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Chris Moore wrote:

Please see my article AN-11. It gives a simplified method (Impulse 
Invariance) of designing with first order filters:

> http://www.sevenwoodsaudio.com/AN11.pdf

Best,

Chris
Jaime Andres Aranguren Cardona wrote:
> 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 Andrés Aranguren Cardona
> j...@ieee.org <mailto:jaime.aranguren%40ieee.org>
> j...@computer.org <mailto:jaime.aranguren%40computer.org> ----- Original Message
----
> From: Amit <a...@yahoo.co.uk <mailto:amit_elns%40yahoo.co.uk>>
> To: smartie_625 <s...@gmail.com 
> <mailto:smartie.625%40gmail.com>>; a...@yahoogroups.com 
> <mailto:audiodsp%40yahoogroups.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 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 <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] = 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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-- 
-----------------------------------------------------------------------
Seven Woods Audio, Inc.                Christopher Moore
Concepts, Products, Circuits for Audio    Analog/Digital
m...@SevenWoodsAudio.com
http://www.SevenWoodsAudio.com

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