to build an application to live stream video from Gstreamer to Chrome browser over LTE network. I need to use this congestion control feature for video bitrate adaptation on the Gstreamer
(sender) side. Can anyone confirm if this feature is available now? Thanks yx _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hi Yong, I am following that space too. I think the commit from Havard has been merged but not yet included in a major release. So you might have to recompile yourself. In my understanding this commit is only one part of a bigger system: https://tools.ietf.org/html/draft-ietf-rmcat-gcc-02 GStreamer sends the TWCC reports and it can help the Chrome remote to send you a smooth video. But on the Gstreamer side, you still need to process these reports to compute the available bandwidth as a sender. Does that makes sense? Paascal On May 13 2020, at 9:49 am, Yong Xin <[hidden email]> wrote:
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Thanks Pascal - it makes sense On Tue, May 19, 2020 at 5:00 AM <[hidden email]> wrote:
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Hi Yong, Ok the full congestion control may be a long way to go but if your use case is just to send a video to a Chrome client there are few an other way to do that. Using goog-remb, you have to hack the SDP offer to add the goog-remb option. Then Chrome will start sending you its own the bandwith estimates On GStreamer side you have to parse the rtcp feedback packets and look for REMB packets Otherwise you can also use the webrtcbin statistics to fall-back when you start loosing packets Cheers, Pascal On May 21 2020, at 12:27 am, Yong Xin <[hidden email]> wrote:
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