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Carte de Schillerkiez
13 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
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Sur d’autres sites (7354)
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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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FFmpeg C API WMAV2 AVCodecParserContext not found even though CLI can parse WMAs on MacOS
3 octobre 2023, par grendellI am following the decode_audio.c example from FFmpeg, but I am unable to initialize a parser for
AV_CODEC_ID_WMAV2.

Test code :


#include 
#include <libavcodec></libavcodec>avcodec.h>

int main() {
 // codec is found successfully
 const AVCodec * codec = avcodec_find_decoder(AV_CODEC_ID_WMAV2);
 if (!codec) {
 fprintf(stderr, "codec not found\n");
 return 1;
 }

 // parser is always NULL
 AVCodecParserContext * parser = av_parser_init(codec->id);
 if (!parser) {
 fprintf(stderr, "parser not found\n");
 return 1;
 }

 av_parser_close(parser);
 return 0;
}


Build commands :


clang -c -I/opt/homebrew/Cellar/ffmpeg/6.0_1/include wma2mp3.c -o obj/wma2mp3.o
clang -L/opt/homebrew/Cellar/ffmpeg/6.0_1/lib -lavcodec obj/wma2mp3.o -o wma2mp3


I'm surprised by the fact that the FFmpeg CLI can perform this operation on the same machine :


% ffmpeg -i test.wma test.mp3
ffmpeg version 6.0 Copyright (c) 2000-2023 the FFmpeg developers
 built with Apple clang version 14.0.3 (clang-1403.0.22.14.1)
 configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/6.0_1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox --enable-neon
 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
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, asf, from 'test.wma':
 Metadata:
 ToolName : Windows Media Encoding Utility
 ToolVersion : 8.00.00.0343
 Duration: 00:00:00.74, start: 0.000000, bitrate: 80 kb/s
 Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, 1 channels, fltp, 48 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (wmav2 (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'test.mp3':
 Metadata:
 ToolName : Windows Media Encoding Utility
 ToolVersion : 8.00.00.0343
 TSSE : Lavf60.3.100
 Stream #0:0: Audio: mp3, 44100 Hz, mono, fltp
 Metadata:
 encoder : Lavc60.3.100 libmp3lame
[libmp3lame @ 0x130706320] Queue input is backward in timeed=N/A 
[mp3 @ 0x1307056e0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 15668 >= 14764
size= 8kB time=00:00:00.97 bitrate= 65.8kbits/s speed= 103x 
video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 4.048112%


I am using an Apple M1 machine running MacOS 13.5.2 (22G91).


Is the CLI using a different mechanism than
av_parser_parse2to perform this conversion, and is there a better way to accomplish this via the C API ?

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Revision 30966 : eviter le moche ’doctype_ecrire’ lors de l’upgrade
17 août 2009, par fil@… — Logeviter le moche ’doctype_ecrire’ lors de l’upgrade
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