"lucy" wrote in message
news:1134502603.115693.72580@g43g2000cwa.googlegroups.com...
> I failed to integrate the exponential kernel against the sinc
> function...
>
> Does anybody know what is the Laplace transform of an ideal low pass
> filter?
If you will admit the Bilateral Lap...
Raymond Toy wrote:
> > > > > > "John" == John writes:
>
>
> John> MATLAB:
> John> fourier(exp(-x-exp(-x)))
>
> John> ans =
>
> John> gamma(1+i*w)
>
> Are you sure it didn't say something like gamma(1+i*w,1)?
>
> I have a table of Laplace transfo...
"Clay S. Turner" wrote:
> [...]
> Integrating sin(x)/x can be done
> as in the above link or it can easily done by contour integration of an
> analytic extention of the original integral.
Isn't the inverse Fourier transform a special case of the inverse
Laplace transform, and the inverse...
"Glen Herrmannsfeldt" wrote in message
news:qSaOa.58648$fG.41271@sccrnsc01...
>
> "Fred Marshall" wrote in message
> news:tj4Na.2297$Jk5.1256042@feed2.centurytel.net...
>
> > I didn't ever say anything about differential equations - although I
know
> > that you had earlier wh...
Jerry, Fred:
[snip]
> > As I tried to say before, which agrees with your notation above, if all
you
> > care about is the time domain, then you can use the real number
notation.
> > That it can be expressed as the sum of complex numbers isn't an issue
> > if that's all you want to d...
On Sat, 9 Aug 2003 09:34:48 -0400, "Clay S. Turner"
wrote:
> Hello Robert,
>
> Yes, I'm referring to the Heaviside expansion theorem. While it is true that
> it doesn't hand Medhi's answer over on a silver platter, but it does allow
> one to go and dig into the problem some. Certainly th...
In article f917933c.0308290847.70804a90@posting.google.com, Gregory Pruden
at gregory@incessant.com wrote on 08/29/2003 12:47:
> After reviewing this paper:
> http://iem.kug.ac.at/~noisternig/iem/bt2003/literature/Mueller_sweep
for some reason i can't get the site to respond, so i'll be ma...
llabakudas@yahoo.com (Lord Labakudas) wrote in message news: ...
> 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 s...
In article vs75rfgustug97@corp.supernews.com, Mike at
electrocole@charter.net wrote on 11/25/2003 13:02:
> I have an "Electronic technician" education and work experience background.
> I want to study DSP on my own. The highest level of math that I presently
> understand is pre-calculus mat...
"Rune Allnor" wrote in message
news:f56893ae.0312101617.4e9da046@posting.google.com...
> 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 frequ...
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...
"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...
AG wrote:
> Hi Tim,
-- snip --
>
> > The term "damping ratio" is much more slippery when you're talking
> > about discrete-time systems. Assuming that I'm not messing up the
> > math, if you find the pole locations of your transfer function, z_0 =
> > e^{jw + q) then the "damping...
In article c1iqi205el@enews3.newsguy.com, Bob Cain at
arcane@arcanemethods.com wrote on 02/25/2004 13:46:
> robert bristow-johnson wrote:
>
> > again, how do you show, that for a filter that has all zeros inside the unit
> > circle (or in the left half s-plane for continuous-time) that s...
In article c1qqaq$cn1$1@mozo.cc.purdue.edu, Joenyim Kim at
jeonyimkim80@yahoo.com wrote on 02/28/2004 14:35:
> Can anybody tell me what is the z-transform of "sinc" function and what is
> its region of convergence?
i thought originally that it's a homework problem, but i wonder if it is
s...
"Till Crueger" wrote in message
news:c41ho7$13u4$1@f1node01.rhrz.uni-bonn.de...
> Hi,
> I have some simple questions about the properties of linear systems. In a
> linear System we have to assume the properties of homogeneity, additivity
> and shift invariance.
> In a DSP book I read...
?ine Canby wrote:
> Hi,
>
> The transfer function for a lossy integrater is
>
> H(z) = z/(z-c)
>
> the magnitude spectrum is given by
>
> M(f) = |z|/|z-c|
>
> so lets say I have a c value of 0.8, how do I plot M(f)? I simply
> tried plugging in values for z ni the ra...
Lee wrote:
> Hi,
>
> Some questions I cannot understand right now. Could you do me a
> favor?Thanks,
>
> Question 1:
> I can understand lowpass filter, highpass filter and bandpass filter.
> But why do we need allpass filter?Since we need all frequency passed,
> why we add an all...
I have a problem with verifying the results for the conversion of a Laplace
transform to a z transform. The Laplace transform is in a table and is:
(b-a)/((s+a)*(s+b))
The z transform for this transfer function is:
( z*(exp(-a*T)-exp(-b*T))/((z-exp(-a*T)*(z-exp(-b*T))
Now let: a=1 b=2 T...
Stephan M. Bernsee wrote:
> > > it looks
> > > like you're correlating (exponentially?) decaying cosines with your
> > > signal, probably with a higher "forgetting factor" for the upper
> > > frequencies, at a stride of one sample (the equivalent to a so-called
> > > "sliding transform").
> ...
Hi,
What is the reason to that dicrete IIR filters are usually
designed using impulse invariant transformation (= Euler
integration) or bilinear transformation (= trapetzoidal
integration) instead of the closed form solution that
can be computed by the matrix exponential function?
At least...
DJTB wrote in message news: ...
> May be this is a stupid question, but can I use the Z Tansform to convert
> any function to a difference equation?
