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  • MediaSPIP Core : La Configuration

    9 novembre 2010, par

    MediaSPIP Core fournit par défaut trois pages différentes de configuration (ces pages utilisent le plugin de configuration CFG pour fonctionner) : une page spécifique à la configuration générale du squelettes ; une page spécifique à la configuration de la page d’accueil du site ; une page spécifique à la configuration des secteurs ;
    Il fournit également une page supplémentaire qui n’apparait que lorsque certains plugins sont activés permettant de contrôler l’affichage et les fonctionnalités spécifiques (...)

  • Ajouter des informations spécifiques aux utilisateurs et autres modifications de comportement liées aux auteurs

    12 avril 2011, par

    La manière la plus simple d’ajouter des informations aux auteurs est d’installer le plugin Inscription3. Il permet également de modifier certains comportements liés aux utilisateurs (référez-vous à sa documentation pour plus d’informations).
    Il est également possible d’ajouter des champs aux auteurs en installant les plugins champs extras 2 et Interface pour champs extras.

  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

Sur d’autres sites (12338)

  • MP4Box / FFMPEG concat loses audio after first clip

    17 novembre 2017, par user1615343

    So I am certainly no expert when it comes to either of these tools, but I have a web-based project that’s executing commands on an Amazon Linux server to concatenate two video files that are uploaded.

    Both files are converted to mp4s first using FFMPEG, and those play perfectly in a browser after conversion :

    ffmpeg -i file1.mpg -c:v libx264 -crf 22 -c:a aac -strict -2 -movflags faststart file2.mp4

    Then, I attempt to combine these two resulting mp4s into a single mp4. I tried using FFMPEG to do this but to no avail. Switching to try MP4Box got me much closer : the videos are concatenated together, but the audio stops playing at the end of the first clip, and the second clip is silent.

    MP4Box -force-cat -keepsys -add file.mp4 -cat file2.mp4 out.mp4

    I’ve tried varying versions of the above command with no better results. Any input is greatly appreciated.

    EDIT : info on .mp4 files using

    ffmpeg -i file1.mp4 -i file2.mp4

    ffmpeg -i 1510189259715DogRunsintoGlassDoor_315a03a8e20acfc.mp4 -i
    1510189273549NewhouseMoonMoonneverseenstairsbeforefunnydog_285a03a8e6aab25.mp4

    ffmpeg version N-61041-g52a2138 Copyright (c) 2000-2014 the FFmpeg
    developers

    built on Mar 2 2014 05:45:04 with gcc 4.6 (Debian 4.6.3-1)

    configuration : —prefix=/root/ffmpeg-static/64bit
    —extra-cflags=’-I/root/ffmpeg-static/64bit/include -static’ —extra-ldflags=’-L/root/ffmpeg-static/64bit/lib -static’ —extra-libs=’-lxml2 -lexpat -lfreetype’ —enable-static —disable-shared —disable-ffserver —disable-doc —enable-bzlib —enable-zlib —enable-postproc —enable-runtime-cpudetect —enable-libx264 —enable-gpl —enable-libtheora —enable-libvorbis —enable-libmp3lame —enable-gray —enable-libass —enable-libfreetype —enable-libopenjpeg —enable-libspeex —enable-libvo-aacenc —enable-libvo-amrwbenc —enable-version3 —enable-libvpx

    libavutil 52. 66.100 / 52. 66.100

    libavcodec 55. 52.102 / 55. 52.102

    libavformat 55. 33.100 / 55. 33.100

    libavdevice 55. 10.100 / 55. 10.100

    libavfilter 4. 2.100 / 4. 2.100

    libswscale 2. 5.101 / 2. 5.101

    libswresample 0. 18.100 / 0. 18.100

    libpostproc 52. 3.100 / 52. 3.100

    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
    ’1510189259715DogRunsintoGlassDoor_315a03a8e20acfc.mp4’ :

    Metadata :

    major_brand : isom

    minor_version : 512

    compatible_brands : isomiso2avc1mp41

    encoder : Lavf55.33.100

    Duration : 00:00:04.92, start : 0.023220, bitrate : 634 kb/s

    Stream #0:0(und) : Video : h264 (High) (avc1 / 0x31637661), yuv420p,
    360x360 [SAR 1:1 DAR 1:1], 501 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc
    (default)

    Metadata :

    handler_name : VideoHandler

    Stream #0:1(und) : Audio : aac (mp4a / 0x6134706D), 44100 Hz, mono,
    fltp, 132 kb/s (default)

    Metadata :

    handler_name : SoundHandler

    Input #1, mov,mp4,m4a,3gp,3g2,mj2, from
    ’1510189273549NewhouseMoonMoonneverseenstairsbeforefunnydog_285a03a8e6aab25.mp4’ :

