| Author |
Message |
rover8898
Guest
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 Posted:
Sat Nov 26, 2005 12:53 am Post subject:
FIR Filter limitation (or not?) |
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Hello,
I am quite new to the field of digital FIR filters (I always worked
with the old-fashionned analog ones, but I digress).
There is an important aspect to FIR filters that eludes me:
A low pass digital FIR filter filters out all frequencies in the
stopband (to the set attenuation). This is quite obvious. Now what
remains unclear to me is that if the digital filter can only recover
sucessfully frequencies up to Fs/2 (Fs=sampling frequency), in order to
either filter them if they lie in the stopband or let them pass if they
lie in the passband, does it (the digital FIR filter) also filter out
frequencies above Fs/2 (since they lie in the stopband)? Or, are the
frequency components of above Fs/2 of the input signal not filtered,
but rather emulate something akin to "noise". Basically, if a digital
FIR lowpass filter can only filter out frequencies up to Fs/2, it would
imply that an a simple [20dB/dec lowpass analog filter] would be able
to filter frequencies that a [-100dB FIR lowpass filter with Fs/Fc=1.1]
would not be able to. This seems quite odd as all litterature indicates
that digital FIR filters far out-perform analog filters. Also, if
digital filters (lp, hp, bp, bs) have a severe limitation to their
upper-frequency-of-proper-operation, it would mean that all digital
filters require an analog lowpass filter at their input that can be
tunned on the fly w.r.t Fs, as to remove the frequency components above
Fs/2 prior to the digital filtering. This seems highly unlikely.
What am I not undestanding properly?
Thanks in advance,
-Roger |
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Jerry Avins
Guest
|
| Posted:
Sat Nov 26, 2005 1:13 am Post subject:
Re: FIR Filter limitation (or not?) |
|
|
rover8898 wrote:
| Quote: | Hello,
I am quite new to the field of digital FIR filters (I always worked
with the old-fashionned analog ones, but I digress).
There is an important aspect to FIR filters that eludes me:
A low pass digital FIR filter filters out all frequencies in the
stopband (to the set attenuation). This is quite obvious. Now what
remains unclear to me is that if the digital filter can only recover
sucessfully frequencies up to Fs/2 (Fs=sampling frequency), in order to
either filter them if they lie in the stopband or let them pass if they
lie in the passband, does it (the digital FIR filter) also filter out
frequencies above Fs/2 (since they lie in the stopband)? Or, are the
frequency components of above Fs/2 of the input signal not filtered,
but rather emulate something akin to "noise". Basically, if a digital
FIR lowpass filter can only filter out frequencies up to Fs/2, it would
imply that an a simple [20dB/dec lowpass analog filter] would be able
to filter frequencies that a [-100dB FIR lowpass filter with Fs/Fc=1.1]
would not be able to. This seems quite odd as all litterature indicates
that digital FIR filters far out-perform analog filters. Also, if
digital filters (lp, hp, bp, bs) have a severe limitation to their
upper-frequency-of-proper-operation, it would mean that all digital
filters require an analog lowpass filter at their input that can be
tunned on the fly w.r.t Fs, as to remove the frequency components above
Fs/2 prior to the digital filtering. This seems highly unlikely.
What am I not undestanding properly?
|
There are no frequencies above Fs/2. Any in the original signal should
have been filtered out (with the analog filters that you're familiar
with) before sampling took place, and any that remain will be aliased
(folded back) around Fs/2. As far as the math describing filter
operation (and sampled signals in general) goes, the frequency universe
is closed and finite.
Jerry
--
Engineering is the art of making what you want from things you can get.
ŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻ |
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Carlos Moreno
Guest
|
| Posted:
Sat Nov 26, 2005 1:15 am Post subject:
Re: FIR Filter limitation (or not?) |
|
|
rover8898 wrote:
| Quote: | Hello,
I am quite new to the field of digital FIR filters (I always worked
with the old-fashionned analog ones, but I digress).
There is an important aspect to FIR filters that eludes me:
A low pass digital FIR filter filters out all frequencies in the
stopband (to the set attenuation). This is quite obvious. Now what
remains unclear to me is that if the digital filter can only recover
sucessfully frequencies up to Fs/2 (Fs=sampling frequency), in order to
either filter them if they lie in the stopband or let them pass if they
lie in the passband, does it (the digital FIR filter) also filter out
frequencies above Fs/2 (since they lie in the stopband)?
|
No, it doesn't. They have been already filtered out. A *digital
signal* has only components in the range -Fs/2 to +Fs/2, or rather,
in the angular frequency range -pi to +pi.
