hej rene, thanks a lot for your explanation, its basically "my" first pipeline, it comes from some wiki+google+irc so no wonder i had so many mis-pasted bits...anyway im still stuck, i've tried what you suggested and got a segfault: gst-launch-0.10 v4lsrc device=/dev/video0 ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=320,height=240 ! theoraenc bitrate=80 ! oggmux name=mux !  shout2send ip=<a href="http://hackitectura.net">hackitectura.net</a> port=8000 password=XXXXX streamname=<a href="http://son0p.tvdescription=">http://son0p.tv description=</a>"alej00d live transmission" mount=son0p.ogg  alsasrc ! audioconvert ! audioresample ! audio/x-raw-float,width=32,rate=22050,channels=2 ! vorbisenc ! queue ! mux. Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: audioclock0 WARNING: Element "shout2send0" warns: gstbasesink.c(1935): gst_base_sink_chain_unlocked (): /pipeline0/shout2send0: Received buffer without a new-segment. Assuming timestamps start from 0. any other thing i should try to avoid the trap? thanks again, /a <div>On 12/17/06, René Stadler <<a href="mailto:mail@renestadler.de">mail@renestadler.de</a>> wrote:<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Am Sonntag, den 17.12.2006, 02:44 +0100 schrieb alejo d: > im having problems getting a video stream with audio using > gst-launch-0.10. > video goes out well, but when i try to get audio from alsa i get a: > > ERROR: from element /pipeline0/alsasrc0: Internal data flow error. > Additional debug info: > gstbasesrc.c(1614): gst_base_src_loop (): /pipeline0/alsasrc0: > > the pipeline im trying is: > > gst-launch-0.10 v4lsrc device=/dev/video0 ! > "video/x-raw-yuv",width=320,height=240 ! ffmpegcolorspace ! theoraenc > bitrate=80 ! oggmux name=mux ! shout2send ip=<a href="http://son0p.tv"> son0p.tv</a> port=8000 > password=XXXXX streamname=<a href="http://son0p.tv">http://son0p.tv</a> description="alej00d live > transmission" mount=son0p.ogg  alsasrc ! > audio/x-raw-int,width=16,depth=16,rate=44100,channels=2 ! audioconvert > mux. There are several things wrong with this pipeline: - The order of the filtered video caps and the colorspace converter should be swapped. - Same thing for audio: Put the converter before the capsfilter. - The audioconverter is not linked to the muxer, you are missing the "!" between "audioconvert" and "mux.". - oggmux does not accept raw audio. Try something like this (completely untested): gst-launch-0.10 v4lsrc device=/dev/video0 ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=320,height=240 ! theoraenc bitrate=80 ! oggmux name=mux ! shout2send ip=<a href="http://son0p.tv">son0p.tv</a> port=8000 password=XXXXX streamname=<a href="http://son0p.tv">http://son0p.tv</a> description="alej00d live transmission" mount=son0p.ogg  alsasrc ! audioconvert ! audioresample ! audio/x-raw-float,width=32,rate=44100,channels=2 ! vorbisenc ! mux. If you use a different audio encoder, don't forget to adjust the audio caps if it uses a different input format. Regards,   René Stadler </blockquote></div>