Recherche avancée

Sur d’autres sites (7372)

• Recording audio with MediaRecorder on iPhone with Safari and Chrome only 1 second long ? Mimetype and FFMPEG problem ?

  9 mai 2023, par Avatar

  I am using MediaRecorder to record the Microphone audio on a website.

  


  Javascript :

  


  var blob;
var blob_url;
var stream;
var recorder;
var chunks;

var media = {
    tag: 'audio',
    type: 'audio/ogg',
    ext: '.ogg',
    gUM: {audio: true}
};

navigator.mediaDevices.getUserMedia(media.gUM).then(_stream => 
{
    stream = _stream;

    recorder = new MediaRecorder(stream);

    recorder.ondataavailable = e => 
    {
        // push data to chunks
        chunks.push(e.data);

        // recording has been stopped
        if(recorder.state == 'inactive') 
        {
            // audio data available
            blob = new Blob(chunks, {type: media.type });
            blob_url = URL.createObjectURL(blob);
            
            // send data to server
            uploadfile_audio();
        }
    };

    if(typeof(recorder)=='undefined')
    {
        alert('No microphone access');
        return;
    }

    chunks = [];
    recorder.start();
}


// when stop button is clicked
recorder.stop();
stream.getTracks().forEach( track => track.stop() );


  


  The audio stream (ogg format) is sent to the server.

  


  Since iPad/iPhone do not play ogg files, the recording file is converted to "mp3" using FFMPEG.

  


  This file is stored on the server.

  

  


  This works on Windows and MAC (Chrome and Safari), also on iPad (Safari) but not properly on iPhone (Chrome/Safari). Version : iPhone iOS 15.1.

  


  On iPhone the recording file is only 0:01 min in length. Size is always 17277 Bytes.

  


  What could be the issue ? (I cannot debug because I don't have a Mac.)

  


  Does the stream get interrupted ? Is the recording stopped after 1 second ?

  


  Update 1 :

  


  I have checked the incoming filesize of the browser-recorded file serverside. It seems to be coming in properly, because there are different sizes such as 184 kB.

  


  My guess is now that FFMPEG cannot handle the incoming file correctly. Which might have the wrong mimetype set in Javascript with type: 'audio/ogg',. Is another format needed ?

  


  The conversion code serverside :

  


  PHP

  


  $mp3file = shell_exec("ffmpeg -i ".$file_locationtmp." -vn -ar 44100 -ac 2 -b:a 128k ".$file_locationtmp.".mp3");


  


  I would need to find out the audio recording format used by iPhone but I couldn't.

  


  I tried to find the supporting mimetypes using https://developer.mozilla.org/en-US/docs/Web/API/MediaRecorder/isTypeSupported - however, it shows that NO mimetypes are supported on iPhone (neither in Chrome nor Safari).

  


  Update 2 :

  


  I used ffprobe to get the recording file information. It says Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 2234 kb/s (default)

  


  Update 3 :

  


  It seems to be a problem with FFMPEG. See my new question How to convert AAC/MP4A to MP3 using FFMPEG in full length ? Audio file gets cut off after 1 second

  


• Permissions issue with Python and ffmpeg on a Mac

  13 avril 2020, par EventHorizon

  I am fairly new to Python ( 4 weeks), and I have been struggling with this all day.

  



  I am using MacOS 10.13, Python 3.7 via Anaconda Navigator 1.9.12 and Spyder 4.0.1.

  



  Somehow (only a noob, remember) I had 2 Anaconda environments. I don't do production code, just research, so I figured I would make life simple and just use the base environment. I deleted the other environment.

  



  I had previously got FFmpeg working and was able to do frame grabs, build mpeg animations, and convert them to gifs for blog posts and such. I had FFmpeg installed in the directories associated with the deleted environment, so it went away.

  



  No worries, I got the git URL, used Terminal to install it in /opt/anaconda3/bin. It's all there and I can run FFmpeg from the Terminal.

