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.