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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
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Stereo master soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Audio
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Elephants Dream - Cover of the soundtrack
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Mis à jour : Octobre 2011
Langue : English
Type : Image
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Mis à jour : Juin 2015
Langue : English
Type : Audio
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#6 Teaser Music
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Langue : English
Type : Audio
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#5 End Title
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Langue : English
Type : Audio
Autres articles (40)
-
Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page. -
De l’upload à la vidéo finale [version standalone]
31 janvier 2010, parLe chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
Upload et récupération d’informations de la vidéo source
Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...) -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...)
Sur d’autres sites (5416)
-
FFMPEG throws error while converting WEBM/MP4 audio files from MediaRecorder to MP3
27 septembre 2022, par Sahil MalikI use MediaRecorder to record the audio from the browser and then upload it to my server (ARM-based Linux machine - AWS Lambda Function, if it matters). Based on the browser type, I get either an MP4 file (for the Safari browser) OR a WEBM file (for every other browser) from the MediaRecorder. The audio is converted to Base64 string and posted to my server with FFMPEG.


MediaRecorder implementation


const audioStream = await navigator.mediaDevices.getUserMedia(mediaConstraints);
const MediaRecorder = window['MediaRecorder'];
const mimeType = MediaRecorder.isTypeSupported('audio/webm') ? 'audio/webm' : 'audio/mp4';
const recordingFileExtension = MediaRecorder.isTypeSupported('audio/webm') ? 'webm' : 'mp4';
mediaRecorder = new MediaRecorder(audioStream, { mimeType });
mediaRecorder.ondataavailable = convertBlobAndUploadChunk;
mediaRecorder.start(30 * 1000); // timeslice needs to be in ms
// When user stops recording
mediaRecorder.stop();
audioStream.getTracks().forEach( t => { t.stop(); });
// To convert the audio blob to string
function convertBlobToBase64(blob) {
 return new Promise((resolve, reject) => {
 const reader = new FileReader();
 reader.readAsDataURL(blob);
 reader.onload = () => {
 const tmpStr = reader.result.toString();
 resolve(tmpStr.substring(tmpStr.indexOf('base64,') + 7));
 };
 reader.onerror = error => reject(error);
 });
}

let blobCount = 0;
async function convertBlobAndUploadChunk(blobEvent) {
 if (!blobEvent.data || blobEvent.data.size === 0) return;

 blobCount++;

 const recordData = JSON.stringify({
 M: {
 blobCount,
 //some other meta data
 },
 D: await convertBlobToBase64(audioBlob),
 });

 await angularHttpClient.post(apiUrl, recordData, new HttpHeaders({
 'Content-Type': 'application/json',
 'x-api-key': apiKey,
 })).toPromise();
}



On my server, I convert the WEBM/MP4 file to an MP3 file for better cross-browser compatibility and to enable scrubbing.


Backend Lambda Implementation


const FFMpegCommand = require('fluent-ffmpeg'); // v2.1.2
new FFMpegCommand()
 .input(originalFile)
 .on('end', (error, stdOut, stdError) => {
 if (error) {
 console.error(error);
 }
 })
 .save(convertedFile);



This works perfectly fine 99.9% of the time but FFMPEG throws one of the following errors for 0.1% of the time :


Error 1 : Invalid data found when processing input


error reading header


ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the 
FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] could not find corresponding trex (id 1)
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] could not find corresponding track id 0
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] trun track id unknown, no tfhd was found
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] error reading header
/tmp/long-filename-of-140chars-to-keep-files-unique.mp4: Invalid data found when processing input



EBML Header parsing failed


ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
[matroska,webm @ 0x7595f40] Format matroska,webm detected only with low score of 1, misdetection possible!
[matroska,webm @ 0x7595f40] EBML header parsing failed
/tmp/long-filename-of-140chars-to-keep-files-unique.webm: Invalid data found when processing input



No specific details


ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
/tmp/long-filename-of-140chars-to-keep-files-unique.webm: Invalid data found when processing input



Error 2 : Output file #0 does not contain any stream


ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
[mpegts @ 0x5d19f40] Format mpegts detected only with low score of 2, misdetection possible!
[mpegts @ 0x5d19f40] Could not detect TS packet size, defaulting to non-FEC/DVHS
Input #0, mpegts, from '/tmp/long-filename-of-140chars-to-keep-files-unique.webm':
Duration: N/A, bitrate: N/A
Output #0, mp3, to '/tmp/long-filename-of-140chars-to-keep-files-unique.mp3':
Output file #0 does not contain any stream



Thanks for reading, any help/suggestion is highly appreciated.


