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Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir
Sur d’autres sites (13209)
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Can open RTSP camera stream in FFMPEG but not in Gstreamer rtspsrc : Bad Request (400)
6 août 2022, par Joran ApixaI have a Panasonic WV-SW559 camera set up as an RTSP stream.


VLC can perfectly open the RTSP stream and display it, as well as FFMPEG.
However, when I try to set up a simple gstreamer pipeline, it does not want to open.
I execute the following command :


gst-launch-1.0 rtspsrc --gst-debug=rtspsrc:5 location="rtspt://admin:12345@192.168.2.148:554/MediaInput/h264/stream_1" ! fakesink



after which I get the following output :


0:00:00.063009504 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:8617:gst_rtspsrc_uri_set_uri:<rtspsrc0> parsing URI
0:00:00.063074922 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:8624:gst_rtspsrc_uri_set_uri:<rtspsrc0> configuring URI
0:00:00.063111485 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:8640:gst_rtspsrc_uri_set_uri:<rtspsrc0> set uri: rtspt://admin:12345@192.168.2.148:554/MediaInput/h264/stream_1
0:00:00.063136642 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:8642:gst_rtspsrc_uri_set_uri:<rtspsrc0> request uri is: rtsp://192.168.2.148:554/MediaInput/h264/stream_1
Setting pipeline to PAUSED ...
0:00:00.064752828 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:8391:gst_rtspsrc_start:<rtspsrc0> starting
0:00:00.064910956 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd OPEN
0:00:00.064938405 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:5598:gst_rtspsrc_loop_send_cmd:<rtspsrc0> not interrupting busy cmd unknown
Pipeline is live and does not need PREROLL ...
0:00:00.065145962 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:8346:gst_rtspsrc_thread:<rtspsrc0> got command OPEN
0:00:00.065182682 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 0
0:00:00.065214662 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4614:gst_rtsp_conninfo_connect:<rtspsrc0> creating connection (rtspt://admin:12345@192.168.2.148:554/MediaInput/h264/stream_1)...
Progress: (open) Opening Stream
0:00:00.065652329 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4625:gst_rtsp_conninfo_connect:<rtspsrc0> sanitized uri rtsp://192.168.2.148:554/MediaInput/h264/stream_1
0:00:00.065710611 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4659:gst_rtsp_conninfo_connect:<rtspsrc0> connecting (rtspt://admin:12345@192.168.2.148:554/MediaInput/h264/stream_1)...
Progress: (connect) Connecting to rtspt://admin:12345@192.168.2.148:554/MediaInput/h264/stream_1
0:00:00.081446411 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:7342:gst_rtspsrc_retrieve_sdp:<rtspsrc0> create options... (async)
0:00:00.081494537 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:7351:gst_rtspsrc_retrieve_sdp:<rtspsrc0> send options...
0:00:00.081575581 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:476:default_before_send:<rtspsrc0> default handler
Progress: (open) Retrieving server options
0:00:00.081618707 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:476:default_before_send:<rtspsrc0> default handler
0:00:00.081671521 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:5964:gst_rtspsrc_try_send:<rtspsrc0> sending message
0:00:00.088226524 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:5866:gst_rtsp_src_receive_response:<rtspsrc0> received response message
0:00:00.088280901 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:5885:gst_rtsp_src_receive_response:<rtspsrc0> got response message 400
0:00:00.088321370 23339 0x55cd528a30 WARN rtspsrc gstrtspsrc.c:6161:gst_rtspsrc_send:<rtspsrc0> error: Unhandled error
0:00:00.088335798 23339 0x55cd528a30 WARN rtspsrc gstrtspsrc.c:6161:gst_rtspsrc_send:<rtspsrc0> error: Bad Request (400)
0:00:00.088454915 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:7514:gst_rtspsrc_retrieve_sdp:<rtspsrc0> free connection
0:00:00.088526323 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4715:gst_rtsp_conninfo_close:<rtspsrc0> closing connection...
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Unhandled error
Additional debug info:
gstrtspsrc.c(6161): gst_rtspsrc_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Bad Request (400)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
0:00:00.088648097 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4721:gst_rtsp_conninfo_close:<rtspsrc0> freeing connection...
Setting pipeline to READY ...
0:00:00.088699505 23339 0x55cd528a30 WARN rtspsrc gstrtspsrc.c:7548:gst_rtspsrc_open:<rtspsrc0> can't get sdp
0:00:00.088747891 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:8346:gst_rtspsrc_thread:<rtspsrc0> got command LOOP
0:00:00.088786121 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 0
0:00:00.088812372 23339 0x55cd528a30 WARN rtspsrc gstrtspsrc.c:5628:gst_rtspsrc_loop:<rtspsrc0> we are not connected
0:00:00.088832841 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:5636:gst_rtspsrc_loop:<rtspsrc0> pausing task, reason flushing
0:00:00.088855394 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd WAIT
0:00:00.088885863 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:5585:gst_rtspsrc_loop_send_cmd:<rtspsrc0> cancel previous request LOOP
0:00:00.088905135 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:5593:gst_rtspsrc_loop_send_cmd:<rtspsrc0> connection flush busy LOOP
0:00:00.088923000 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 1
0:00:00.088996595 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd CLOSE
0:00:00.089030346 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:5593:gst_rtspsrc_loop_send_cmd:<rtspsrc0> connection flush busy WAIT
0:00:00.089045971 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 1
0:00:00.089085660 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:8346:gst_rtspsrc_thread:<rtspsrc0> got command CLOSE
0:00:00.089109462 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 0
0:00:00.089129619 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:7569:gst_rtspsrc_close:<rtspsrc0> TEARDOWN...
Setting pipeline to NULL ...
0:00:00.089211288 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:7574:gst_rtspsrc_close:<rtspsrc0> not ready, doing cleanup
0:00:00.089263997 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:7637:gst_rtspsrc_close:<rtspsrc0> closing connection...
0:00:00.089300769 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:8422:gst_rtspsrc_stop:<rtspsrc0> stopping
0:00:00.089330926 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd WAIT
0:00:00.089333374 23339 0x55cd528a30 DEBUG rtspsrc gstrtspsrc.c:2058:gst_rtspsrc_cleanup:<rtspsrc0> cleanup
0:00:00.089354260 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:5593:gst_rtspsrc_loop_send_cmd:<rtspsrc0> connection flush busy CLOSE
0:00:00.089419939 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 1
0:00:00.089493430 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:7569:gst_rtspsrc_close:<rtspsrc0> TEARDOWN...
0:00:00.089520931 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:7574:gst_rtspsrc_close:<rtspsrc0> not ready, doing cleanup
0:00:00.089539212 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:7637:gst_rtspsrc_close:<rtspsrc0> closing connection...
0:00:00.089556192 23339 0x55cd6d9180 DEBUG rtspsrc gstrtspsrc.c:2058:gst_rtspsrc_cleanup:<rtspsrc0> cleanup
Freeing pipeline ...
</rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0></rtspsrc0>


