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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
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 (...) -
Mise à jour de la version 0.1 vers 0.2
24 juin 2013, parExplications des différents changements notables lors du passage de la version 0.1 de MediaSPIP à la version 0.3. Quelles sont les nouveautés
Au niveau des dépendances logicielles Utilisation des dernières versions de FFMpeg (>= v1.2.1) ; Installation des dépendances pour Smush ; Installation de MediaInfo et FFprobe pour la récupération des métadonnées ; On n’utilise plus ffmpeg2theora ; On n’installe plus flvtool2 au profit de flvtool++ ; On n’installe plus ffmpeg-php qui n’est plus maintenu au (...)
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17 avril 2024, par Erin -
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


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How to transcode to another video parameters ? [on hold]
23 mai 2014, par user3668381Google, man pages and any docs I found didn t shown anything relevant so...
I want to be able to concatene video with ffmpeg, this part is simple, but fail (freeze or massive frame dropping) if the videos don t have the same properties.
But for now, I didn t found anything else but trying to set a lot of options, expecting to get the good properties... But when they aren t rounded down (or up), you just can t set them (tbr, tbn...).
So my question is, is there any hidden option in ffmpeg to take the properties of another video (so -copy won t work) as the properties of the transcode.
Illustration :
This is the video from which I wan t to copy the parameters :
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'cdr.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
creation_time : 2013-07-04 11:04:27
encoder : Lavf54.11.100
Duration: 00:06:26.96, start: 0.000000, bitrate: 804 kb/s
Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 854x480 [SAR 1280:1281 DAR 16:9], 753 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc (default)
Metadata:
creation_time : 2013-07-04 11:04:27
handler_name : VideoHandler
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 44 kb/s (default)
Metadata:
creation_time : 2013-07-04 11:04:27
handler_name : SoundHandler
At least one output file must be specifiedFor now, my command is
ffmpeg -i video.mp4 -c:v h264 -c:a libfdk_aac -aspect 16:9 -b:v 753k -b:a 44k output.mp4
But it turn out that output.mp4 reveal :
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf55.21.100
Duration: 00:00:07.11, start: 0.046440, bitrate: 685 kb/s
Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 854x480 [SAR 1280:1281 DAR 16:9], 637 kb/s, 15 fps, 15 tbr, 15360 tbn, 30 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 45 kb/s (default)
Metadata:
handler_name : SoundHandler
At least one output file must be specifiedAs we can see, audio bitrate is rounded, tbn is off the chart, general bitrate isn t the same and on and on and on...
Is there any better way but to add options again and again and hope that nothing will be rounded ? Something like
ffmpeg -i video.mp4 -use_properties_of model.mp4 output.mp4
?