[Bug 746035] rtpjitterbuffer: buffering mode=slave outputs audio buffers with duplicate pts

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=746035

--- Comment #6 from Sebastian Dröge (slomo) <slomo at coaxion.net> ---
The problem here is that we get a burst of data in the beginning.
The way to handle this is a bit complicated :)

- The jitterbuffer has to check when it first filled up to $latency amount of
RTP timestamps
- If the slope is >> 1.0 then (i.e. we received all these buffers much faster
than real time), the real latency of the jitterbuffer is the time it took to
fill up the buffer and this should be used in the replies to the LATENCY query
- Once the slope goes back to normal, i.e. close to 1.0, we skew once
- Whenever the slope is much bigger than 1.0 (also later), we won't skew until
the slope goes back to normal. Skewing would just completely break everything
:)


As an end result, we would start playback faster (i.e. GStreamer latency is
small) but the sender-to-receiver latency would have the latency value
configured on the jitterbuffer.

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