[Bug 794544] rtpbuffer: expose gst_rtp_buffer_initialize_header function

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=794544

--- Comment #7 from Olivier Crête <olivier.crete at ocrete.ca> ---
(In reply to Mathieu Duponchelle from comment #4)
> The use case for this is:
> 
> alawenc ! rtppcmapay , where the payloader provides a buffer pool, creating
> buffers with gst_buffer_new_allocate , with a single memory, then setting
> the memory offset to header_len, which means the encoder will write right
> after the header and no allocation of GstMemory will occur at gst_buffer_map
> time, gst_rtp_buffer_new_allocate doesn't really work for this.

I don't understand why it doesn't work? You can do
gst_buffer_new_allocate(payload_len), then do
gst_buffer_resize(offset=header_len), and then if the buffer comes back as
expected, then you can just size it back?

Are you planning to do this in GstRTPBaseAudioPayload ?


(In reply to Sebastian Dröge (slomo) from comment #6)
> (In reply to Olivier Crête from comment #3)
> > As we required to header to be in only one memory block, the header layout
> > is fixed.
> 
> I also thought this first but it's not true fortunately.

In which case is it not true?

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