MPEG video format
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I am having a problem with the output video aspect ratio from mpeg2dec. A sample that illustrates the problem is available at http://www.w6rz.net/parkrun1280_18mbps.ts., although it also happens on live DVB streams. The video is supposed to be 1920x1080, but during caps negotiation, it is offered as 1280x1080. The header appears to have 1280x1080, but aspect code 3, which is 16:9. The result is that after scaling the picture appears roughly 5:4, rather than 16:9. Should the mpeg decoder be putting out the 1920x1080 caps, or should the other elements of the stream be looking at some other caps? ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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On Mon, 2009-03-23 at 20:57 -0700, Dan Taylor wrote:
> I am having a problem with the output video aspect ratio from mpeg2dec. > > A sample that illustrates the problem is available at > http://www.w6rz.net/parkrun1280_18mbps.ts., > although it also happens on live DVB streams. > > The video is supposed to be 1920x1080, but during caps negotiation, it > is offered as 1280x1080. > The header appears to have 1280x1080, but aspect code 3, which is 16:9. > > The result is that after scaling the picture appears roughly 5:4, rather > than 16:9. Here I see the following caps: video/x-raw-yuv, format=(fourcc)I420, width=(int)1280, height=(int)1080, pixel-aspect-ratio=(fraction)3/2, framerate=(fraction)30000/1001 Which is: pixel_width: 1280 pixel_height: 1080 Framerate: 30000/10001 Pixel aspect ratio (PAR) : 3/2 The relationship between display-aspect-ratio (DAR) and PAR is as follows: pixel_width / pixel_height * PAR = DAR We therefore have DAR = 1280 / 1080 * (3/2) = 16/9 Everything is correct (a Display Aspect Ratio of 16:9) Edward > > Should the mpeg decoder be putting out the 1920x1080 caps, or should the > other elements of the > stream be looking at some other caps? > > > ------------------------------------------------------------------------------ > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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Hi,
I would like to stream the video taken from my webcam and the sound taken from my micro. I tried that : To send on Windows : -------------------- gst-launch -v gstrtpbin name=rtpbin \ dshowvideosrc ! decodebin name=dec \ dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X ts-offset=0 name=vrtpsink \ rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X sync=false async=false name=vrtcpsink \ udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ dshowaudiosrc ! queue ! audioresample ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X ts-offset=0 name=artpsink \ rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X sync=false async=false name=artcpsink \ udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 To receive on Linux : --------------------- gst-launch -v gstrtpbin name=rtpbin latency=200 \ udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X sync=false async=false This works well. So I tried to encode with Vorbis and not Alaw. I tried that : To send on Windows : -------------------- gst-launch -v gstrtpbin name=rtpbin \ dshowvideosrc ! decodebin name=dec \ dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X ts-offset=0 name=vrtpsink \ rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X sync=false async=false name=vrtcpsink \ udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ dshowaudiosrc ! queue ! audioresample ! audioconvert ! vorbisenc ! rtpvorbispay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X ts-offset=0 name=artpsink \ rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X sync=false async=false name=artcpsink \ udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 To receive on Linux : --------------------- gst-launch -v gstrtpbin name=rtpbin latency=200 \ udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)VORBIS" port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpvorbisdepay ! vorbisdec ! audioconvert ! audioresample ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X sync=false async=false And I have this error : WARNING: from element /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not decode stream. Additional debug info: gstrtpvorbisdepay.c(605): gst_rtp_vorbis_depay_process (): /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not switch codebooks What can I do to resolve this problem ? Thank you. ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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On Tue, 2009-03-24 at 10:08 +0100, [hidden email] wrote:
> Hi, > > I would like to stream the video taken from my webcam and the sound taken > from my micro. > > I tried that : > > To send on Windows : > -------------------- > > gst-launch -v gstrtpbin name=rtpbin \ > dshowvideosrc ! decodebin name=dec \ > dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! > rtpbin.send_rtp_sink_0 \ > rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X ts-offset=0 > name=vrtpsink \ > rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X sync=false > async=false name=vrtcpsink \ > udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ > dshowaudiosrc ! queue ! audioresample ! audioconvert ! alawenc ! > rtppcmapay ! rtpbin.send_rtp_sink_1 \ > rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X ts-offset=0 > name=artpsink \ > rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X sync=false > async=false name=artcpsink \ > udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 > > > > To receive on Linux : > --------------------- > > gst-launch -v gstrtpbin name=rtpbin latency=200 \ > udpsrc > caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" > port=5000 ! rtpbin.recv_rtp_sink_0 \ > rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ > udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ > rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X > sync=false async=false \ > udpsrc > caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" > port=5002 ! rtpbin.recv_rtp_sink_1 \ > rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! > alsasink \ > udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ > rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X > sync=false async=false > > > This works well. > > So I tried to encode with Vorbis and not Alaw. > > I tried that : > To send on Windows : > -------------------- > > gst-launch -v gstrtpbin name=rtpbin \ > dshowvideosrc ! decodebin name=dec \ > dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! > rtpbin.send_rtp_sink_0 \ > rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X ts-offset=0 > name=vrtpsink \ > rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X sync=false > async=false name=vrtcpsink \ > udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ > dshowaudiosrc ! queue ! audioresample ! audioconvert ! vorbisenc ! > rtpvorbispay ! rtpbin.send_rtp_sink_1 \ > rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X ts-offset=0 > name=artpsink \ > rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X sync=false > async=false name=artcpsink \ > udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 > > > To receive on Linux : > --------------------- > > gst-launch -v gstrtpbin name=rtpbin latency=200 \ > udpsrc > caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" > port=5000 ! rtpbin.recv_rtp_sink_0 \ > rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ > udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ > rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X > sync=false async=false \ > udpsrc > caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)VORBIS" > port=5002 ! rtpbin.recv_rtp_sink_1 \ > rtpbin. ! rtpvorbisdepay ! vorbisdec ! audioconvert ! audioresample > ! alsasink \ > udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ > rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X > sync=false async=false > > > And I have this error : > WARNING: from element > /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not decode > stream. > Additional debug info: > gstrtpvorbisdepay.c(605): gst_rtp_vorbis_depay_process (): > /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: > Could not switch codebooks > Wim > > > What can I do to resolve this problem ? > > Thank you. > > > ------------------------------------------------------------------------------ > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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Can you explain me how to send the vorbis codebook config in the caps
please ? I searched on the Internet but I didn't find how. Thank you for your help. > On Tue, 2009-03-24 at 10:08 +0100, [hidden email] wrote: >> Hi, >> >> I would like to stream the video taken from my webcam and the sound >> taken >> from my micro. >> >> I tried that : >> >> To send on Windows : >> -------------------- >> >> gst-launch -v gstrtpbin name=rtpbin \ >> dshowvideosrc ! decodebin name=dec \ >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! >> rtpbin.send_rtp_sink_0 \ >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >> ts-offset=0 >> name=vrtpsink \ >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >> sync=false >> async=false name=vrtcpsink \ >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! alawenc ! >> rtppcmapay ! rtpbin.send_rtp_sink_1 \ >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >> ts-offset=0 >> name=artpsink \ >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >> sync=false >> async=false name=artcpsink \ >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >> >> >> >> To receive on Linux : >> --------------------- >> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >> udpsrc >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X >> sync=false async=false \ >> udpsrc >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >> rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! >> alsasink \ >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X >> sync=false async=false >> >> >> This works well. >> >> So I tried to encode with Vorbis and not Alaw. >> >> I tried that : >> To send on Windows : >> -------------------- >> >> gst-launch -v gstrtpbin name=rtpbin \ >> dshowvideosrc ! decodebin name=dec \ >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! >> rtpbin.send_rtp_sink_0 \ >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >> ts-offset=0 >> name=vrtpsink \ >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >> sync=false >> async=false name=vrtcpsink \ >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! vorbisenc ! >> rtpvorbispay ! rtpbin.send_rtp_sink_1 \ >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >> ts-offset=0 >> name=artpsink \ >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >> sync=false >> async=false name=artcpsink \ >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >> >> >> To receive on Linux : >> --------------------- >> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >> udpsrc >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X >> sync=false async=false \ >> udpsrc >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)VORBIS" >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >> rtpbin. ! rtpvorbisdepay ! vorbisdec ! audioconvert ! >> audioresample >> ! alsasink \ >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X >> sync=false async=false >> >> >> And I have this error : >> WARNING: from element >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not >> decode >> stream. >> Additional debug info: >> gstrtpvorbisdepay.c(605): gst_rtp_vorbis_depay_process (): >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: >> Could not switch codebooks >> > You forgot to send the vorbis codebook config in the caps. > > Wim >> >> >> What can I do to resolve this problem ? >> >> Thank you. >> >> >> ------------------------------------------------------------------------------ >> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are >> powering Web 2.0 with engaging, cross-platform capabilities. Quickly and >> easily build your RIAs with Flex Builder, the Eclipse(TM)based >> development >> software that enables intelligent coding and step-through debugging. >> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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On Tue, 2009-03-24 at 12:17 +0100, [hidden email] wrote:
