In my search to create this pipeline it would seem to me that I could connect the depay and pay elements together such as:
./test-launch "( udpsrc port=5000 caps=\"application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)H264, payload=\(int\)96, ssrc=\(guint\)2396357661, clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" ! rtph264depay ! rtph264pay name=pay0 pt=96 )"
This doesn't work but it seems like it should. What do I need between the depay and pay to make it fly? Is there an even simpler pipe that would achieve the desired results?
Morris
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Il giorno lun, 28/02/2011 alle 15.43 -0500, Morris Ford ha scritto: > In my search to create this pipeline it would seem to me that I could > connect the depay and pay elements together such as: > > > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp, media= > \(string\)video, clock-rate=\(int\)90000, encoding-name=\(string > \)H264, payload=\(int\)96, ssrc=\(guint\)2396357661, clock-base= > \(guint\)2297066863, seqnum-base=\(guint\)49439\" ! rtph264depay ! > rtph264pay name=pay0 pt=96 )" this should work, please post the debug messages (export GST_DEBUG="*:5") > > > This doesn't work but it seems like it should. What do I need between > the depay and pay to make it fly? Is there an even simpler pipe that > would achieve the desired results? > > > Morris > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
In reply to this post by morrisford
Hi,
On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]> wrote: > In my search to create this pipeline it would seem to me that I could > connect the depay and pay elements together such as: > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp, > media=\(string\)video, clock-rate=\(int\)90000, > encoding-name=\(string\)H264, payload=\(int\)96, ssrc=\(guint\)2396357661, > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" ! rtph264depay > ! rtph264pay name=pay0 pt=96 )" looks like you're missing sink elements. Depending on what you want you mean for "making it fly" it might already be flying or not ;). Regards > This doesn't work but it seems like it should. What do I need between the > depay and pay to make it fly? Is there an even simpler pipe that would > achieve the desired results? > Morris > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > > gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
I did gst-inspect on the pay and depay elements and the depay sources exactly the same thing that the pay sinks so I assumed that the two should talk. My understanding is that the gst-rtsp-server code in test-launch.c has the rest of what is needed to make the pipeline work.
Morris
On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]> wrote: Hi, _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Very late (and short) reply. I'm sorry I didn't remember that you were
using the rtsp server. Could you please post the debug log for the failing cases? Regards On Fri, Mar 4, 2011 at 2:40 PM, Morris Ford <[hidden email]> wrote: > I did gst-inspect on the pay and depay elements and the depay sources > exactly the same thing that the pay sinks so I assumed that the two should > talk. My understanding is that the gst-rtsp-server code in test-launch.c has > the rest of what is needed to make the pipeline work. > Morris > > On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]> wrote: >> >> Hi, >> >> On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]> >> wrote: >> > In my search to create this pipeline it would seem to me that I could >> > connect the depay and pay elements together such as: >> > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp, >> > media=\(string\)video, clock-rate=\(int\)90000, >> > encoding-name=\(string\)H264, payload=\(int\)96, >> > ssrc=\(guint\)2396357661, >> > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" ! >> > rtph264depay >> > ! rtph264pay name=pay0 pt=96 )" >> >> looks like you're missing sink elements. Depending on what you want >> you mean for "making it fly" it might already be flying or not ;). >> >> Regards >> >> > This doesn't work but it seems like it should. What do I need between >> > the >> > depay and pay to make it fly? Is there an even simpler pipe that would >> > achieve the desired results? >> > Morris >> > _______________________________________________ >> > gstreamer-devel mailing list >> > [hidden email] >> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >> > >> > >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > > gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Here is the debug output:
morrisford@ubuntu:~/gst-rtsp-server/gst-rtsp-0.10.7/examples$ ./test-launch --gst-debug=2 "( udpsrc port=5000 caps=\"application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)H264, payload=\(int\)96, ssrc=\(guint\)2396357661\" ! rtph264depay ! rtph264pay name=pay0 pt=96 )"
0:00:11.641343170 5193 0x8667008 WARN rtspmedia rtsp-media.c:806:alloc_udp_ports: multiudpsink version found without send-duplicates property 0:00:11.645912928 5193 0x8667008 WARN bin gstbin.c:2330:gst_bin_do_latency_func:<media-pipeline> failed to query latency
** (lt-test-launch:5193): WARNING **: ignoring stream 0 without media type On Thu, Mar 10, 2011 at 1:28 AM, Marco Ballesio <[hidden email]> wrote: Very late (and short) reply. I'm sorry I didn't remember that you were _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Dear all,
I have the same problem. It is not possible to publish an RTSP stream from an udpsrc without transcoding/reencoding the rtp source stream? Best Regards, Angel
2011/3/10 Morris Ford <[hidden email]> Here is the debug output: -- Ángel Martín Navas _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hombre, Angel!
K tal?
Un abrazo,
Mattias
Sent from my iPhone
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hi to all.
