Hi,
  As a side note, this whole jitterbuffer is going to be replaced soon
by completely new code that actually works :) But the test app will be
very useful.

Regards,
Philippe


On Thu, 18 Jan 2007 12:20:00 +0200 (EET)
Kai Vehmanen <[EMAIL PROTECTED]> wrote:

> Hi,
> 
> On Thu, 18 Jan 2007, Scott Zuk wrote:
> 
> > I've run into a bug with the jitter buffer implementation in rtpbin that
> > causes the jitter buffer to stop working when there is some packet loss in
> > the rtp stream.  I have attached a simple testcase and a patch that I 
> > believe
> > corrects the problem.
> [...]
> > I believe the problem is in gst_rtp_jitter_buffer_add_to_queue with the 
> > 'late
> > packet' case.  There should be a condition that checks that the timestamp of
> > the newly received packet is actually less than jbuffer->min_ts otherwise 
> > the
> 
> thanks for the patch! I just committed this to the darcs tree.
> 
> > Anyway, please try out my patch, it has some extra debugging code to print 
> > out
> > the data in the queue that probably isn't needed but helped with debugging.
> > Without the patch you will see the jitter buffer get "stuck" when packets 
> > are
> > lost, but with the patch the jitter buffer is able cope with larger gaps in
> > rtp seqnum/timestamp caused by packet loss.
> 
> Hmm, the test app would be a nice addition to the default set of utils 
> built.
> 
> > I also had to fix some pretty
> > evil errors in priv_compare_rtp_seq_lt and priv_delta_rtp_ts.  Half of 16 
> > bit
> > range is 1<<15 not 1<<7 (which is only 128), and half of 32 bit range is
> > 1<<31 not G_MAXUINT16.  Looks like somebody was thinking in base 10 ;)
> 
> Ugh, yes, those are not quite right (either comments or code need fixing, 
> preferably the latter ;)). 128 seqs is still multiple seconds worth of 
> media, so the code does work most of the time, but yeah, a stupid, stupid 
> bug.
> 
> -- 
>   under work: Sofia-SIP at http://sofia-sip.sf.net
> 
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