> Can you explain me how to send the vorbis codebook config in the caps > please ? You just put them in the caps. Maybe you should read this document that tells you how to negotiate between RTP sender and receiver: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README Wim > > I searched on the Internet but I didn't find how. > > > Thank you for your help. > > > > On Tue, 2009-03-24 at 10:08 +0100, [hidden email] wrote: > >> Hi, > >> > >> I would like to stream the video taken from my webcam and the sound > >> taken > >> from my micro. > >> > >> I tried that : > >> > >> To send on Windows : > >> -------------------- > >> > >> gst-launch -v gstrtpbin name=rtpbin \ > >> dshowvideosrc ! decodebin name=dec \ > >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! > >> rtpbin.send_rtp_sink_0 \ > >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X > >> ts-offset=0 > >> name=vrtpsink \ > >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X > >> sync=false > >> async=false name=vrtcpsink \ > >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ > >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! alawenc ! > >> rtppcmapay ! rtpbin.send_rtp_sink_1 \ > >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X > >> ts-offset=0 > >> name=artpsink \ > >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X > >> sync=false > >> async=false name=artcpsink \ > >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 > >> > >> > >> > >> To receive on Linux : > >> --------------------- > >> > >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ > >> udpsrc > >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" > >> port=5000 ! rtpbin.recv_rtp_sink_0 \ > >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ > >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ > >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X > >> sync=false async=false \ > >> udpsrc > >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" > >> port=5002 ! rtpbin.recv_rtp_sink_1 \ > >> rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! > >> alsasink \ > >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ > >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X > >> sync=false async=false > >> > >> > >> This works well. > >> > >> So I tried to encode with Vorbis and not Alaw. > >> > >> I tried that : > >> To send on Windows : > >> -------------------- > >> > >> gst-launch -v gstrtpbin name=rtpbin \ > >> dshowvideosrc ! decodebin name=dec \ > >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! > >> rtpbin.send_rtp_sink_0 \ > >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X > >> ts-offset=0 > >> name=vrtpsink \ > >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X > >> sync=false > >> async=false name=vrtcpsink \ > >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ > >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! vorbisenc ! > >> rtpvorbispay ! rtpbin.send_rtp_sink_1 \ > >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X > >> ts-offset=0 > >> name=artpsink \ > >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X > >> sync=false > >> async=false name=artcpsink \ > >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 > >> > >> > >> To receive on Linux : > >> --------------------- > >> > >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ > >> udpsrc > >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" > >> port=5000 ! rtpbin.recv_rtp_sink_0 \ > >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ > >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ > >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X > >> sync=false async=false \ > >> udpsrc > >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)VORBIS" > >> port=5002 ! rtpbin.recv_rtp_sink_1 \ > >> rtpbin. ! rtpvorbisdepay ! vorbisdec ! audioconvert ! > >> audioresample > >> ! alsasink \ > >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ > >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X > >> sync=false async=false > >> > >> > >> And I have this error : > >> WARNING: from element > >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not > >> decode > >> stream. > >> Additional debug info: > >> gstrtpvorbisdepay.c(605): gst_rtp_vorbis_depay_process (): > >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: > >> Could not switch codebooks > >> > > You forgot to send the vorbis codebook config in the caps. > > > > Wim > >> > >> > >> What can I do to resolve this problem ? > >> > >> Thank you. > >> > >> > >> ------------------------------------------------------------------------------ > >> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > >> powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > >> easily build your RIAs with Flex Builder, the Eclipse(TM)based > >> development > >> software that enables intelligent coding and step-through debugging. > >> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > >> _______________________________________________ > >> gstreamer-devel mailing list > >> [hidden email] > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------------ > > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > > software that enables intelligent coding and step-through debugging. > > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > > _______________________________________________ > > gstreamer-devel mailing list > > [hidden email] > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------------ > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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In reply to this post by Edward Hervey
Looks like I need to add pixel-aspect-ratio caps handling to fbdevsink.
