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Médias (9)

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  • 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

  • L’utiliser, en parler, le critiquer

    10 avril 2011

    La première attitude à adopter est d’en parler, soit directement avec les personnes impliquées dans son développement, soit autour de vous pour convaincre de nouvelles personnes à l’utiliser.
    Plus la communauté sera nombreuse et plus les évolutions seront rapides ...
    Une liste de discussion est disponible pour tout échange entre utilisateurs.

  • Contribute to translation

    13 avril 2011

    You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
    To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
    MediaSPIP is currently available in French and English (...)

Sur d’autres sites (6558)

  • Save video to disk from WebRTC MediaStream in Node

    27 novembre 2020, par SAGBO Aimé

    I'm building an app where the user can connect to the server through a WebRTC (I'm using simple-peer library both server-side and client-side to set the peer-to-peer connection).
Once the client and the server are connected, the client app stream the user camera and micro to the server.

    


    Now, I want to save the streamed data to the filesystem server-side as an MP4 video file.

    


    I hear about ffmpeg and fluent-ffmpeg to achieve this but i don't know how to use them.

    


      

    • Server side code to set up the peer connection
    • 


    


    const Peer = require("simple-peer");
const wrtc = require("wrtc");

const peer = new Peer({ initiator: false, wrtc: wrtc, trickle: false });

peer.on("error", (err: any) => console.log("error", err));

  peer.on("signal", (data: any) => {
    if (data.type === "offer" || data.type === "answer")
      dispatchMessage(JSON.stringify(data));
    // if (data.renegotiate || data.transceiverRequest) return;
  });

  peer.on("connect", () => {
    console.log("CONNECTED");
    peer.send(JSON.stringify("HELLO DEER PEER FROM SERVER"));
  });

  peer.on("data", (data: any) => {
    console.log("data: ", data);
  });

  peer.on("stream", (stream: MediaStream) => {
    console.log("-------Stream received", stream);
  });

  peer.on("track", (track: MediaStreamTrack) => {
    console.log("-------trackEvent:", track);
  });


    


      

    • Client-side code
    • 


    


    const stream = await window.navigator.mediaDevices.getUserMedia({
    video: { width: { ideal: 4096 }, height: { ideal: 2160 }},
    audio: true,
});

const p = new SimplePeer({
    initiator: isInitiator,  
    trickle: false  
});

stream.getTracks().forEach(track => p.addTrack(
    track,  
    stream  
));

// Here I set up the listeners for the peer connection


    


  • ffmpeg legitimate decoding errors

    20 juillet 2017, par Gideon Oduro

    My issue is as follows, i’m sending a H.264 encoded video captured with the help of WebRTC over WebSocket. The idea is to perform server side analysis and object tracking.

    navigator.mediaDevices.getUserMedia(constraint).then((stream) => {
     isVideoElement(target, stream)
     mediaRecorder = recorder(stream, {mimeType: 'video/webm; codecs=H264'})
     mediaRecorder.ondataavailable = (blob) => socket.send(blob.data)
     mediaRecorder.start('2000');
    })

    On the server side, data is being received as ByteBuffer :

    override fun handleBinaryMessage(session: WebSocketSession, msg: BinaryMessage) {
       analysis(msg.payload)
    }

    Im using the following resources (resource_1, resource_2) to try to convert my ByteBuffer to a OpenCv frame :

    fun startPreview(data: ByteBuffer) {
    avcodec_register_all()
    val pack = avcodec.AVPacket()
    pack.data(BytePointer(data))
    avcodec.av_init_packet(pack)

    val videoData = BytePointer(data)

    val codec = avcodec.avcodec_find_decoder(avcodec.AV_CODEC_ID_H264)
    val videoCodecContext = avcodec.avcodec_alloc_context3(codec)

    videoCodecContext.width(1280)
    videoCodecContext.height(720)

    videoCodecContext.pix_fmt(avutil.AV_PIX_FMT_YUV420P)
    videoCodecContext.codec_type(avutil.AVMEDIA_TYPE_VIDEO)
    videoCodecContext.extradata(videoData)
    videoCodecContext.extradata_size(data.capacity())
    videoCodecContext.flags2(videoCodecContext.flags2() or avcodec.CODEC_FLAG2_CHUNKS)

    avcodec.avcodec_open2(videoCodecContext, codec, null as PointerPointer<*>?)

    val decodedFrameLength = avcodec.avcodec_receive_frame(videoCodecContext, avutil.AVFrame())

    println(decodedFrameLength)

    }

    Im then receiving decodedFrameLength of -35 indicating a decoding error, cant figure out how to proceed from here ?

  • No audio output using FFmpeg

    26 mars 2022, par John Mergene Arellano

    I am having problem on Live stream output. I am streaming from mobile app to Node JS server to RTMP. Video output of the live stream is working but not the audio. There is no audio output from live stream.

    


    From my client side, I am sending a stream using the Socket.IO library. I captured the video and audio using getUserMedia API.

    


    navigator.mediaDevices.getUserMedia(constraints).then((stream) => {
    window.videoStream = video.srcObject = stream;
    let mediaRecorder = new MediaRecorder(stream, {
        videoBitsPerSecond : 3 * 1024 * 1024
    });
    mediaRecorder.addEventListener('dataavailable', (e) => {
        let data = e.data;
        socket.emit('live', data);
    });
    mediaRecorder.start(1000);
});


    


    Then my server will receive the stream and write it to FFmpeg.

    


    client.on('live', (stream)=>{
   if(ffmpeg)
       ffmpeg.stdin.write(stream);
});


    


    I tried watching the live video in VLC media player. There is a 5 seconds delay and no audio output.

    


    Please see below for FFmpeg options I used :

    


    ffmpeg = this.CHILD_PROCESS.spawn("ffmpeg", [
   '-f',
   'lavfi',
   '-i', 'anullsrc',
   '-i','-',
   '-c:v', 'libx264', '-preset', 'veryfast', '-tune', 'zerolatency',
   '-c:a', 'aac', '-ar', '44100', '-b:a', '64k',
   '-y', //force to overwrite
   '-use_wallclock_as_timestamps', '1', // used for audio sync
   '-async', '1', // used for audio sync
   '-bufsize', '1000',
   '-f',
   'flv',
   `rtmp://127.0.0.1:1935/live/stream` ]);


    


    What is wrong with my setup ? I need to fix the command so that the live stream will output both video and audio.

    


    I tried streaming to youtube RTMP but still no audio. I am expecting to have an output of video and audio from the getUserMedia API.

    


    What is wrong with my setup ? I need to fix the command so that the live stream will output both video and audio.

    


    I tried streaming to youtube RTMP but still no audio. I am expecting to have an output of video and audio from the getUserMedia API.