Nyquist
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A.E lover - 20:19 09-08-07
Hi all,
(1) I know the famous Nyquist Condition, f(t) with bandwidth B is
sampled without aliasing if Fs> 2B.
Today I read another thing called Nyquist Condition which says:
for a continuous time signal x(t), take x(t) convolute with itself and
then sample the obtained signal at Fs,
...
On 2 Apr., 12:36, Randy Yates wrote:
> dbd writes:
> > [...]
> > .How can a 'properly designed' decimation filter not anti-alias when
> > .there is input signal content above the Nyquist frequency of the
> > .downsampled sampling frequency?
>
> Dale,
>
> Here's an interes...
On Wed, 2 Apr 2008 04:12:06 -0700 (PDT), Andor
wrote:
> On 2 Apr., 12:36, Randy Yates wrote:
> > dbd writes:
> > > [...]
> > > .How can a 'properly designed' decimation filter not anti-alias when
> > > .there is input signal content above the Nyquist frequency of the
> > > ...
11:17 17-08-05
Hello,
I read in the text book ("Digital signal Processing Principles,
Algorithm and APplications" J. G. Proakis % D. G. Manolakis, page 30)
that the Nyquist frequency (rate) is a double of the highest frequency
of the signal.
In the web, I learn that it is the double of the bandwith. Moreover ...
abariska@student.ethz.ch wrote:
> > > Since Fs = Ts = 1, I don't see a difference between sinc(t+1/7) and
> > > sinc(n+1/7) ?
> >
...
> You claim that the infinitely long sequence
>
> b[n] = sinc(n + 1/7),
>
> for all n, is linear-phase. That could be true. Do you have an ide...
I (vaguely) heard that sampling complex-valued data does not abide by the
Nyquist rate criteria, i.e., the sampling rate fs can go lower than Nyquist
rate and it still can avoid aliasing and reconstruct perfectly...
Is that true? Any theory behind it?
Thanks a lot
...
RobR - 09:17 14-11-07
Hello,
there is a question that bugs me for quite a long time:
You can read about Nyquist constrain online, that to reconstruct all
frequencies within a signal, it has to be sampled with at least twice the
bandwidth _or_ maximum frequency.
Maybe this _or_ is already the problem...
Let's se...
It is generally credited that the Sampling Theorem is due to fistly
the Mathematician Whittaker and Shannon and the Russian Kotelnikov. I
have no doubt that Whittaker was first but was Shannon aware of
Whittakers work? Also where does the Russian engineer fit in? What
role did Nyquist play. We t...
On Fri, 4 Apr 2008 10:58:49 -0700 (PDT), robert bristow-johnson
wrote:
(snipped by Lyons)
>
> possibly you're right. but i think the term is nearly universally
> understood as Fs/2 (or z = -1 on the unit circle or pi in normalized
> omega, but not "1" as stupid MATLAB scales it) and i...
Don Pearce - 10:30 25-08-07
On Sat, 25 Aug 2007 06:16:10 -0800, floyd@apaflo.com (Floyd L.
Davidson) wrote:
> >
> > I guessed you would think it was correct. You can't sample at a rate
> > equal to twice the frequency you are sampling. The wanted signal has
> > collided with its image and you can't disambiguate them. T...
Don Pearce - 02:33 21-08-07
On Mon, 20 Aug 2007 22:14:45 -0800, floyd@apaflo.com (Floyd L.
Davidson) wrote:
> > source that you claim is authoritative and impeccable. Kindly go and
> > read what it has to say on the Nyquist rate and come back and repeat
> > that claim without blushing. Actually I'm betting you won't blus...
W. Dale Hall - 01:59 12-08-07
Luna Moon wrote:
> Hi all,
>
> If I have the signal not in closed form, but in form of some
> collected data.
>
> The sampling(or collection) of such signal data points are very
> costly.
>
> Thus we want to minimize the number of samples(data collection).
>
> Is there a wa...
