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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 <s...@gmail.com>
To: a...@yahoogroups.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

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