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The Wayback Machine - https://web.archive.org/web/20120615051141/http://www.dsprelated.com:80/comp.dsp/keyword/Z_Transform.php
Hi,
I am learning about digital decimation. The problem is like this:
Z-transform of input sequence and filter aree X(z), H(z) respectively.
After the filter H(z), there is a 2 decimation. From one book talking
about decimation, it says the Z-transform of output sequence after
decimation is:
...
Dear all,
For demonstrating the effect of a multipath channel on a wireless comm.
system, I am trying to implement an equalizer working according to
Zero-Forcing criterion.
However I think I have some vaque points in the theory so I would be most
happy if you can advice/correct me on the below i...
Hi,
If I have been given a z-transform and I know that the z transform has
a pole on the unit circle at a certain angle, does it mean that the
fourier transform does not exist at all, because I have read a paper
where the author tries to derive the fourier transform from the z
transform even wh...
On Mar 26, 6:34 am, "Steven G. Johnson" wrote:
> On Mar 26, 2:31 am, "Steven G. Johnson" wrote:
>
> > For z on the unit circle, the chirp z-transform algorithm consists of
> > three steps: multiply the input signal by a chirp, convolve with a
> > chirp (i.e. FFT and multiply by th...
Hi all,
I've been searching for the best way to do this programmically, but I
can't seem to come up with a simple solution.
I have a program that calculates filter coefficients correctly based
on this z-transform:
H(z) = (b0+b1*z^-1+b2*z^-2)/(1+a1*z^-1+a2*z^-2)
I want to be able to dis...
Hi all, I would like to have your opignon, which transform is better :
Z-transform or Fourier transform for implementation of reed solomon
codec? if it will be implemented on a DSP processor,
what would be the performance? will I get a high speed with Z-
transform or Fourier transform ?
Thanks
...
Ok, say i have a single pole lowpass filter, 1 pole, no zero.
y[0] = c.x[0] + (1-c).y[-1]
This is basicly a single pole moving from (0,0) to (1,0) on the real axis.
And normalized at DC.
Now by doing this seperately after the lowpass.
highpass = x[0] - y[0]
You get a high pass. But ...
Communications_engineer writes:
> Hello, can we use Z-transform on continuous time signals to find their
> z-domain analysis. (what do we really realize in z-domain and how is
> it different from frequency domain analysis) and also can we use
> Laplace for discrete time signals.
>
> ...
Hi
i have a free online article : root selection methods in flood
analysis
http://www.hydrol-earth-syst-sci.net/7/151/2003/hess-7-151-2003.html
and there is a z transform for the unit hydrograph for a single
reservoir
(Nash-cascade) i can't figure out how its made !
in the time domaine si...
Hi All,
I have studied three diff kinds of transforms, The laplace transform, the
z transform and the fourier transform. As per my understanding the usage of
the above transforms are:
Laplace Transforms are used primarily in continuous signal studies, more
so in realizing the analog circuit equi...
Question regarding Power Series method of finding IZT
-------------------------------------------------------
Hello. I want to clear a doubt regarding Power Series method for Inverse
Z-transform
If I have X(z) and I'm given a RoC that says that X(z) is a two-sided
sequence and I'm supposed to f...
On Feb 6, 4:45=A0am, jim wrote:
> dbd wrote:
> > Why would you want an approximate a non-
> > periodic operator with something you feel obliged to consider
> > periodic??
>
> Convenience I guess.
>
> Substitute 3 for the Pi term in the DFT and you have an operator that is
> a...
"Craig" ha scritto nel messaggio
news:82396605.0307030813.233f549c@posting.google.com...
> I guess I am just a little confused with the constant notation
> switching, I am following Crochiere, since it is what I have available
> to me. The notation is rather abnoxious, and it isn't not ...
Hi,
Years ago I obtained a really good explanation ofhe Z Transform called
"introduction to the Z transform and its derivation" by Karwoski. This
was a TRW app note. I have since lost my copy and was amazed to find
that I could not find it on the internet. That division of TRW that
was respon...
I tried taking a signal of 1102 samples, 44.1kHz, 16bit, mono containing a
fundamental = 83.2 Hz harmonic tone and applied the function as follows in
matlab:
s = wavread(...)
hl = lagrange( 3, 0.2 )
nl = [zeros(530,1);1]
yfd = filter( hl, 1, s )
ynd = filter( nl, 1, s )
y = 1 - ynd.*yfd
...
Dear members:
Plz tell me what is the point I am wrong. It is my exam.
I have to plot the magnitude of the freq response a Low Pass filter:
y[n]=1/3*(x[n]+x[n-1]+x[n-2]);
I used Z transform and found
Y(z)/X(z)=H(z)=1/3*(z^2+z+1)/z^2.
So there are two zeros at z1=-1/2 +sqrt(3)/2 a...
Hello,
I wrote a program in Java that does a DFT on raw 8bit samples stored in
memory from a RF ADC (Post processing). This allows me to easily adjust the
span (zoom) when I'm viewing the spectrum. Works great, but slow as hell. So
I'm now trying an FFT with a Chirp Z-Transform so I can zoom i...
Hi friends!
I got a basic doubt in the theoritical dsp. Hope some one can
help me. My actual question is:
Consider a sequence x(n) whose z-transform is X(z) and ROC is
characterized by Rx. Consider another sequence y(n) with z-transform
Y(z) and ROC Ry. Now
suppose that Rx and Ry...
Hello DSP folks,
This question has been bothering me for a long time since I took my
Microwave Engineering course:
Suppose we have a linear system H(z) we can easily find its poles and
zeros and perform stability analyis of the system. Does such a thing
exist for general network analysis us...
Liz wrote:
> As a signal-processing person, I am trying to wade through some
> heavy-duty math papers and having a problem.
