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The Great Big Beautiful Tomorrow
28 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Texte
Autres articles (84)
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Configuration spécifique pour PHP5
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Sur d’autres sites (4392)
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Unable to convert .mp3 to .m4a using ffmpeg [closed]
4 mai 2013, par rahulgI am fully aware the legal constraints in using
libfaac
but this is just for my testing purpose.I have compiled
ffmpeg
withfaac
enabled. So when I tried to convert an .mp3 to a .m4a here is the error that I am getting. Please provide a resolution to this problem. I tried it on two different sources of .mp3, still I am getting the same error.[user@ip-10-161-13-26 ~]$ ffmpeg -i Kalimba.mp3 -c:a libfaac Kalimba.m4a
ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
built on May 4 2013 09:33:27 with gcc 4.4.6 20120305 (Red Hat 4.4.6-4)
configuration: --enable-libfaac --enable-nonfree --disable-yasm
libavutil 51. 54.100 / 51. 54.100
libavcodec 54. 23.100 / 54. 23.100
libavformat 54. 6.100 / 54. 6.100
libavdevice 54. 0.100 / 54. 0.100
libavfilter 2. 77.100 / 2. 77.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 15.100 / 0. 15.100
[mp3 @ 0x2464680] Header missing
[mp3 @ 0x2463100] max_analyze_duration 5000000 reached at 5015510
[mp3 @ 0x2463100] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'Kalimba.mp3':
Metadata:
publisher : Ninja Tune
track : 1
album : Ninja Tuna
artist : Mr. Scruff
album_artist : Mr. Scruff
title : Kalimba
genre : Electronic
composer : A. Carthy and A. Kingslow
date : 2008
Duration: 00:05:50.60, start: 0.000000, bitrate: 191 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s
Stream #0:1: Video: mjpeg, yuvj420p, 512x512, 90k tbr, 90k tbn, 90k tbc
Metadata:
title : thumbnail
comment : Cover (front)
Output #0, ipod, to 'Kalimba.m4a':
Metadata:
publisher : Ninja Tune
track : 1
album : Ninja Tuna
artist : Mr. Scruff
album_artist : Mr. Scruff
title : Kalimba
genre : Electronic
composer : A. Carthy and A. Kingslow
date : 2008
Stream #0:0: Video: none, q=2-31, 128 kb/s, 90k tbn
Metadata:
title : thumbnail
comment : Cover (front)
Stream #0:1: Audio: none, 0 channels, 128 kb/s
Stream mapping:
Stream #0:1 -> #0:0 (mjpeg -> ?)
Stream #0:0 -> #0:1 (mp3 -> libfaac)
Encoder (codec none) not found for output stream #0:0MP3 file is at http://db.tt/HtpEBpFU
Also while using Faac independently I get this weird error for any file.
Freeware Advanced Audio Coder
FAAC 1.28
Couldn't open input file sample.mp3 -
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.wav
and 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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ffmpeg split video H264 stutters on Windows Media Player
9 mars 2019, par Juan FoegenI am converting and splitting wmv and mp4 into new mp4 files. I am using ffmpeg to do the conversion with simply :
ffmpeg -i inputfile.wmv -vcodec libx264 -ss 00:00:10 -t 00:00:15 outputfile.mp4So when I create my split video it plays fine on other video players, like VLC, but stutters when initially playing on Windows Media Player and even shows a blank screen for a moment and then after about 5 seconds, the video will play normally. If I then select play again without reloading, Windows Media Player will play the video fine, with no stuttering.
If I REMOVE the start parameter when doing the conversion, then the resulting video will play without issues on all my players, INCLUDING Windows Media Player.
I have tried changing the frame rate, the bit rate and altered different optimization options mentioned here - https://trac.ffmpeg.org/wiki/Encode/H.264
I need the split video to play on Windows machines on Windows Media Player and on the web streamed using the video element. This is why I chose H264 codec and MP4.
Has anyone had issues playing ffmpeg split videos using H264 codec with Windows Media Player ? I am using the latest version of ffmpeg. Anyone have any other suggestions on any different settings I could try to adjust ?
Thanks