One of the open-source applications which can be used as a Sound-Editor, is named ‘Audacity’. And in an earlier posting, I had written that this application may apply certain effects, which first involve performing a Fourier Transform of some sort on sampling-windows, which then manipulate the frequency-coefficients, and which then invert the Fourier Transform, to result in time-domain sound samples again.

On closer inspection of Audacity, I’ve recently come to realize that its programmers have *avoided* going that route, as often as possible. They may have designed effects which sound more natural as a result, but which follow how traditional analog methods used to process sound.

In some places, this has actually led to criticism of Audacity, let’s say because the users have discovered, that a low-pass or a high-pass filter would not maintain phase-constancy. But in traditional audio work, low-pass or high-pass filters always used to introduce phase-shifts. Audacity simply brings this into the digital realm.

I just seem to be remembering *certain other* sound editors, that used the Fourier Transforms extensively.

Dirk