cutoff and sampling frequency

Hai,

Basically i am little bit confused with FIR filter cutoff frequency
and sampling frequency relation....Both the terms are very much
familier to me separately but i dont know how to relate each other??

Few standard sampling frequencies for audio applications are
8,44.1,48,96,192Khz...For this sampling frequencies wat should be the
minimum cutoff frequency for LP,HP,BP,BR FIR filter....

example fs=8khz...fc=???

regards,
faz

Hai,

Continuation to the above questions....As per sampling theorem fs>2fm
(where fm-maximum input freq.) to avoid aliasing...

Should i have to follow this relation for cutoff frequency also ....i
mean fs>2fc(where fc is the cutoff frequency) or in wat way i should
assume the sampling frequency for any specific cutoff frequency will
they related or not??

regards,
faz


On Jun 1, 4:35&#4294967295;pm, faz <fazulu.v...@gmail.com> wrote:
> Hai,
>
> Basically i am little bit confused with FIR filter cutoff frequency
> and sampling frequency relation....Both the terms are very much
> familier to me separately but i dont know how to relate each other??
>
> Few standard sampling frequencies for audio applications are
> 8,44.1,48,96,192Khz...For this sampling frequencies wat should be the
> minimum cutoff frequency for LP,HP,BP,BR FIR filter....
>
> example fs=8khz...fc=???
>
> regards,
> faz

faz wrote:
> Hai,
> 
> Basically i am little bit confused with FIR filter cutoff frequency
> and sampling frequency relation....Both the terms are very much
> familier to me separately but i dont know how to relate each other??
> 
> Few standard sampling frequencies for audio applications are
> 8,44.1,48,96,192Khz...For this sampling frequencies wat should be the
> minimum cutoff frequency for LP,HP,BP,BR FIR filter....
> 
> example fs=8khz...fc=???

fc = any frequency up to fs/2.

Consider only signals that extend to (or very near to) DC.

Any sample rate is possible provided that it exceeds twice the highest 
frequency if the signal. That requirement is often met by using an 
analog low-pass filter before the signal is digitized.

The designer chooses the cutoff frequency of a filter to suit his 
purpose. It is a fraction of the sample rate, so that the same filter 
used in a system with, say, twice the sample rate will have double the 
cutoff frequency.

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

faz wrote:
> Hai,
> 
> Continuation to the above questions....As per sampling theorem fs>2fm
> (where fm-maximum input freq.) to avoid aliasing...
> 
> Should i have to follow this relation for cutoff frequency also ....i
> mean fs>2fc(where fc is the cutoff frequency) or in wat way i should
> assume the sampling frequency for any specific cutoff frequency will
> they related or not??

For a sample frequency fs, all components of the signal are less than 
fs/2. Of course, the cutoff frequency and all other significant 
frequencies also need to be less than fs/2 in order to make sense.

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

faz wrote:
> Hai,
> 
> Basically i am little bit confused with FIR filter cutoff frequency
> and sampling frequency relation....Both the terms are very much
> familier to me separately but i dont know how to relate each other??
> 
> Few standard sampling frequencies for audio applications are
> 8,44.1,48,96,192Khz...For this sampling frequencies wat should be the
> minimum cutoff frequency for LP,HP,BP,BR FIR filter....
> 
> example fs=8khz...fc=???
> 
> regards,
> faz

It's more complicated than that.  The term "Cut off frequency" makes it 
sound like the signal will be, well, cut off.  In reality a filter's 
stated cut off frequency may often be where the signal is a mere 3dB 
down from the filter's nominal pass frequency value.

In fact, no finite-length, finite delay filter will give you "perfect" 
cut off -- the best you can ever do is reach a compromise where the 
filter attenuates the signals enough at the aliasing frequencies that 
your sound is adequate -- and different people, in different situations, 
have different definitions of "adequate".

Read this, see if it helps you out: 
http://www.wescottdesign.com/articles/Sampling/sampling.html.

