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  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

  • Gestion des droits de création et d’édition des objets

    8 février 2011, par

    Par défaut, beaucoup de fonctionnalités sont limitées aux administrateurs mais restent configurables indépendamment pour modifier leur statut minimal d’utilisation notamment : la rédaction de contenus sur le site modifiables dans la gestion des templates de formulaires ; l’ajout de notes aux articles ; l’ajout de légendes et d’annotations sur les images ;

Sur d’autres sites (9064)

  • Is it possible to encode an audio stream with the exact length (to millisecond) ?

    27 avril 2019, par v010dya

    I am looking to create an output file with a length exactly 30 seconds in either mp3, aac, or vorbis codecs. I have used avconv (and also tried ffmpeg) with the following command :

    avconv  -i input_file_here -c:a aac -b:a 320k -t 30 out.mp4

    This produces the output, stating that it has encoded exactly 30 seconds, but then :

    $ avprobe out.mp4
    avprobe version v13_dev0-1648-gc4642788e, Copyright (c) 2007-2018 the Libav developers
     built on Apr 24 2019 15:49:35 with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'out.mp4':
     Metadata:
       major_brand     : isom
       minor_version   : 512
       compatible_brands: isomiso2mp41
       encoder         : Lavf57.83.100
     Duration: 00:00:30.02, start: 0.023220, bitrate: 323 kb/s
       Stream #0:0(und): Audio: aac (LC) [mp4a / 0x6134706D]
         44100 Hz, stereo, fltp, 321 kb/s (default)
       Metadata:
         handler_name    : SoundHandler
    # avprobe output

    When i encode in mp3 it correctly displays that it does more than 30 seconds during the encoding. When i try vorbis it does slightly less than 30 seconds.

    Is it possible to achieve the length of 30 seconds to the millisecond using any tool (i prefer either avconv).

  • av_interleaved_write_frame() : Connection reset by peer - Using ffmpeg connected to local nginx

    10 avril 2019, par haxpor

    I have a local nginx web server installed with nginx-rtmp-module.

    I have 2 RTMP sources that are fed into local network at rtmp://127.0.0.1/live-video, and rtmp://127.0.0.1/live-audio. Then these twos will be combined and fed into rtmp://127.0.0.1/live again. The latter one will be the one that I will be using.

    The reason I did just that is I want to do noise reduction via sox.

    The setup is as follows

    1. rtmp://127.0.0.1/live-video - it uses ffmpeg to capture only screen (video) then output to such URL with command

      ffmpeg -analyzeduration 0 -video_size 1280x1080 -framerate 25 -f x11grab -i :0.0 -vcodec libx264 -flags +global_header -preset ultrafast -minrate 7200 -maxrate 8k -vsync 1 -f flv -metadata streamName=ZombieHeroLiveStream rtmp://127.0.0.1/live-video
    2. rtmp://127.0.0.1/live-audio - it uses sox to do noise reduction then uses ffmpeg to feed into such URL with command

      ffmpeg -f alsa -ac 1 -i default -preset ultrafast -f flac - | sox - -p noisered /tmp/noise.prof 0.21 | ffmpeg -analyzeduration 0 -thread_queue_size 2 -i - -f flv -preset ultrafast rtmp://127.0.0.1/live-audio
    3. rtmp://127.0.0.1/live - it uses ffmpeg to combine the above two sources then output into such URL, this is the one I tested viewing via VLC (which is works fine) with command

      ffmpeg -analyzeduration 0 -thread_queue_size 512 -rtmp_live live -flags +global_header -i "rtmp://127.0.0.1/live-video" -analyzeduration 0 -thread_queue_size 512 -rtmp_live live -flags +global_header -i "rtmp://127.0.0.1/live-audio" -c:v copy -c:a copy -map 0:v -map 1:a -async 1 -vsync 1 -video_size 1280x1080 -fflags +genpts -framerate 30 -preset ultrafast -minrate 7200 -maxrate 8k -flags +global_header -segment_list_flags +live -rtmp_live live -f flv -metadata streamName=ZombieHeroLiveStream "rtmp://127.0.0.1/live"

    The setup works fine, I tested and confirmed viewing via VLC targeting to URL of 3. The problem is that for some times after all threes above up and running, 1. will exit and thus make the whole encoding stop. The error shown is

    av_interleaved_write_frame(): Connection reset by peer29.88 bitrate=2021.8kbits/s dup=1791 drop=0 speed=0.998x    
       Last message repeated 2 times
    ...

    Error writing trailer of rtmp://127.0.0.1/live-video: Connection reset by peer

    No matter how I change and modify command line’s flags/options, it will end up like error above. Please note, I’ve modified flags quite a lot thus a certain flags might be possibly not needed.

    Full error log from 1. is here.

    Configuration of nginx server is here. It’s pretty much basic, but I will further use it to relay to multiple target RTMP servers later.

    I’m on Ubuntu 18.04, 4.18.0-17-generic with 8 GB of RAM, 4 CPU Cores with ffmpeg

    ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
     built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)

    , and with Nginx

    nginx version: nginx/1.14.2
    built by gcc 7.3.0 (Ubuntu 7.3.0-27ubuntu1~18.04)
    built with OpenSSL 1.1.0g  2 Nov 2017 (running with OpenSSL 1.1.1b  26 Feb 2019)
    TLS SNI support enabled
    configure arguments: --with-http_ssl_module --add-module=../nginx-rtmp-module-1.2.1

    So again the question is how can I fix such issue ?
    Any suggestion would be appreciated.

    Important edit : I just found out I entered a wrong duplicated command for 3. Now it has been corrected ! Sorry about that.

    Update : I’m able to record audio with ffmpeg then do noise filter with sox, then stream to target rtmp with ffmpeg. But noise reduction doesn’t satisfy me yet. See above at 2. for its updated command. And according to this, sox has problem in understand a few bytes so I use flac format in piping instead. Still my question remained as this update is for improvement for command in 2.

  • Making a thumbnail using fluent-ffmpeg returns ffprobe

    29 avril 2019, par user10204157

    I am getting the following error :

    Error: ffprobe exited with code 1
    ffprobe version 4.1.1 Copyright (c) 2007-2019 the FFmpeg developers
     built with gcc 8.2.1 (GCC) 20190212
     configuration: --enable-gpl --enable-version3 --enable-sdl2 --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. 22.100 / 56. 22.100
     libavcodec     58. 35.100 / 58. 35.100
     libavformat    58. 20.100 / 58. 20.100
     libavdevice    58.  5.100 / 58.  5.100
     libavfilter     7. 40.101 /  7. 40.101
     libswscale      5.  3.100 /  5.  3.100
     libswresample   3.  3.100 /  3.  3.100
     libpostproc    55.  3.100 / 55.  3.100
    [tcp @ 0000020813d01740] Connection to tcp://stemuli.blob.core.windows.net:443 failed: Error number -138 occurred  <------
    https://stemuli.blob.core.windows.net/stemuli/mentor-lesson-video-76cbf390-4033-4a92-b127-7df22d5885f8.MOV: Unknown error

    Does this mean that azure’s connection is just being cut off ?