No. There are some restrictions, but none are very serious
since you start out with filters and transfer functions.
First of all, the Z ...
Hello Randy,
I experienced what you have discovered, and basically I handle converting
analog filter designs based on mapping a frequency point in the analog
domain to a point in the discrete domain.
So starting with the bilinear transform
z-1
s = c ----
z+1
A...
Archive-name: dsp-faq/part1
Last-modified: Tue Oct 19 2004
URL: http://www.bdti.com/faq/
FAQs (Frequently asked questions with answers) on Digital Signal Processing
The world-wide web version of the comp.dsp FAQ is maintained and
sponsored by Berkeley Design Technology, Inc....
[BG] Responses embedded below...
"lucy" wrote in message
news:clfaoc$19j$1@news.Stanford.EDU...
> In the title, "w" denotes Omega, which is 2*pi*f; "f" is the variable in
> frequency domain.
>
> I am trying to understand Oppenheim's Signal & Systems and Discrete-Time
> Signal pro...
Hi all,
I am studying a digital phase locked loop.
The closed loop filter of this loop has the following Z-Transform :
H(z) = ((K1+K2)*z^-1 - K1*z^-2) / (1 + (K1+K2-2)*z^-1 + (1-K1)*z^-2)
I would like to know the damping factor and the natural pulse of the
equivalent time continuous fi...
"Robert Israel" wrote in message
news:cmmoma$d38$1@nntp.itservices.ubc.ca...
> In article ,
> Bob Adams wrote:
> > I made an error in my previous post; please use this one instead!
>
> > I am having difficulty solving the Fourier Transform of the following
> > complex time sig...
A Sound Mathematical Basis For Sampling - Lesson 3
--------------------------------------------------
Good Morning, once again, Boys and Girls!
I'm sorry that I got called away yesterday; SWMBO,
indeed, MBO!
Today I'll derive for you the mathematics of sampling,
based on an analysis of t...
"Tim Wescott" wrote in message
news:wYGdnbPA6obFKjTe4p2dnA@web-ster.com...
> Bhaskar Thiagarajan wrote:
> > Hi all
> >
> > I'm working on trying to model a non-linear system (described by a
second
> > order differential eqn) into a discrete IIR filter.
>
> Whoa! Stop right ...
The AES paper I wrote in the 80's that RBJ referenced showed that if
you have a "periodically-missing" sample, you can recover a bandlimited
signal from the non-uniform samples as long as the bandwidth is less
that 1/2 of the AVERAGE sampling rate. It's a pretty simple idea based
on M-band filte...
> When I was in third year of Uni, we did a course on complex maths that
> really threw me. It was all about analytic functions, the
> Cauchy-Riemann equations, etc. and involved all sorts of integrations
> of curves in the complex plane. Now I'd always been good at maths, but
> what with a...
Hi all,
I understood the two stability conditions:
One is BIBO criteria in Fourier analysis... it says that if the impulse
response is not absolute integrable, i.e. if Integrate(|h(t)|, t from -inf
to inf)=inf then the system is not BIBO stable...
From this criteria, an ideal low pass f...
Atmapuri wrote:
> Hi!
>
> I am looking for a method that can take an s domain
> transfer function and use FFT/IFFT to obtain the
> discrete time domain impulse response.
>
> Are there any aproximations that allow that?
>
> Thanks!
> Atmapuri.
Use a Laplace transform. s doma...
reading my old copy of 'Modern Control Engineering' by Ogata, 1970,
(ok, no longer 'modern'), pg 117 describes a method of multivariable
control using a 'transfer matrix' of Laplace transfer functions where
Gij(s) is the transfer function from ith input to jth output.
Would this be practical t...
"maxyp" wrote in message
news:-9Odne7iy4Whp7zfRVn-1A@giganews.com...
> Hi,
>
> I need to plot the magnitude frequency response of a filter without using
> the DFT method.
>
> The input to the filter (2nd order IIR) is a sequence of the form cos(nw)
> (n and w vary), and I want to ...
in article 1109882654.091294.118040@g14g2000cwa.googlegroups.com, Clay at
physics@bellsouth.net wrote on 03/03/2005 15:44:
> robert bristow-johnson wrote:
>
> >
> > hey Clay, would you like a section on how, for minimum-phase filters, that
> > the phase response (in radians) is the ne...
in article zIOdnSILWryu8vHfRVn-ug@comcast.com, James Van Buskirk at
not_valid@comcast.net wrote on 04/25/2005 00:16:
> Recall that the O.P.
> did not claim to be a expert on the mathematics of DSP.
maybe he doesn't like bragging. are you claiming to be an expert on either?
> It seems ...
It's pretty easy to figure out who was responsible for the Fourier
transform, ditto for the Laplace.
Does anybody out there know who dreamed up the z transform (Please tell
me it wasn't someone named 'Z')?
-------------------------------------------
Tim Wescott
Wescott Design Services
h...
I understand the mathematical differences between the two - e.g. -
a) LT is more general b/c it is a function of a complex variable 's',
whereas FT is a function of an imaginary variable (real part = 0)
b) LT converges for a larger range of functions
But when to use which? I have used LT ...
in article VVLIe.108880$Pf3.64017@fe2.news.blueyonder.co.uk, ma at
ma@nowhere.com wrote on 08/05/2005 11:51:
> Where can I find some mathematical modeling? I need to know the theory and
> then use the code as a learning practice.
you want theory? i'll give you theory.
below is about as...