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Autres articles (29)
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Installation en mode ferme
4 février 2011, parLe mode ferme permet d’héberger plusieurs sites de type MediaSPIP en n’installant qu’une seule fois son noyau fonctionnel.
C’est la méthode que nous utilisons sur cette même plateforme.
L’utilisation en mode ferme nécessite de connaïtre un peu le mécanisme de SPIP contrairement à la version standalone qui ne nécessite pas réellement de connaissances spécifique puisque l’espace privé habituel de SPIP n’est plus utilisé.
Dans un premier temps, vous devez avoir installé les mêmes fichiers que l’installation (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
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 (...)
Sur d’autres sites (5358)
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FFMPEG DASH Issues
5 octobre 2022, par user726720I have a couple of questions regarding ffmpeg.


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First is i can't seem to get the frame rate correctly.




I'm expecting 25, i have tried both , both give me the same result.


-r 25 and -filter:v fps=25


Here is my complete command im using


ffmpeg -re -i file.mxf -pix_fmt yuv420p -vsync 1 -map 0:v:0 -map 0:a:0 -c:a aac -c:v libx264 -use_template 1 -use_timeline 1 -init_seg_name "init-stream$RepresentationID$-$Bandwidth$.mp4" -media_seg_name "chunk-stream$RepresentationID$-$Number%05d$.$ext$" -b:v 2000k -b:a 48k -ac 2 -profile:v main -level:v 3.0 -s 640x360 -filter:v fps=25 -vsync passthrough -increment_tc 1 -adaptation_sets "id=0,streams=v id=1,streams=a" -g 100 -keyint_min 100 -seg_duration 5 -frag_duration 5 -dash_segment_type auto -f dash stream.mpd


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My second question is related to the use of audio, with the above command im acheiving the below results, with the use of AAC




But my target is to acheive the below, how do i go about this




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- How do i change the muxing mode to dash






Like the below










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FFMPEG UDP to DASH I am wondering if it could be written more efficiently ?
18 mars 2019, par c7borgI have the below working but it is quite cpu intensive I’ve just moved to ffmpeg 3.4 and was wondering if it could be written more efficently ?
The below takes a multicast stream from our local LAN avoids the choppy footage by using the scenecut then adjusts the audio with the async to keep it in time and uses the yadif to deinterlace to provide better quality. This command/script also trims the maximum number of segments otherwise with a live stream it would fill up the directory.
If anyone has any improvements I’d much appreciate it
I also add subtitles using -filter_complex "[0:v][0:s]overlay" but this conflicts with the -vf yadif option.
ffmpeg -i \
"udp://@239.192.4.5:1234?overrun_nonfatal=1&fifo_size=50000000" \
-acodec aac -strict -2 -vcodec libx264 \
-vf yadif \
-af aresample=async=1 \
-x264opts 'keyint=25:scenecut=-1' \
-window_size 10 -extra_window_size 10 \
-f dash /var/www/html/stream/out.mpdIf it can’t be written more efficiently at least this may help others as it took me a long time to get this far. For reference I use shaka player in chromium for the client side
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Error audio loading when runing Whisper Open AI model
9 juin 2024, par John mickThe problem I'm trying to solve is that I can't run Whisper model for some audio, it says something related to audio decoding.


payload.wav: Invalid data found when processing input.
raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e


I tried using the
micro-machines.wavand it works fine but when i used other audio it gives me an error.

import whisper

model = whisper.load_model("base")
text=model.transcribe('micro-machines.wav',fp16=False)
print(text)
text=model.transcribe('payload.wav',fp16=False)
print(text)


Error I'm getting for payload :


d:\...\venv\lib\site-packages\whisper\transcribe.py:79: UserWarning: FP16 is not supported on CPU; using FP32 instead
 warnings.warn("FP16 is not supported on CPU; using FP32 instead") 
Traceback (most recent call last):
 File "d:\...\venv\lib\site-packages\whisper\audio.py", line 42, in load_audio
 ffmpeg.input(file, threads=0) 
 File "d:\...\venv\lib\site-packages\ffmpeg\_run.py", line 325, in run 
 raise Error('ffmpeg', out, err) 
ffmpeg._run.Error: ffmpeg error (see stderr output for detail) 

The above exception was the direct cause of the following exception:

Traceback (most recent call last):
 File "C:\....\Python\Python39\lib\runpy.py", line 197, in _run_module_as_main
 return _run_code(code, main_globals, None,
 File "C:\.....\Python\Python39\lib\runpy.py", line 87, in _run_code
 exec(code, run_globals)
 File "D:\...\venv\Scripts\whisper.exe\__main__.py", line 7, in <module>
 File "d:\...\venv\lib\site-packages\whisper\transcribe.py", line 314, in cli
 result = transcribe(model, audio_path, temperature=temperature, **args)
 File "d:\...\venv\lib\site-packages\whisper\transcribe.py", line 85, in transcribe
 mel = log_mel_spectrogram(audio)
 File "d:\...\venv\lib\site-packages\whisper\audio.py", line 111, in log_mel_spectrogram
 audio = load_audio(audio)
 File "d:\...\venv\lib\site-packages\whisper\audio.py", line 47, in load_audio
 raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
RuntimeError: Failed to load audio: ffmpeg version 6.0-essentials_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 12.2.0 (Rev10, Built by MSYS2 project)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enab
le-gmp --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxv
id --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf 
--enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libo
pencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enab
le-librubberband
 libavutil 58. 2.100 / 58. 2.100
 libavcodec 60. 3.100 / 60. 3.100
 libavformat 60. 3.100 / 60. 3.100
 libavdevice 60. 1.100 / 60. 1.100
 libavfilter 9. 3.100 / 9. 3.100
 libswscale 7. 1.100 / 7. 1.100
 libswresample 4. 10.100 / 4. 10.100
 libpostproc 57. 1.100 / 57. 1.100
payload.wav: Invalid data found when processing input
</module>

I tried searching for solutions and I found one which says It appears that the code failed to load the audio file for some reason and even failed to display that error because e.stderr did not contain a valid UTF-8 string


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