[gstreamer-bugs] [Bug 599292] Synchronizing two RTP streams from different devices using RTCP is inaccurate.

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=599292
  GStreamer | gst-plugins-good | git

--- Comment #12 from Wim Taymans <wim.taymans at gmail.com> 2010-01-21 09:40:01 UTC ---
The patch is similar to yours except that it takes normalized values for the
timestamps and the clock times (starting from 0).

The idea is to capture a pair of running_time (of a buffer using the segment)
and the RTP timestamp in that buffer. The timestamp represents the time when
the buffer was captured and when it should be played back (after adding
latency).

To make the RTCP packet, we sample a pair of pipeline running_time (using the
pipeline clock + base_time) and NTP time (using the system clock). We can
calculate the diff between the last buffer running_time and the pipeline
running_time and add this diff to the last RTP timestamp.

The end result is that we then have the RTP time for the current NTP time,
which is what was needed. 

I think your patch attempted to do that but without normalizing the buffer
timestamps (which can be anything) and the clock absolute times (which can also
be anything). Above patch normalizes those values to work in more situations.

I guess I'll have to actually try the patch then, now :)

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