Thanks. > -----Original Message----- > From: Edward Hervey [mailto:[hidden email]] > Sent: Tuesday, March 24, 2009 1:17 AM > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] MPEG video format > > On Mon, 2009-03-23 at 20:57 -0700, Dan Taylor wrote: > > I am having a problem with the output video aspect ratio > from mpeg2dec. > > > > A sample that illustrates the problem is available at > > http://www.w6rz.net/parkrun1280_18mbps.ts., > > although it also happens on live DVB streams. > > > > The video is supposed to be 1920x1080, but during caps > negotiation, it > > is offered as 1280x1080. > > The header appears to have 1280x1080, but aspect code 3, > which is 16:9. > > > > The result is that after scaling the picture appears > roughly 5:4, rather > > than 16:9. > > Here I see the following caps: video/x-raw-yuv, > format=(fourcc)I420, > width=(int)1280, height=(int)1080, pixel-aspect-ratio=(fraction)3/2, > framerate=(fraction)30000/1001 > > Which is: > pixel_width: 1280 > pixel_height: 1080 > Framerate: 30000/10001 > Pixel aspect ratio (PAR) : 3/2 > > The relationship between display-aspect-ratio (DAR) and PAR is as > follows: > pixel_width / pixel_height * PAR = DAR > > We therefore have DAR = 1280 / 1080 * (3/2) = 16/9 > > Everything is correct (a Display Aspect Ratio of 16:9) > > Edward > > > > > Should the mpeg decoder be putting out the 1920x1080 caps, > or should the > > other elements of the > > stream be looking at some other caps? > > > > > > > -------------------------------------------------------------- > ---------------- > > Apps built with the Adobe(R) Flex(R) framework and Flex > Builder(TM) are > > powering Web 2.0 with engaging, cross-platform > capabilities. Quickly and > > easily build your RIAs with Flex Builder, the > Eclipse(TM)based development > > software that enables intelligent coding and step-through debugging. > > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > > _______________________________________________ > > gstreamer-devel mailing list > > [hidden email] > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > -------------------------------------------------------------- > ---------------- > Apps built with the Adobe(R) Flex(R) framework and Flex > Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. > Quickly and > easily build your RIAs with Flex Builder, the > Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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In reply to this post by Wim Taymans
I read the document, but the use of the caps is still dark for me.
