# Polyphase decimation matlab [closed]

I am new to DSP, and currently trying to implement a simple polyphase decimation program in matlab.

M is decimation factor, x is input (for now it is a square wave 8096 samples), y_polyDec is result from polyphase, output is result from regular filter then decimation.

code :

FilterBank_LPF = fir1(127, 1/M);
y_polyDec = zeros(8096/M, 1);

for i = 1:M

p(:, i) = FilterBank_LPF(i:M:end);
x(:, i) = input(M-i+1:M:end);
y_polyDec = filter(p(:, i), 1, x(:, i))+ y_polyDec;

end

out_LPF = filter(FilterBank_LPF, 1, input);
output = downsample(out_LPF, M);

%===========================================


If I compare "output" and "y_polyDec" (either comparing the decimated sample values or frequency spectrum using fvtool) I notice there are differences in the values. I was wondering if my code is wrong, or am I still missing something.

Thank you

## closed as off-topic by lennon310, Wandering Logic, jonsca, Naresh, Jason RMay 6 '14 at 17:19

This question appears to be off-topic. The users who voted to close gave this specific reason:

• "General programming questions are off-topic here, but can be asked on Stack Overflow." – lennon310, Wandering Logic, jonsca, Naresh, Jason R
If this question can be reworded to fit the rules in the help center, please edit the question.

The problem is in the input signals to your polyphase filters. They should be delayed and subsampled versions of the input signal, like in the following code fragment:

M=8;    % downsampling factor
L=256;  % length of input signal, integer multiple of M
x=randn(L,1);  % input signal
h = fir1(127, 1/M);
yp = zeros(L/M, 1);

for i = 1:M,
xtmp = [zeros(i-1,1); x];  % delayed input signal
tmp = filter( h(i:M:end), 1, xtmp(1:M:L) );  % polyphase filtering
yp = yp + tmp;  % accumulate outputs of polyphase filters
end

% compare with standard filtering
y = filter ( h, 1, x);
yM = downsample( y, M );

n=1:L/M;
plot(n,(yM-yp)/max(yM))