Hello All,
I had a recent situation where I needed to write a paper explaining the
why's and wherefores of Johnson noise. So if you are interested, the
following link will take you to my paper.
http://www.claysturner.com/dsp/Johnson-Nyquist%20Noise.pdf
Any and all comments welcome.
...
Is there a Nyquist sampling theory for spatial dimensions?
Suppose point A to point B is about 1 mile, how many sensors I should place
to get a fair estimation of the property of the whole range?
...
Floyd L. Davidson wrote:
> Jerry Avins wrote:
> > Floyd L. Davidson wrote:
> >
> > ...
> >
> > > Nyquist rate:
> > > The reciprocal of the Nyquist interval, i.e., the
> > > minimum theoretical sampling rate that fully
> > > describes a given signal, i.e., ena...
Hello all,
I noticed that all FFT routines transform a time array into a
frequency array of the following form:
F[0], F[n/2], real:F[1], imag:F[1], real:F[2], imag:F[2], etc..
Now the question is, why does "F[n/2]" (the nyquist freq) appear as the
second term, and more importantly, what i...
Hello all,
I am really confused with how people calculate the spatial feq values
in FFT:
Assume I have a continuous signal f(x), in spatial domain. I need to
know at what freq (cycles/inch) the flactuation is the most dominant
(highest). Assume that freq is f_d.
Firstly I sample it wit...
Greg Berchin - 12:31 24-08-07
> When is decimating by N not equivalent to decimating by 2*N followed
> by interpolating by 2?
I think, except for a handful of pathological cases, any time
bandwidth > = Fs/4N.
For example, try it with a sine wave (NOT a cosine wave) of frequency
Fs/4N. Decimate by N and Nyquist is sti...
I need a digital filter or sequence of digital filters with the
following
response:
Amplitude flat across all frequencies up to the Nyquist (folding)
frequency.
Phase starts at 0 degrees at low fregs linearly increases with
frequency until
it is 180 at the Nyquist (folding) frequency.
...
Hello,
I have a signal consisting of 4 harmonics (200k,400k,600k,and 800k Hz)
and dc component.The signal is very pure and SNR better than 60 dB. I
have to sample it in sub Nyquisit rate(lesser than 1600k). What
sampling rate should i choose so that there no alaising. I cannot
chose higher samp...
> Consider your idee fixee: if every second sample of a set of valid
> samples is discarded, the result is still a set of valid samples. How
> many times in a row would you apply that theorem? Why stop there?
I'm not saying it's "the same". If I said that then I expressed myself wrong.
What...
I understand Nyquist specifying a minimum sampling rate to determine
the high frequency component of a signal.
What happens at at the other end of the spectrum?
I.E. Is there a minimum time window required?
E.G. If the signal has a significant 1 Hz component and sample window
was .1 sec...
VelociChicken - 13:02 01-03-08
Hello, I'm currently using an FFT to zereo out all the frequences above half
Nyquist for my application. It makes perfect cutoff, but I have to use
buffers and the old overlap/add to remove clicks.
My question to you good folks - is there a clever 'trick' to cut frequencies
off above this s...
RF - 15:44 12-01-08
I'd like to ask a couple very basic questions. Being hands-on rather than
mathematically-focused I'm trying to visualize sampling in a project I'm
planning to start.
1. By sampling say a 3.1 kHz band-limited voice channel at the Nyquist rate
am I guaranteed to capture *all* the information...
I've seen a lot of posts over the last year or so that indicate a lack
of understanding of the implications of the Nyquist theory, and just
where the Nyquist rate fits into the design of sampled systems.
So I decided to write a short little article to make it all clear.
It's a little longe...
14:01 29-06-05
Lets say I have a filter with a cutoff at 100Hz and my sampling rate is
1kHz. This means my cutoff is at 0.1 in normalized frequency terms. But
it seems that the matlab freqz function plots my cutoff as being at
0.2, because it defines normalized as being relative to the nyquist. Is
this not rea...
bulk - 05:46 12-09-06
A very basic question:
Take two properly sampled signals (more than nyquist). Now I mulitply
the two sample streams. From a continous time view point it is easy to
see that the product could have frequencies for which the initial
sampling rate wouldnt be enough.
