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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 f=
ind out methods to obtain them, or use MATLAB (or equivalent tool) to obtai=
n them.

Regards,
=20
Jaime Andr=E9s Aranguren Cardona
j...@ieee.org
j...@computer.org

----- Original Message ----
From: Amit <a...@yahoo.co.uk>
To: smartie_625 <s...@gmail.com>; a...@yahoogroups.com
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 a=
round 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 frequen=
cy fs/2 sampled at fs). Multiplying this with your FIR set would be equival=
ent to convolving in the frequency domain, which would result in translatin=
g the frequency spectrum of original filter by pi.

Amit=20

----- Original Message ----
From: smartie_625 <smartie.625@ gmail.com>
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] =3D b0*x[n] + b1*x[n-1]

What will be bo and b1 for the low pass and high pass filters?

Thanks,
Abhishek

=20=20

=20

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