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<font face="Times New Roman, Times, serif">I've been mulling over
this idea of using the Jetson board to do an FFT of the entire
(mostly) sample stream to then make the frequency-domain signal
available to applications. Seems very intriguing and I love
hearing about entirely new ways of approaching problems.<br>
<br>
I'm curious though. When designing a codec, one chooses the
parameters for the FFT very carefully. The sampling rate is given
to you by the sample stream coming in (though you may decimate to
change that) but you pick the size of the FFT, the overlap between
blocks, and the window function all to make the FFT bins show up
just how you want them.<br>
<br>
With this new architecture, all those choices are made up front
and all the codecs running behind it have to use them. So my
question is, is there a way in the frequency domain to convert
from one to another? Can you simply average a bunch of FFT bins
together to get larger bin sizes and interpolate between bins if
you want smaller? Or do you have to do an IFFT back to the time
domain and then FFT again to change anything?<br>
<br>
I usually get about halfway through a DSP book before I get bogged
down so maybe this is all covered in later chapters.<br>
<br>
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