Interesting, I'll definitely have to try out the new code whenever it is 
ready.  In the mean time, at least the current jitter buffer will be a bit 
more useful.

Regards,
~Scott

On Thursday 18 January 2007 06:51, Philippe Khalaf wrote:
> 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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