0.2 Hz DSP filter

[Mook Johnson]

I'm trying to replace a 0.2Hz 2nd order LP active filter with a similar DSP
filter. It doesn't need to have a very sharp cutoff but the .6 Hz
information needs to be down by ~30dB. It should continued down to -50 to
60 db from there.

My question is how do I determine the number of taps required? if taps are
roughly equivalent to the order then 2 or three should get me. However.
with a 100Hz sample rate from the A2D my ScopeFIR says I need ~700 taps. Am
I doing something wrong?

What is the design algorithm for a analog guy that wants to replace analog
filters with a digital one?


Also, I'm using a ADSP2181 processor. How do I calculate how many clock
cycles it takes to execute a single tap on the filter?


Thanks

 

[Tim Wescott]

Mook Johnson wrote:

I'm trying to replace a 0.2Hz 2nd order LP active filter with a similar DSP
filter. It doesn't need to have a very sharp cutoff but the .6 Hz
information needs to be down by ~30dB. It should continued down to -50 to
60 db from there.

My question is how do I determine the number of taps required? if taps are
roughly equivalent to the order then 2 or three should get me. However.
with a 100Hz sample rate from the A2D my ScopeFIR says I need ~700 taps. Am
I doing something wrong?

Yes -- you're designing a FIR filter when you want an IIR filter.
Expect to use 2x taps as the filter order for an IIR, but you need a
much lower order filter.

What is the design algorithm for a analog guy that wants to replace analog
filters with a digital one?

Search for "IIR Filter" on the net. Be way careful about the degree of

precision -- if you use the typical bilinear filter forms you're going
to need (100Hz/0.2Hz)^2 more precision than your input data, or 18 bits.
This can be a bummer when you have 16-bit data. You can use a
state-space filter that trades off more computation for lower precision
(i.e. you'll get it down to 9 bits instead of 18, but you'll need six
multiplies for a 2nd-order filter instead of 4).

Also, I'm using a ADSP2181 processor. How do I calculate how many clock
cycles it takes to execute a single tap on the filter?

For an FIR filter with 16-bit taps and 16-bit data you can expect some

setup (10 to 50 clocks) then one clock per tap. IIR filters tend to be
more difficult to code effectively. You could probably do this with an
8051 and some clever coding tricks, so unless you're really running that
2181 slowly you just don't have to worry.

Thanks

You're welcome. You may want to ask the question again on comp.dsp if

you're not happy with what you've heard here.

--

Tim Wescott
Wescott Design Services
http://www.wescottdesign.com

 

[CBarn24050]

Subject: Re: 0.2 Hz DSP filter
From: John Larkin

I'm trying to replace a 0.2Hz 2nd order LP active filter with a similar DSP
filter.

Look on the microchip site, there are several application notes on implementing
digital filters.

 

[Rene Tschaggelar]

Mook Johnson wrote:

I'm trying to replace a 0.2Hz 2nd order LP active filter with a similar DSP
filter. It doesn't need to have a very sharp cutoff but the .6 Hz
information needs to be down by ~30dB. It should continued down to -50 to
60 db from there.

My question is how do I determine the number of taps required? if taps are
roughly equivalent to the order then 2 or three should get me. However.
with a 100Hz sample rate from the A2D my ScopeFIR says I need ~700 taps. Am
I doing something wrong?

What is the design algorithm for a analog guy that wants to replace analog
filters with a digital one?

Also, I'm using a ADSP2181 processor. How do I calculate how many clock
cycles it takes to execute a single tap on the filter?

IMO the same sampling theorem holds as for doing FFTs.
The sampling memory has to be about as long as a period
of the lowest frequency you wish to process. So these
100Hz divided by 700 go down to 0.15Hz.
This doesn't mean that you need all 700 taps unequal
zero. There are decimatiuon algorithms that allow you
to process only every N-th sample, IIRC.

I'd recommend digging up some theory.

Rene
--
Ing.Buero R.Tschaggelar - http://www.ibrtses.com
& commercial newsgroups - http://www.talkto.net

 

[Tony]

On Thu, 27 Jan 2005 17:06:24 GMT, "Nicholas O. Lindan" <see@sig.com>
wrote:

Mook Johnson wrote:

I'm trying to replace a 0.2Hz 2nd order LP active filter with a similar DSP
filter. It doesn't need to have a very sharp cutoff but the .6 Hz
information needs to be down by ~30dB. It should continued down to -50 to
60 db from there.

What is the design algorithm for a analog guy that wants to replace analog
filters with a digital one?

If you search with google you should find a few DSP design programs. At least
one of these has a demo version available, limited to 7 sections IIRC, but
it should help you get started. Your filter spec seems to indicate
60db/decade -
that's pretty sharp, BTW.

It depends of your perspective - 3 pole filters aren't usually
regarded as bleeding edge stuff. But at such low frequencies, one
would hope for a correspondingly low sampling rate, or else a
double-precision implementation.

Tony (remove the "_" to reply by email)