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    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
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    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • 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.
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    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

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  • nginx RTMP to HLS : FFMPG error when trying multiple bitrate output [closed]

    28 mai 2014, par user3685074

    I’m currently trying to convert my RTMP Livestream into a HLS with 3 quality-settings.

    I followed this guide

    I’ve compiled my own FFMPEG and it’s working if I just convert 1 file.
    It seems libx264 isn’t able to do multiple encodings at the same time ?

    I’m using these command :

            exec /usr/local/bin/ffmpeg -i rtmp://localhost/src/$name 
    
        -c:a libfdk_aac -b:a 32k   -c:v libx264 -b:v 128K -f flv rtmp://localhost/hls/$name_low 
    
        -c:a libfdk_aac -b:a 64k   -c:v libx264 -b:v 256K -f flv rtmp://localhost/hls/$name_mid 
    
        -c:a libfdk_aac -b:a 128k  -c:v libx264 -b:v 512K -f flv rtmp://localhost/hls/$name_hi  2>>/tmp/ffmpeg.log;

    this is the output :

        ffmpeg version N-63519-g61917a1 Copyright (c) 2000-2014 the FFmpeg developers
    
      built on May 28 2014 18:06:42 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
    
      configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libmp3lame --enable-librtmp --enable-libspeex --enable-libfdk_aac
    
      libavutil      52. 87.100 / 52. 87.100
    
      libavcodec     55. 65.100 / 55. 65.100
    
      libavformat    55. 41.100 / 55. 41.100
    
      libavdevice    55. 13.101 / 55. 13.101
    
      libavfilter     4.  5.100 /  4.  5.100
    
      libswscale      2.  6.100 /  2.  6.100
    
      libswresample   0. 19.100 /  0. 19.100
    
      libpostproc    52.  3.100 / 52.  3.100
    
    Metadata:
    
      Server                NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
      width                 1280.00
    
      height                720.00
    
      displayWidth          1280.00
    
      displayHeight         720.00
    
      duration              0.00
    
      framerate             25.00
    
      fps                   25.00
    
      videodatarate         390.00
    
      videocodecid          0.00
    
      audiodatarate         27.00
    
      audiocodecid          11.00
    
    Input #0, flv, from 'rtmp://localhost/src/test':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
      Duration: 00:00:00.00, start: 0.080000, bitrate: N/A
    
        Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 399 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc
    
        Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 27 kb/s
    
    [libx264 @ 0x5260380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
    
    [libx264 @ 0x5260380] profile High, level 3.1
    
    [libx264 @ 0x5260380] 264 - core 142 r2431 f23da7c - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 lookahead_threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=128 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    
    [libx264 @ 0x525a920] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
    
    Output #0, flv, to 'rtmp://localhost/hls/test_low':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
        Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720, q=-1--1, 128 kb/s, 25 fps, 90k tbn, 25 tbc
    
        Metadata:
    
          encoder         : Lavc55.65.100 libx264
    
        Stream #0:1: Audio: aac (libfdk_aac), 16000 Hz, mono, s16, 32 kb/s
    
        Metadata:
    
          encoder         : Lavc55.65.100 libfdk_aac
    
    Output #1, flv, to 'rtmp://localhost/hls/test_mid':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
        Stream #1:0: Video: h264, yuv420p, 1280x720, q=-1--1, 256 kb/s, 25 fps, 90k tbn, 25 tbc
    
        Metadata:
    
          encoder         : Lavc55.65.100 libx264
    
        Stream #1:1: Audio: aac, 16000 Hz, mono, s16
    
        Metadata:
    
          encoder         : Lavc55.65.100 libfdk_aac
    
    Output #2, flv, to 'rtmp://localhost/hls/test_hi':
    
      Metadata:
    
        Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
    
        displayWidth    : 1280
    
        displayHeight   : 720
    
        fps             : 25
    
        profile         : 
    
        level           : 
    
        Stream #2:0: Video: h264, yuv420p, 1280x720, q=-1--1, 25 fps, 90k tbn, 25 tbc
    
        Metadata:
    
          encoder         : Lavc55.65.100 libx264
    
        Stream #2:1: Audio: aac, 16000 Hz, mono, s16
    
        Metadata:
    
          encoder         : Lavc55.65.100 libfdk_aac
    
    Stream mapping:
    
      Stream #0:0 -> #0:0 (h264 -> libx264)
    
      Stream #0:1 -> #0:1 (libspeex -> libfdk_aac)
    
      Stream #0:0 -> #1:0 (h264 -> libx264)
    
      Stream #0:1 -> #1:1 (libspeex -> libfdk_aac)
    
      Stream #0:0 -> #2:0 (h264 -> libx264)
    
      Stream #0:1 -> #2:1 (libspeex -> libfdk_aac)
    
    Error while opening encoder for output stream #1:0 - maybe incorrect parameters such as bit_rate, rate, width or height

    I hope you can help me and sorry for my bad english.

    Greetz
    Kevin

  • ffmpeg screencasting with webcam

    6 septembre 2017, par curiouser

    I want to use the screen view of the computer in real-time for a fake webcam. I did the video action on the bottom. But the frames like the video (2.34) are forming on top of each other. How can I solve this problem.

    https://www.youtube.com/watch?v=6AZRiW3hHrw

    Load the module
    
sudo modprobe v4l2loopback exclusive_caps=1
    

    
Find the dummy device
    
v4l2-ctl --list-devices
    

    
Start the virtual-webcam (change "/dev/video1" to reflect your system)
    
ffmpeg -f x11grab -r 15 -s 1920x1080 -i :0.0+0,0 -vcodec rawvideo -pix_fmt yuv420p -threads 0 -f v4l2 /dev/video1

  • How i can slice audio file with using node.js

    11 février 2018, par Maxim Cherevatov

    tell me please how i can slice audio file with using node.js ? Now i read the documentation for ffmpeg module, but don’t understand how to slice audio file with using this module.
    I found this code, but it gives an error error : NaN

    ffmpeg('music/ant.mp3')
    
    .setStartTime('00:00:03')
    
    .setDuration('10')
    
    .output('music/ant.mp3')
    
    .on('end', function(err) {
    
        if(!err)
    
        {
    
            console.log('conversion Done');
    
        }
    
    })
    
    .on('error', function(err){
    
        console.log('error: ', +err);
    
    }).run();

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