> There may be some problems... > > - What about noise-like signals ?
The brain treats all sound in a fourier representation, noise also. Don't you think noise sounds like thousands of different beeps? The computer representation would be no different. > - For a resolution of 0.1 Hz you need 10 seconds of sound. Only a real > sine wave lasting the whole 10 s will tranform as a 'peak', everything > else will be smeared out. Hmm, are you sure? I would agree on that for a 0.1 Hz signal, which is not the case. Assume the sound contains a 100.0 Hz signal, then the analyzed amplitude (using sin and cos for phase independency) should be higher for 100.0 Hz than for 99.9 or 100.1, even for short periods. But I'm a newbie to DSP... What I actually aim at doing is a "sampler" program/plugin with pitch scaling, using this "complete fourier transform" approach to overcome the problems listed in: http://www.dspdimension.com/html/timepitch.html Any suggestions?
