[reliablereason]

use this instead: ffmpeg -i original.wav -i decoded.wav -filter_complex "[1:a]aresample=async=1,volume=-1.0[inverted];[0:a][inverted]amix=inputs=2:weights=1 1" difference.wav

But honestly the only thing you get is something that subjectively sounds exactly the same, but lower volume. Probably due to the fact that subjective sound experience is more related to the fourier transform of the waves than it is to the waves themselves.

[gwbas1c]

It's because mp3 dramatically changes phase. As a result, merely mixing the inverted original won't leave you with what's filtered out.

That technique will work with simpler compression techniques, like companding. (Companding is basically doing the digital equivalent of the old Dolby NR button from the cassette days.)