rhnlogic@yahoo.com wrote:
> Jerry Avins wrote:
> 
>>rhnlogic@yahoo.com wrote:
>>
>>>Is there any rule-of-thumb on what the phase response needs to look
>>>like near any artificially sharp transition in a frequency domain
>>>filter in order to reduce ripple (caused by the sharp transition)
>>>in the continuous frequency response, if that's at all possible?
>>
>>We both know that linear-phase filters can be made arbitrarily sharp,
>>so I don't understand what you're asking for.
> 
> 
> I was refering not to arbitrarily sharp filters, but to artificially
> sharp filters, such as the one the OP was using which quantizes
> the magnitude response to either 1.0 or zero (obvious not a flat
> filter).  My question was about whether allowing degrees of freedom
> in the phase response would help anything (assuming more than 1 bin
> of 1.0 coefficients, or course).  e.g. would his box filter get
> any "better" if he removed all linear phase restrictions?

Those filters aren't arbitrarily sharp. They can't be sharper than the 
bin spacing, even in one's dreams. A plot of the continuous frequency 
response shows poor attenuation between the zeroed-out bins and between 
the unity-gain bins near the transition. Using a longer FFT to put the 
bins closer together raises the frequency of the Gibbs oscillations, 
just as adding more frequency terms to a synthesized square wave does, 
but it doesn't affect the amplitudes.

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

Jerry Avins wrote:
> rhnlogic@yahoo.com wrote:
> > Is there any rule-of-thumb on what the phase response needs to look
> > like near any artificially sharp transition in a frequency domain
> > filter in order to reduce ripple (caused by the sharp transition)
> > in the continuous frequency response, if that's at all possible?
>
> We both know that linear-phase filters can be made arbitrarily sharp,
> so I don't understand what you're asking for.

I was refering not to arbitrarily sharp filters, but to artificially
sharp filters, such as the one the OP was using which quantizes
the magnitude response to either 1.0 or zero (obvious not a flat
filter).  My question was about whether allowing degrees of freedom
in the phase response would help anything (assuming more than 1 bin
of 1.0 coefficients, or course).  e.g. would his box filter get
any "better" if he removed all linear phase restrictions?


Thanks.
-- 
rhn A.T nicholson d.O.t C-o-M

tjuii wrote:
>>Why did you ask your original question?
>>
>>Jerry
>>-- 
>>Engineering is the art of making what you want from things you can get.
>>�����������������������������������������������������������������������
>>
> 
> 
> Because I want to know more than I do right now.  Plus, I didn't
> understand what my colleague was talking about when he said that the
> method I was using wasn't a good one.  I still think he was thinking that
> I was using a rectangular window in the time domain.  THAT I do know is
> not a very effective thing to do.
> 
> Thanks all for the discussion/explanations.
> 
> TJ
> 
> Oh, almost forgot.  I asked this question buried in one of my replies
> above, but forgot about.  Is there a rule-of-thumb about how fast to
> sample in comparison to the frequencies of interest (aside from the
> Nyquist issue, i.e. sample at twice the frequency of the frequency of
> interest)?

The sampled signal needs to n]be filtered to meet the Nyquist criterion, 
The sample rate needs to accommodate the nature if the imperfect filter. 
25% oversampling is marginal. For servo systems that need low lag, 5x 
oversampling is good and more is better. Context governs.

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

rhnlogic@yahoo.com wrote:
> Jerry Avins wrote:
> 
>>tjuii wrote:
>>
>>>>If you are specifying your filter in the frequency domain
>>>>what are you doing about the phase portion of that
>>>>specification?
> 
> ...
> 
>>>I am not doing anything about the phase.  The filter function of zeros and
>>>ones just leaves the phase of the remaining frequencies unchanged when
>>>multiply by the FFT of the signal, right (i.e. both real and imaginary
>>>parts of the FFT are the same before and after the multiplication, for the
>>>remaining freqs.)?  Should I be doing something different with the phase?
>>>What I am doing seems to be working well for my purposes, but I'd love to
>>>know if there is something else that I need to think about.
>>
>>You aren't getting that "infinitely fast roll-off and a perfectly flat
>>pass-band", ...
> 
> 
> Is there any rule-of-thumb on what the phase response needs to look
> like near any artificially sharp transition in a frequency domain
> filter in order to reduce ripple (caused by the sharp transition)
> in the continuous frequency response, if that's at all possible?

