[gst-devel] problem sending audio/mpeg over RTP
Antoine Tremblay
hexa00 at gmail.com
Tue Jun 6 23:54:16 CEST 2006
hehe now that I think of it it should work with the default ....but still
it's a good thing to set it just to be sure...
try to run it with --gst-debug-level=5 or any level and check what's
happening juste before the error...
On 6/6/06, Antoine Tremblay <hexa00 at gmail.com> wrote:
>
> Well you're missing some args in udpsink /udpsrc
>
> Should be like udpsink host=localhost port=7777
>
> udpsrc port=7777
>
> the rest seems ok at 1st glance...
>
> Regards
>
> Antoine
>
>
>
> On 6/6/06, Bebjak, Michal <michal.bebjak at siemens.com> wrote:
> >
> >
> > Hi,
> >
> > I'm developing an client-server aplication which should use GStreamer. I
> > want to endcode the audio into MP3 and send it oved RTP to the client. I
> > first tried to run this following commands:
> >
> > server terminal: gst-launch-0.10 -v audiotestsrc ! lame ! rtpmpapay !
> > udpsink
> > client terminal: gst-launch-0.10 -v udpsrc ! 'application/x-rtp,
> > media=audio, payload=96, media=(string)audio, clock-rate=(int)90000,
> > encoding-name=(string)MPA' !
> > rtpmpadepay !
> > 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100' ! mad !
> > audioconvert ! volume volume=0.2 ! autoaudiosink
> >
> >
> > They both work independently but when I try to send the audio from
> > server to client the client crashes. The terminal output is:
> >
> >
> > $ gst-launch-0.10 -v udpsrc ! 'application/x-rtp, media=audio,
> > payload=96, media=(string)audio, clock-rate=(int)90000,
> > encoding-name=(string)MPA' !
> > rtpmpadepay ! 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100' !
> > mad ! audioconvert ! volume volume=0.2 ! autoaudiosink
> > Setting pipeline to PAUSED ...
> > Pipeline is live and does not need PREROLL ...
> > Setting pipeline to PLAYING ...
> > New clock: GstSystemClock
> > /pipeline0/capsfilter0.sink: caps = application/x-rtp,
> > media=(string)audio, payload=(int)96, clock-rate=(int)90000,
> > encoding-name=(string)MPA
> > /pipeline0/capsfilter0.src: caps = application/x-rtp,
> > media=(string)audio, payload=(int)96, clock-rate=(int)90000,
> > encoding-name=(string)MPA
> > /pipeline0/rtpmpadepay0.sink: caps = application/x-rtp,
> > media=(string)audio, payload=(int)96, clock-rate=(int)90000,
> > encoding-name=(string)MPA
> > ERROR: from element /pipeline0/udpsrc0: Internal data flow error.
> > Additional debug info:
> > gstbasesrc.c(1318): gst_base_src_loop (): /pipeline0/udpsrc0:
> > streaming task paused, reason error
> > Execution ended after 1513470000 ns.
> > Setting pipeline to PAUSED ...
> > Setting pipeline to READY ...
> > /pipeline0/rtpmpadepay0.sink: caps = NULL
> > /pipeline0/capsfilter0.sink: caps = NULL
> > /pipeline0/capsfilter0.src: caps = NULL
> > Setting pipeline to NULL ...
> > FREEING pipeline ...
> >
> > Can someone please help me to make it work?? Thanks a lot!!
> >
> > Michael
> >
> >
> > _______________________________________________
> > gstreamer-devel mailing list
> > gstreamer-devel at lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> >
>
>
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