I need to transform the phase (and not amplitude) of a sound and create a new sound file, however ifft() of the transformed array appears complex and cannot be written to audio. Below is my code:
% read the audio file and fourier transform it
[x, Fs] = audioread('Sample.m4a');
y = fft(x,Fs); % Compute DFT of x
Xf_mag = abs(y);
% the phase of each resulting frequency bin was randomly shuffled
Xf_phase = angle(y);
Xf_phase_new=shuffle(Xf_phase);
% The phase-shuffled frequency domain data were then back-transformed to
% the time domain using an inverse Fourier transform
Xf_new = Xf_mag.*exp(1i*Xf_phase_new);
x_new = ifft(Xf_new);
% write a new sound file
filename = 'test.m4a';
audiowrite(filename,x_new,Fs);
I get the following error:
Error using audiowrite (line 170)
Expected input to be real.

 採用された回答

Mathieu NOE
Mathieu NOE 2021 年 6 月 23 日

0 投票

hello
take the real part after you do the ifft :
x_new = real(ifft(Xf_new));

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質問済み:

2021 年 6 月 22 日

回答済み:

2021 年 6 月 23 日

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