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Emballe médias : à quoi cela sert ?
4 février 2011, parCe plugin vise à gérer des sites de mise en ligne de documents de tous types.
Il crée des "médias", à savoir : un "média" est un article au sens SPIP créé automatiquement lors du téléversement d’un document qu’il soit audio, vidéo, image ou textuel ; un seul document ne peut être lié à un article dit "média" ; -
List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...) -
Librairies et logiciels spécifiques aux médias
10 décembre 2010, parPour un fonctionnement correct et optimal, plusieurs choses sont à prendre en considération.
Il est important, après avoir installé apache2, mysql et php5, d’installer d’autres logiciels nécessaires dont les installations sont décrites dans les liens afférants. Un ensemble de librairies multimedias (x264, libtheora, libvpx) utilisées pour l’encodage et le décodage des vidéos et sons afin de supporter le plus grand nombre de fichiers possibles. Cf. : ce tutoriel ; FFMpeg avec le maximum de décodeurs et (...)
Sur d’autres sites (8805)
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ffmpeg failed to load audio file
14 avril 2024, par Vaishnav GhengeFailed to load audio: ffmpeg version 5.1.4-0+deb12u1 Copyright (c) Failed to load audio: ffmpeg version 5.1.4-0+deb12u1 Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 12 (Debian 12.2.0-14)
 configuration: --prefix=/usr --extra-version=0+deb12u1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libglslang --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librist --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --disable-sndio --enable-libjxl --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-libplacebo --enable-librav1e --enable-shared
 libavutil 57. 28.100 / 57. 28.100
 libavcodec 59. 37.100 / 59. 37.100
 libavformat 59. 27.100 / 59. 27.100
 libavdevice 59. 7.100 / 59. 7.100
 libavfilter 8. 44.100 / 8. 44.100
 libswscale 6. 7.100 / 6. 7.100
 libswresample 4. 7.100 / 4. 7.100
 libpostproc 56. 6.100 / 56. 6.100
/tmp/tmpjlchcpdm.wav: Invalid data found when processing input


backend :



@app.route("/transcribe", methods=["POST"])
def transcribe():
 # Check if audio file is present in the request
 if 'audio_file' not in request.files:
 return jsonify({"error": "No file part"}), 400
 
 audio_file = request.files.get('audio_file')

 # Check if audio_file is sent in files
 if not audio_file:
 return jsonify({"error": "`audio_file` is missing in request.files"}), 400

 # Check if the file is present
 if audio_file.filename == '':
 return jsonify({"error": "No selected file"}), 400

 # Save the file with a unique name
 filename = secure_filename(audio_file.filename)
 unique_filename = os.path.join("uploads", str(uuid.uuid4()) + '_' + filename)
 # audio_file.save(unique_filename)
 
 # Read the contents of the audio file
 contents = audio_file.read()

 max_file_size = 500 * 1024 * 1024
 if len(contents) > max_file_size:
 return jsonify({"error": "File is too large"}), 400

 # Check if the file extension suggests it's a WAV file
 if not filename.lower().endswith('.wav'):
 # Delete the file if it's not a WAV file
 os.remove(unique_filename)
 return jsonify({"error": "Only WAV files are supported"}), 400

 print(f"\033[92m{filename}\033[0m")

 # Call Celery task asynchronously
 result = transcribe_audio.delay(contents)

 return jsonify({
 "task_id": result.id,
 "status": "pending"
 })


@celery_app.task
def transcribe_audio(contents):
 # Transcribe the audio
 try:
 # Create a temporary file to save the audio data
 with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_audio:
 temp_path = temp_audio.name
 temp_audio.write(contents)

 print(f"\033[92mFile temporary path: {temp_path}\033[0m")
 transcribe_start_time = time.time()

 # Transcribe the audio
 transcription = transcribe_with_whisper(temp_path)
 
 transcribe_end_time = time.time()
 print(f"\033[92mTranscripted text: {transcription}\033[0m")

 return transcription, transcribe_end_time - transcribe_start_time

 except Exception as e:
 print(f"\033[92mError: {e}\033[0m")
 return str(e)


frontend :


useEffect(() => {
 const init = () => {
 navigator.mediaDevices.getUserMedia({audio: true})
 .then((audioStream) => {
 const recorder = new MediaRecorder(audioStream);

 recorder.ondataavailable = e => {
 if (e.data.size > 0) {
 setChunks(prevChunks => [...prevChunks, e.data]);
 }
 };

 recorder.onerror = (e) => {
 console.log("error: ", e);
 }

 recorder.onstart = () => {
 console.log("started");
 }

 recorder.start();

 setStream(audioStream);
 setRecorder(recorder);
 });
 }

 init();

 return () => {
 if (recorder && recorder.state === 'recording') {
 recorder.stop();
 }

 if (stream) {
 stream.getTracks().forEach(track => track.stop());
 }
 }
 }, []);

 useEffect(() => {
 // Send chunks of audio data to the backend at regular intervals
 const intervalId = setInterval(() => {
 if (recorder && recorder.state === 'recording') {
 recorder.requestData(); // Trigger data available event
 }
 }, 8000); // Adjust the interval as needed