The above is independent of anything else -- samples at a rate of
Fs represents one (and only one) signal with bandlimited spectrum;
if you sample a signal with wider bandwidth, then you obtain *a
different* signal (or, a digital signal that represents a different
continuous-time signal)
Or, are the
| Quote: | frequency components of above Fs/2 of the input signal not filtered,
but rather emulate something akin to "noise". Basically, if a digital
FIR lowpass filter can only filter out frequencies up to Fs/2, it would
imply that an a simple [20dB/dec lowpass analog filter]
|
Digital filters don't really work on a 20dB/dec basis -- it's completely
different with digital filters (well, not completely)
| Quote: | to filter frequencies that a [-100dB FIR lowpass filter with Fs/Fc=1.1]
would not be able to. This seems quite odd as all litterature indicates
that digital FIR filters far out-perform analog filters.
|
Well they do (arguably, at least) -- they do give you a better
control, much better precision, as you don't rely on the precision
of physical properties that are affected by the environment (such
as temperature), and that have a *huge* variance. Digital filters
work with numbers and high-precision clock signals -- the result is
a "virtually absolute* precision in the filtering itself.
But as everything in engineering/technology: there is no magic
silver bullet that is perfect in every regard. Digital filters do
suffer from certain "limitations" when compared to analog filters;
one could safely argue that the advantages of digital filters *far*
outweight the disadvantages.
| Quote: | digital filters (lp, hp, bp, bs) have a severe limitation to their
upper-frequency-of-proper-operation, it would mean that all digital
filters require an analog lowpass filter at their input that can be
tunned on the fly w.r.t Fs, as to remove the frequency components above
Fs/2 prior to the digital filtering. This seems highly unlikely.
|
It's not the digital filters -- it's the digitizing process of an
analog signal what requires this filtering; it's not always done
with analog signals; some times it's easier to sample the signal
at a far higher rate, then apply a digital low-pass filter, and
then "sample" the resulting signal (since it is already in the
digital domain, we're talking simply about keeping one every N
samples, or "downsample" by a factor of N)
| Quote: | What am I not undestanding properly?
|
Perhaps you need to better visualize how *discrete-time signals*
work; how does the spectral repersentation works with these signals,
and how that relates to the spectral representation of analog signals.
It seems like your difficulties all lead back to these issues.
HTH,
Carlos
-- |
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Fred Marshall
Guest
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| Posted:
Sat Nov 26, 2005 1:15 am Post subject:
Re: FIR Filter limitation (or not?) |
|
|
"rover8898" <rover8898@hotmail.com> wrote in message
news:1132944801.814391.106900@g43g2000cwa.googlegroups.com...
| Quote: | Hello,
I am quite new to the field of digital FIR filters (I always worked
with the old-fashionned analog ones, but I digress).
There is an important aspect to FIR filters that eludes me:
A low pass digital FIR filter filters out all frequencies in the
stopband (to the set attenuation). This is quite obvious. Now what
remains unclear to me is that if the digital filter can only recover
sucessfully frequencies up to Fs/2 (Fs=sampling frequency), in order to
either filter them if they lie in the stopband or let them pass if they
lie in the passband, does it (the digital FIR filter) also filter out
frequencies above Fs/2 (since they lie in the stopband)? Or, are the
frequency components of above Fs/2 of the input signal not filtered,
but rather emulate something akin to "noise". Basically, if a digital
FIR lowpass filter can only filter out frequencies up to Fs/2, it would
imply that an a simple [20dB/dec lowpass analog filter] would be able
to filter frequencies that a [-100dB FIR lowpass filter with Fs/Fc=1.1]
would not be able to. This seems quite odd as all litterature indicates
that digital FIR filters far out-perform analog filters. Also, if
digital filters (lp, hp, bp, bs) have a severe limitation to their
upper-frequency-of-proper-operation, it would mean that all digital
filters require an analog lowpass filter at their input that can be
tunned on the fly w.r.t Fs, as to remove the frequency components above
Fs/2 prior to the digital filtering. This seems highly unlikely.
What am I not undestanding properly?
