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

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Hamza AKYILDIZ
Hamza AKYILDIZ 2021-5-14
回答： Paul Hoffrichter 2021-5-14
[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 个评论
Daniel Pollard
Daniel Pollard 2021-5-14
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 2021-5-14
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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采纳的回答

Paul Hoffrichter
Paul Hoffrichter 2021-5-14
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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