[music-dsp] FFT - Some beginner questions
am at andre-michelle.com
Fri Mar 7 05:57:45 EST 2008
since this is my first posting, I like to introduce myself a bit. My
name is Andre Michelle. I used to be a flash developer for 10 years
and changed recently to audio dsp programming - also in Flash. The
FlashPlugin doesn't provide a native way to synthesize audio, however
I found a dirty workaround which does the task well enough. You can
find some of my early experiments in my laboratory: http://lab.andre-michelle.com
I am now building a more complex audio application in Flash, where we
recently launch a private Beta at http://www.hobnox.com If you want me
to send you an invitation to join the Beta, send me an email offlist.
Enough of mine, here are my questions.
- FFT -
I spend a lot of time getting behind FFT, but some questions have been
unanswered while reading several articles. What I understand is, that
FFT transforms a time-domain signal to a frequency-domain signal and
vice versa. So far so good.
The funny thing is that after a forward transformation the signal is
mirrored at N/2. In my experiments I could easily set the right part
to zero (real/imag) after FFT-forward, without getting any difference
after FFT-inverse. Is that true?
The FFT (forward) returns discrete frequency array (where frequency =
bandIndex * sampleRate / N).
Is it possible to get the amplitude and phase from a frequency that is
BETWEEN any of those partials?
Is it possible to apply a sinus with a certain frequency (also between
a partial) and phase to a complex array for inverse FFT ?
I would be pleased for any advise.
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