AES67 sender pipeline options

[James Cowdery]

I see the RTPs markers are the symptom not the cause of the discontinuities.
I went back my code just before I added rtpbin and it doesn't have the
regular 40ms discontinuities. It also doesn't have the fast preroll and just
starts immediately at 1 packet per ms.
That code plays back OK with an 8ms fudge factor to get initial
timestamp-offset right.
I'll need RTCP at some point but for now I've ditched it for now and work on
getting that clean and robust.

Thanks for advice on setting the payloader timestamp offset. The method I
have working is accurate enough for now but I'll look at these methods to
try and get rid of the fudge factors. From reading SMPTE ST 2110 it seems
the RTP has to be a sample of the audio clock i.e. the SDP offset should be
0. That matches all the streams I've seen.

James



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