Thanks in advance,
-Roger
|
Roger,
When a signal is sampled, its Fourier Transform becomes periodic at the
sampling frequency.
Start with a continuous signal with bandwidth practically less than B.
The signal will have a Fourier Transform with the following properties:
The even part of the signal will have a transform that is real and even.
The odd part of the signal will have a transform that is odd and imaginary.
Both parts will be effectively zero at outside the range -B to +B.
The real part mirrors at zero because it's even.
The imaginary part mirrors and reverses sign at zero because it's odd.
Now, sample this signal:
When it's sampled, the Fourier Transform above repeats at fs, -fs,
2fs, -2fs, .....
That is, it becomes periodic with period fs.
So, you can see if B is greater than fs/2 that sampling (and the periodicity
in frequency it causes) will cause the repeating spectra to overlap and
superimpose - which creates ambiguity or aliasing.
Note that I although the time sequence is sampled, I have not sampled the
Fourier Transform here - it is not a Discrete Fourier Transform. I've kept
it continuous for purposes of discussion and because that seems in order
with the questions you're asking.
Because of the periodicity in the Fourier Transform of any signal, it gives
the appearance that the underlying signal has a bandwidth that is limited to
fs/2. Taking the inverse (continuous) transform can be done with an inverse
computation of a Fourier Series - yielding an infinite set of discrete
coefficients. Doing an infinite summation type of Fourier Transform will
yield the same result. Accordingly, there are no apparent components that
exist outside the interval -fs/2 to fs/2 because the frequency information
just repeats ... and repeats.
We treat the frequency response of a FIR filter in pretty much the same way.
The frequency response is periodic, repeating at intervals of 1/T, the
reciprocal of the unit delay of the filter - which is very often also taken
as fs.
Accordingly, the stop band is *not defined* outside -fs/2 to fs/2. In fact,
it repeats so the response at fs/2 - delta is the same as the response
at -fs/2 - delta and the magnitude response at fs/2 - delta is the same as
the magnitude response at fs/2 + delta (where delta < fs/2).
You can't have a lowpass filter with Fs/Fc=1.1. Fc has to be less than fs/2
and probably not much beyond 0.4fs to make reasonable sense. So fs/fc > 2.x
| Quote: | it would mean that all digital
filters require an analog lowpass filter at their input that can be
tunned on the fly w.r.t Fs, as to remove the frequency components above
Fs/2 prior to the digital filtering.
|
Not exactly. Samplers require the application of an effective analog
lowpass filter ahead of the sampler to avoid aliasing / ambiguity in the
Fourier Transform. Most samplers are at a fixed frequency so there isn't
generally an "on the fly" requirement. After the sampling takes place,
there are samples which can be put into a digital filter without further
concern for antialiasing filtering - unless one is engaged in sample rate
changes / multirate dsp.
To answer your practical question about frequency content above fs/2:
Yes, in the real world the pre-sampler filter won't be perfect and there
will be some components of the signal that alias into [-fs/2 to fs/2] from
above fs/2. You may consider these to be "noise" or have some other
treatment for them..... The job of the pre-sampler filter is to make them
acceptably small.
Fred |
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Guest
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| Posted:
Mon Nov 28, 2005 1:16 am Post subject:
Re: FIR Filter limitation (or not?) |
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|
1) Frequencies above Fs/2 will appear, aliased to other frequencies.
Uusally an antialias filter is used - this MUST be analog, and should
filter out all frequencies above Fs/2 so that they are not a problem.
But in undersampling, you can have a high frequency signal (ie above
Fs/2) that aliases, to a known alias) frequency, and can work on that
quite comfortably. Provided only that the signal has no two frequencies
whose aliases are the same, and that your sample/hold is fast enough to
not itself act as a lowpass filter.
2) I find it unhelpful when people say that 'digital filters are
superior' etc. It depends on your criteria. FIR filters can guarantee
linear phase. Certain types of FIR filter can guarantee to be (almost,
probably) the closest match to what you desire by some meaure (eg LMS).
But sometimes you don't want that.
Try the RIEE analog filters (for gramophones) implemented with a
digital filter - analog ones are often specified with extremely sharp
rolloffs that are really hard to get with digital.
3) Yes, all digital filters that work on signals that come from the
analog world do requir an analog filter at their front end, that
antialiases. You can sometimes sample very much faster than you really
need to, so relaxing the requirements on that analog filter, but you
can't do away with it. And yes, if the digital filter changes its
sample rate then the analog filter should do too. Unless you were
sampling very fast, in which case you amy not care.
