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  • 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

  • Gestion générale des documents

    13 mai 2011, par

    MédiaSPIP ne modifie jamais le document original mis en ligne.
    Pour chaque document mis en ligne il effectue deux opérations successives : la création d’une version supplémentaire qui peut être facilement consultée en ligne tout en laissant l’original téléchargeable dans le cas où le document original ne peut être lu dans un navigateur Internet ; la récupération des métadonnées du document original pour illustrer textuellement le fichier ;
    Les tableaux ci-dessous expliquent ce que peut faire MédiaSPIP (...)

  • 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 (4885)

  • Live stream the screen and microphone to azure media services with ffmpeg

    16 octobre 2020, par Panzer Ihnen

    I'm able to stream a video of my screen to Azure Media Services with ffmpeg, but when I add the audio input, the stream stops with this error :

    


    ffmpeg: frame=   27 fps=0.0 q=0.0 size=       0kB time=00:00:00.36 bitrate=   8.4kbits/s dup=0 drop=7 speed=0.703x
ffmpeg: frame=   43 fps= 42 q=0.0 size=       0kB time=00:00:00.36 bitrate=   8.4kbits/s dup=0 drop=7 speed=0.356x    
ffmpeg: frame=   58 fps= 38 q=27.0 size=      37kB time=00:00:00.36 bitrate= 818.6kbits/s dup=0 drop=7 speed=0.24x    
ffmpeg: av_interleaved_write_frame(): End of file
ffmpeg:     Last message repeated 1 times
ffmpeg: [flv @ 0633fec0] Failed to update header with correct duration.
ffmpeg: [flv @ 0633fec0] Failed to update header with correct filesize.
ffmpeg: Error writing trailer of rtmp://rtmpurl/stream: End of file
ffmpeg: frame=   59 fps= 32 q=26.0 Lsize=      37kB time=00:00:00.39 bitrate= 764.5kbits/s dup=0 drop=7 speed=0.214x    
ffmpeg: video:35kB audio:2kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
ffmpeg: [libx264 @ 0633c6c0] frame I:1     Avg QP:18.27  size: 30491
ffmpeg: [libx264 @ 0633c6c0] frame P:16    Avg QP:20.68  size:  4210
ffmpeg: [libx264 @ 0633c6c0] frame B:42    Avg QP:23.25  size:   152
ffmpeg: [libx264 @ 0633c6c0] consecutive B-frames:  5.1%  0.0%  0.0% 94.9%
ffmpeg: [libx264 @ 0633c6c0] mb I  I16..4: 12.7% 36.9% 50.4%
Ffmpeg Exited
ffmpeg: [libx264 @ 0633c6c0] mb P  I16..4:  0.4%  1.0%  0.6%  P16..4: 45.6% 10.5% 12.1%  0.0%  0.0%    skip:29.8%
ffmpeg: [libx264 @ 0633c6c0] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8: 15.5%  0.3%  0.0%  direct: 0.3%  skip:83.9%  L0:46.5% L1:51.0% BI: 2.5%
ffmpeg: [libx264 @ 0633c6c0] 8x8 transform intra:41.1% inter:59.1%
ffmpeg: [libx264 @ 0633c6c0] coded y,uvDC,uvAC intra: 79.9% 96.5% 88.6% inter: 7.8% 15.8% 5.4%
ffmpeg: [libx264 @ 0633c6c0] i16 v,h,dc,p: 20% 13%  6% 61%
ffmpeg: [libx264 @ 0633c6c0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 29% 21% 22%  4%  4%  5%  3%  3%  8%
ffmpeg: [libx264 @ 0633c6c0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 38% 16% 10%  5%  7%  7%  5%  6%  6%
ffmpeg: [libx264 @ 0633c6c0] i8c dc,h,v,p: 54% 18% 23%  5%
ffmpeg: [libx264 @ 0633c6c0] Weighted P-Frames: Y:0.0% UV:0.0%
ffmpeg: [libx264 @ 0633c6c0] ref P L0: 55.1%  2.9% 29.5% 12.6%
ffmpeg: [libx264 @ 0633c6c0] ref B L0: 79.5% 14.0%  6.6%
ffmpeg: [libx264 @ 0633c6c0] ref B L1: 93.2%  6.8%
ffmpeg: [libx264 @ 0633c6c0] kb/s:333.49
ffmpeg: Conversion failed!