Can somebody help me concretely by explaining me how to resolve my problem ? I also tried to do the same thing with the Speex plugin : - alawenc --> speexenc - rtppcmapay --> rtpspeexpay But after these messages : Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock I suddenly have an exception. > On Tue, 2009-03-24 at 12:17 +0100, [hidden email] wrote: >> Can you explain me how to send the vorbis codebook config in the caps >> please ? > > You just put them in the caps. Maybe you should read this document that > tells you how to negotiate between RTP sender and receiver: > > http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README > > Wim > > >> >> I searched on the Internet but I didn't find how. >> >> >> Thank you for your help. >> >> >> > On Tue, 2009-03-24 at 10:08 +0100, [hidden email] wrote: >> >> Hi, >> >> >> >> I would like to stream the video taken from my webcam and the sound >> >> taken >> >> from my micro. >> >> >> >> I tried that : >> >> >> >> To send on Windows : >> >> -------------------- >> >> >> >> gst-launch -v gstrtpbin name=rtpbin \ >> >> dshowvideosrc ! decodebin name=dec \ >> >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay >> ! >> >> rtpbin.send_rtp_sink_0 \ >> >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >> >> ts-offset=0 >> >> name=vrtpsink \ >> >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >> >> sync=false >> >> async=false name=vrtcpsink \ >> >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >> >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! alawenc ! >> >> rtppcmapay ! rtpbin.send_rtp_sink_1 \ >> >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >> >> ts-offset=0 >> >> name=artpsink \ >> >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >> >> sync=false >> >> async=false name=artcpsink \ >> >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >> >> >> >> >> >> >> >> To receive on Linux : >> >> --------------------- >> >> >> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >> >> udpsrc >> >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >> >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >> >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >> >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >> >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X >> >> sync=false async=false \ >> >> udpsrc >> >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" >> >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >> >> rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! >> audioresample ! >> >> alsasink \ >> >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >> >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X >> >> sync=false async=false >> >> >> >> >> >> This works well. >> >> >> >> So I tried to encode with Vorbis and not Alaw. >> >> >> >> I tried that : >> >> To send on Windows : >> >> -------------------- >> >> >> >> gst-launch -v gstrtpbin name=rtpbin \ >> >> dshowvideosrc ! decodebin name=dec \ >> >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay >> ! >> >> rtpbin.send_rtp_sink_0 \ >> >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >> >> ts-offset=0 >> >> name=vrtpsink \ >> >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >> >> sync=false >> >> async=false name=vrtcpsink \ >> >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >> >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! vorbisenc >> ! >> >> rtpvorbispay ! rtpbin.send_rtp_sink_1 \ >> >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >> >> ts-offset=0 >> >> name=artpsink \ >> >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >> >> sync=false >> >> async=false name=artcpsink \ >> >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >> >> >> >> >> >> To receive on Linux : >> >> --------------------- >> >> >> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >> >> udpsrc >> >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >> >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >> >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >> >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >> >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X >> >> sync=false async=false \ >> >> udpsrc >> >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)VORBIS" >> >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >> >> rtpbin. ! rtpvorbisdepay ! vorbisdec ! audioconvert ! >> >> audioresample >> >> ! alsasink \ >> >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >> >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X >> >> sync=false async=false >> >> >> >> >> >> And I have this error : >> >> WARNING: from element >> >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not >> >> decode >> >> stream. >> >> Additional debug info: >> >> gstrtpvorbisdepay.c(605): gst_rtp_vorbis_depay_process (): >> >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: >> >> Could not switch codebooks >> >> >> > You forgot to send the vorbis codebook config in the caps. >> > >> > Wim >> >> >> >> >> >> What can I do to resolve this problem ? >> >> >> >> Thank you. >> >> >> >> >> >> ------------------------------------------------------------------------------ >> >> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) >> are >> >> powering Web 2.0 with engaging, cross-platform capabilities. Quickly >> and >> >> easily build your RIAs with Flex Builder, the Eclipse(TM)based >> >> development >> >> software that enables intelligent coding and step-through debugging. >> >> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >> >> _______________________________________________ >> >> gstreamer-devel mailing list >> >> [hidden email] >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> > ------------------------------------------------------------------------------ >> > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) >> are >> > powering Web 2.0 with engaging, cross-platform capabilities. Quickly >> and >> > easily build your RIAs with Flex Builder, the Eclipse(TM)based >> development >> > software that enables intelligent coding and step-through debugging. >> > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >> > _______________________________________________ >> > gstreamer-devel mailing list >> > [hidden email] >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> >> >> >> ------------------------------------------------------------------------------ >> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are >> powering Web 2.0 with engaging, cross-platform capabilities. Quickly and >> easily build your RIAs with Flex Builder, the Eclipse(TM)based >> development >> software that enables intelligent coding and step-through debugging. >> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------------ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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I just want to remind that i'm trying to do that on Windows and not on
