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    10 avril 2011

    Vous 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, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
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  • Publier sur MédiaSpip

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    Puis-je poster des contenus à partir d’une tablette Ipad ?
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Sur d’autres sites (13209)

  • Can open RTSP camera stream in FFMPEG but not in Gstreamer rtspsrc : Bad Request (400)

    6 août 2022, par Joran Apixa

    I 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&#xA;0:00:00.063074922 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:8624:gst_rtspsrc_uri_set_uri:<rtspsrc0> configuring URI&#xA;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&#xA;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&#xA;Setting pipeline to PAUSED ...&#xA;0:00:00.064752828 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:8391:gst_rtspsrc_start:<rtspsrc0> starting&#xA;0:00:00.064910956 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd OPEN&#xA;0:00:00.064938405 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:5598:gst_rtspsrc_loop_send_cmd:<rtspsrc0> not interrupting busy cmd unknown&#xA;Pipeline is live and does not need PREROLL ...&#xA;0:00:00.065145962 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:8346:gst_rtspsrc_thread:<rtspsrc0> got command OPEN&#xA;0:00:00.065182682 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 0&#xA;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)...&#xA;Progress: (open) Opening Stream&#xA;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&#xA;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)...&#xA;Progress: (connect) Connecting to rtspt://admin:12345@192.168.2.148:554/MediaInput/h264/stream_1&#xA;0:00:00.081446411 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:7342:gst_rtspsrc_retrieve_sdp:<rtspsrc0> create options... (async)&#xA;0:00:00.081494537 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:7351:gst_rtspsrc_retrieve_sdp:<rtspsrc0> send options...&#xA;0:00:00.081575581 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:476:default_before_send:<rtspsrc0> default handler&#xA;Progress: (open) Retrieving server options&#xA;0:00:00.081618707 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:476:default_before_send:<rtspsrc0> default handler&#xA;0:00:00.081671521 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:5964:gst_rtspsrc_try_send:<rtspsrc0> sending message&#xA;0:00:00.088226524 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:5866:gst_rtsp_src_receive_response:<rtspsrc0> received response message&#xA;0:00:00.088280901 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:5885:gst_rtsp_src_receive_response:<rtspsrc0> got response message 400&#xA;0:00:00.088321370 23339   0x55cd528a30 WARN                 rtspsrc gstrtspsrc.c:6161:gst_rtspsrc_send:<rtspsrc0> error: Unhandled error&#xA;0:00:00.088335798 23339   0x55cd528a30 WARN                 rtspsrc gstrtspsrc.c:6161:gst_rtspsrc_send:<rtspsrc0> error: Bad Request (400)&#xA;0:00:00.088454915 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:7514:gst_rtspsrc_retrieve_sdp:<rtspsrc0> free connection&#xA;0:00:00.088526323 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:4715:gst_rtsp_conninfo_close:<rtspsrc0> closing connection...&#xA;ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Unhandled error&#xA;Additional debug info:&#xA;gstrtspsrc.c(6161): gst_rtspsrc_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:&#xA;Bad Request (400)&#xA;ERROR: pipeline doesn&#x27;t want to preroll.&#xA;Setting pipeline to PAUSED ...&#xA;0:00:00.088648097 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:4721:gst_rtsp_conninfo_close:<rtspsrc0> freeing connection...&#xA;Setting pipeline to READY ...&#xA;0:00:00.088699505 23339   0x55cd528a30 WARN                 rtspsrc gstrtspsrc.c:7548:gst_rtspsrc_open:<rtspsrc0> can&#x27;t get sdp&#xA;0:00:00.088747891 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:8346:gst_rtspsrc_thread:<rtspsrc0> got command LOOP&#xA;0:00:00.088786121 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 0&#xA;0:00:00.088812372 23339   0x55cd528a30 WARN                 rtspsrc gstrtspsrc.c:5628:gst_rtspsrc_loop:<rtspsrc0> we are not connected&#xA;0:00:00.088832841 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:5636:gst_rtspsrc_loop:<rtspsrc0> pausing task, reason flushing&#xA;0:00:00.088855394 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd WAIT&#xA;0:00:00.088885863 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:5585:gst_rtspsrc_loop_send_cmd:<rtspsrc0> cancel previous request LOOP&#xA;0:00:00.088905135 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:5593:gst_rtspsrc_loop_send_cmd:<rtspsrc0> connection flush busy LOOP&#xA;0:00:00.088923000 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 1&#xA;0:00:00.088996595 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd CLOSE&#xA;0:00:00.089030346 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:5593:gst_rtspsrc_loop_send_cmd:<rtspsrc0> connection flush busy WAIT&#xA;0:00:00.089045971 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 1&#xA;0:00:00.089085660 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:8346:gst_rtspsrc_thread:<rtspsrc0> got command CLOSE&#xA;0:00:00.089109462 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 0&#xA;0:00:00.089129619 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:7569:gst_rtspsrc_close:<rtspsrc0> TEARDOWN...&#xA;Setting pipeline to NULL ...&#xA;0:00:00.089211288 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:7574:gst_rtspsrc_close:<rtspsrc0> not ready, doing cleanup&#xA;0:00:00.089263997 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:7637:gst_rtspsrc_close:<rtspsrc0> closing connection...&#xA;0:00:00.089300769 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:8422:gst_rtspsrc_stop:<rtspsrc0> stopping&#xA;0:00:00.089330926 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:5567:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd WAIT&#xA;0:00:00.089333374 23339   0x55cd528a30 DEBUG                rtspsrc gstrtspsrc.c:2058:gst_rtspsrc_cleanup:<rtspsrc0> cleanup&#xA;0:00:00.089354260 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:5593:gst_rtspsrc_loop_send_cmd:<rtspsrc0> connection flush busy CLOSE&#xA;0:00:00.089419939 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:4748:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 1&#xA;0:00:00.089493430 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:7569:gst_rtspsrc_close:<rtspsrc0> TEARDOWN...&#xA;0:00:00.089520931 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:7574:gst_rtspsrc_close:<rtspsrc0> not ready, doing cleanup&#xA;0:00:00.089539212 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:7637:gst_rtspsrc_close:<rtspsrc0> closing connection...&#xA;0:00:00.089556192 23339   0x55cd6d9180 DEBUG                rtspsrc gstrtspsrc.c:2058:gst_rtspsrc_cleanup:<rtspsrc0> cleanup&#xA;Freeing pipeline ...&#xA;</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>

    &#xA;

    Does anyone have an idea on why this could occur ?

    &#xA;

  • ffmpef cannot open a simple microsoft wav file exported with Audacity

    23 juillet 2013, par sebpiq

    I 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&#39;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 &#39;analyzeduration&#39; and &#39;probesize&#39; options
    steps-stereo-16b-44khz.wav: could not find codec parameters

    If 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.

  • 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 Isidro

    Is 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.