1


Hello!
I want to equalize, change tempo and split an mp3 file with ffmpeg. I have managed to split it correctly (after finding that in windows I have to escape de % in the file format) but I am stuck in the tempo problem. I have tried the following command:

ffmpeg -i out02.mp3 -af  "atempo=0.90" output.mp3

But I get the error:

ffmpeg version 4.3 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200621
  configuration: --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gn
utls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enabl
e-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencor
e-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnap
py --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --ena
ble-libvpx --enable-libwavpack --enable-libwebp --enable-libxml2 --enable-libzim
g --enable-lzma --enable-zlib --enable-gmp --enable-libvmaf --enable-libvorbis -
-enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libaom --en
able-libgsm --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cud
a-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dx
va2 --enable-libopenmpt --enable-amf
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
Input #0, mp3, from 'out02.mp3':
  Metadata:
    album           : Cuentos y Relatos
    artist          : JMT
    genre           : Podcast
    title           : La Noche de Margaret Rose
    track           : 296
    date            : 2020
    encoder         : Lavf58.45.100
  Duration: 00:05:00.02, start: 0.011995, bitrate: 64 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 64 kb/s
    Metadata:
      encoder         : Lavc57.75
File 'output.mp3' already exists. Overwrite? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (mp3_mf))
Press [q] to stop, [?] for help
[mp3_mf @ 00828d40] could not find any MFT for the given media type
[mp3_mf @ 00828d40] could not create MFT
Error initializing output stream 0:0 -- Error while opening encoder for output s
tream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Conversion failed!

I'm not an expert in sound. Why do I get this problem? Could you give me some advice to break through this problem?
Thank you.

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2

Your ffmpeg is giving priority to the MediaFoundation MP3 encoder (mp3_mf) instead the more common encoder libmp3lame. Not sure why–personally I would have given libmp3lame more priority so it would be the default if available for MP3 encoding.

mp3_mf is probably more limited than libmp3lame, but I didn't investigate your particular error.

Tell it to use libmp3lame instead:

ffmpeg -i out02.mp3 -af "atempo=0.90" -c:a libmp3lame output.mp3

This may have been fixed in the git master branch or in 4.3.1, so alternatively you can try downloading a newer version. But I don't use Windows and didn't check.

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