Sync=true Adding 6 seconds latency

[Eric-Stone]

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[Sebastian Dröge-3]

On Tue, 2016-10-25 at 17:52 -0700, Eric-Stone wrote:

> Hi,
>
> I'm working on lowering the latency of RTMP playback with  gst-unity-bridge
> > <https://github.com/ua-i2cat/gst-unity-bridge>  . I've found that by setting
> sync=0, playback is almost instant, but the video drops a LOT of frames.
>
> My question is, how do I either get a lower latency with the sync param
> enabled, or how can I get the sync param to buffer the video less?
>
> Any suggestions/advice would be more than appreciated!

You need to check where exactly the latency comes from. Also having
more details about the actual pipeline that is used in your case would
be useful to be able to do some guesses.

6 seconds sounds a lot though.

--
Sebastian Dröge, Centricular Ltd · http://www.centricular.com
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[rland]

Maybe this post can provide some help?

http://gstreamer-devel.966125.n4.nabble.com/gstplayer-how-to-let-all-of-the-plugins-are-not-cached-and-processing-data-as-soon-as-possible-td4679724.html

[Nicolas Dufresne-4]

Le mardi 25 octobre 2016 à 17:52 -0700, Eric-Stone a écrit :
> I'm working on lowering the latency of RTMP playback with  gst-unity-bridge
> > <https://github.com/ua-i2cat/gst-unity-bridge>  . I've found that by setting
> sync=0, playback is almost instant, but the video drops a LOT of frames.
>
> My question is, how do I either get a lower latency with the sync param
> enabled, or how can I get the sync param to buffer the video less?
>
> Any suggestions/advice would be more than appreciated!

You should describe what you are doing. I see in your code that you
play with pipeline base time, could be your error. Latency depends on
elements you place in your pipeline.

Nicolas
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[Eric-Stone]

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[Nicolas Dufresne-4]

Le jeudi 27 octobre 2016 à 19:54 -0700, Eric-Stone a écrit :
> Would anyone happen to know if it's possible to use rtpjitterbuffer
> with an
> rtmp stream? Or are there any alternatives that might be able to
> smooth the
> playback of this stream?

No, RTMP is not implementing RTP. rtpjitterbuffer works with RTP
packet, by reading the RTP payloaded timing information. Also,
rtpjitterbuffer is for live source, rtmpsrc is not a live source in
GStreamer term.

What you describe here is a mix of network and transmitter jitter. You
need a queue that is as big as the worst "normal" jitter. For playback,
we usually pick something close to 2 or 3 seconds. If you need lower
latency, don't use TCP type of streaming. If that is not possible, you
might want to tweak your transmitter/server to never burst, and enable
google tcp traffic control.

Nicolas
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