-
How to improve web camera streaming latency to v4l2loopback device with ffmpeg ?
11 mars, par Made by MosesI'm trying to stream my iPhone camera to my PC on LAN.


What I've done :


- 

-
HTTP server with html page and streaming script :


I use WebSockets here and maybe WebRTC is better choice but it seems like network latency is good enough






async function beginCameraStream() {
 const mediaStream = await navigator.mediaDevices.getUserMedia({
 video: { facingMode: "user" },
 });

 websocket = new WebSocket(SERVER_URL);

 websocket.onopen = () => {
 console.log("WS connected");

 const options = { mimeType: "video/mp4", videoBitsPerSecond: 1_000_000 };
 mediaRecorder = new MediaRecorder(mediaStream, options);

 mediaRecorder.ondataavailable = async (event) => {
 // to measure latency I prepend timestamp to the actual video bytes chunk
 const timestamp = Date.now();
 const timestampBuffer = new ArrayBuffer(8);
 const dataView = new DataView(timestampBuffer);
 dataView.setBigUint64(0, BigInt(timestamp), true);
 const data = await event.data.bytes();

 const result = new Uint8Array(data.byteLength + 8);
 result.set(new Uint8Array(timestampBuffer), 0);
 result.set(data, 8);

 websocket.send(result);
 };

 mediaRecorder.start(100); // Collect 100ms chunks
 };
}



- 

-
Server to process video chunks






import { serve } from "bun";
import { Readable } from "stream";

const V4L2LOOPBACK_DEVICE = "/dev/video10";

export const setupFFmpeg = (v4l2device) => {
 // prettier-ignore
 return spawn("ffmpeg", [
 '-i', 'pipe:0', // Read from stdin
 '-pix_fmt', 'yuv420p', // Pixel format
 '-r', '30', // Target 30 fps
 '-f', 'v4l2', // Output format
 v4l2device, // Output to v4l2loopback device
 ]);
};

export class FfmpegStream extends Readable {
 _read() {
 // This is called when the stream wants more data
 // We push data when we get chunks
 }
}

function main() {
 const ffmpeg = setupFFmpeg(V4L2LOOPBACK_DEVICE);
 serve({
 port: 8000,
 fetch(req, server) {
 if (server.upgrade(req)) {
 return; // Upgraded to WebSocket
 }
 },
 websocket: {
 open(ws) {
 console.log("Client connected");
 const stream = new FfmpegStream();
 stream.pipe(ffmpeg?.stdin);

 ws.data = {
 stream,
 received: 0,
 };
 },
 async message(ws, message) {
 const view = new DataView(message.buffer, 0, 8);
 const ts = Number(view.getBigUint64(0, true));
 ws.data.received += message.byteLength;
 const chunk = new Uint8Array(message.buffer, 8, message.byteLength - 8);

 ws.data.stream.push(chunk);

 console.log(
 [
 `latency: ${Date.now() - ts} ms`,
 `chunk: ${message.byteLength}`,
 `total: ${ws.data.received}`,
 ].join(" | "),
 );
 },
 },
 });
}

main();



After I try to open the v4l2loopback device


cvlc v4l2:///dev/video10



picture is delayed for at least 1.5 sec which is unacceptable for my project.


Thoughts :


- 

- Problem doesn't seems to be with network latency




latency: 140 ms | chunk: 661 Bytes | total: 661 Bytes
latency: 206 ms | chunk: 16.76 KB | total: 17.41 KB
latency: 141 ms | chunk: 11.28 KB | total: 28.68 KB
latency: 141 ms | chunk: 13.05 KB | total: 41.74 KB
latency: 199 ms | chunk: 11.39 KB | total: 53.13 KB
latency: 141 ms | chunk: 16.94 KB | total: 70.07 KB
latency: 139 ms | chunk: 12.67 KB | total: 82.74 KB
latency: 142 ms | chunk: 13.14 KB | total: 95.88 KB



150ms is actually too much for 15KB on LAN but there can some issue with my router


- 

- As far as I can tell it neither ties to ffmpeg throughput :




Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pipe:0':
 Metadata:
 major_brand : iso5
 minor_version : 1
 compatible_brands: isomiso5hlsf
 creation_time : 2025-03-09T17:16:49.000000Z
 Duration: 00:00:01.38, start:
0.000000, bitrate: N/A
 Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuvj420p(pc), 1280x720, 4012 kb/s, 57.14 fps, 29.83 tbr, 600 tbn, 1200 tbc (default)
 Metadata:
 rotate : 90
 creation_time : 2025-03-09T17:16:49.000000Z
 handler_name : Core Media Video
 Side data:
 displaymatrix: rotation of -90.00 degrees

Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> rawvideo (native))

[swscaler @ 0x55d8d0b83100] deprecated pixel format used, make sure you did set range correctly

Output #0, video4linux2,v4l2, to '/dev/video10':
 Metadata:
 major_brand : iso5
 minor_version : 1
 compatible_brands: isomiso5hlsf
 encoder : Lavf58.45.100

Stream #0:0(und): Video: rawvideo (I420 / 0x30323449), yuv420p, 720x1280, q=2-31, 663552 kb/s, 60 fps, 60 tbn, 60 tbc (default)
 Metadata:
 encoder : Lavc58.91.100 rawvideo
 creation_time : 2025-03-09T17:16:49.000000Z
 handler_name : Core Media Video
 Side data:
 displaymatrix: rotation of -0.00 degrees

frame= 99 fps=0.0 q=-0.0 size=N/A time=00:00:01.65 bitrate=N/A dup=50 drop=0 speed=2.77x
frame= 137 fps=114 q=-0.0 size=N/A time=00:00:02.28 bitrate=N/A dup=69 drop=0 speed=1.89x
frame= 173 fps= 98 q=-0.0 size=N/A time=00:00:02.88 bitrate=N/A dup=87 drop=0 speed=1.63x
frame= 210 fps= 86 q=-0.0 size=N/A time=00:00:03.50 bitrate=N/A dup=105 drop=0 speed=1.44x
frame= 249 fps= 81 q=-0.0 size=N/A time=00:00:04.15 bitrate=N/A dup=125 drop=0 speed=1.36
frame= 279 fps= 78 q=-0.0 size=N/A time=00:00:04.65 bitrate=N/A dup=139 drop=0 speed=1.31x



- 

-
I also tried to write the video stream directly to
video.mp4
file and immediately open it withvlc
but it only can be successfully opened after 1.5 sec.

-
I've tried to use OBS v4l2 input source instead of vlc but the latency is the same








Update №1


When i try to stream actual
.mp4
file toffmpeg
it works almost immediately with 0.2sec delay to spin up the ffmpeg itself :

cat video.mp4 | ffmpeg -re -i pipe:0 -pix_fmt yuv420p -f v4l2 /dev/video10 & ; sleep 0.2 && cvlc v4l2:///dev/video10



So the problem is apparently with streaming process


-
-
how to add audio using ffmpeg when recording video from browser and streaming to Youtube/Twitch ?
26 juillet 2021, par Tosh VelagaI have a web application I am working on that allows the user to stream video from their browser and simultaneously livestream to both Youtube and Twitch using ffmpeg. The application works fine when I don't need to send any of the audio. Currently I am getting the error below when I try to record video and audio. I am new to using ffmpeg and so any help would be greatly appreciated. Here is also my repo if needed : https://github.com/toshvelaga/livestream


Here is my node.js server with ffmpeg


const child_process = require('child_process') // To be used later for running FFmpeg
const express = require('express')
const http = require('http')
const WebSocketServer = require('ws').Server
const NodeMediaServer = require('node-media-server')
const app = express()
const cors = require('cors')
const path = require('path')
const logger = require('morgan')
require('dotenv').config()

app.use(logger('dev'))
app.use(cors())

app.use(express.json({ limit: '200mb', extended: true }))
app.use(
 express.urlencoded({ limit: '200mb', extended: true, parameterLimit: 50000 })
)

var authRouter = require('./routes/auth')
var compareCodeRouter = require('./routes/compareCode')

app.use('/', authRouter)
app.use('/', compareCodeRouter)

if (process.env.NODE_ENV === 'production') {
 // serve static content
 // npm run build
 app.use(express.static(path.join(__dirname, 'client/build')))

 app.get('*', (req, res) => {
 res.sendFile(path.join(__dirname, 'client/build', 'index.html'))
 })
}

const PORT = process.env.PORT || 8080

app.listen(PORT, () => {
 console.log(`Server is starting on port ${PORT}`)
})

const server = http.createServer(app).listen(3000, () => {
 console.log('Listening on PORT 3000...')
})


const wss = new WebSocketServer({
 server: server,
})

wss.on('connection', (ws, req) => {
 const ffmpeg = child_process.spawn('ffmpeg', [
 // works fine when I use this but when I need audio problems arise
 // '-f',
 // 'lavfi',
 // '-i',
 // 'anullsrc',

