Hello,
Also, the values of b0 and b1 (or -b1, for the HPF) depend on the frequency
response that you need to achieve. On a Signals & Systems' book you will
find out methods to obtain them, or use MATLAB (or equivalent tool) to obtain
them.
Regards,
Jaime Andr Aranguren Cardona
j...@ieee.org
j...@computer.org
----- Original Message ----
From: Amit
To: smartie_625 ; a...
Sent: Monday, December 3, 2007 11:36:37 PM
Subject: Re: [audiodsp] High pass and low pass FIR filter
Hi,
Multiply any FIR coefficient set with this sequence [...-1, +1, -1, +1...] and
in effect you would mirror image the frequency response of the filter around
fs/4. In your case if [b0, b1] are the filter coefficients for a LPF then [b0,
-b1] would translate it to a HPF.
[...-1, +1, -1, +1...] is a cosine signal cos(pi*n) (Cosine wave of frequency
fs/2 sampled at fs). Multiplying this with your FIR set would be equivalent to
convolving in the frequency domain, which would result in translating the
frequency spectrum of original filter by pi.
Amit
----- Original Message ----
From: smartie_625
To: audiodsp@yahoogroup s.com
Sent: Sunday, 2 December, 2007 5:43:59 AM
Subject: [audiodsp] High pass and low pass FIR filter
Hi,
I am new to DSP. I want to design a low pass and a high pass FIR
filter of order 2 (moving average filter) to implement the same on
Xilinx V2P board.
What will be the difference in coefficients for the low pass and high
pass filter?
To elaborate, the equation for the output is:
y[n] = b0*x[n] + b1*x[n-1]
What will be bo and b1 for the low pass and high pass filters?
Thanks,
Abhishek
