Recherche avancée

Médias (1)

Mot : - Tags -/net art

Autres articles (106)

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

  • 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.

Sur d’autres sites (11512)

  • broken ffmpeg default settings detected

    8 août 2016, par Ramakrishna

    I am getting broken ffmpeg error while VideoWrite using X264 Fourcc codec.I have install all the dependencies.How can I rectify this problem.The sample code that I have been using is as follows.

    VideoWriter oVideoWriter ("path.mp4", CV_FOURCC('X','2','6','4'), 15, frameSize, false);

    Operating system : Ubuntu 14.04 64-bit

    Console Error :

    [libx264 @ 0x8d6220] broken ffmpeg default settings detected
    [libx264 @ 0x8d6220] use an encoding preset (e.g. -vpre medium)
    [libx264 @ 0x8d6220] preset usage: -vpre <speed> -vpre <profile>
    [libx264 @ 0x8d6220] speed presets are listed in x264 --help
    [libx264 @ 0x8d6220] profile is optional; x264 defaults to high
    Could not open codec 'libx264': Unspecified errorOpenCV Error: Unsupported format or combination of formats (Your version of Gstreamer doesn't support this codec acutally or needed plugin missing.) in CvVideoWriter_GStreamer::open, file /home/mbox140/Development/opencv-2.4.9/modules/highgui/src/cap_gstreamer.cpp, line 518
    terminate called after throwing an instance of 'cv::Exception'
     what():  /home/mbox140/Development/opencv-2.4.9/modules/highgui/src/cap_gstreamer.cpp:518: error: (-210) Your version of Gstreamer doesn't support this codec acutally or needed plugin missing. in function CvVideoWriter_GStreamer::open
    </profile></speed>

    Suggest me any solution for this.

  • 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.

  • Video Streaming from an ip camera h264

    4 décembre 2012, par user1764164

    I wish to develop an application for android which can receive video frames and display them in h264 encoded format. I was able to find solutions using mjpeg but i wish to use javacv which contains ffmpeg . Javacv also allows me to add motion detection and color enhancement algorithms to my frames. Please point me to references and solutions which would enable me to further the development of my application