-- 

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

Do you need to implement control loops in software?
"Applied Control Theory for Embedded Systems" gives you just what it says.
See details at http://www.wescottdesign.com/actfes/actfes.html

On Jun 1, 5:37 am, faz <fazulu.v...@gmail.com> wrote:
> Continuation to the above questions....As per sampling theorem fs>2fm
> (where fm-maximum input freq.) to avoid aliasing...
>
> Should i have to follow this relation for cutoff frequency also ....i
> mean fs>2fc(where fc is the cutoff frequency) or in wat way i should
> assume the sampling frequency for any specific cutoff frequency will
> they related or not??

Filters which can be constructed at reasonable cost or operate
with a finite time delay are not infinitely sharp.  Therefore
you have to leave a gap between 2fc and fs which depends on
the steepness (and other characteristics) of the filter you
choose, or else you will have aliasing and/or distortion near
the filter's skirt.


IMHO. YMMV.

Ron N wrote:
> On Jun 1, 5:37 am, faz <fazulu.v...@gmail.com> wrote:
>> Continuation to the above questions....As per sampling theorem fs>2fm
>> (where fm-maximum input freq.) to avoid aliasing...
>>
>> Should i have to follow this relation for cutoff frequency also ....i
>> mean fs>2fc(where fc is the cutoff frequency) or in wat way i should
>> assume the sampling frequency for any specific cutoff frequency will
>> they related or not??
> 
> Filters which can be constructed at reasonable cost or operate
> with a finite time delay are not infinitely sharp.  Therefore
> you have to leave a gap between 2fc and fs which depends on
> the steepness (and other characteristics) of the filter you
> choose, or else you will have aliasing and/or distortion near
> the filter's skirt.
> 
> 
> IMHO. YMMV.

I know what you mean but, for the record, the gap is between fc and fs/2.

Regards,
John

John Monro wrote:
> Ron N wrote:
(snip)

>> Filters which can be constructed at reasonable cost or operate
>> with a finite time delay are not infinitely sharp.  Therefore
>> you have to leave a gap between 2fc and fs which depends on
>> the steepness (and other characteristics) of the filter you
>> choose, or else you will have aliasing and/or distortion near
>> the filter's skirt.

> I know what you mean but, for the record, the gap is between fc and fs/2.

For a digital filter, it is only the ratio of the filter frequencies
to the sample rate that matters.  For analog filters, the filter
complexity (order) depends on the ratio of the cutoff rate to the
frequency.   Since most CD players use oversampling and digital
filters, it is more likely 16fc and 8fs.  (Or whatever the
current numbers are.)

-- glen

glen herrmannsfeldt wrote:
> John Monro wrote:
>> Ron N wrote:
> (snip)
> 
>>> Filters which can be constructed at reasonable cost or operate
>>> with a finite time delay are not infinitely sharp.  Therefore
>>> you have to leave a gap between 2fc and fs which depends on
>>> the steepness (and other characteristics) of the filter you
>>> choose, or else you will have aliasing and/or distortion near
>>> the filter's skirt.
> 
>> I know what you mean but, for the record, the gap is between fc and fs/2.
> 
> For a digital filter, it is only the ratio of the filter frequencies
> to the sample rate that matters.  For analog filters, the filter
> complexity (order) depends on the ratio of the cutoff rate to the
> frequency.   Since most CD players use oversampling and digital
> filters, it is more likely 16fc and 8fs.  (Or whatever the
> current numbers are.)
> 
> -- glen
> 

Glen,I can't grasp the point you are making here.  It appears that you 
are saying that the filter order is 16fc and 8fs, which is clearly not 
correct.

Regards,
John

John Monro wrote:
(snip)

>> For a digital filter, it is only the ratio of the filter frequencies
>> to the sample rate that matters.  For analog filters, the filter
>> complexity (order) depends on the ratio of the cutoff rate to the
>> frequency.   Since most CD players use oversampling and digital
>> filters, it is more likely 16fc and 8fs.  (Or whatever the
>> current numbers are.)

> Glen,I can't grasp the point you are making here.  It appears that you 
> are saying that the filter order is 16fc and 8fs, which is clearly not 
> correct.

Yes, I didn't explain it very well.  It is usual in CD players
to use digital filters to generate a signal with a much higher
sampling rate, and use analog filters on the result.
Described in terms of the original sample rate, fs, the
filter cutoff will be much higher.  I don't know what the
current state-of-the-art is for CD players, though.

-- glen