Linux... Thank you in advance. > I read the document, but the use of the caps is still dark for me. > > Can somebody help me concretely by explaining me how to resolve my problem > ? > > I also tried to do the same thing with the Speex plugin : > - alawenc --> speexenc > - rtppcmapay --> rtpspeexpay > > But after these messages : > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstAudioSrcClock > > I suddenly have an exception. > > > >> On Tue, 2009-03-24 at 12:17 +0100, [hidden email] wrote: >>> Can you explain me how to send the vorbis codebook config in the caps >>> please ? >> >> You just put them in the caps. Maybe you should read this document that >> tells you how to negotiate between RTP sender and receiver: >> >> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README >> >> Wim >> >> >>> >>> I searched on the Internet but I didn't find how. >>> >>> >>> Thank you for your help. >>> >>> >>> > On Tue, 2009-03-24 at 10:08 +0100, [hidden email] wrote: >>> >> Hi, >>> >> >>> >> I would like to stream the video taken from my webcam and the sound >>> >> taken >>> >> from my micro. >>> >> >>> >> I tried that : >>> >> >>> >> To send on Windows : >>> >> -------------------- >>> >> >>> >> gst-launch -v gstrtpbin name=rtpbin \ >>> >> dshowvideosrc ! decodebin name=dec \ >>> >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay >>> ! >>> >> rtpbin.send_rtp_sink_0 \ >>> >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >>> >> ts-offset=0 >>> >> name=vrtpsink \ >>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >>> >> sync=false >>> >> async=false name=vrtcpsink \ >>> >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >>> >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! alawenc ! >>> >> rtppcmapay ! rtpbin.send_rtp_sink_1 \ >>> >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >>> >> ts-offset=0 >>> >> name=artpsink \ >>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >>> >> sync=false >>> >> async=false name=artcpsink \ >>> >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >>> >> >>> >> >>> >> >>> >> To receive on Linux : >>> >> --------------------- >>> >> >>> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >>> >> udpsrc >>> >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >>> >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >>> >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >>> >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X >>> >> sync=false async=false \ >>> >> udpsrc >>> >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" >>> >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >>> >> rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! >>> audioresample ! >>> >> alsasink \ >>> >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X >>> >> sync=false async=false >>> >> >>> >> >>> >> This works well. >>> >> >>> >> So I tried to encode with Vorbis and not Alaw. >>> >> >>> >> I tried that : >>> >> To send on Windows : >>> >> -------------------- >>> >> >>> >> gst-launch -v gstrtpbin name=rtpbin \ >>> >> dshowvideosrc ! decodebin name=dec \ >>> >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay >>> ! >>> >> rtpbin.send_rtp_sink_0 \ >>> >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >>> >> ts-offset=0 >>> >> name=vrtpsink \ >>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >>> >> sync=false >>> >> async=false name=vrtcpsink \ >>> >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >>> >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! vorbisenc >>> ! >>> >> rtpvorbispay ! rtpbin.send_rtp_sink_1 \ >>> >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >>> >> ts-offset=0 >>> >> name=artpsink \ >>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >>> >> sync=false >>> >> async=false name=artcpsink \ >>> >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >>> >> >>> >> >>> >> To receive on Linux : >>> >> --------------------- >>> >> >>> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >>> >> udpsrc >>> >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >>> >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >>> >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >>> >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.X.X >>> >> sync=false async=false \ >>> >> udpsrc >>> >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)VORBIS" >>> >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >>> >> rtpbin. ! rtpvorbisdepay ! vorbisdec ! audioconvert ! >>> >> audioresample >>> >> ! alsasink \ >>> >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.X.X >>> >> sync=false async=false >>> >> >>> >> >>> >> And I have this error : >>> >> WARNING: from element >>> >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not >>> >> decode >>> >> stream. >>> >> Additional debug info: >>> >> gstrtpvorbisdepay.c(605): gst_rtp_vorbis_depay_process (): >>> >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: >>> >> Could not switch codebooks >>> >> >>> > You forgot to send the vorbis codebook config in the caps. >>> > >>> > Wim >>> >> >>> >> >>> >> What can I do to resolve this problem ? >>> >> >>> >> Thank you. >>> >> >>> >> >>> >> ------------------------------------------------------------------------------ >>> >> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) >>> are >>> >> powering Web 2.0 with engaging, cross-platform capabilities. Quickly >>> and >>> >> easily build your RIAs with Flex Builder, the Eclipse(TM)based >>> >> development >>> >> software that enables intelligent coding and step-through debugging. >>> >> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >>> >> _______________________________________________ >>> >> gstreamer-devel mailing list >>> >> [hidden email] >>> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > >>> > >>> > ------------------------------------------------------------------------------ >>> > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) >>> are >>> > powering Web 2.0 with engaging, cross-platform capabilities. Quickly >>> and >>> > easily build your RIAs with Flex Builder, the Eclipse(TM)based >>> development >>> > software that enables intelligent coding and step-through debugging. >>> > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >>> > _______________________________________________ >>> > gstreamer-devel mailing list >>> > [hidden email] >>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are >>> powering Web 2.0 with engaging, cross-platform capabilities. Quickly >>> and >>> easily build your RIAs with Flex Builder, the Eclipse(TM)based >>> development >>> software that enables intelligent coding and step-through debugging. >>> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >>> _______________________________________________ >>> gstreamer-devel mailing list >>> [hidden email] >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are >> powering Web 2.0 with engaging, cross-platform capabilities. Quickly and >> easily build your RIAs with Flex Builder, the Eclipse(TM)based >> development >> software that enables intelligent coding and step-through debugging. >> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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I don't succeed to use the Vorbis or Speex plugin.