    


    My ffmpeg command is :

    


    -f dshow -i audio="Microphone device" -strict -2 -c:a aac -f gdigrab -framerate 30 -offset_x 161 -offset_y 203 -video_size 430x322 -show_region 1 -i title="Window Title" -b:v 415K -g 60 -keyint_min 60 -b:a 32K -ar 22050 -filter:a "volume=0.8" -pix_fmt yuv420p -c:v libx264 -preset medium -bufsize 400k -maxrate 400k -f flv rtmp://rtmpurl/stream


    


  • How to create a video file webm from chunks by media recorder api using ffmpeg

    17 octobre 2020, par Caio Nakai

    I'm trying to create a webm video file from blobs generated by MediaRecorderAPI in a NodeJS server using FFMPEG. I'm able to create the .webm file but it's not playable, I ran this command $ ffmpeg.exe -v error -i lel.webm -f null - >error.log 2>&1 to generate an error log, the error log file contains this :

    


    


    [null @ 000002ce7501de40] Application provided invalid, non monotonically increasing dts to muxer in stream 0 : 1 >= 1

    


    [h264 @ 000002ce74a727c0] Invalid NAL unit size (804 > 74).

    


    [h264 @ 000002ce74a727c0] Error splitting the input into NAL units.

    


    Error while decoding stream #0:0 : Invalid data found when processing input

    


    


    This is my web server code

    


    const app = require("express")();
const http = require("http").createServer(app);
const io = require("socket.io")(http);
const fs = require("fs");
const child_process = require("child_process");

app.get("/", (req, res) => {
  res.sendFile(__dirname + "/index.html");
});

io.on("connection", (socket) => {
  console.log("a user connected");

  const ffmpeg = child_process.spawn("ffmpeg", [
    "-i",
    "-",
    "-vcodec",
    "copy",
    "-f",
    "flv",
    "rtmpUrl.webm",
  ]);

  ffmpeg.on("close", (code, signal) => {
    console.log(
      "FFmpeg child process closed, code " + code + ", signal " + signal
    );
  });

  ffmpeg.stdin.on("error", (e) => {
    console.log("FFmpeg STDIN Error", e);
  });

  ffmpeg.stderr.on("data", (data) => {
    console.log("FFmpeg STDERR:", data.toString());
  });

  socket.on("message", (msg) => {
    console.log("Writing blob! ");
    ffmpeg.stdin.write(msg);
  });

  socket.on("stop", () => {
    console.log("Stop recording..");
    ffmpeg.kill("SIGINT");
  });
});

http.listen(3000, () => {
  console.log("listening on *:3000");
});



    


    And this is my client code, using HTML, JS :

    


