[music-dsp] FFT - Some beginner questions

[Andre Michelle]

Hi,


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.

1)
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?

2)
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?

Or reverse:
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.

bye

--
Andre Michelle
http://blog.andre-michelle.com