Digital Filter with arbitrary amplitude shape?

Hi there,

Here is a topic that we are looking for the solution. We are looking
for a convenient way to implement the digital filters with arbitrary
amplitude shape. By "arbitrary" I mean the amplitude may vary within
+/- 6dB. The application is for sensor calibration. In the
instrumentation, each sensor may have its own frequency
characteristics. The variation is FIXED for each sensor. it would be
nice if we can find a digital filtering method that such variation is
compensated. A dynamic signal analyzer with DSP would be the ideal
platform to implement such an algorithm.

The filter must be implemented in the time domain, preferred with
linear phase.

James
www.go-ci.com

DigitalSignal wrote:
> Hi there,
> 
> Here is a topic that we are looking for the solution. 
> We are looking
> for a convenient way to implement the digital filters with arbitrary
> amplitude shape.
> The filter must be implemented in the time domain, preferred with
> linear phase.

Sounds like a perfect application for Parks-McClellan algorithm.



Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com

DigitalSignal wrote:
> Hi there,
> 
> Here is a topic that we are looking for the solution. We are looking
> for a convenient way to implement the digital filters with arbitrary
> amplitude shape. By "arbitrary" I mean the amplitude may vary within
> +/- 6dB. The application is for sensor calibration. In the
> instrumentation, each sensor may have its own frequency
> characteristics. The variation is FIXED for each sensor. it would be
> nice if we can find a digital filtering method that such variation is
> compensated. A dynamic signal analyzer with DSP would be the ideal
> platform to implement such an algorithm.
> 
> The filter must be implemented in the time domain, preferred with
> linear phase.

I believe that ScopeFIR at http://iowegian.com/ will do what you want.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

"Vladimir Vassilevsky" <antispam_bogus@hotmail.com> wrote in message 
news:3yLQj.1175$506.1160@newssvr27.news.prodigy.net...
>
>
> DigitalSignal wrote:
>> Hi there,
>>
>> Here is a topic that we are looking for the solution. We are looking
>> for a convenient way to implement the digital filters with arbitrary
>> amplitude shape.
>> The filter must be implemented in the time domain, preferred with
>> linear phase.
>
> Sounds like a perfect application for Parks-McClellan algorithm.

The existing Parks-McClellan program will do it if the number of passbands 
is large enough for you.  Otherwise, the same type of Remez algorithm 
program implementation but one using a continuous frequency amplitude 
function as the objective function / filter design will certainly do what 
you want.

With the Parks-McClellan program you can maybe specify frequency response in 
octaves .. or whatever .. that is, you may want do it in such a log 
frequency sense.

I'd be a little concerned that the impulse response would be what you want 
unless you limit the length of the filter.  The length of the filter and the 
degree to which you match the desired response go together.

Fred 

On Apr 26, 10:14&#4294967295;pm, "Fred Marshall" <fmarshallx@remove_the_x.acm.org>
wrote:
> "Vladimir Vassilevsky" <antispam_bo...@hotmail.com> wrote in message
>
> news:3yLQj.1175$506.1160@newssvr27.news.prodigy.net...
>
>
>
> > DigitalSignal wrote:
> >> Hi there,
>
> >> Here is a topic that we are looking for the solution. We are looking
> >> for a convenient way to implement the digital filters with arbitrary
> >> amplitude shape.
> >> The filter must be implemented in the time domain, preferred with
> >> linear phase.
>
> > Sounds like a perfect application for Parks-McClellan algorithm.
>
> The existing Parks-McClellan program will do it if the number of passbands
> is large enough for you. &#4294967295;Otherwise, the same type of Remez algorithm
> program implementation but one using a continuous frequency amplitude
> function as the objective function / filter design will certainly do what
> you want.
>
> With the Parks-McClellan program you can maybe specify frequency response in
> octaves .. or whatever .. that is, you may want do it in such a log
> frequency sense.
>
> I'd be a little concerned that the impulse response would be what you want
> unless you limit the length of the filter. &#4294967295;The length of the filter and the
> degree to which you match the desired response go together.
>
> Fred

consider also the phase (group delay) response...that may or may not
be important to your application...

Mark

On Apr 28, 6:48&#4294967295;am, Mark <makol...@yahoo.com> wrote:
> On Apr 26, 10:14&#4294967295;pm, "Fred Marshall" <fmarshallx@remove_the_x.acm.org>
> wrote:
>
>
>
>
>
> > "Vladimir Vassilevsky" <antispam_bo...@hotmail.com> wrote in message
>
> >news:3yLQj.1175$506.1160@newssvr27.news.prodigy.net...
>
> > > DigitalSignal wrote:
> > >> Hi there,
>
> > >> Here is a topic that we are looking for the solution. We are looking
> > >> for a convenient way to implement the digital filters with arbitrary
> > >> amplitude shape.
> > >> The filter must be implemented in the time domain, preferred with
> > >> linear phase.
>
> > > Sounds like a perfect application for Parks-McClellan algorithm.
>
> > The existing Parks-McClellan program will do it if the number of passbands
> > is large enough for you. &#4294967295;Otherwise, the same type of Remez algorithm
> > program implementation but one using a continuous frequency amplitude
> > function as the objective function / filter design will certainly do what
> > you want.
>
> > With the Parks-McClellan program you can maybe specify frequency response in
> > octaves .. or whatever .. that is, you may want do it in such a log
> > frequency sense.
>
> > I'd be a little concerned that the impulse response would be what you want
> > unless you limit the length of the filter. &#4294967295;The length of the filter and the
> > degree to which you match the desired response go together.
>
> > Fred
>
> consider also the phase (group delay) response...that may or may not
> be important to your application...
>
> Mark- Hide quoted text -
>
> - Show quoted text -