    Metadata :

    major_brand : isom

    minor_version : 512

    compatible_brands : isomiso2avc1mp41

    encoder : Lavf55.33.100

    Duration : 00:00:18.79, start : 0.023220, bitrate : 455 kb/s

    Stream #1:0(und) : Video : h264 (High) (avc1 / 0x31637661), yuv420p,
    362x360 [SAR 1:1 DAR 181:180], 320 kb/s, 29.94 fps, 29.94 tbr, 11976
    tbn, 59.88 tbc (default)

    Metadata :

    handler_name : VideoHandler

    Stream #1:1(eng) : Audio : aac (mp4a / 0x6134706D), 44100 Hz, stereo,
    fltp, 129 kb/s (default)

    Metadata :

    handler_name : SoundHandler

    At least one output file must be specified

  • AWS Lambda making video thumbnails

    9 décembre 2017, par Jesus

    I want make thumbnails from videos uploaded to S3, I know how to make it with Node.js and ffmpeg.

    According to this forum post I can add libraries :

    ImageMagick is the only external library that is currently provided by
    default, but you can include any additional dependencies in the zip
    file you provide when you create a Lambda function. Note that if this
    is a native library or executable, you will need to ensure that it
    runs on Amazon Linux.

    But how can I put static ffmpeg binary on aws lambda ?

    And how can I call from Node.js this static binary (ffmpeg) with AWS Lambda ?

    I’m newbie with amazon AWS and Linux

    Can anyone help me ?

  • Live audio using ffmpeg, javascript and nodejs

    8 novembre 2017, par klaus

    I am new to this thing. Please don’t hang me for the poor grammar. I am trying to create a proof of concept application which I will later extend. It does the following : We have a html page which asks for permission to use the microphone. We capture the microphone input and send it via websocket to a node js app.

    JS (Client) :

    var bufferSize = 4096;
    var socket = new WebSocket(URL);
    var myPCMProcessingNode = context.createScriptProcessor(bufferSize, 1, 1);
    myPCMProcessingNode.onaudioprocess = function(e) {
     var input = e.inputBuffer.getChannelData(0);
     socket.send(convertFloat32ToInt16(input));
    }

    function convertFloat32ToInt16(buffer) {
     l = buffer.length;
     buf = new Int16Array(l);
     while (l--) {
       buf[l] = Math.min(1, buffer[l])*0x7FFF;
     }
     return buf.buffer;
    }

    navigator.mediaDevices.getUserMedia({audio:true, video:false})
                                   .then(function(stream){
                                     var microphone = context.createMediaStreamSource(stream);
                                     microphone.connect(myPCMProcessingNode);
                                     myPCMProcessingNode.connect(context.destination);
                                   })
                                   .catch(function(e){});

    In the server we take each incoming buffer, run it through ffmpeg, and send what comes out of the std out to another device using the node js ’http’ POST. The device has a speaker. We are basically trying to create a 1 way audio link from the browser to the device.

    Node JS (Server) :

    var WebSocketServer = require('websocket').server;
    var http = require('http');
    var children = require('child_process');

    wsServer.on('request', function(request) {
     var connection = request.accept(null, request.origin);
     connection.on('message', function(message) {
       if (message.type === 'utf8') { /*NOP*/ }
       else if (message.type === 'binary') {
         ffm.stdin.write(message.binaryData);
       }
     });
     connection.on('close', function(reasonCode, description) {});
     connection.on('error', function(error) {});
    });

    var ffm = children.spawn(
       './ffmpeg.exe'
      ,'-stdin -f s16le -ar 48k -ac 2 -i pipe:0 -acodec pcm_u8 -ar 48000 -f aiff pipe:1'.split(' ')
    );

    ffm.on('exit',function(code,signal){});

    ffm.stdout.on('data', (data) => {
     req.write(data);
    });

    var options = {
     host: 'xxx.xxx.xxx.xxx',
     port: xxxx,
     path: '/path/to/service/on/device',
     method: 'POST',
     headers: {
      'Content-Type': 'application/octet-stream',
      'Content-Length': 0,
      'Authorization' : 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx',
      'Transfer-Encoding' : 'chunked',
      'Connection': 'keep-alive'
     }
    };

    var req = http.request(options, function(res) {});

    The device supports only continuous POST and only a couple of formats (ulaw, aiff, wav)

    This solution doesn’t seem to work. In the device speaker we only hear something like white noise.

    Also, I think I may have a problem with the buffer I am sending to the ffmpeg std in -> Tried to dump whatever comes out of the websocket to a .wav file then play it with VLC -> it plays everything in the record very fast -> 10 seconds of recording played in about 1 second.

    I am new to audio processing and have searched for about 3 days now for solutions on how to improve this and found nothing.

    I would ask from the community for 2 things :

    1. Is something wrong with my approach ? What more can I do to make this work ? I will post more details if required.

    2. If what I am doing is reinventing the wheel then I would like to know what other software / 3rd party service (like amazon or whatever) can accomplish the same thing.

    Thank you.