 '-i',
 '-',

 '-f',
 'flv',
 '-c',
 'copy',
 `${process.env.TWITCH_STREAM_ADDRESS}`,
 '-f',
 'flv',
 '-c',
 'copy',
 `${process.env.YOUTUBE_STREAM_ADDRESS}`,
 // '-f',
 // 'flv',
 // '-c',
 // 'copy',
 // `${process.env.FACEBOOK_STREAM_ADDRESS}`,
 ])

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

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

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

 ws.on('message', (msg) => {
 console.log('DATA', msg)
 ffmpeg.stdin.write(msg)
 })

 ws.on('close', (e) => {
 console.log('kill: SIGINT')
 ffmpeg.kill('SIGINT')
 })
})

const config = {
 rtmp: {
 port: 1935,
 chunk_size: 60000,
 gop_cache: true,
 ping: 30,
 ping_timeout: 60,
 },
 http: {
 port: 8000,
 allow_origin: '*',
 },
}

var nms = new NodeMediaServer(config)
nms.run()



Here is my frontend code that records the video/audio and sends to server :


import React, { useState, useEffect, useRef } from 'react'
import Navbar from '../../components/Navbar/Navbar'
import './Dashboard.css'

const CAPTURE_OPTIONS = {
 audio: true,
 video: true,
}

function Dashboard() {
 const [mute, setMute] = useState(false)
 const videoRef = useRef()
 const ws = useRef()
 const mediaStream = useUserMedia(CAPTURE_OPTIONS)

 let liveStream
 let liveStreamRecorder

 if (mediaStream && videoRef.current && !videoRef.current.srcObject) {
 videoRef.current.srcObject = mediaStream
 }

 const handleCanPlay = () => {
 videoRef.current.play()
 }

 useEffect(() => {
 ws.current = new WebSocket(
 window.location.protocol.replace('http', 'ws') +
 '//' + // http: -> ws:, https: -> wss:
 'localhost:3000'
 )

 ws.current.onopen = () => {
 console.log('WebSocket Open')
 }

 return () => {
 ws.current.close()
 }
 }, [])

 const startStream = () => {
 liveStream = videoRef.current.captureStream(30) // 30 FPS
 liveStreamRecorder = new MediaRecorder(liveStream, {
 mimeType: 'video/webm;codecs=h264',
 videoBitsPerSecond: 3 * 1024 * 1024,
 })
 liveStreamRecorder.ondataavailable = (e) => {
 ws.current.send(e.data)
 console.log('send data', e.data)
 }
 // Start recording, and dump data every second
 liveStreamRecorder.start(1000)
 }

 const stopStream = () => {
 liveStreamRecorder.stop()
 ws.current.close()
 }

 const toggleMute = () => {
 setMute(!mute)
 }

 return (
 <>
 <navbar></navbar>
 <div style="{{" classname="'main'">
 <div>
 
 </div>
 <div classname="'button-container'">
 <button>Go Live</button>
 <button>Stop Recording</button>
 <button>Share Screen</button>
 <button>Mute</button>
 </div>
 </div>
 >
 )
}

const useUserMedia = (requestedMedia) => {
 const [mediaStream, setMediaStream] = useState(null)

 useEffect(() => {
 async function enableStream() {
 try {
 const stream = await navigator.mediaDevices.getUserMedia(requestedMedia)
 setMediaStream(stream)
 } catch (err) {
 console.log(err)
 }
 }

 if (!mediaStream) {
 enableStream()
 } else {
 return function cleanup() {
 mediaStream.getVideoTracks().forEach((track) => {
 track.stop()
 })
 }
 }
 }, [mediaStream, requestedMedia])

 return mediaStream
}

export default Dashboard