Yes phase is important. Thanks for mentioning it. I will do some
research about Remez filter.

James
www.go-ci.com

"DigitalSignal" <digitalsignal999@yahoo.com> wrote in message 
news:bfc41b9a-b72d-46a8-a28f-4a30e7e3e2db@q1g2000prf.googlegroups.com...
On Apr 28, 6:48 am, Mark <makol...@yahoo.com> wrote:
> On Apr 26, 10:14 pm, "Fred Marshall" <fmarshallx@remove_the_x.acm.org>
> wrote:

>Yes phase is important. Thanks for mentioning it. I will do >some research 
>about Remez filter.


You did say "linear phase" so the length of the filter is directly related 
to the delay which is flat.

Fred 

On Sat, 26 Apr 2008 12:17:02 -0700 (PDT), DigitalSignal
<digitalsignal999@yahoo.com> wrote:

>Hi there,
>
>Here is a topic that we are looking for the solution. We are looking
>for a convenient way to implement the digital filters with arbitrary
>amplitude shape. By "arbitrary" I mean the amplitude may vary within
>+/- 6dB. The application is for sensor calibration. In the
>instrumentation, each sensor may have its own frequency
>characteristics. The variation is FIXED for each sensor. it would be
>nice if we can find a digital filtering method that such variation is
>compensated. A dynamic signal analyzer with DSP would be the ideal
>platform to implement such an algorithm.
>
>The filter must be implemented in the time domain, preferred with
>linear phase.
>
>James
>www.go-ci.com

Hi,
  I'm not sure if it will help you but you might take 
a look at the article:

"Frequency Response Compensation With DSP" in the 
"DSP Tips & Tricks" column of the July 2003 IEEE 
Signal Processing magazine.

Good Luck,
[-Rick-]

On Apr 29, 8:17&#4294967295;pm, Rick Lyons <R.Lyons@_BOGUS_ieee.org> wrote:
> On Sat, 26 Apr 2008 12:17:02 -0700 (PDT), DigitalSignal
>
>
>
>
>
> <digitalsignal...@yahoo.com> wrote:
> >Hi there,
>
> >Here is a topic that we are looking for the solution. We are looking
> >for a convenient way to implement the digital filters with arbitrary
> >amplitude shape. By "arbitrary" I mean the amplitude may vary within
> >+/- 6dB. The application is for sensor calibration. In the
> >instrumentation, each sensor may have its own frequency
> >characteristics. The variation is FIXED for each sensor. it would be
> >nice if we can find a digital filtering method that such variation is
> >compensated. A dynamic signal analyzer with DSP would be the ideal
> >platform to implement such an algorithm.
>
> >The filter must be implemented in the time domain, preferred with
> >linear phase.
>
> >James
> >www.go-ci.com
>
> Hi,
> &#4294967295; I'm not sure if it will help you but you might take
> a look at the article:
>
> "Frequency Response Compensation With DSP" in the
> "DSP Tips & Tricks" column of the July 2003 IEEE
> Signal Processing magazine.
>
> Good Luck,
> [-Rick-]- Hide quoted text -
>
> - Show quoted text -

Hi Rick, this is beautiful! It is exactly what I need! It is a huge
mistake that IEEE SP stops publishing the DSP Tips and Tricks.

James
www.go-ci.com

On Apr 26, 3:17 pm, DigitalSignal <digitalsignal...@yahoo.com> wrote:
> Hi there,
>
> Here is a topic that we are looking for the solution. We are looking
> for a convenient way to implement the digital filters with arbitrary
> amplitude shape. By "arbitrary" I mean the amplitude may vary within
> +/- 6dB. The application is for sensor calibration. In the
> instrumentation, each sensor may have its own frequency
> characteristics. The variation is FIXED for each sensor. it would be
> nice if we can find a digital filtering method that such variation is
> compensated. A dynamic signal analyzer with DSP would be the ideal
> platform to implement such an algorithm.
>
> The filter must be implemented in the time domain, preferred with
> linear phase.
>
> Jameswww.go-ci.com

This is a textbook Parks-McClellan (Remez) application.

It's well described in any of the good DSP textbooks.

Chris