20 Jun
2003
20 Jun
'03
3:41 p.m.
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?