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  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (10689)

  • libavformat/mpegtsenc : allow to set service_type in sdt

    2 février 2015, par dhead666
    libavformat/mpegtsenc : allow to set service_type in sdt
    

    This adds an option to set the service type in mpegts as defined in ETSI 300 468.

    I added what I believe are the most useful service types as pre defined values,
    the others can be sent by using their hexdecimal form directly (e.g. -mpegts_service_type digital_radio, -mpegts_service_type 0x07).

    I’ve been using this patch in order to pipe internet radio stream (originally as HLS/m3u8) from ffmpeg to tvheadend,
    when the service type set right tvheadend recognize the mpegts stream as a radio channel.

    The patch in its original form was written by linuxstb from freenode’s hts channel which allowed me pushing it upstream.

    This close issue 4118.

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] doc/muxers.texi
    • [DH] libavformat/mpegtsenc.c
  • FFMPEG av_read_frame() return an I/O error (-5) ?

    7 mars 2019, par vominhtien961476

    I’m using ffmpeg library to capture screen (desktop). Everything works fine when I run my app in debug/release mode. The I/O error comes when this app run in c# window service mode (win 10 OS). I googled but still don’t know why the error come. In the service mode there is no problem when connecting to some rtsp link (Camera) except desktop capture.

    AVInputFormat * pAVInputFormat = av_find_input_format("gdigrab");
    if (pAVInputFormat != NULL)
    {
      int err = avformat_open_input(&amp;pFormatCtx, "desktop", pAVInputFormat, NULL);
      if(err== 0)
      {
         err = av_read_frame(&amp;pFormatCtx, &amp;packet);
         if (err &lt; 0)
         {
             char errs[AV_ERROR_MAX_STRING_SIZE];
             av_make_error_string(errs, AV_ERROR_MAX_STRING_SIZE, err );
             String^ errStr = gcnew String(errs);
             WriteLog("Error in function av_read_frame() return = " + errStr);
             return -1;
        }
      }
    }

  • ffmpeg rtsp stream to YouTube livestream not doing anything

    29 mai 2018, par felixosth

    I’m using C# to initiate ffmpeg. I’ve a onvif bridge server for a CCTV VMS and I’m building a application to enable the user to livestream any CCTV camera to YouTube.

    The rtsp stream to the camera looks like this :

    rtsp ://onvif:bridge@localhost:554/live/xxxxx-xxxguidtocameraxxx-xxxxx

    I’m new to ffmpeg and I’m using this arg :

    -f lavfi -i anullsrc -rtsp_transport udp -i camerastreamurl -tune zerolatency -vcodec libx264 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp ://x.rtmp.youtube.com/live2/streamkey -loglevel debug

    It seems like ffmpeg isn’t finding the camera stream, only the fake audio one. It just freezes.

    This is the result of the debug log :

    ffmpeg version N-91172-gebf85d3190 Copyright (c) 2000-2018 the FFmpeg developers
     built with gcc 7.3.0 (GCC)
     configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
     libavutil      56. 18.102 / 56. 18.102
     libavcodec     58. 19.104 / 58. 19.104
     libavformat    58. 17.100 / 58. 17.100
     libavdevice    58.  4.100 / 58.  4.100
     libavfilter     7. 24.100 /  7. 24.100
     libswscale      5.  2.100 /  5.  2.100
     libswresample   3.  2.100 /  3.  2.100
     libpostproc    55.  2.100 / 55.  2.100
    Splitting the commandline.
    Reading option '-f' ... matched as option 'f' (force format) with argument 'lavfi'.
    Reading option '-i' ... matched as input url with argument 'anullsrc'.
    Reading option '-rtsp_transport' ... matched as AVOption 'rtsp_transport' with argument 'udp'.
    Reading option '-i' ... matched as input url with argument 'rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b'.
    Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
    Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
    Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument '+'.
    Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.
    Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'aac'.
    Reading option '-strict' ...Routing option strict to both codec and muxer layer
    matched as AVOption 'strict' with argument 'experimental'.
    Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
    Reading option 'rtmp://x.rtmp.youtube.com/live2/xxxxxxxx' ... matched as output url.
    Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
    Finished splitting the commandline.
    Parsing a group of options: global .
    Applying option loglevel (set logging level) with argument debug.
    Successfully parsed a group of options.
    Parsing a group of options: input url anullsrc.
    Applying option f (force format) with argument lavfi.
    Successfully parsed a group of options.
    Opening an input file: anullsrc.
    detected 8 logical cores
    [AVFilterGraph @ 0000027a34bad7c0] query_formats: 2 queried, 3 merged, 0 already done, 0 delayed
    [Parsed_anullsrc_0 @ 0000027a34badb80] sample_rate:44100 channel_layout:'stereo' nb_samples:1024
    [lavfi @ 0000027a34babc80] All info found
    Input #0, lavfi, from 'anullsrc':
     Duration: N/A, start: 0.000000, bitrate: 705 kb/s
       Stream #0:0, 1, 1/44100: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s
    Successfully opened the file.
    Parsing a group of options: input url rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
    Successfully parsed a group of options.
    Opening an input file: rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
    [tcp @ 0000027a34bb5980] No default whitelist set

    Edit :
    I’m getting this result with minimal args :
    ffmpeg result