[gst-devel] status of rtp packetizers

[Edgard Lima]

I have already change the following plugins so they are compliant with 
the 'todo' proposed.
Since, apply that changes implies in removing the old ones I would like 
know if everybody agree with it and I can commit?

gstrtpg711enc - gstg711payload
gstrtpg711dec - gstg711depayload

gstrtpgsmenc - gstgsmpayload
gstrtpgsmdec - gstgsmdepayload

gstrtpamrenc - gstamrpayload
gstrtpamrdec - gstamrdepayload

gstrtpmpaenc - gstmpapayload
gstrtpmpadec - gstmpadepayload

New element: gstspeexpayload and gstspeexdepayload

[]s
Edgard (alima)
edgard.lima at indt.org.br


Kai Vehmanen wrote:

> Hello all,
>
> and more RTP mails. It seems that there are quite a lot of gst/RTP 
> apps (and elements, especially for session handling) in active 
> development now. There's at least Flumotion, Farsight, Tapioca, Sofsip 
> (console mode SIP VoIP client using gst/rtpbin/jrtp, this what I'm 
> personally working with), the Xeris RTP/RTSP server, ... any others?
>
> But I guess we all agree that nobody should be making their own 
> versions of packetizers for codecs (and hopefully we can agree upon 
> session handling, muxing, etc later on). So to that end, here's some 
> material for comments. I've been talking with people from various 
> group (especially Philippe, Andre and Zeeshan) about the status of the 
> (de)payloaders, and here's a list of guidelines and/or open items. Of 
> course, Wim has designed the current base classes, so hopefully this 
> post is inline with the original design. Please comment if you are 
> working on this area, or there is something clearly wrong:
>
> generic todo
> ------------
>
> - updating the base classes
> - porting the current in-tree packetizers to the guidelines (especially
>   to use the base-classes)
> - renaming of the packetizers in gst-plugins-good-cvs/gst/rtp/ (I've
>   understood that this has been ok'ed on #gstreamer some time ok)
>     - when's a good time to do this from 1.0 pov...?
>
> payloaders
> ----------
>
> - should implement the GstBaseRTPPayload base class
>     - use gst_basertppayload_push() et al
>     - OPEN: harmonize the way the subclasses are implemented (setting
>       clock rate, caps, etc)
> - naming
>     - CODECpayload (=> "gsmpayload", "gstgsmpayload.c")
> - SDP
>     - static caps for src pads should be defined according
>       to gst-plugins-good-cvs/gst/rtp/README
>     - note: payloader should set the PT in the caps
>       if fixed (e.g. PT=8 for PCMA)
>
> depayloaders
> ------------
>
> - implement the GstBaseRTPDepayload base class
>     - setting the clock_rate, PT, ... (see above)
>     - OPEN: harmonize the way the subclasses are implemented (setting
>       clock rate, caps, etc)
> - SDP
>     - static caps for sink pads should be defined according to
>       gst-plugins-good-cvs/gst/rtp/README
>     - the sink pad caps should contain at least 'media',
>       'clock-rate' and 'encoding-name'
>     - note: depayloaders should accept all PT's (even if the
>       codec in question has a fixed PT allocated by its payload RFC)
> - naming
>     - CODECdepayload (=> "gsmdepayload", gstgsmdepayload.c)
> - base-class todo
>     - improve the jitter buffer (separate queue-delay, and
>       max-size params that cover overflows both in queue_release
>       and in add_to_queue)
>     - similar services as gst_basertppayload_set_outcaps() for
>       payloaders for setting caps
>
> -- 
>  under work: Sofia-SIP at http://sofia-sip.sf.net
>
>
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