Recherche avancée
Médias (1)
-
Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (53)
-
Formulaire personnalisable
21 juin 2013, parCette page présente les champs disponibles dans le formulaire de publication d’un média et il indique les différents champs qu’on peut ajouter. Formulaire de création d’un Media
Dans le cas d’un document de type média, les champs proposés par défaut sont : Texte Activer/Désactiver le forum ( on peut désactiver l’invite au commentaire pour chaque article ) Licence Ajout/suppression d’auteurs Tags
On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire. (...) -
Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...) -
Qu’est ce qu’un masque de formulaire
13 juin 2013, parUn masque de formulaire consiste en la personnalisation du formulaire de mise en ligne des médias, rubriques, actualités, éditoriaux et liens vers des sites.
Chaque formulaire de publication d’objet peut donc être personnalisé.
Pour accéder à la personnalisation des champs de formulaires, il est nécessaire d’aller dans l’administration de votre MediaSPIP puis de sélectionner "Configuration des masques de formulaires".
Sélectionnez ensuite le formulaire à modifier en cliquant sur sont type d’objet. (...)
Sur d’autres sites (11229)
-
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


-
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


-
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.


Syndiquer tout le site
