Comment by midjji

You also need to make sure that the gibbs phenomena does not cause the filtered signal to leave the range of the representation. So a prefiltered signal is less than 1, but the post filtered signal wont be. Meaning if 1 is the cap for your audio output, then enjoy a brand new category of distortion. But its a terrible way in general ofc, just optimize an appropriate, zero phase filter with an unaffected passband and minimum distortion. Its trivially easy, and there is no excuse to just use the terribly shitty short "classic" filters common in audio processing or graphics as implemented by skilled programmers who dont know the difference between dft and fft. (which is always easy to tell as they use the term fft as if it was synonymous with frequency transform estimates)