Chris
=========================
Chris Bore
BORES Signal Processing
www.bores.com |
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Jerry Avins
Guest
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| Posted:
Mon Nov 28, 2005 9:15 am Post subject:
Re: FIR Filter limitation (or not?) |
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chris_bore@yahoo.co.uk wrote:
...
| Quote: | Try the RIEE analog filters (for gramophones) implemented with a
digital filter - analog ones are often specified with extremely sharp
rolloffs that are really hard to get with digital.
|
What is RIEE? I know what RIAA is, and the recording compensation curve
specified in its name is intended to be minimum phase, best implemented
by two R-C time constants.
....
Jerry
--
Engineering is the art of making what you want from things you can get.
ŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻ |
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Fred Marshall
Guest
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| Posted:
Tue Nov 29, 2005 1:16 am Post subject:
Re: FIR Filter limitation (or not?) |
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|
"rover8898" <rover8898@hotmail.com> wrote in message
news:1133216698.967625.187630@g47g2000cwa.googlegroups.com...
| Quote: | Hello everyone,
OK. So I got the point that the response of a digital filter just
repeats itself at a period of Fs. Also it seems that a digital signal
has only frequency components from -Fs/2 to +Fs/2. If I undestood
correctly, this is because all signals when digitized, are, in the
frequency domain a series of the frequency spectrum of the undigitized
signal interspaced at intervals of Fs. Depending on the bandwidth of
the undigitized signal and the Fs, absence or presence of aliasing will
be determined.
So basically, if aliasing occurs, the *digital signal* (-Fs/2 to +Fs/2)
will have frequency components of the original signal that are above
Fs/2. That is where those unwanted "high frequencies" end up; they
double back in the relevant [-Fs/2 to +Fs/2] frequency range if there
is aliasing .If there isn't any aliasing, it implies that the input
signal was already properly cleaned up. And if the input signal
contains frequencies components up to Fbw, and we decide to sample the
input signal at Fbw/10 because 95% of the relevant signal lies under
Fbw/30, we would be commiting a huge no-no for the frequency components
from [Fbw/20 to Fbw] will fold back and corrupt the retrieved data,
despite these components being quite weak in strength.Rigth ?
A low pass digital FIR filter whose [Fpassband=0.45*Fs and
Fstopband=0.47*Fs] serves no practical purpose if its purpose for being
is to attenuate high frequencies. Right?
It will supress only 6-10% of the input frequencies (depending on the
transitions bands). In other words:
the [0-0.45 0.55-1.45 1.55-2.45 2.55-3.45....]*Fs frequencies of the
undigitized input signal will pass unhindered through the digital
lowpass FIR filter. It certaintly not a viable antialiasing filter.
Rigth?
-Roger
|
Right. Good study. The antialiasing filtering needs to be done
pre-sampling. If you're doing sample rate conversion, then you could well
*also* need to do antialiasing filtering in the discrete time world - and
*there* you use digital filters.
Fred |
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rover8898
Guest
|
| Posted:
Tue Nov 29, 2005 1:16 am Post subject:
Re: FIR Filter limitation (or not?) |
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Hello everyone,
OK. So I got the point that the response of a digital filter just
repeats itself at a period of Fs. Also it seems that a digital signal
has only frequency components from -Fs/2 to +Fs/2. If I undestood
correctly, this is because all signals when digitized, are, in the
frequency domain a series of the frequency spectrum of the undigitized
signal interspaced at intervals of Fs. Depending on the bandwidth of
the undigitized signal and the Fs, absence or presence of aliasing will
be determined.
So basically, if aliasing occurs, the *digital signal* (-Fs/2 to +Fs/2)
will have frequency components of the original signal that are above
Fs/2. That is where those unwanted "high frequencies" end up; they
double back in the relevant [-Fs/2 to +Fs/2] frequency range if there
is aliasing .If there isn't any aliasing, it implies that the input
signal was already properly cleaned up. And if the input signal
contains frequencies components up to Fbw, and we decide to sample the
input signal at Fbw/10 because 95% of the relevant signal lies under
Fbw/30, we would be commiting a huge no-no for the frequency components
from [Fbw/20 to Fbw] will fold back and corrupt the retrieved data,
despite these components being quite weak in strength.Rigth ?
