Hi all,
I am converting an audio signal to freq domain and back to time domain at different fft sizes to see the trade-off between time and freq resolution.
I thought that if I take the FFT of a very large window, I will get good frequency resolution but smear the time resolution. But when I take the FFT of a very long window (say 2^17 samples, which is roughly 3 sec), the following conversion reconstructs the signal nearly perfectly:
s = fft(x); % d contains 2^17 samples x2 = ifft(s);
I know these are inverse functions, but shouldn't there be some loss incurred? I expected the result to sound like a bank of sinewaves but playing back x2 it sounds like the original. What am I doing wrong? Thanks!
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