Similar as udpsrc but to pass RTP packets directly to gstreamer

[bojan74]

I will continue with this thread to not open new one.
As it was suggested to me I changed udpsrc with appsrc.

In both versions (using udpsrc or appsrc) I am using same input content,
difference is that with:
- udpsrc version - I am sending content to some port and udpsrc read it and
send further ... what ends with filesink where I get mp4 sile
- appsrc version - I am feeding pipeline directly with
g_signal_emit_by_name(appsrc, "push-buffer", buffer, &ret) where I pack
content in gst_buffer

In both cases I get mp4 file with similar size (+-1kB) so it looks that
whole content comes to end. And I can play both files.

But with appsrc version everything is ok till about 55% of file but then
playing of the mp4 file jump to the end. It looks like 45% of video is
deleted or more probably have some problem with headers cause size of both
files is similar and also length in seconds displayed in mp4 player.
In other words if video is 10s length, everything is ok till about 6s, after
that playing jump directly to 10s (end).

First I though that maybe is problem cause I am not feeding pipeline enough
fast but I didn't notice any change if I am calling g_signal_emit_by_name
faster. I also didn't notice any change if I am feeding with more content,
for example if length of video is 1 minute. Also in this case about first
35s will be ok, but then will jump till end.
I will understand that some queues could be full but then I guess they will
be full at same absolute time no matter how long is input. Also if packets
will be dropped then size of file at the end will probably be shorter?
At this case it would be easier to me to investigate where packets are lost,
but in my situation it looks like some timing headers are not correct. But
again what gets wrong after first 55% of file.

Any idea where is the problem, what to check to ...?
Thank you




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