  



  My problem : When I attempt to run a module that previously worked fine, I get the following message :

  



  [Errno 13] Permission denied : '/opt/anaconda3/bin/ffmpeg'

  



  In my module I set the default location of FFmpeg : plt.rcParams['animation.ffmpeg_path'] = '/opt/anaconda3/bin/ffmpeg'

  



  In my module I have the following lines :

  



  writer = animation.FFMpegWriter(fps=frameRate, metadata=metadata)
writer.setup(fig, "animation.mp4", 100)


  



  This calls matplotlib's 'animation.py', which runs the following :

  



  def setup(self, fig, outfile, dpi=None):
    '''
    Perform setup for writing the movie file.

    Parameters
    ----------
    fig : `~matplotlib.figure.Figure`
        The figure object that contains the information for frames
    outfile : str
        The filename of the resulting movie file
    dpi : int, optional
        The DPI (or resolution) for the file.  This controls the size
        in pixels of the resulting movie file. Default is fig.dpi.
    '''
    self.outfile = outfile
    self.fig = fig
    if dpi is None:
        dpi = self.fig.dpi
    self.dpi = dpi
    self._w, self._h = self._adjust_frame_size()

    # Run here so that grab_frame() can write the data to a pipe. This
    # eliminates the need for temp files.
    self._run()

def _run(self):
    # Uses subprocess to call the program for assembling frames into a
    # movie file.  *args* returns the sequence of command line arguments
    # from a few configuration options.
    command = self._args()
    _log.info('MovieWriter.run: running command: %s', command)
    PIPE = subprocess.PIPE
    self._proc = subprocess.Popen(
        command, stdin=PIPE, stdout=PIPE, stderr=PIPE,
        creationflags=subprocess_creation_flags)


  



  Everything works fine up to the last line (i.e. 'command' looks like a well-formatted FFmpeg command line, PIPE returns -1) but subprocess.Popen() bombs out with the error message above.

  



  I have tried changing file permissions - taking a sledgehammer approach and setting everything in /opt/anaconda3/bin/ffmpeg to 777, read, write, and execute. But that doesn't seem to make any difference. I really am clueless when it comes to Apple's OS, file permissions, etc. Any suggestions ?

  


• Convert Webrtc track stream to URL (RTSP/UDP/RTP/Http) in Video tag

  19 juillet 2020, par Zeeshan Younis

  I am new in WebRTC and i have done client/server connection, from client i choose WebCam and post stream to server using Track and on Server side i am getting that track and assign track stream to video source. Everything till now fine but problem is now i include AI(Artificial Intelligence) and now i want to convert my track stream to URL maybe UDP/RTSP/RTP etc. So AI will use that URL for object detection. I don't know how we can convert track stream to URL.
Although there is a couple of packages like https://ffmpeg.org/ and RTP to Webrtc etc, i am using Nodejs, Socket.io and Webrtc, below you can check my client and server side code for getting and posting stream, i am following thi github code https://github.com/Basscord/webrtc-video-broadcast.
Now my main concern is to make track as a URL for video tag, is it possible or not or please suggest, any help would be appreciated.

  


  Server.js

  


  This is nodejs server code


  


  


  const express = require("express");
const app = express();

let broadcaster;
const port = 4000;

const http = require("http");
const server = http.createServer(app);

const io = require("socket.io")(server);
app.use(express.static(__dirname + "/public"));

io.sockets.on("error", e => console.log(e));
io.sockets.on("connection", socket => {
  socket.on("broadcaster", () => {
    broadcaster = socket.id;
    socket.broadcast.emit("broadcaster");
  });
  socket.on("watcher", () => {
    socket.to(broadcaster).emit("watcher", socket.id);
  });
  socket.on("offer", (id, message) => {
    socket.to(id).emit("offer", socket.id, message);
  });
  socket.on("answer", (id, message) => {
    socket.to(id).emit("answer", socket.id, message);
  });
  socket.on("candidate", (id, message) => {
    socket.to(id).emit("candidate", socket.id, message);
  });
  socket.on("disconnect", () => {
    socket.to(broadcaster).emit("disconnectPeer", socket.id);
  });
});
server.listen(port, () => console.log(`Server is running on port ${port}`));

  


  


  



  Broadcast.js
This is the code for emit stream(track)


  


