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我正在嘗試在Matlab中建立一個帶有不同頻段的濾波器。過濾後，低頻被推遲，因爲你可以在下面的圖片看到：濾波後發生低頻移位
過濾前： 
過濾後： 
你能幫助我理解爲什麼會這樣？
我使用的代碼如下：
[x,Fs] = audioread('drum-loop.wav');
% define constants - - - - - - - - - - - - - - - - - - - - - - - - - - - -
ntaps = 255; % number of filter-taps
% sampling frequencies for each branch
fs_A = 48000; %
fs_B = 24000; %
fs_C = 6000; %
fs_D = 1500; %
% decimation for each branch
dec_A = 1; %
dec_B = 2; %
dec_C = 4; %
dec_D = 4; %
% decimation filters - - - - - - - - - - - - - - - - - - - - - - - - - - -
% ref: https://es.mathworks.com/help/dsp/ref/mfilt.html
decfilter_B = mfilt.firdecim(dec_B); % decimate signal for branch B
x_B = filter(decfilter_B, x); % apply decimation filter
decfilter_C = mfilt.firdecim(dec_C); % decimate signal for branch C
x_C = filter(decfilter_C, x_B); % apply decimation filter
decfilter_D = mfilt.firdecim(dec_D); % decimate signal for branch D
x_D = filter(decfilter_D, x_C); % apply decimation filter
% interpolation filters - - - - - - - - - - - - - - - - - - - - - - - - -
% ref: https://es.mathworks.com/help/dsp/ref/mfilt.html
intfilter_B = mfilt.firinterp(dec_B); % decimate signal for branch B
intfilter_C = mfilt.firinterp(dec_C); % decimate signal for branch C
intfilter_D = mfilt.firinterp(dec_D); % decimate signal for branch D
% define filters - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
% refs:
% https://es.mathworks.com/help/signal/ref/fir1.html
% http://es.mathworks.com/matlabcentral/answers/17260-fir1-basic-question
% http://es.mathworks.com/help/pdf_doc/signal/signal_tb.pdf
ovlp1 = 1000;
ovlp2 = 500;
ovlp3 = 250;
% branch A: band pass filter.
coeff_A = [fs_B/2-ovlp1 (fs_A/2-1)]/(fs_A/2);
bp_A = fir1(ntaps,coeff_A);
% branch B: band pass filter.
coeff_B = [fs_C/2-ovlp2 (fs_B/2-1)]/(fs_B/2);
bp_B = fir1(ntaps,coeff_B);
% branch C: band pass filter.
coeff_C = [fs_D/2-ovlp3 (fs_C/2-1)]/(fs_C/2);
bp_C = fir1(ntaps,coeff_C);
% branch D: band pass filter.
coeff_D = (fs_D/2-1)/(fs_D/2);
bp_D = fir1(ntaps,coeff_D,'low');
% apply filters
x_A_f = filter(bp_A,1,x);
x_B_f = filter(bp_B,1,x_B);
x_C_f = filter(bp_C,1,x_C);
x_D_f = filter(bp_D,1,x_D);
% summation filter
y = x_A_f + filter(intfilter_B, x_B_f) + ...
filter(intfilter_B, filter(intfilter_C, x_C_f)) + ...
filter(intfilter_B, filter(intfilter_C, filter(intfilter_D, x_D_f)));
y = y/max(y); % normalize
% % compute fft
figure;
subplot(3,1,1);
plot(x);
NFFT = 2^nextpow2(length(x)); % Next power of 2 from length of y (in samples)
Y = fft(x,NFFT)/Fs;
f = Fs/2*linspace(0,1,NFFT/2+1);
subplot(3,1,2);spectrogram(x,blackman(128),60,128,1e3,'yaxis')
subplot(3,1,3);plot(f,2*angle(Y(1:NFFT/2+1)))
% % compute fft
figure;
subplot(3,1,1);
plot(y);
NFFT = 2^nextpow2(length(y)); % Next power of 2 from length of y (in samples)
Y = fft(y,NFFT)/Fs;
f = Fs/2*linspace(0,1,NFFT/2+1);
subplot(3,1,2);spectrogram(y,blackman(128),60,128,1e3,'yaxis')
subplot(3,1,3);plot(f,2*angle(Y(1:NFFT/2+1)))
audiowrite('output.wav',y,48000);