Matlab男声变女声

FFT实现长序列卷积，256点

FIR滤波器

-------------------------------------------------------Code-------------------------------------------------------------

clear;

clc;

[y,fs]=audioread('C:\Users\shinelon\Documents\录音\number2.wav');

figure

plot(y);

y = y(:, 1)';

for i=0:1:1158

%sound(y,fs);

y_temp=y((i*256+1):1:(i+1)*256);

S=fft(y_temp,256);

%figure

%subplot(2,1,1);

%plot(y);

%title('原语音信号波形');

%subplot(2,1,2);

%plot(abs(S));

%title('原语音信号频谱');

 

Hd = FIR1;

female_freq = filter(Hd,S);

%figure(2);

%subplot(2,1,1);

%plot(abs(female_freq));

%title('转化后语音信号频谱');

    if i==0

       female = real(ifft(female_freq,256))';

    else

        temp = real(ifft(female_freq,256))';

        female =[female;temp];

    end

end

figure(2);

plot(female);

audiowrite('C:\Users\shinelon\Documents\录音\transform.wav',female*100,fs);

 

 

 

 

 

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function Hd = FIR1

%FIR1 Returns a discrete-time filter object.

 

% MATLAB Code

% Generated by MATLAB(R) 9.1 and the DSP System Toolbox 9.3.

% Generated on: 06-Nov-2022 23:42:27

 

% Equiripple Highpass filter designed using the FIRPM function.

 

% All frequency values are in Hz.

Fs = 48000;  % Sampling Frequency

 

Fstop = 150;             % Stopband Frequency

Fpass = 1000;            % Passband Frequency

Dstop = 0.0001;          % Stopband Attenuation

Dpass = 0.057501127785;  % Passband Ripple

dens  = 20;              % Density Factor

 

% Calculate the order from the parameters using FIRPMORD.

[N, Fo, Ao, W] = firpmord([Fstop, Fpass]/(Fs/2), [0 1], [Dstop, Dpass]);

 

% Calculate the coefficients using the FIRPM function.

b  = firpm(N, Fo, Ao, W, {dens});

Hd = dfilt.dffir(b);

 

% [EOF]