>
> Suppose that you have a signal-processing network that is
> represented by a transfer function (either Laplace or Z, doesn't
> matter right now). Suppose tha...
"Matt Timmermans" wrote in message news: ...
> Again, I have no rational polynomials.
Actually, there may be a chance that you do. The Z transform of a FIR filter
*is* a rational polynomial, though with only a numerator and no denominator.
Rune
...
On 12 Nov 2003 13:58:47 -0800, allnor@tele.ntnu.no (Rune Allnor)
wrote:
> > Does anyone know why the two different editions are available?
> > And what differnces there may be betwen the two? According to one
> > customer review at amazon there seems to be quite substantial
> > differences ...
thanks for your responses. I like the pid controller thing, the "PID
without a PhD" is quite straightforward and i think it could be of use for
my cause
...
Rune Allnor wrote:
> Does anyone know how to compute the DFT coefficients efficiently
> in a narrow frequency band (few but more than one bins)? I guess
> such an algorithm would be a cousin of the Goertzel algorithm?
Tom Loredo wrote:
> FractionalFFT:
>
> http://citeseer.nj.nec.c...
Hello!
I have some questions about the z-transform and what to use it for in
for example ARMA-filters. I know it is used to find poles and zeros,
but what else?
Consider an ARMA filter:
y(t)+a1*y(t-1)+a2*y(t-2)=x(t)+c1*x(t-1)+c2*x(t-2)
After z-tranformation it can be written: Y(z)=H(z)...
Bob Cain wrote in message news: ...
> Wouldn't it be
> correct to believe that if the result of the calculation is
> a least mean square approximation to the component's actual
> impulse response having a particular length and delay that
> the phase information would be optimally prese...
MCTimes@21cn.com (Hakuna M. C.) wrote in message news: ...
> Hi all,
> I am using a frequency operator to act as a differentiator like
>
> i*w d/dt
> here w is the frequency defined in fourier space.
Almost. What you state is valid for continuous functions. With matlab
(and a...
Hi all.
I am working with this problem that involves modelling the total
phase of a signal, i.e the phase can take on any value and is not
restricted to the interval [0, 2pi> .
Part of the analysis involves a reflection sequence on the form
Q
r(n) = sum A_q*d(n-m_q)
...
On Wed, 11 Jan 2006 16:44:48 -0500, Stan Pawlukiewicz
wrote:
> robert bristow-johnson wrote:
>
> (big snip)
>
> s" that the data
> > passed to it is one period of a discrete, infinite, and periodic
> > sequence of numbers that has period length of N.
> >
> > i fail to see this ...
On 10 Mar 2006 05:22:41 -0800, "Srini" wrote:
> When I try to apply fft to sequence lengths which are not powers of 2 -
> I decided to extend the data with zeros to the next higher power of 2
> and then apply fft. Problem is the transform has more coefficients than
> the input. We cannot ju...
"Stan Pawlukiewicz" wrote in message
news:dq3u8h$nan$1@newslocal.mitre.org...
> robert bristow-johnson wrote:
>
> (big snip)
>
> s" that the data
> > passed to it is one period of a discrete, infinite, and periodic
> > sequence of numbers that has period length of N.
> >
> > ...
On Fri, 16 Jan 2004 02:33:36 +0000, Anand wrote:
> I am trying to implement J.17 de-emphasis cure in a 32 bit processor.
> I have converted J.17 S-domain transfer function using Bilinear of
> MATLAB. The maltlab plot of manitude responce matches with the table
> biven in the std. But when ...
On Sep 3, 11:21 am, Andor wrote:
> Randy Yates wrote:
> > Randy Yates writes:
> > > In general, the frequency response of a digital filter (IIR or FIR)
> > > is determined by evaluating H(z) at z = e^{j*2*pi*f*Ts}, where Ts is
> > > the sample period and f is the frequency at whi...
"Jerry Avins" wrote in message
news:40096010$0$6092$61fed72c@news.rcn.com...
> PROVENTEK MINDCRAFT AB wrote:
>
> > Hi folks,
> >
> > I'm working with an dsp audio application and I desperately
> > need an algorithm for tone control.
> >
> > The filter is described in my spe...
On Tue, 20 Jan 2004 08:51:40 -0600, "Shawn Steenhagen"
wrote:
> Guys,
Hi Shawn,
> I didn't see the article in question, but I believe if they are talking
> about a "sliding Goertzel", then k must be an integer, otherwise you don't
> get a good zero/pole cancellation and the filter des...
"Luiz Carlos" wrote in message
news:3fd8f66b.0401230509.38272b12@posting.google.com...
> Martin,
>
> Somebody here said: sin(x)/x. (Now obvious!)
> So, I'll ask for something a little bit different:
> I want an example for a causal signal that has bandlimited spectrum.
Luiz Carlos...
"Lee Southern" wrote in message
news:457ffbd.0401291253.4501b82@posting.google.com...
> I am a complete newbie to DSP...
>
> I have a requirement to implement (in software) a "first-order lowpass
> filter with a cut-off frequency of 1.6Hz and a gain of 0dB". No
> mention is made of ...
"Ray Andraka" wrote in message
news:401FA657.2575343D@andraka.com...
> > what is the complexity of FFT computed on block lenght other then power
> > of 2 ?
> > Is it still NlogN ?
>
> roughly, but it depends on the radix
Even if you had a prime-length FFT, you could do it in O(...
I read the article and downloaded the code (below). Problem is every
compiler I have tried (CodeComposer, gnu, VisualC++) has problems with
the notations:
double[][] xxxx (I indicate occurrences in the code below with ' 256) && ((N+M) 128) && ((N+M) 64 ) && ((N+M) 32 ) && ((N+M) 1...
