[gst-devel] caps of rtp stream

[Tomasz Grobelny]

I have such a commandline:

$ LANG=C gst-launch -v udpsrc port=9012 caps="application/x-rtp, 
clock-rate=(int)8000, encoding-name=(string)SPEEX" ! rtpspeexdepay ! 
speexdec ! alsasink

and it works just fine (I get audio in my speakers). But I would like to write 
a filter that would modify raw udp packets. First I wanted to decouple udpsrc 
from setting caps like that:

$ LANG=C gst-launch -v udpsrc port=9012 ! capsfilter caps="application/x-rtp, 
clock-rate=(int)8000, encoding-name=(string)SPEEX" ! rtpspeexdepay ! 
speexdec ! fakesink

but it doesn't work as expected:

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
notification: This doesn't look like a Speex file
ERROR: from element /pipeline0/speexdec0: Could not decode stream.
Additional debug info:
gstspeexdec.c(555): speex_dec_chain_parse_header (): /pipeline0/speexdec0:
couldn't read header
Execution ended after 1122214 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
FREEING pipeline ...

What element (and how) should I use to assign caps to a stream just like caps 
property of udpsrc does? Thanks in advance,
-- 
Regards,
Tomasz Grobelny