[Bug 644404] [rtspsrc/jitterbuffer] position increasing while buffering

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=644404
  GStreamer | gst-plugins-good | git

--- Comment #17 from Mark Nauwelaerts <mnauw at users.sourceforge.net> 2011-12-08 11:59:56 UTC ---
So finally got around to this latest version, which seems to make it all Just
Work without messing with buffer timestamps (and corresponding disadvantages)
and has mostly advantages (e.g. rendering starts/stops along with buffering,
and position reporting is kept properly).

As before, disadvantage is some event'ing, custom for now but could be made
more standard of course.  Only real disadvantage appears to be that "live
aware" elements can no longer assume that a passing buffer will be rendered at
its running_time, but that assumption was already broken when latency was
introduced (and this adds a variable latency to the setting, which buffering
and/or underlying network delays basically are).

As such, if needed, or for completeness, the "variable latency" could also be
reported upstream (as is latency currently) with some event (such as extending
the current latency event with some field or whatever) for the benefit of
whatever interested upstream element.

All in all, while maybe not the very cleanest, it is still pretty clean and
likely simplier than e.g. trying to squeeze in some BUFFERING state (between
PAUSED and PLAYING) and playing around with (re)distributing updated base_time
in between transitions.

0.11 might be a good time to standardize some of this, though maybe already
possible in 0.10 ...

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