Please, may I have some help ? > I just want to remind that i'm trying to do that on Windows and not on > Linux... > > Thank you in advance. > >> I read the document, but the use of the caps is still dark for me. >> >> Can somebody help me concretely by explaining me how to resolve my >> problem >> ? >> >> I also tried to do the same thing with the Speex plugin : >> - alawenc --> speexenc >> - rtppcmapay --> rtpspeexpay >> >> But after these messages : >> Setting pipeline to PAUSED ... >> Pipeline is live and does not need PREROLL ... >> Setting pipeline to PLAYING ... >> New clock: GstAudioSrcClock >> >> I suddenly have an exception. >> >> >> >>> On Tue, 2009-03-24 at 12:17 +0100, [hidden email] wrote: >>>> Can you explain me how to send the vorbis codebook config in the caps >>>> please ? >>> >>> You just put them in the caps. Maybe you should read this document that >>> tells you how to negotiate between RTP sender and receiver: >>> >>> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README >>> >>> Wim >>> >>> >>>> >>>> I searched on the Internet but I didn't find how. >>>> >>>> >>>> Thank you for your help. >>>> >>>> >>>> > On Tue, 2009-03-24 at 10:08 +0100, [hidden email] wrote: >>>> >> Hi, >>>> >> >>>> >> I would like to stream the video taken from my webcam and the sound >>>> >> taken >>>> >> from my micro. >>>> >> >>>> >> I tried that : >>>> >> >>>> >> To send on Windows : >>>> >> -------------------- >>>> >> >>>> >> gst-launch -v gstrtpbin name=rtpbin \ >>>> >> dshowvideosrc ! decodebin name=dec \ >>>> >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! >>>> rtph264pay >>>> ! >>>> >> rtpbin.send_rtp_sink_0 \ >>>> >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >>>> >> ts-offset=0 >>>> >> name=vrtpsink \ >>>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >>>> >> sync=false >>>> >> async=false name=vrtcpsink \ >>>> >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >>>> >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! alawenc >>>> ! >>>> >> rtppcmapay ! rtpbin.send_rtp_sink_1 \ >>>> >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >>>> >> ts-offset=0 >>>> >> name=artpsink \ >>>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >>>> >> sync=false >>>> >> async=false name=artcpsink \ >>>> >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >>>> >> >>>> >> >>>> >> >>>> >> To receive on Linux : >>>> >> --------------------- >>>> >> >>>> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >>>> >> udpsrc >>>> >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >>>> >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >>>> >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >>>> >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >>>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 >>>> host=192.168.X.X >>>> >> sync=false async=false \ >>>> >> udpsrc >>>> >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" >>>> >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >>>> >> rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! >>>> audioresample ! >>>> >> alsasink \ >>>> >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >>>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 >>>> host=192.168.X.X >>>> >> sync=false async=false >>>> >> >>>> >> >>>> >> This works well. >>>> >> >>>> >> So I tried to encode with Vorbis and not Alaw. >>>> >> >>>> >> I tried that : >>>> >> To send on Windows : >>>> >> -------------------- >>>> >> >>>> >> gst-launch -v gstrtpbin name=rtpbin \ >>>> >> dshowvideosrc ! decodebin name=dec \ >>>> >> dec. ! queue ! x264enc byte-stream=true bitrate=300 ! >>>> rtph264pay >>>> ! >>>> >> rtpbin.send_rtp_sink_0 \ >>>> >> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.X.X >>>> >> ts-offset=0 >>>> >> name=vrtpsink \ >>>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.X.X >>>> >> sync=false >>>> >> async=false name=vrtcpsink \ >>>> >> udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ >>>> >> dshowaudiosrc ! queue ! audioresample ! audioconvert ! >>>> vorbisenc >>>> ! >>>> >> rtpvorbispay ! rtpbin.send_rtp_sink_1 \ >>>> >> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.X.X >>>> >> ts-offset=0 >>>> >> name=artpsink \ >>>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.X.X >>>> >> sync=false >>>> >> async=false name=artcpsink \ >>>> >> udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 >>>> >> >>>> >> >>>> >> To receive on Linux : >>>> >> --------------------- >>>> >> >>>> >> gst-launch -v gstrtpbin name=rtpbin latency=200 \ >>>> >> udpsrc >>>> >> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >>>> >> port=5000 ! rtpbin.recv_rtp_sink_0 \ >>>> >> rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ >>>> >> udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ >>>> >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 >>>> host=192.168.X.X >>>> >> sync=false async=false \ >>>> >> udpsrc >>>> >> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)VORBIS" >>>> >> port=5002 ! rtpbin.recv_rtp_sink_1 \ >>>> >> rtpbin. ! rtpvorbisdepay ! vorbisdec ! audioconvert ! >>>> >> audioresample >>>> >> ! alsasink \ >>>> >> udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ >>>> >> rtpbin.send_rtcp_src_1 ! udpsink port=5007 >>>> host=192.168.X.X >>>> >> sync=false async=false >>>> >> >>>> >> >>>> >> And I have this error : >>>> >> WARNING: from element >>>> >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not >>>> >> decode >>>> >> stream. >>>> >> Additional debug info: >>>> >> gstrtpvorbisdepay.c(605): gst_rtp_vorbis_depay_process (): >>>> >> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: >>>> >> Could not switch codebooks >>>> >> >>>> > You forgot to send the vorbis codebook config in the caps. >>>> > >>>> > Wim >>>> >> >>>> >> >>>> >> What can I do to resolve this problem ? >>>> >> >>>> >> Thank you. >>>> >> >>>> >> >>>> >> ------------------------------------------------------------------------------ >>>> >> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) >>>> are >>>> >> powering Web 2.0 with engaging, cross-platform capabilities. >>>> Quickly >>>> and >>>> >> easily build your RIAs with Flex Builder, the Eclipse(TM)based >>>> >> development >>>> >> software that enables intelligent coding and step-through >>>> debugging. >>>> >> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >>>> >> _______________________________________________ >>>> >> gstreamer-devel mailing list >>>> >> [hidden email] >>>> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> > >>>> > >>>> > ------------------------------------------------------------------------------ >>>> > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) >>>> are >>>> > powering Web 2.0 with engaging, cross-platform capabilities. Quickly >>>> and >>>> > easily build your RIAs with Flex Builder, the Eclipse(TM)based >>>> development >>>> > software that enables intelligent coding and step-through debugging. >>>> > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >>>> > _______________________________________________ >>>> > gstreamer-devel mailing list >>>> > [hidden email] >>>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> > >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) >>>> are >>>> powering Web 2.0 with engaging, cross-platform capabilities. Quickly >>>> and >>>> easily build your RIAs with Flex Builder, the Eclipse(TM)based >>>> development >>>> software that enables intelligent coding and step-through debugging. >>>> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> [hidden email] >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> ------------------------------------------------------------------------------ >>> Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are >>> powering Web 2.0 with engaging, cross-platform capabilities. Quickly >>> and >>> easily build your RIAs with Flex Builder, the Eclipse(TM)based >>> development >>> software that enables intelligent coding and step-through debugging. >>> Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com >>> _______________________________________________ >>> gstreamer-devel mailing list >>> [hidden email] >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> > > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------------ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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