So in any DSP system is one ...
Ted wrote:
> Dear Group,
>
> I have an elementary question. If a sine-wave is sampled such that the
> samples fall at the times when the value of the wave is zero (meaning
> at 0 and pi).
>
> The sampling frequency thus is twice the frquency of the wave. Is this
> proper sampling...
To all you DSP mavens out there.
I am looking for some references to distortions introduced by sampling =
errors,
especially sampling close to the Nyquist criterion. In all the =
literature I
have seen, reconstruction of a waveform is guaranteed if the sampling =
meets the
Nyquist criterio...
Ron N. wrote:
> On Aug 1, 9:16 pm, Jerry Avins wrote:
(snip)
> > You know it's CW if it has no sidebands: i.e., it has a line spectrum.
> Morse code modulated CW does have sidebands. The
> bandwidth required for typical human demodulation is
> around 2 to 4 times the WPM, although ...
19:13 31-07-06
Michael wrote:
> If I have a time signal which is periodic, and I use FFT to obtain the
> spectrum, which should be discrete, will this FFT procedure be approximate?
> I am wondering about this because I've heard that FFT is only an
> approximation to the true Fourier transform...
There...
just to be clear guys, truly "white" noise has infinite power, and i
doubt that Chris's random numbers have infinite variance.
random number generators usually output uniform PDF pseudo-random
numbers that are "virtually" independent of each other. these numbers
are hypothetically "sampled"...
I have an FIR filter with the system equation y[n] =
0.25x[n]+0.5x[n-1]+0.25x[n-2] which gives an impulse response of h = [0.25
0.5 0.25]
without resorting to Z-transform analysis, how can I work out the gain of
the filter at DC and at the half nyquist frequency?
I think the gain of the fil...
This is due to Whittaker
E. T. Whittaker, On the functions which are represented by the expansions
of the interpolation theory, Proc. Roy. Soc. Edinburgh 35 (1915), 181-194
http://www-gap.dcs.st-and.ac.uk/~history/Mathematicians/Whittaker.html
and to Kotelnikov
V. A. Kotelnikov, ...
Gary Marsh - 11:02 06-02-07
Randy Yates wrote:
> Since you have the s-domain response, I would use the binlinear transformation.
And the resultant pre-emphasis filter will have (1) infinite gain at
nyquist (not a good thing) and (2) due to frequency warping, it probably
won't have an ideal response.
I fought with...
I'll try to answer from an analog guys perspective.
12 bits gives a quantizing noise floor of about -72 dBc.
This Q noise is spread across the entire Nyquist bandwidth.
If you double the sampling rate, the Q noise floor is still -72 dBc but
the Nyquist bandwidth has doubled so while the tota...
I am trying to implement FM algorithms for computer music applications. The
language I'm using (Nyquist) has a primitive FM oscillator called fmosc. I
want to use modulators in series which is a very common technique in some
Yamaha synths. In pseudo code it is the composition fmosc(fmosc(osc())...
fredct - 09:22 19-10-07
Thanks, Jerry, Tim,
You've been quite helpful.
First let me say that I understand what he's trying to show - that
undersampling can alias a signal down to baseband, or at least a lower
frequency.
I'm trying to use this book though to understand a few related things that
relate to my work. L...
in article 5JqdnR5udeMrETXfRVn-vw@rcn.net, Jerry Avins at jya@ieee.org wrote
on 06/09/2005 14:38:
> Middle coefficient non zero, but all other odd-numbered coefficients.
> Round-off errors usually cause computer programs to return very small
> numbers instead of zero. Ignore those.
also,...
jjmai - 20:57 19-12-06
Let's say you have a perfect sine wave at frequency w.
According to Nyquist, in order to be able to recover the sine wave, you
need to have a sampling rate of at least 2w.
So if you decide to sample at 2w, you end up with 2 samples for each cycle
of this sine wave.
If you sample at the peaks ...
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