A low pass digital FIR filter whose [Fpassband=0.45*Fs and
Fstopband=0.47*Fs] serves no practical purpose if its purpose for being
is to attenuate high frequencies. Right?
It will supress only 6-10% of the input frequencies (depending on the
transitions bands). In other words:
the [0-0.45 0.55-1.45 1.55-2.45 2.55-3.45....]*Fs frequencies of the
undigitized input signal will pass unhindered through the digital
lowpass FIR filter. It certaintly not a viable antialiasing filter.
Rigth?
-Roger |
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rover8898
Guest
|
| Posted:
Tue Nov 29, 2005 1:16 am Post subject:
Re: FIR Filter limitation (or not?) |
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Hello,
| Quote: | It's not the digital filters -- it's the digitizing process of an
analog signal what requires this filtering; it's not always done
with analog signals; some times it's easier to sample the signal
at a far higher rate, then apply a digital low-pass filter, and
then "sample" the resulting signal (since it is already in the
digital domain, we're talking simply about keeping one every N
samples, or "downsample" by a factor of N)
|
The filtering can be done with a [oversampling at higher sample rate] +
[digital low pass filter] +[downsampling] scheme, I guess. But if the
input signal (prior to A/D) has frequency components beyond the
[oversampling rate/2] threshold, then there will aliasing, digital
lowpass filter or not. Is there a reason why a precautionary ~broad
analog filter cannot be placed ahead of the A/D (aside from cost and
maybe gadget size) ?
-Roger |
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Jerry Avins
Guest
|
| Posted:
Tue Nov 29, 2005 1:16 am Post subject:
Re: FIR Filter limitation (or not?) |
|
|
rover8898 wrote:
| Quote: | Hello,
It's not the digital filters -- it's the digitizing process of an
analog signal what requires this filtering; it's not always done
with analog signals; some times it's easier to sample the signal
at a far higher rate, then apply a digital low-pass filter, and
then "sample" the resulting signal (since it is already in the
digital domain, we're talking simply about keeping one every N
samples, or "downsample" by a factor of N)
The filtering can be done with a [oversampling at higher sample rate] +
[digital low pass filter] +[downsampling] scheme, I guess. But if the
input signal (prior to A/D) has frequency components beyond the
[oversampling rate/2] threshold, then there will aliasing, digital
lowpass filter or not. Is there a reason why a precautionary ~broad
analog filter cannot be placed ahead of the A/D (aside from cost and
maybe gadget size) ?
|
Aliasing from frequencies above the [oversampling rate/2] threshold
won't hurt if the aliases are above the downsampled Fs/2, as those will
be removed by the digital downsampling (decimation) filter. A
"precautionary ~broad analog filter" should be placed ahead of the A.D
in most cases. In Servo work, delay is often more harmful than aliasing,
so the antialias filter is omitted.
Jerry
--
Engineering is the art of making what you want from things you can get.
ŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻ |
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Carlos Moreno
Guest
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| Posted:
Tue Nov 29, 2005 9:16 am Post subject:
Re: FIR Filter limitation (or not?) |
|
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rover8898 wrote:
| Quote: | Hello,
It's not the digital filters -- it's the digitizing process of an
analog signal what requires this filtering; it's not always done
with analog signals; some times it's easier to sample the signal
at a far higher rate, then apply a digital low-pass filter, and
then "sample" the resulting signal (since it is already in the
digital domain, we're talking simply about keeping one every N
samples, or "downsample" by a factor of N)
The filtering can be done with a [oversampling at higher sample rate] +
[digital low pass filter] +[downsampling] scheme, I guess. But if the
input signal (prior to A/D) has frequency components beyond the
[oversampling rate/2] threshold, then there will aliasing, digital
lowpass filter or not. Is there a reason why a precautionary ~broad
analog filter cannot be placed ahead of the A/D (aside from cost and
maybe gadget size) ?
|
The technique of sampling at a much higher frequency and then do
a low-pass digital filtering does not replace the analog filter;
it just makes the analog filter trivial -- if you know that your
signal has valuable spectral contents up to 20 kHz and want to
sample at 44.1 kHz, the analog filter required would have to be
very precise and very "wall-like" -- it would be quite hard to
design an analog filter with such a "wall" frequency response
and without truly ugly phase distorsion (mostly at frequencies
near the cutoff).
