nyquist sampling rate

On 2009-05-06, aitezaz.abd@gmail.com <aitezaz.abd@gmail.com> wrote:
> Now, the next question that i would like to ask is how much
> oversampled a signal should be (as a rule of thumb) so that we can
> make practically feasible anti aliasing filters(subject to current
> technology of analog filters).

A better question is to compare the cost (in terms of board area, parts
cost, design time, manufacturability, etc) of a better analog filter
versus a faster ADC and more digital processing capability.  Remember
that you only have to process the data at the input rate enough to
digitally lowpass filter it and decimate it down to the actual bandwidth.

To answer your original question look at any article on analog filters
and the rolloff of filters as a function of filter order.

-- 
Ben Jackson AD7GD
<ben@ben.com>
http://www.ben.com/

aitezaz.abd@gmail.com wrote:
> On May 6, 5:41 am, Ben Jackson <b...@ben.com> wrote:
>> On 2009-05-05, aitezaz....@gmail.com <aitezaz....@gmail.com> wrote:
>>
>>> i have a signal of bandwidth 2 MHz on a carrier of 5 MHz. That means
>>> the Nyquist sampling rate is (5+2/2)*2 = 12 MHz. But I have heard that
>>> it is not a good idea to sample on or near nyquist rate
>> I asked a very similar question a while ago, only in the context of a
>> DAC.  In the same way that you quickly rejected "4MHz" as your sampling
-- snip --
> Now, the next question that i would like to ask is how much
> oversampled a signal should be (as a rule of thumb) so that we can
> make practically feasible anti aliasing filters(subject to current
> technology of analog filters).

There aren't any good ones.  Between what you might want to do, what is 
available, what your skills are, what the skills of the folks you're 
working with, whether you're trying to optimize for engineering cost or 
manufacturing cost -- there's too many widely-varying good answers for 
there to be a rule of thumb.

It's work.  Understand the problem, understand what's available, 
understand how to wrangle the math, circuits and software to get what 
you want, then assess what works for your particular problem -- and keep 
in mind that it may be different for the next product, or the next 
production cycle of the existing product.

-- 

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

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

I am assuming that you are receiving only one carrier (unlike a FM
receiver, where you receive many different carriers for different
stations).

Did you translate your message spectrum from 0 Hz to 5 Mhz while
modulating it?

If so, then to get your message signal back, you (ideally) need to sample
it at 2*2 = 4 Mhz.(and not twice the carrier+message BW) Ant then pass it
through an ideal Low Pass Filter of cutoff = 1Mhz. The reason being that
you will get other images of message spectrum as well after sampling, but
they will not ALIAS as long as you sample at twice their bandwidth.

Please correct me if I'm wrong.

Thank you. 

>On May 6, 5:41=A0am, Ben Jackson <b...@ben.com> wrote:
>> On 2009-05-05, aitezaz....@gmail.com <aitezaz....@gmail.com> wrote:
>>
>> > i have a signal of bandwidth 2 MHz on a carrier of 5 MHz. That means
>> > the Nyquist sampling rate is (5+2/2)*2 =3D 12 MHz. But I have heard
tha=
>t
>> > it is not a good idea to sample on or near nyquist rate
>>
>> I asked a very similar question a while ago, only in the context of a
>> DAC. =A0In the same way that you quickly rejected "4MHz" as your
sampling
>> rate because your data is not perfectly bandpass filtered you can
reject
>> 12MHz because your data is not perfectly lowpass filtered. =A0Whatever
>> analog filter you have on your input has a transition band.
=A0Wherever
>> the rolloff of that filter crosses the limit of your tolerance for
>> aliasing is the true bandwidth of your input (it's a little more
tricky
>> than that because the first bit of noise above Nyquist folds back onto
>> the extra bit of sampling bandwidth you don't care about anyway).
>>
>> --
>> Ben Jackson AD7GD
>> <b...@ben.com>http://www.ben.com/
>
>Thanks for replies guys.
>Yes Ben you got it right and I have found the same answer on
>http://www.dspdesignline.com/howto/207500439
>We cannot sample the signal on twice the maximum frequency present in
>the signal because in this case the anti aliasing filter would be
>impossible to realize. Instead, we should sample it at twice the ws
>where ws is the starting frequency of stop band.
>Now, the next question that i would like to ask is how much
>oversampled a signal should be (as a rule of thumb) so that we can
>make practically feasible anti aliasing filters(subject to current
>technology of analog filters).
>
>Thanks for your time
>