[gst-devel] problem sending audio/mpeg over RTP

Antoine Tremblay hexa00 at gmail.com
Tue Jun 6 23:51:02 CEST 2006


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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20060606/e7a04829/attachment.htm>


More information about the gstreamer-devel mailing list