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

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[x,fs]=audioread('some.wav');
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 commentaires
Daniel Pollard
Daniel Pollard le 14 Mai 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
Hamza AKYILDIZ le 14 Mai 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.

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Paul Hoffrichter
Paul Hoffrichter le 14 Mai 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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