    &#xA;&#xA;  &#xA;    &#xA;    &#xA;    &#xA;  &#xA;  <code class="echappe-js">&lt;script src='http://stackoverflow.com/socket.io/socket.io.js'&gt;&lt;/script&gt;&#xA;  &lt;script&gt;&amp;#xA;    const socket = io();&amp;#xA;    let mediaRecorder = null;&amp;#xA;    const startRecording = (someStream) =&gt; {&amp;#xA;      const mediaStream = new MediaStream();&amp;#xA;      const videoTrack = someStream.getVideoTracks()[0];&amp;#xA;      const audioTrack = someStream.getAudioTracks()[0];&amp;#xA;      console.log(&quot;Video trac &quot;, videoTrack);&amp;#xA;      console.log(&quot;audio trac &quot;, audioTrack);&amp;#xA;      mediaStream.addTrack(videoTrack);&amp;#xA;      mediaStream.addTrack(audioTrack);&amp;#xA;&amp;#xA;      const recorderOptions = {&amp;#xA;        mimeType: &quot;video/webm;codecs=h264&quot;,&amp;#xA;        videoBitsPerSecond: 3 * 1024 * 1024,&amp;#xA;      };&amp;#xA;&amp;#xA;      mediaRecorder = new MediaRecorder(mediaStream, recorderOptions);&amp;#xA;      mediaRecorder.start(1000); // 1000 - the number of milliseconds to record into each Blob&amp;#xA;      mediaRecorder.ondataavailable = (event) =&gt; {&amp;#xA;        console.debug(&quot;Got blob data:&quot;, event.data);&amp;#xA;        if (event.data &amp;amp;&amp;amp; event.data.size &gt; 0) {&amp;#xA;          socket.emit(&quot;message&quot;, event.data);&amp;#xA;        }&amp;#xA;      };&amp;#xA;    };&amp;#xA;&amp;#xA;    const getVideoStream = async () =&gt; {&amp;#xA;      try {&amp;#xA;        const stream = await navigator.mediaDevices.getUserMedia({&amp;#xA;          video: true,&amp;#xA;          audio: true,&amp;#xA;        });&amp;#xA;        startRecording(stream);&amp;#xA;        myVideo.srcObject = stream;&amp;#xA;      } catch (e) {&amp;#xA;        console.error(&quot;navigator.getUserMedia error:&quot;, e);&amp;#xA;      }&amp;#xA;    };&amp;#xA;&amp;#xA;    const stopRecording = () =&gt; {&amp;#xA;      mediaRecorder.stop();&amp;#xA;      socket.emit(&quot;stop&quot;);&amp;#xA;    };&amp;#xA;  &lt;/script&gt;&#xA;  &#xA;    

    hello world
    

    &#xA;
    


    &#xA;
    


    &#xA;&#xA;    &#xA;    &lt;script&gt;&amp;#xA;      const myVideo = document.getElementById(&quot;myvideo&quot;);&amp;#xA;      myVideo.muted = true;&amp;#xA;    &lt;/script&gt;&#xA;  &#xA;&#xA;&#xA;

    &#xA;

    Any help is appreciated !

    &#xA;

  • How to stream RTMP to Azure Media Services ?

    26 octobre 2020, par Abbas

    I'm trying to stream my camera to Azure Media Services LiveEvent. I'm using Media Services' REST-API to obtain the ingest URL, however the docs don't mention how to stream RTMP from an Android Phone.

    &#xA;

    So far I've tried quiet a few Android RTMP publishing libraries available on Git but each one of them fails at establishing a connection. The list of libraries I've tried so far :

    &#xA;

    &#xA;

    I've also tried streaming from an mp4 video file using ffmpeg inspired from this SO Answer :

    &#xA;

    ffmpeg -re -i video.mp4 -vcodec libx264 -profile:v main -preset:v medium -r 30 -g 60 -keyint_min 60 -sc_threshold 0 -b:v 2500k -maxrate 2500k -bufsize 2500k -filter:v scale="trunc(oha/2)2:720" -sws_flags lanczos&#x2B;accurate_rnd -acodec aac -b:a 96k -ar 48000 -ac 2 -f flv rtmp://&#xA;

    &#xA;

    But I'm getting :

    &#xA;

    rtmp://: I/O error&#xA;

    &#xA;

    Am I missing something ?

    &#xA;

    Is it even at all possible to stream to an ingest URL without a middle tier as suggested by many Azure people is the way to go ?

    &#xA;

    Edit : I've successfully streamed to YouTube Live Streaming using two RTMP libraries and so I'm now pretty sure the issue is not with the RTMP streaming libraries but with how the Azure Live Streaming works. I'm definitely missing a step here.

    &#xA;

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