I have a a low pass filter described by the following transfer function:
h[n] = (w_c/Pi) * sinc( n * w_c / Pi ), where is w_c is the cutoff frequency
I have to convert this low-pass filter to a band-pass filter.
I have a a low pass filter described by the following transfer function:
h[n] = (w_c/Pi) * sinc( n * w_c / Pi ), where is w_c is the cutoff frequency
I have to convert this low-pass filter to a band-pass filter.
You h[n]
transforms into a rect
in frequency domain. To make it band pass you need to move its central frequency higher.
To do this, multiply h[n]
by exp(j*w_offset*n)
, where w_offset
is the amount to shift. If w_offset
is positive, then you shift towards higher frequencies.
Multiplication in time domain is convolution in frequency domain. Since exp(j*w_offset*n)
turns into impulse function centred on w_offset
, the multiplication shifts the H(w)
by w_offset
.
See Discrete Time Fourier Transform for more details.
Note: such a filter will not be symmetric about 0, which means it will have complex values. To make it symmetric, you need to add h[n]
multiplied by exp(-j*w_offset*n)
:
h_bandpass[n] = h[n](exp(j*w_offset*n)+exp(-j*w_offset*n))
Since cos(w*n) = (exp(j*w*n)+exp(-j*w*n))/2
we get:
h_bandpass[n] = h[n]cos(w_offset*n)
This filter then has purely real values.
Let f[n]
be the signal you get from the low-pass filter with w_c
at the lower bound of the desired band. You can get the frequencies above this lower bound by subtracting f[n]
from the original signal. This is the input you want for the second low-pass filter.
The short answer is that you will multiply by a complex exponential in the time domain. Multiplication in the time domain will shift the signal in the frequency domain.
Matlab code:
n_taps = 100;
n = 1:n_taps;
h = ( w_c / Pi ) * sinc( ( n - n_taps / 2) * w_c / Pi ) .* ...
exp( i * w_offset * ( n - n_taps / 2) );
p.s. I happened to have just implemented this exact functionality for school a couple of weeks ago.
Here is code for creating your own band pass filter using the windowing method:
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
% Function: Create bandpass filter using windowing method
% Purpose: Simple method for creating filter taps ( useful when more elaborate
% filter design libraries are not available )
%
% @author Trevor B. Smith, 24MAR2009
%
% @param n_taps How many taps are in your output filter
% @param omega_p1 The lower cutoff frequency for your passband filter
% @param omega_p2 The upper cutoff frequency for your passband filter
% @return h_bpf_hammingWindow The filter coefficients for your passband filter
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
function h_bpf_hammingWindow = BPF_hammingWindow(n_taps,omega_p1,omega_p2)
% Error checking
if( ( omega_p2 == omega_p1 ) || ( omega_p2 < omega_p1 ) || ( n_taps < 10 ) )
str = 'ERROR - h_bpf_hammingWindow(): Incorrect input parameters'
h_bpf_hammingWindow = -1;
return;
end
% Compute constants from function parameters
length = n_taps - 1; % How many units of T ( i.e. how many units of T, sampling period, in the continuous time. )
passbandLength = omega_p2 - omega_p1;
passbandCenter = ( omega_p2 + omega_p1 ) / 2;
omega_c = passbandLength / 2; % LPF omega_c is half the size of the BPF passband
isHalfSample = 0;
if( mod(length,2) == 1 )
isHalfSample = 1/2;
end
% Compute hamming window
window_hamming = hamming(n_taps);
% Compute time domain samples
n = transpose(-ceil(length/2):floor(length/2));
h1 = sinc( (1/pi) * omega_c * ( n + isHalfSample ) ) * pi .* exp( i * passbandCenter * ( n + isHalfSample ) );
% Window the time domain samples
h2 = h1 .* window_hamming;
if 1
figure; stem(h2); figure; freqz(h2);
end
% Return filter coefficients
h_bpf_hammingWindow = h2;
end % function BPF_hammingWindow()
Example on how to use this function:
h_bpf_hammingWindow = BPF_hammingWindow( 36, pi/4, 3*pi/4 );
freqz(h_bpf_hammingWindow); % View the frequency domain