[gst-devel] Should RTSPSRC correct for time drift when converting the RTP timestamp to gstreamer clock?

[Edward Hervey]

On Wed, 2011-01-26 at 13:25 -0500, Loren Rogers wrote:
> Hey gang,
> I have a gstreamer program that records audio from an RTP audio
> source.  I then timestamp the audio frame on the recorder based on the
> gstreamer clock (RTSPSRC should have already done the RTP TS
> ->gstreamer clock TS conversion).  I am finding that the time drifts
> (i.e. the most recent gstreamer-clock-based timestamp on this audio
> frame might be increasingly early or late relative to timestamp the
> current recorder thinks it should be) when timestamping the frames end
> to end.  I am wondering if this drift may be linked to the clock drift
> between the audio device and my recording device.  Should the RTSPSRC
> element already be correcting for such time drift when converting the
> RTP timestamp to the gstreamer clock?

  The drift should normally be taken into account (i.e. detecting that
the remote clock is faster or slower than the local gstreamer clock
being used) and compensated in rtspsrc (actually in the jitterbuffer
contained within that element).
  What you might be experiencing though is jitter which is somewhat
compensated but not completely.

  A small test to do to see if it's indeed jitter you're getting to
replace your encoding part with " ! queue ! autoaudiosink". If it plays
back fine (no noise/pops) then it means you're experiencing jitter
(which the audiosinks automatically correct up to 40ms difference).

  If it's jitter indeed, you need to insert a audiorate with a specified
tolerance value before your encoder. Try with values ranging from 10ms
to 40ms until it records chunks of 10seconds without glitches. That's
because encoders expect a perfect incoming audio stream (without any
jitter/gaps/overlaps).

    Edward

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