We both know that linear-phase filters can be made arbitrarily sharp, so 
I don't understand what you're asking for.

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

>Why did you ask your original question?
>
>Jerry
>-- 
>Engineering is the art of making what you want from things you can get.
>�����������������������������������������������������������������������
>

Because I want to know more than I do right now.  Plus, I didn't
understand what my colleague was talking about when he said that the
method I was using wasn't a good one.  I still think he was thinking that
I was using a rectangular window in the time domain.  THAT I do know is
not a very effective thing to do.

Thanks all for the discussion/explanations.

TJ

Oh, almost forgot.  I asked this question buried in one of my replies
above, but forgot about.  Is there a rule-of-thumb about how fast to
sample in comparison to the frequencies of interest (aside from the
Nyquist issue, i.e. sample at twice the frequency of the frequency of
interest)?

		
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Jerry Avins wrote:
> tjuii wrote:
> >>If you are specifying your filter in the frequency domain
> >>what are you doing about the phase portion of that
> >>specification?
...
> > I am not doing anything about the phase.  The filter function of zeros and
> > ones just leaves the phase of the remaining frequencies unchanged when
> > multiply by the FFT of the signal, right (i.e. both real and imaginary
> > parts of the FFT are the same before and after the multiplication, for the
> > remaining freqs.)?  Should I be doing something different with the phase?
> > What I am doing seems to be working well for my purposes, but I'd love to
> > know if there is something else that I need to think about.
>
> You aren't getting that "infinitely fast roll-off and a perfectly flat
> pass-band", ...

Is there any rule-of-thumb on what the phase response needs to look
like near any artificially sharp transition in a frequency domain
filter in order to reduce ripple (caused by the sharp transition)
in the continuous frequency response, if that's at all possible?


Thanks.
-- 
rhn A.T nicholson D.o.T c-O-m

tjuii wrote:
>>If you are specifying your filter in the frequency domain 
>>what are you doing about the phase portion of that 
>>specification?
>>
>>
>>Bob
>>-- 
> 
> 
> 
> I am not doing anything about the phase.  The filter function of zeros and
> ones just leaves the phase of the remaining frequencies unchanged when
> multiply by the FFT of the signal, right (i.e. both real and imaginary
> parts of the FFT are the same before and after the multiplication, for the
> remaining freqs.)?  Should I be doing something different with the phase? 
> What I am doing seems to be working well for my purposes, but I'd love to
> know if there is something else that I need to think about.

You aren't getting that "infinitely fast roll-off and a perfectly flat 
pass-band", but if it's working well for your purposes, keep it up. Why 
did you ask your original question?

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

>If you are specifying your filter in the frequency domain 
>what are you doing about the phase portion of that 
>specification?
>
>
>Bob
>-- 


I am not doing anything about the phase.  The filter function of zeros and
ones just leaves the phase of the remaining frequencies unchanged when
multiply by the FFT of the signal, right (i.e. both real and imaginary
parts of the FFT are the same before and after the multiplication, for the
remaining freqs.)?  Should I be doing something different with the phase? 
What I am doing seems to be working well for my purposes, but I'd love to
know if there is something else that I need to think about.

Thanks for all your help, everyone.
		
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Bob Cain wrote:

   ...

> "Bin" is a terrible misnomer for frequency domain samples which 
> continues to persist for some reason.

I agree, but I use it for lack of a better concise term. "Sample of the 
Underlying Continuous Spectrum" could give us SUCS, but people would 
construe it as plural. Besides, that's not the whole story except in 
filters.

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

Bob Cain wrote:

..
> "Bin" is a terrible misnomer for frequency domain samples which 
> continues to persist for some reason.
> 
..

freqles?

Richard Dobson