  


  const peerConnections = {};
const config = {
  iceServers: [
    {
      urls: ["stun:stun.l.google.com:19302"]
    }
  ]
};

const socket = io.connect(window.location.origin);

socket.on("answer", (id, description) => {
  peerConnections[id].setRemoteDescription(description);
});

socket.on("watcher", id => {
  const peerConnection = new RTCPeerConnection(config);
  peerConnections[id] = peerConnection;

  let stream = videoElement.srcObject;
  stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));

  peerConnection.onicecandidate = event => {
    if (event.candidate) {
      socket.emit("candidate", id, event.candidate);
    }
  };

  peerConnection
    .createOffer()
    .then(sdp => peerConnection.setLocalDescription(sdp))
    .then(() => {
      socket.emit("offer", id, peerConnection.localDescription);
    });
});

socket.on("candidate", (id, candidate) => {
  peerConnections[id].addIceCandidate(new RTCIceCandidate(candidate));
});

socket.on("disconnectPeer", id => {
  peerConnections[id].close();
  delete peerConnections[id];
});

window.onunload = window.onbeforeunload = () => {
  socket.close();
};

// Get camera and microphone
const videoElement = document.querySelector("video");
const audioSelect = document.querySelector("select#audioSource");
const videoSelect = document.querySelector("select#videoSource");

audioSelect.onchange = getStream;
videoSelect.onchange = getStream;

getStream()
  .then(getDevices)
  .then(gotDevices);

function getDevices() {
  return navigator.mediaDevices.enumerateDevices();
}

function gotDevices(deviceInfos) {
  window.deviceInfos = deviceInfos;
  for (const deviceInfo of deviceInfos) {
    const option = document.createElement("option");
    option.value = deviceInfo.deviceId;
    if (deviceInfo.kind === "audioinput") {
      option.text = deviceInfo.label || `Microphone ${audioSelect.length + 1}`;
      audioSelect.appendChild(option);
    } else if (deviceInfo.kind === "videoinput") {
      option.text = deviceInfo.label || `Camera ${videoSelect.length + 1}`;
      videoSelect.appendChild(option);
    }
  }
}

function getStream() {
  if (window.stream) {
    window.stream.getTracks().forEach(track => {
      track.stop();
    });
  }
  const audioSource = audioSelect.value;
  const videoSource = videoSelect.value;
  const constraints = {
    audio: { deviceId: audioSource ? { exact: audioSource } : undefined },
    video: { deviceId: videoSource ? { exact: videoSource } : undefined }
  };
  return navigator.mediaDevices
    .getUserMedia(constraints)
    .then(gotStream)
    .catch(handleError);
}

function gotStream(stream) {
  window.stream = stream;
  audioSelect.selectedIndex = [...audioSelect.options].findIndex(
    option => option.text === stream.getAudioTracks()[0].label
  );
  videoSelect.selectedIndex = [...videoSelect.options].findIndex(
    option => option.text === stream.getVideoTracks()[0].label
  );
  videoElement.srcObject = stream;
  socket.emit("broadcaster");
}

function handleError(error) {
  console.error("Error: ", error);
}

  


  


  



  RemoteServer.js
This code is getting track and assign to video tag


  


  


  let peerConnection;
const config = {
  iceServers: [
    {
      urls: ["stun:stun.l.google.com:19302"]
    }
  ]
};

const socket = io.connect(window.location.origin);
const video = document.querySelector("video");

socket.on("offer", (id, description) => {
  peerConnection = new RTCPeerConnection(config);
  peerConnection
    .setRemoteDescription(description)
    .then(() => peerConnection.createAnswer())
    .then(sdp => peerConnection.setLocalDescription(sdp))
    .then(() => {
      socket.emit("answer", id, peerConnection.localDescription);
    });
  peerConnection.ontrack = event => {
    video.srcObject = event.streams[0];
  };
  peerConnection.onicecandidate = event => {
    if (event.candidate) {
      socket.emit("candidate", id, event.candidate);
    }
  };
});

socket.on("candidate", (id, candidate) => {
  peerConnection
    .addIceCandidate(new RTCIceCandidate(candidate))
    .catch(e => console.error(e));
});

socket.on("connect", () => {
  socket.emit("watcher");
});

socket.on("broadcaster", () => {
  socket.emit("watcher");
});

socket.on("disconnectPeer", () => {
  peerConnection.close();
});

window.onunload = window.onbeforeunload = () => {
  socket.close();
};