So, instead, if you sample at, say, 8 times the intended rate
(i.e., at 8 x 44.1 kHz), then a very simple, perhaps first-
order RC filter with cutoff at 50 kHz would do a more-than-
excellent job, since now you only need to worry to eliminate
frequencies above 4x44.1, or approx. 170 kHz -- that's trivial,
since the RC already has a good attenuation at that point,
and also, the audio signal really has very low contents at
those ultra-high frequencies.
The part that you really were worried about -- a wall-like
cutoff above 20 but below 22, that you get with a digital
filter with nice phase response.
Notice, however, that I'm not describing a universal technique
that is applied unconditionally in every design -- it's just
that it may be very practical and easy, so you do encounter it
quite often -- for speech, for instance, where you want a
sampling rate of 8 kHz (typically), it's quite easy to do the
sampling at a higher rate and then bring it down after it is
in the digital domain.
HTH,
Carlos
-- |
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Jerry Avins
Guest
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| Posted:
Tue Nov 29, 2005 5:16 pm Post subject:
Re: FIR Filter limitation (or not?) |
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rover8898 wrote:
| Quote: | Jerry,
Yes you are rigth. Aliasing ocurring above the downsampled Fs/2 should
be moot for it will removed later with the decimation digital filter.
However, (ever being accused of being too cautious), one can never
predict ALL parasitics that will find their way onto the input signal,
even if only sligthly. The VERY high frequency parasitics (RF and
above) are usually difficult to predict; depends on environmental
geometry and such ... .Therefore, since at least in theory, all high
frequency components should fold back and thus cause aliasing ...well,
I doubt that we will be lucky enough that all parasitics will fold back
only between [downsampled Fs/2] and [oversampling Fs/2].
Hence, this leads me back to my precautionnary "broad analog filter"
solution (ahead of A/D) that would (at least in theory) filter the very
high unforseen frequencies of the input signal that would cause at
least some minor aliasing (how minor? probably below 1%, but better
safe than sorry I say ).
I would think that in today's world of cell phone, sattelite, radio,
wireless gadgets ...., one would "clean" a signal as much as possible,
cost-permitting obviously.
As for Servo work concerning the harmfull effect of the delay of an
analog antialiasing filter, well I would have to look into that for I
am sure there is merit into that as well.
|
Roger,
Engineering is compromise. No filter is perfect. Alias components
significantly smaller than one LSB are generally harmless, and going to
great lengths to make them smaller can degrade other aspects of a
design. "Better safe than sorry" is a fine motto, but one must not put
oneself in the position of being sorry for trying to be too safe.
Jerry
--
You know that the outhouse is in the right place if öżö
it seems too close in summer and too far in winter. ş
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rover8898
Guest
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| Posted:
Tue Nov 29, 2005 5:16 pm Post subject:
Re: FIR Filter limitation (or not?) |
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| Quote: | but one must not put oneself in the position of being sorry for trying to be too safe
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Been there. Done that. :-).
Ok. Gotcha. Have to learn to comprise between solutions for potential
problems and solutions for problable problems.
-Thanks
Roger |
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rover8898
Guest
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| Posted:
Tue Nov 29, 2005 5:17 pm Post subject:
Re: FIR Filter limitation (or not?) |
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Jerry,
Yes you are rigth. Aliasing ocurring above the downsampled Fs/2 should
be moot for it will removed later with the decimation digital filter.
However, (ever being accused of being too cautious), one can never
predict ALL parasitics that will find their way onto the input signal,
even if only sligthly. The VERY high frequency parasitics (RF and
above) are usually difficult to predict; depends on environmental
geometry and such ... .Therefore, since at least in theory, all high
frequency components should fold back and thus cause aliasing ...well,
I doubt that we will be lucky enough that all parasitics will fold back
only between [downsampled Fs/2] and [oversampling Fs/2].
Hence, this leads me back to my precautionnary "broad analog filter"
solution (ahead of A/D) that would (at least in theory) filter the very
high unforseen frequencies of the input signal that would cause at
least some minor aliasing (how minor? probably below 1%, but better
safe than sorry I say ).
I would think that in today's world of cell phone, sattelite, radio,
wireless gadgets ...., one would "clean" a signal as much as possible,
cost-permitting obviously.
As for Servo work concerning the harmfull effect of the delay of an
analog antialiasing filter, well I would have to look into that for I
am sure there is merit into that as well.
-Roger |
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