[gstreamer-bugs] [Bug 633088] New: [rtspsrc] tweak default jitterbuffer buffer mode

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=633088
  GStreamer | gst-plugins-good | git

           Summary: [rtspsrc] tweak default jitterbuffer buffer mode
    Classification: Desktop
           Product: GStreamer
           Version: git
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-good
        AssignedTo: gstreamer-bugs at lists.sourceforge.net
        ReportedBy: mnauw at users.sourceforge.net
         QAContact: gstreamer-bugs at lists.sourceforge.net
      GNOME target: ---
     GNOME version: ---


Created an attachment (id=173159)
 --> (https://bugzilla.gnome.org/attachment.cgi?id=173159)
jitterbuffer slave mode output timestamps log

Default buffer mode passed down by rtspsrc is SLAVE, which works well for
real-time low latency (i.e. live) streams.  However, BUFFER is otherwise
suitable for (non-live) streaming mode.

In particular, when dealing with non-live streams (e.g. youtube stuff), clock
slaving in jitterbuffer does not really make sense and leads to timestamps
distortions in case of bursts.  Attached log shows some jitterbuffer output
timestamps that result from repeatedly pausing/playing some stream; the effect
is that at each play resuming, some amount of time is "lost", which causes the
data to run out well before actual EOS.  In fact, also without repeated
pause/play, the initial part is often lost, leading to shorter playback than
actual duration.

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