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Topic: Encoding 32bit float WAV (over 0dB) to MP3 in foobar (Read 813 times) previous topic - next topic
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Encoding 32bit float WAV (over 0dB) to MP3 in foobar

If I convert a 32bit float WAV with over 0dBFS content to MP3 in foobar2000, it clips.
Doing the same conversion in Audacity results in no clipping.
Both use LAME, same version. What's going on and is there a way to make it work in f2k?

Re: Encoding 32bit float WAV (over 0dB) to MP3 in foobar

Reply #1 –
The command line frontend lame.exe clips the signal for some reason. Someone could remove this misfeature from it to solve the problem.
Until then if you want to preserve full input range you can try using LAME through ffmpeg: https://hydrogenaud.io/index.php?topic=114335.msg942256#msg942256.
 

Re: Encoding 32bit float WAV (over 0dB) to MP3 in foobar

Reply #2 –
I see, thanks. Indeed, the ffmpeg command works fine.

I opened a bug report for lame.exe at sourceforge.