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hello,

i am reading data with fmod_readdata. i realized, that there are differences in the data when reading the same track as .wav and .mp3.
the mp3 file contained a lot of zeros at the beginning and the end.
that’s why i get different results for the same track when i run a bpm-algorithm on it.
i tried to get rid of the zeros in the mp3 at the beginning and i came closer to the result from the wav-file.
is this the correct way? should i also try to get rid of the zeros at the end? and why do i get zeros in the mp3 pcm data?

i would really like the same results with mp3s like i have with wav files

i am thankful for your help :)

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MP3 encoders generally do padding at the start and end of the file, so when it’s decoded you will get different output.

You could try using FMODs FSB encoder using MP3, we do our best to mitigate that silence. Also it helps if the source wav file is a multiple of 1152 samples (this is the MPEG frame size) and will avoid some padding.

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thank you. where do i find the fmod’s fsb encoder? i was looking in fmod ex’s documentation, but just found the System::createCodec. is this the correct method i need? where do i put it?

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Use FMOD Sound Bank Generator (fsbankex.exe). It’s part of the FMOD programmer API installation.

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[quote="peter":2xatwcj6]Use FMOD Sound Bank Generator (fsbankex.exe). It’s part of the FMOD programmer API installation.[/quote:2xatwcj6]

ahh, i see. ok thats nice. But my software should cope with mp3 files internally. however, i will try it.

thanks :)

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You can create FSB programatically using fsbanklib inside the tools/fsbanklib directory.

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[quote="peter":eaqr13jw]You can create FSB programatically using fsbanklib inside the tools/fsbanklib directory.[/quote:eaqr13jw]

thank you. i will look further into this. pls check your pm 😉

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