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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. -
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Mise à disposition des fichiers
14 avril 2011, parPar défaut, lors de son initialisation, MediaSPIP ne permet pas aux visiteurs de télécharger les fichiers qu’ils soient originaux ou le résultat de leur transformation ou encodage. Il permet uniquement de les visualiser.
Cependant, il est possible et facile d’autoriser les visiteurs à avoir accès à ces documents et ce sous différentes formes.
Tout cela se passe dans la page de configuration du squelette. Il vous faut aller dans l’espace d’administration du canal, et choisir dans la navigation (...)
Sur d’autres sites (7140)
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FFmpeg : encoding a PCM audio file to AAC using AAC codec info from another file
7 septembre 2022, par Siddharth KumarI have two audio files :


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audio_0.wav: PCM audioaudio_1.aac: encoded audio using some parameters






I'm trying to achieve the following outcome with FFmpeg :


Create a new file
audio_2.aacwhich contains audio fromaudio_0.wavencoded using the codec parameters fromaudio_1.aac.

If I run the following :

ffmpeg -i audio_0.wav -i audio_1.aac -c copy audio_2.aacthen I get streams from both muxed together.
I do not want content fromaudio1.aac. Any pointers to solve this would be appreciated.

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File contains data in an unknown format. (m4a load from librosa)
2 août 2022, par Moriyama AikoSo I am currently working on a DNN that takes in m4a files. I have ffmpeg, it creates a few batches and then dies on this error :


Traceback (most recent call last):
 File "/users/work/s163838/./main.py", line 126, in <module>
 
 File "/users/work/s163838/./main.py", line 96, in main
 print("e")
 File "/apl/tryton/python/3.9.5/lib/python3.9/site-packages/torch/utils/data/dataloader.py", line 521, in __next__
 data = self._next_data()
 File "/apl/tryton/python/3.9.5/lib/python3.9/site-packages/torch/utils/data/dataloader.py", line 1203, in _next_data
 return self._process_data(data)
 File "/apl/tryton/python/3.9.5/lib/python3.9/site-packages/torch/utils/data/dataloader.py", line 1229, in _process_data
 data.reraise()
 File "/apl/tryton/python/3.9.5/lib/python3.9/site-packages/torch/_utils.py", line 425, in reraise
 raise self.exc_type(msg)
EOFError: Caught EOFError in DataLoader worker process 0.
Original Traceback (most recent call last):
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/librosa/core/audio.py", line 164, in load
 y, sr_native = __soundfile_load(path, offset, duration, dtype)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/librosa/core/audio.py", line 195, in __soundfile_load
 context = sf.SoundFile(path)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/soundfile.py", line 629, in __init__
 self._file = self._open(file, mode_int, closefd)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/soundfile.py", line 1183, in _open
 _error_check(_snd.sf_error(file_ptr),
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/soundfile.py", line 1357, in _error_check
 raise RuntimeError(prefix + _ffi.string(err_str).decode('utf-8', 'replace'))
RuntimeError: Error opening 'vox2/dev/aac/id08194/QnBYPze-x9A/00079.m4a': File contains data in an unknown format.

During handling of the above exception, another exception occurred:

Traceback (most recent call last):
 File "/apl/tryton/python/3.9.5/lib/python3.9/site-packages/torch/utils/data/_utils/worker.py", line 287, in _worker_loop
 data = fetcher.fetch(index)
 File "/apl/tryton/python/3.9.5/lib/python3.9/site-packages/torch/utils/data/_utils/fetch.py", line 44, in fetch
 data = [self.dataset[idx] for idx in possibly_batched_index]
 File "/apl/tryton/python/3.9.5/lib/python3.9/site-packages/torch/utils/data/_utils/fetch.py", line 44, in <listcomp>
 data = [self.dataset[idx] for idx in possibly_batched_index]
 File "/users/work/s163838/vox_celeb_loader.py", line 53, in __getitem__
 load(speaker2utt1, self.num_samples)
 File "/users/work/s163838/vox_celeb_loader.py", line 13, in load
 wav, sr = librosa.load(path, sr=16000)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/librosa/util/decorators.py", line 88, in inner_f
 return f(*args, **kwargs)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/librosa/core/audio.py", line 170, in load
 y, sr_native = __audioread_load(path, offset, duration, dtype)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/librosa/core/audio.py", line 226, in __audioread_load
 reader = audioread.audio_open(path)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/audioread/__init__.py", line 111, in audio_open
 return BackendClass(path)
 File "/users/kdm/s163838/.local/lib/python3.9/site-packages/audioread/rawread.py", line 65, in __init__
 self._file = aifc.open(self._fh)
 File "/apl/tryton/python/3.9.5/lib/python3.9/aifc.py", line 917, in open
 return Aifc_read(f)
 File "/apl/tryton/python/3.9.5/lib/python3.9/aifc.py", line 358, in __init__
 self.initfp(f)
 File "/apl/tryton/python/3.9.5/lib/python3.9/aifc.py", line 314, in initfp
 chunk = Chunk(file)
 File "/apl/tryton/python/3.9.5/lib/python3.9/chunk.py", line 63, in __init__
 raise EOFError
EOFError

</listcomp></module>

I am using this command


wav, sr = librosa.load(path, sr=16000)


is it just a broken file ? How do I skip such then ? Or is it something about loading a m4a file even with ffmpeg and the desired output when tested on a single m4a file ?


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FFmpeg output video file much smaller than uncompressed input audio file, using option to preserve original audio quality
17 mars 2021, par JanI attempt to create a video slideshow from a number of image files and an audio file in 2 steps :


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- Create a temporary video file from a sequence of image files
- Add an audio file to the temporary video file with a delay of 5 seconds






The audio file is an uncompressed stereo wav file, encoded with a sample rate of 44100 Hz and a bit depth of 32 bits, with a size of 40.1 MB. To preserve the lossless quality of the input audio file I use the option -c:a aac -b:a 192k as per Slideshow Wiki. However, the final output video file has a size of only 4.49 MB.


How can the output video file be about 10 times smaller than the input audio file and still preserve the original lossless quality ?


My code :


ffmpeg -f concat -i slide-sequence.txt -c:v libx264 -r 30 -filter_complex format=yuv420p temp.mp4
ffmpeg -i temp.mp4 -i audio.wav -af "adelay=5000|5000" -c:v copy -c:a aac -b:a 192k out.mp4


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