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Question about Z transform of decimation

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: ...

complex valued Equalizer - zero forcing criterion

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...

If a z transform has a pole on the unit circle does that signal have a fourier transform?

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...

Re: Inverse chirp z

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...

Calculate frequency response of Z transform with known frequency bins

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...

reed solomon: z transform vs fourier transform

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 ...

Single pole lowpass --> highpass - z transform?

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 ...

Re: uniform filter bank implementation.

"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 ...

Re: DFT VS DTFT

Fred Marshall wrote: > "Tom" wrote in message > news:slrnbho1p5.aeh.T.Otermans_REMOVE_THIS@noritake.basement... > > > praveen wrote: > > > > > Hello, > > > > > > I wanted to know the difference between discrete fourier transform and > > > discrete time fourier transform. > > > ...

Looking for old TRW app note "Intro to the Z Transform".

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...

Re: transfer function confusion and MATLAB

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 ...

Confusion with magnitude plot of a freg response of a filter, plz help

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...

windowing DFT vs FFT (newbie)

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...

Z-Transform - ROC

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...

S-Parameters, Z-Transforms and Stability.

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...

Re: Math versus signal processing; terminology differences?

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...

Re: Minimum Phase Impulse Response

"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 ...

Re: Oppenheim & Schafer

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 ...

Re: using z transform for a discrete time filter

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 ...

Re: Vernier FFT

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...

Question about the z-transform for ARMA modelling

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)...

Re: Why minimising in the mean-error sense.

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...

Re: numerical differentiation using FFT

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...

Total phase of a sum of complex numbers

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) ...

Re: Anthropomorphisms and the inherent periodicity of the DFT

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 ...

Re: FFT for sequences of arbitrary lengths

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...

Re: Anthropomorphisms and the inherent periodicity of the DFT

"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. > > > > ...

Re: J.17 s -> z conversion

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 ...

Re: Gain of an IIR Filter

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...

Re: Audio application problem

"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...

Re: Yet another article claiming Goertzel k must be integer

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...

Re: Fast response filter?

"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...

Re: Need help with lowpass filter

"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 ...

Re: Complexity of not 2^n FFT

"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(...

ESP article Chirp Z transform questions/problems

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...

Re: equivalence between Z-transform and Laplace-Transform

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...

Re: 1D Discrete wavelet transform(DWT)

Umutesi Faith wrote: > > More info: http://www.math.hmc.edu/faculty/ward/wavelets/pdfs/m185l2.pdf > > I have been at this link, well i guess the calculations of these > coefficients are not as easy as i thought. It shows quite simple way of > getting the scaling and wavelet coefficients in...

what is the z-transform of sinc function?

Can anybody tell me what is the z-transform of "sinc" function and what is its region of convergence? Thanks a lot, -Joenyim ...

[Q] How can the chirp-z transform be used in resampling?

DSP Gurus: I'm familiar with the classic interpolate-filter-decimate approach to multirate DSP. I've heard that the chirp-z transform can be used to resample a signal also. Can anyone provide additional information or references (other than Google search results)? Thanks, OUP ...

Minimum phase filter design using cepstrum methods?

I've run across a few web pages describing how to convert an arbitrary FIR filter to a minimum phase variant by the use of cepstral methods: e.g. > wn = [ones(1,m); 2*ones((n+odd)/2-1,m) ; ones(1-rem(n,2),m); > zeros((n+od d)/2-1,m)]; > y = real(ifft(exp(fft(wn.*real(ifft(log(abs(fft(x...