音频处理matlab，将原有音频进行抽样后，进行FFT傅里叶变换后再重构，查看IFFT后的变化

close all clear clc centralsample=4096; N_win=[64 256 1024]; Const_threshold=-50; Var_threshold=[-70 -40 -20]; %%%%%%%%%%%%% [y1,f1]=audioread('aeiou.wav'); win=4096;N_window=4096; frames=floor(size(y1,1)./N_window); length_data=frames*N_window; Data_syn=zeros(length_data,1); centra=centralsample*24; % legend('y = x','y = cos(x)*x','y = sin(x)*x^{2}'); % specular=stpt_frame(nt_total,win,win,1,centralsample,10.^(-3),fs); % subplot(2,1,2) % plot(specular,'k'); % hold on % title('Spectrum'); % xlabel('Frequency/Hz'); % ylabel('Amplitude'); %%%%%%%%%%%%%%%%%%%%%%%%%%%%%% stpt_frame function %%%%%%%%%%%%%%%%%%%% x=y1;w=win;N=win;H=1;threshold=-50;centralsample=centra; % function y = stpt_frame(x, w, N, H, centralsample,threshold, ~) % Analysis/synthesis of a sound using the peaks % of the short-time Fourier transform % Authors: J. Bonada, X. Serra, X. Amatriain, A. Loscos % x: A frame from the input sound, w: analysis window (odd size), N: FFT size, H: hop size, % t: threshold in negative dB, y: output sound % %-------------------------------------------------------------------------- % This source code is provided without any warranties as published in % DAFX book 2nd edition, copyright Wiley & Sons 2011, available at % http://www.dafx.de. It may be used for educational purposes and not % for commercial applications without further permission. %-------------------------------------------------------------------------- % M = length(w); % analysis window size M = w; % analysis window size N2 = (N)/2; % size of positive spectrum soundlength = length(x); % length of input sound aray hM = (M-1)/2; % half analysis window size fftbuffer = zeros(N,1); % initialize buffer for FFT yw = zeros(M,1); % initialize output sound frame y = zeros(M,1); % initialize output array % w = w/sum(w); % normalize analysis window sw = hanning(M); % synthesis window sw = sw./sum(sw); pend = centralsample; % last sample to start a frame pin = pend-M+1; % initialize sound pointer at the middle of analysis window %while pin<pend %-----analysis-----% xw = x(pin:pend).*sw(1:M); % window the input sound fftbuffer(:) = 0; % reset buffer fftbuffer(1:(M+1)/2) = xw((M+1)/2:M); % zero-phase fftbuffer fftbuffer((M+1)/2+1:N) = xw(1:(M-1)/2); X = fft(fftbuffer); % compute the FFT mX = 20*log10(abs(X(1:N2))); % magnitude spectrum of positive frequencies pX = unwrap(angle(X(1:N2))); % unwrapped phase spectrum plot(mX); ploc = 1 + find((mX(2:N2-1)>threshold) .* (mX(2:N2-1)>mX(3:N2)) ... .* (mX(2:N2-1)>mX(1:N2-2))); % peaks pmag = mX(ploc); % magnitude of peaks pphase = pX(ploc); % phase of peaks y=mX; %-----synthesis-----% Y = zeros(N,1); % initialize output spectrum y2=zeros(M,1); Y(ploc) = 10.^(pmag/20).*exp(1i.*pphase); % generate positive freq. Y(N+2-ploc) = 10.^(pmag/20).*exp(-1i.*pphase); % generate negative freq. fftbuffer = real(ifft(Y)); % real part of the inverse FFT yw((M+1)/2:M) = fftbuffer(1:(M+1)/2); % undo zero phase window yw(1:(M-1)/2) = fftbuffer(N-(M-1)/2+1:N); y2(1:M) = y2(1:M) + H*N*sw.*yw(1:M); % overlap-add % pin = pin+H; % advance sound pointer % % % subplot(2,1,2) % plot(y,'k'); % hold on % title('Spectrum'); % xlabel('Frequency/Hz'); % ylabel('Amplitude'); %% % end audiowrite('test.wav',y2,f1); sound(y2,f1);