[gst-devel] saving rtsp stream tracks

gather bzbz gbzbz at yahoo.com
Tue Dec 22 07:54:55 CET 2009


Hi, Aurelien,

Thanks for the hint. Really appreciate it. 

I ran the said pipeline, it failed with the following errors

*****************************************************************

/GstPipeline:pipeline0/GstCapsFilter:capsfilter2: caps = application/x-rtp, media=(string)audio
/GstPipeline:pipeline0/GstCapsFilter:capsfilter3: caps = application/x-rtp, media=(string)video
/GstPipeline:pipeline0/GstCapsFilter:capsfilter3.GstPad:src: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-level-id=(string)41
/GstPipeline:pipeline0/GstCapsFilter:capsfilter3.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-level-id=(string)1
/GstPipeline:pipeline0/GstRTSPSrc:S.GstGhostPad:recv_rtp_src_1_3488930930_97: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-l1
/GstPipeline:pipeline0/GstRTSPSrc:S.GstGhostPad:recv_rtp_src_1_3488930930_97.GstProxyPad:proxypad7: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mod1
/GstPipeline:pipeline0/GstRTSPSrc:S.GstGhostPad:recv_rtp_src_1_3488930930_97: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-l1
/GstPipeline:pipeline0/GstRTSPSrc:S/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_src_1_3488930930_97: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(s1
/GstPipeline:pipeline0/GstRTSPSrc:S/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_src_1_3488930930_97.GstProxyPad:proxypad6: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264,1
/GstPipeline:pipeline0/GstFakeSink:fakesink1.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)97, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-level-id=(string)42801
New clock: GstSystemClock
/GstPipeline:pipeline0/GstCapsFilter:capsfilter4: caps = application/x-rtp, media=(string)audio
/GstPipeline:pipeline0/GstRTSPSrc:S.GstGhostPad:recv_rtp_src_1_520794757_14: caps = application/x-rtp, media=(string)video, payload=(int)14, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-le1
/GstPipeline:pipeline0/GstRTSPSrc:S.GstGhostPad:recv_rtp_src_1_520794757_14.GstProxyPad:proxypad9: caps = application/x-rtp, media=(string)video, payload=(int)14, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode1
/GstPipeline:pipeline0/GstRTSPSrc:S.GstGhostPad:recv_rtp_src_1_520794757_14: caps = application/x-rtp, media=(string)video, payload=(int)14, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-le1
/GstPipeline:pipeline0/GstRTSPSrc:S/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_src_1_520794757_14: caps = application/x-rtp, media=(string)video, payload=(int)14, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(st1
/GstPipeline:pipeline0/GstRTSPSrc:S/GstRtpBin:rtpbin0.GstGhostPad:recv_rtp_src_1_520794757_14.GstProxyPad:proxypad8: caps = application/x-rtp, media=(string)video, payload=(int)14, clock-rate=(int)90000, encoding-name=(string)H264, 1
0:00:02.744148481  1885    0xba050 WARN               basesrc gstbasesrc.c:2334:gst_base_src_loop:<udpsrc4> error: Internal data flow error.
0:00:03.073260000  1885    0xba050 WARN               basesrc gstbasesrc.c:2334:gst_base_src_loop:<udpsrc4> error: streaming task paused, reason not-linked (-1)

*****************************************************************

The strange thing is that, after rtspsrc, the track with payload 14 (according to the rtsp SDP media session) is supposed to have "media=audio", but somehow the rtsp pipe thinks it is "media=video payload=14 encoding-name=H.264). Thus, I think the whole capfilter things are messed up. I am totally lost here. Any pads here for rtspsrc? Please help!.

--- On Mon, 12/21/09, Aurelien Grimaud <gstelzz at yahoo.fr> wrote:

> From: Aurelien Grimaud <gstelzz at yahoo.fr>
> Subject: Re: [gst-devel] saving rtsp stream tracks
> To: "Discussion of the development of GStreamer" <gstreamer-devel at lists.sourceforge.net>
> Date: Monday, December 21, 2009, 12:53 PM
> Hi,
> 
> Le 21/12/2009 08:02, gather bzbz a écrit :
> > Hi, I try to use gstreamer to receive rtsp stream to
> files. The stream contains a track1 for audio and track2 for
> video. VLC works very well with the stream. When I use the
> following command pipeline
> > " gst-launch rtspsrc location=rtsp://<ip> 
> debug=true ! fakesink ", I can see that gstreamer actually
> gets the SDP part right. See following info.
> >
> >    
> Try gst-launch -v rtspsrc name=src src. !
> application/x-rtp, media=audio 
> ! fakesink src. ! application/x-rtp, media=video !
> fakesink
> 
> Aurelien
> > *************************************************
> >   medias:
> >    media 0:
> >     media:   
>    'audio'
> >     port:     
>   '0'
> > 
>    num_ports:   '4294967295'
> >     proto:   
>    'RTP/AVP'
> >     formats:
> >      format  '14'
> >     information: '(NULL)'
> >     connections:
> >      nettype:      'IN'
> >      addrtype: 
>    'IP4'
> >      address:     
> '0.0.0.0'
> >      ttl:       
>   '0'
> >      addr_number:  '0'
> >     key:
> >      type:   
>    '(NULL)'
> >      data:   
>    '(NULL)'
> >     attributes:
> >      attribute 'control' : 'track1'
> >    media 1:
> >     media:   
>    'video'
> >     port:     
>   '0'
> > 
>    num_ports:   '4294967295'
> >     proto:   
>    'RTP/AVP'
> >     formats:
> >      format  '97'
> >     information: '(NULL)'
> >     connections:
> >      nettype:      'IN'
> >      addrtype: 
>    'IP4'
> >      address:     
> '0.0.0.0'
> >      ttl:       
>   '0'
> >      addr_number:  '0'
> >     key:
> >      type:   
>    '(NULL)'
> >      data:   
>    '(NULL)'
> >     attributes:
> >      attribute 'rtpmap' : '97
> H264/90000'
> >      attribute 'fmtp' : '97
> packetization-mode=1;profile-level-id=<num>;sprop-parameter-sets=<str>'
> >      attribute 'control' : 'track2'
> >
> > *************************************************
> > How can I save the audio track1 to a file and video
> track2 to another so I can do some post-processings? Thanks
> a lot for your help!!!
> >
> >
> >
> >
> >
> ------------------------------------------------------------------------------
> > This SF.Net email is sponsored by the Verizon
> Developer Community
> > Take advantage of Verizon's best-in-class app
> development support
> > A streamlined, 14 day to market process makes app
> distribution fast and easy
> > Join now and get one step closer to millions of
> Verizon customers
> > http://p.sf.net/sfu/verizon-dev2dev
> > _______________________________________________
> > gstreamer-devel mailing list
> > gstreamer-devel at lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> >
> >    
> 
> 
> ------------------------------------------------------------------------------
> This SF.Net email is sponsored by the Verizon Developer
> Community
> Take advantage of Verizon's best-in-class app development
> support
> A streamlined, 14 day to market process makes app
> distribution fast and easy
> Join now and get one step closer to millions of Verizon
> customers
> http://p.sf.net/sfu/verizon-dev2dev 
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> 


      




More information about the gstreamer-devel mailing list