We have solved this issue modifying gstreamer rtsp server: We have a rtsp server which streams video that is received from rtp and saves it simultaneously to a temp file: gst_rtsp_media_factory_set_launch (factory, "gstrtpbin name=rtpbin udpsrc port=5000 ! application/x-rtp,media=video,clock-rate=90000,encoding-name=H264 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay byte-stream=true name=videodepay ! queue ! tee name=videotee ! queue ! rtph264pay name=pay0 pt=96 udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=127.0.0.1 sync=false async=false udpsrc name=audioudpsrc port=5002 ! application/x-rtp, media=(string)audio, clock-rate=(int)22050, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, streamtype=(string)5, profile-level-id=(string)2, mode=(string)AAC-hbr, config=(string)1388, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, ssrc=(uint)3831044409, payload=(int)96, clock-base=(uint)3295317201, seqnum-base=(uint)50289 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpmp4gdepay ! queue ! tee name=audiotee ! queue ! rtpmp4gpay name=pay1 pt=97 audiotee. ! decodebin ! lame ! flvmux name=muxer ! filesink location=/tmp/video.flv videotee. ! queue ! muxer. "); We didn't get the caps when we made the rtph264depay ! rtph264pay name=pay0 pt=96 We solved it hardcoding the rtsp-sdp.c file with our rtp received caps: I mean: if(stream->caps==NULL) { g_warning("NO CAPS"); stream->caps=gst_caps_from_string("application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, ssrc=(uint)3142824526, payload=(int)96, clock-base=(uint)1706805505, seqnum-base=(uint)54578"); } Hope it helps. Regards On Fri, May 13, 2011 at 2:04 PM, MATTIAS BARTHEL <[hidden email]> wrote: > Hombre, Angel! > K tal? > Un abrazo, > Mattias > > Sent from my iPhone > On May 13, 2011, at 13:53, "Angel Martin" <[hidden email]> wrote: > > Dear all, > > I have the same problem. > > It is not possible to publish an RTSP stream from an udpsrc without > transcoding/reencoding the rtp source stream? > > Best Regards, > > Angel > > > 2011/3/10 Morris Ford <[hidden email]> >> >> Here is the debug output: >> morrisford@ubuntu:~/gst-rtsp-server/gst-rtsp-0.10.7/examples$ >> ./test-launch --gst-debug=2 "( udpsrc port=5000 caps=\"application/x-rtp, >> media=\(string\)video, clock-rate=\(int\)90000, >> encoding-name=\(string\)H264, payload=\(int\)96, ssrc=\(guint\)2396357661\" >> ! rtph264depay ! rtph264pay name=pay0 pt=96 )" >> 0:00:11.641343170 5193 0x8667008 WARN rtspmedia >> rtsp-media.c:806:alloc_udp_ports: multiudpsink version found without >> send-duplicates property >> 0:00:11.645912928 5193 0x8667008 WARN bin >> gstbin.c:2330:gst_bin_do_latency_func:<media-pipeline> failed to query >> latency >> ** (lt-test-launch:5193): WARNING **: ignoring stream 0 without media type >> >> On Thu, Mar 10, 2011 at 1:28 AM, Marco Ballesio <[hidden email]> >> wrote: >>> >>> Very late (and short) reply. I'm sorry I didn't remember that you were >>> using the rtsp server. Could you please post the debug log for the >>> failing cases? >>> >>> Regards >>> >>> On Fri, Mar 4, 2011 at 2:40 PM, Morris Ford <[hidden email]> >>> wrote: >>> > I did gst-inspect on the pay and depay elements and the depay sources >>> > exactly the same thing that the pay sinks so I assumed that the two >>> > should >>> > talk. My understanding is that the gst-rtsp-server code in >>> > test-launch.c has >>> > the rest of what is needed to make the pipeline work. >>> > Morris >>> > >>> > On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]> >>> > wrote: >>> >> >>> >> Hi, >>> >> >>> >> On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]> >>> >> wrote: >>> >> > In my search to create this pipeline it would seem to me that I >>> >> > could >>> >> > connect the depay and pay elements together such as: >>> >> > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp, >>> >> > media=\(string\)video, clock-rate=\(int\)90000, >>> >> > encoding-name=\(string\)H264, payload=\(int\)96, >>> >> > ssrc=\(guint\)2396357661, >>> >> > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" ! >>> >> > rtph264depay >>> >> > ! rtph264pay name=pay0 pt=96 )" >>> >> >>> >> looks like you're missing sink elements. Depending on what you want >>> >> you mean for "making it fly" it might already be flying or not ;). >>> >> >>> >> Regards >>> >> >>> >> > This doesn't work but it seems like it should. What do I need >>> >> > between >>> >> > the >>> >> > depay and pay to make it fly? Is there an even simpler pipe that >>> >> > would >>> >> > achieve the desired results? >>> >> > Morris >>> >> > _______________________________________________ >>> >> > gstreamer-devel mailing list >>> >> > [hidden email] >>> >> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >>> >> > >>> >> > >>> >> _______________________________________________ >>> >> gstreamer-devel mailing list >>> >> [hidden email] >>> >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >>> > >>> > >>> > _______________________________________________ >>> > gstreamer-devel mailing list >>> > [hidden email] >>> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >>> > >>> > >>> _______________________________________________ >>> gstreamer-devel mailing list >>> [hidden email] >>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >> >> >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >> > > > > -- > > Ángel Martín Navas > Investigador / Researcher > Televisión Digital y Servicios Multimedia / Digital TV & Multimedia Services > > Vicomtech - Visual Interaction Communication Technologies > > Mikeletegi Pasealekua, 57 - Parque Tecnológico > 20009 Donostia - San Sebastián - Spain > Tel: +[34] 943 30 92 30 > Fax: +[34] 943 30 93 93 > e-mail: [hidden email] > > www.vicomtech.org > > *** member of IK4 Research Alliance **** > www.ik4.es > *** member of GraphicsMedia.net **** > > www.graphicsmedia.net > > > > ----------------------------------------------------- > Vicomtech is an ISO 9001:2000 certified institute > ----------------------------------------------------- > > > Este mensaje se dirige exclusivamente a su destinatario. > La información incluida en el presente correo es confidencial sometida a > secreto profesional, especialmente en lo que respecta a los datos de > carácter personal, cuya divulgación está prohibida, en virtud de la > > legislación vigente. 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