 return () => {
 if (intervalId) {
 console.log("Interval cleared");
 clearInterval(intervalId);
 }
 };
 }, [recorder]);

 useEffect(() => {
 const processAudio = async () => {
 if (chunks.length > 0) {
 // Send the latest chunk to the server for transcription
 const latestChunk = chunks[chunks.length - 1];

 const audioBlob = new Blob([latestChunk]);
 convertBlobToAudioFile(audioBlob);
 }
 };

 void processAudio();
 }, [chunks]);

 const convertBlobToAudioFile = useCallback((blob: Blob) => {
 // Convert Blob to audio file (e.g., WAV)
 // This conversion may require using a third-party library or service
 // For example, you can use the MediaRecorder API to record audio in WAV format directly
 // Alternatively, you can use a library like recorderjs to perform the conversion
 // Here's a simplified example using recorderjs:

 const reader = new FileReader();
 reader.onload = () => {
 const audioBuffer = reader.result; // ArrayBuffer containing audio data

 // Send audioBuffer to Flask server or perform further processing
 sendAudioToFlask(audioBuffer as ArrayBuffer);
 };

 reader.readAsArrayBuffer(blob);
 }, []);

 const sendAudioToFlask = useCallback((audioBuffer: ArrayBuffer) => {
 const formData = new FormData();
 formData.append('audio_file', new Blob([audioBuffer]), `speech_audio.wav`);

 console.log(formData.get("audio_file"));

 fetch('http://34.87.75.138:8000/transcribe', {
 method: 'POST',
 body: formData
 })
 .then(response => response.json())
 .then((data: { task_id: string, status: string }) => {
 pendingTaskIdsRef.current.push(data.task_id);
 })
 .catch(error => {
 console.error('Error sending audio to Flask server:', error);
 });
 }, []);


I was trying to pass the audio from frontend to whisper model which is in flask app


-
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


-
python subprocess ffmpeg return code = 69
13 juin 2023, par Tim ChenI try to call ffmpeg through the
subprocess.run(['ffmpeg', '-i', file_name, output_file_name], capture_output=True, text=True)command in python to convert the audio file incoming from the front end to wav format file. The backend code is as follows, using python+fastapi :

@app.post("/api/upload/convert")
async def convert_upload_file(request: Request, file: UploadFile = File(...)):
 token = uuid.uuid4().hex
 tmpFileName = os.path.join(os.path.dirname(__file__), token)
 with open(tmpFileName, "wb") as buffer:
 buffer.write(await file.read())
 await file.seek(0)
 output_path = tmpFileName + '-output.wav'
 command = ['ffmpeg', '-i', tmpFileName, output_path]
 result = subprocess.run(command, capture_output=True, text=True)


This code usually works, but there are some scenarios where it doesn't work. The audio file is recorded by js code (specifically
navigator.mediaDevices.getUserMedia({audio: true})).
The code of the audio recorded in windows chrome can run normally and get the converted wav file, but the audio recorded from ios15 safari for more than 3 seconds cannot be converted, promptingreturncode=69. The error message is as follows :

CompletedProcess(args=['ffmpeg', '-i', '5cfb52c503a646bda0f422b517c8014a', '5cfb52c503a646bda0f422b517c8014a-output.wav'], returncode=69, stdout='', stderr="
ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '5cfb52c503a646bda0f422b517c8014a':
 Metadata:
 major_brand : iso5
 minor_version : 1
 compatible_brands: isomiso5hlsf
 creation_time : 2023-06-11T16:36:53.000000Z
 Duration: 00:00:07.06, start: 0.000000, bitrate: 187 kb/s
 Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 184 kb/s (default)
 Metadata:
 creation_time : 2023-06-11T16:36:53.000000Z
 handler_name : Core Media Audio
 vendor_id : [0][0][0][0]
Stream mapping:
 Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '5cfb52c503a646bda0f422b517c8014a-output.wav':
 Metadata:
 major_brand : iso5
 minor_version : 1
 compatible_brands: isomiso5hlsf
 ISFT : Lavf58.76.100
 Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s (default)
 Metadata:
 creation_time : 2023-06-11T16:36:53.000000Z
 handler_name : Core Media Audio
 vendor_id : [0][0][0][0]
 encoder : Lavc58.134.100 pcm_s16le
size= 2kB time=00:00:00.00 bitrate=N/A speed=N/A 
[aac @ 0x55f1f8f19fc0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x55f1f8f19fc0] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x55f1f8f19fc0] If you want to help, upload a sample of this file to https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[aac @ 0x55f1f8f19fc0] Multiple frames in a packet.
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Number of bands (18) exceeds limit (13).
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Prediction is not allowed in AAC-LC.
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.


For the abnormal code, I tried to execute
ffmpeg -i input output.wavafter fastapi handle request on the command line andsubprocess.run(['ffmpeg', '-i', file_name, output_path], capture_output =True, text=True), all succeeded, which means that the final file must be normal, otherwise the subsequent verification work will get the same error.

This confuses me, is there some information I'm missing ?


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