[gst-devel] saving rtsp stream tracks
gather bzbz
gbzbz at yahoo.com
Tue Dec 22 10:24:05 CET 2009
Wim,
Thanks so much. I am very new to gstreamer framework. Writing an app sounds "scary" to me.... :(. Any sample out there on rtspsrc video+audio. I will try to give it a shot.....
Meanwhile, is there any way to overwrite or manually correct the audio part in the pipeline script, in case the gst-launch does not do it correctly?
--- On Tue, 12/22/09, Wim Taymans <wim.taymans at gmail.com> wrote:
> From: Wim Taymans <wim.taymans at gmail.com>
> Subject: Re: [gst-devel] saving rtsp stream tracks
> To: "Discussion of the development of GStreamer" <gstreamer-devel at lists.sourceforge.net>
> Date: Tuesday, December 22, 2009, 12:24 AM
> On Mon, 2009-12-21 at 22:54 -0800,
> gather bzbz wrote:
> > 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!.
>
> gst-launch is not smart enough to link this pipeline
> because it
> negotiates a format after creating the pad. You'll have to
> write an
> application.
>
> Wim
>
> >
> > --- 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!!!
> > > >
> > > >
> > > >
> > > >
> > > >
> > >
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