[gst-devel] rtp plugin

[Li Juan GUO]

Hi Flavio,

The following are some examples you have send to me.

gst-launch-0.8 udpsrc multicast_group=239.255.42.42 ! rmpeg12audiodec ! mad
! alsasink
gst-launch-0.8 filesrc location="test.mp3" ! rmpeg12audioenc rtpheader=1 !
udpsink host=239.255.42.42

gst-launch-0.8 udpsrc multicast_group=239.255.42.42 ! rmp3adudec ! mad !
alsasink
gst-launch-0.8 filesrc location="test.mp3" ! rmp3aduenc rtpheader=1 !
udpsink host=239.255.42.42

Could you tell me that rmpeg12audiodec, rmpeg12audioenc,
rmp3adudec,rmp3aduenc,etc are rtp plugins or not? 

In my understanding, rtp is general protocol and codec independent. So I
wonder why to write different plugin for different media format, for
example, rmpeg12audiodec and rmpeg12audioenc for mpeg, rmp3adudec and
rmp3aduenc for mp3.  
Is there any special reason (performance, efficiency  or any other
requirements specific to media format )to choose such method? why cannot
implement rtp plugin in the manner similar to udp plugin, i.e. rtpsrc and
rtpsink? 

If these plugin are rtp plugins, which rtp library do they depend on? I
download code from CVS:
http://cvs.sourceforge.net/viewcvs.py/farsight/gst-plugins/gst/rlive/, and
read some files, such as gstraacaudiodec.c, gstraacaudioenc.c, I donnot find
the disposal related to rtp packet. 

I also read the files in gstreamer,
\gst-plugins-0.8.9\gst\rtp\gstrtpgsmenc.c and gstrtpgsmparse.c. In these two
files, RTP packet disposal is put in the the functions gst_rtpgsmenc_chain
and gst_rtpgsmparse_chain. 
In gstreamer, it also define different plugin for Raw audio and GSM audio,
is it obligatory? 

Thanks a lot.

Best Regards
Lijuan



-----Original Message-----
From: Flavio Oliveira [mailto:flavyobr at yahoo.com.br] 
Sent: Wednesday, June 29, 2005 8:58 PM
To: Li Juan GUO
Subject: Re: [gst-devel] rtp plugin


Hi Juan,

You can try the RLive Plugin, you get it on Farsight
CVS: http://cvs.sourceforge.net/viewcvs.py/farsight/gst-plugins/gst/rlive/

Media Types supported are:

-> MPEG I or II Audio
-> MPEG I or II Video
-> MP3 ADU-Interleaved
-> AMR NB/WB
-> AC3
-> AAC (Support for ADTS Header)

I am sending a file attached with pipelines to test 
each Media Type supported. We are using the UDP Plugin

(GStreamer 0.8), but IS REQUIRED TO CHANGE THE MTU
SIZE
TO 1500. It is the biggest sized IP packet that can 
normally traverse the internet without getting
fragmented, if it is not used, the RLIVE Plugin ISN'T
GOING TO WORK PROPERLY, because formats such as MP3
and AC3 has packets bigger than 1200.  I am sending
the UDP Plugin with the change! Please, let me know if
you have any question.

Best Regards, Flavio

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