after applying a filter -i design- to a real signal, it returns complex signal after ifft

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Hamza AKYILDIZ on 14 May 2021
Answered: Paul Hoffrichter on 14 May 2021
n=length(x);
t = ((0:n-1)*(fs/n));
y=fft(x);
f = ((0:n-1)*(fs/n));
y0 = fftshift(y);
f0 = ((-n/2:n/2-1)*(fs/n));
noise = find(abs(y0) == max(abs(y0)));
noiseW= [2*pi*f0(noise(1)) 2*pi*f0(noise(2))];
z1=exp(1i*noiseW(1));
z2=exp(1i*noiseW(2));
z=exp(1i*2*pi*f0);
H=(1-z1*(z.^-1)).*(1-z2*(z.^-1));
out0=y0.*(H.');
out=ifftshift(out0);
signal=ifft(out);
audiowrite('output.wav',final,fs);
2 CommentsShow NoneHide None
Daniel Pollard on 14 May 2021
I'm not sure I understand the problem. The FFT (and the IFFT) produce complex outputs by their definition. Could you give some more details? What were you expecting to happen?
Hamza AKYILDIZ on 14 May 2021
some.wav file is a real signal. after i apply the filter H to it, i expect it to be still real after ifft.

Paul Hoffrichter on 14 May 2021
Your imaginary part only is non-zero due to the usual floating point roundoffs and truncations. Fix this using round:
signal=ifft(out);
signalRound = round(signal, 10);

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