Hi.
I have a pcap file which contains mpeg4-generic audio carried in RTP I use the following pipeline gst-launch filesrc location=audio.rtp.pcap ! pcapparse ! "application/x-rtp, payload=96, media=audio, clock-rate=30" ! rtpmp4adepay ! "audio/mpeg, mpegversion=4" ! qtmux ! fakesink Below is the error I get Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstQTMux:qtmux0: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Additional debug info: gstqtmux.c(1815): gst_qt_mux_add_buffer (): /GstPipeline:pipeline0/GstQTMux:qtmux0: format wasn't negotiated before buffer flow on pad audio_00 Execution ended after 300000 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... Am I missing on some caps that needs to be added to the pipeline? Thanks Shiva ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Update
I have tried the following pipeline and I get a similar error gst-launch filesrc location=audio.pcap ! pcapparse ! "application/x-rtp, payload=96, clock-rate=44100, encoding-name=MP4A-LATM, cpresent=0, config=NULL" ! gstrtpjitterbuffer ! rtpmp4adepay ! "audio/mpeg, mpegversion=4, stream-format=raw, channels=2, rate=44100" ! qtmux ! fakesink |
Hi,
try getting the proper playload type and port to set in the pcapparse element. You achieve it easily by opening your dump with Wireshark. Besides, which sw did you use to collect the network dump? And which command? Regards On Fri, Dec 17, 2010 at 2:13 AM, shiva varma <[hidden email]> wrote: > > Update > I have tried the following pipeline and I get a similar error > > gst-launch filesrc location=audio.pcap ! pcapparse ! "application/x-rtp, > payload=96, clock-rate=44100, encoding-name=MP4A-LATM, cpresent=0, > config=NULL" ! gstrtpjitterbuffer ! rtpmp4adepay ! "audio/mpeg, > mpegversion=4, stream-format=raw, channels=2, rate=44100" ! qtmux ! fakesink > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problems-converting-audio-pcap-using-pcapparse-rtpmp4depay-and-qtmux-plugins-tp3090099p3091917.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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