Does anyone have an idea on why this could occur ?


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ffmpef cannot open a simple microsoft wav file exported with Audacity
23 juillet 2013, par sebpiqI have exported a sound file to microsoft wav using Audacity.
I am trying to open this file with ffmpeg :ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg
and here's the ouput I get :
fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
libavutil 52. 18.100 / 52. 18.100
libavcodec 54. 92.100 / 54. 92.100
libavformat 54. 63.104 / 54. 63.104
libavdevice 54. 3.103 / 54. 3.103
libavfilter 3. 42.103 / 3. 42.103
libswscale 2. 2.100 / 2. 2.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 2.100 / 52. 2.100
[dca @ 0x7fd30c013600] Not a valid DCA frame
... SNIP ...
[dca @ 0x7fd5bc013600] Invalid bit allocation index
[dca @ 0x7fd5bc013600] error decoding block
Last message repeated 3 times
[dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
[dca @ 0x7fd5bc013600] error decoding block
[wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
[wav @ 0x7fd5bc013000] decoding for stream 0 failed
[wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
Consider increasing the value for the 'analyzeduration' and 'probesize' options
steps-stereo-16b-44khz.wav: could not find codec parametersIf I export the same file to .ogg or .aiff, no problem, the following works fine :
ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg
Any idea what could be wrong ?
A link to my wav file so you can try to reproduce.
NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.
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Is it possible to use Emgu CV 3.0 to open a live stream with rstp protocol with ffmpeg h264
1er février 2017, par Rezell IsidroIs it possible to use this code in capturing a video stream from an ip camera ?
Capture cap = new Capture("rtsp://192.168.42.1:554/live");
imageBox1.Image = cap.QueryFrame();because my image box is displaying nothing but when i tried viewing it to VLC Media Player the ip address worked. Please help.
I also tried it with VLCPlugin v2 instead of using imageBox and the ip address still work..
I also tried it with iSpy and it worked under ffmpeg(h264), maybe the problem is there ? I’m using Visual Studio Ultimate 2010, Emgu CV 3.x. and I’m using Please help. Been working on this for long now.