Is it possible to replace an FIR with an IIR filter?

Started by Oliver Faust in comp.dsp14 years ago 2 replies

Dear group, I'm new here. I subscribed to comp.dsp because I'm puzzled by a statement from Simon Haykin in his book: Adaptive Filter...

Dear group, I'm new here. I subscribed to comp.dsp because I'm puzzled by a statement from Simon Haykin in his book: Adaptive Filter Theory (fourth edition) In section 1.6 page 52 he describes the Wold decomposition. There he writes: "... A solution of particular interest is an all-zero filter that is minimum phase, which means that all the zeros of the polynomial B(z) lie inside the ...


Re: Digital Equalizer Design for a Radio Receiver

Started by Anonymous in comp.dsp14 years ago

I was thinking of something like: http://www.w9gr.com/ "Anonymous" wrote in message news:... > > "Mark" wrote in message > ...

I was thinking of something like: http://www.w9gr.com/ "Anonymous" wrote in message news:... > > "Mark" wrote in message > news:1139417980.646987.93170@g44g2000cwa.googlegroups.com... > > Clark, > > > > The AF (adaptive filter) you are describing automatically adapts the > > overall bandwidth of the signal...OK...good... but does it actuall


Opinion - Adaptive Filtering of Uncorrelated Sequences

Started by M. Wirtzfeld in comp.dsp14 years ago

Hello, I have several questions to ask regarding the expected behaviour of a linear adaptive filter applied in a system identification...

Hello, I have several questions to ask regarding the expected behaviour of a linear adaptive filter applied in a system identification arrangement using two uncorrelated sequences for both the reference and desired signals. I'll describe the experiment, the observed results, and then put forward my questions. In Matlab I create two uncorrelated sequences using the following code: SI...


Adapting a lattice filter

Started by Lalin in comp.dsp13 years ago 6 replies

Hi, I am planning to use a lattice adaptive filter for system identification. I am unable to find method to update the km?s in the lattice. ...

Hi, I am planning to use a lattice adaptive filter for system identification. I am unable to find method to update the km?s in the lattice. What have I missed? We can convert any FIR to a lattice and can adapt a FIR as needed. Why can?t we only adapt the lattice ? Thank you, Lalin


QRD-LSL

Started by mobi in comp.dsp13 years ago

Hi all, I am having problems regarding weight extraction in QRD-LSL algorithm i.e., I want to know the unknown filter coefficients of the...

Hi all, I am having problems regarding weight extraction in QRD-LSL algorithm i.e., I want to know the unknown filter coefficients of the QRD-LSL adaptive filter. I have implemented QRD-LSL algorithm, given on page 666 of symon hakin book on adaptive filtering. On page 686 he mentions weight extraction, which states that in order to compute the weight vector we need backward prediction coeff...


Need help to get article:"Multidelay block frequency domain adaptive filter", J.S. Soo, K.K. Pang

Started by ogb2 in comp.dsp13 years ago

"Multidelay block frequency domain adaptive filter" Soo, J.-S.; Pang, K.K. Acoustics, Speech, and Signal Processing [see also IEEE Transactions...

"Multidelay block frequency domain adaptive filter" Soo, J.-S.; Pang, K.K. Acoustics, Speech, and Signal Processing [see also IEEE Transactions on Signal Processing], IEEE Transactions on Volume 38, Issue 2, Feb 1990 Page(s):373 - 376.


FDAF (Frequency domain adaptive filter)

Started by Jing in comp.dsp13 years ago 2 replies

Hi, I am trying to implement frequency domain adaptive filter FDAF, and want to compare its performance with that of BLMS, can anybody let mw...

Hi, I am trying to implement frequency domain adaptive filter FDAF, and want to compare its performance with that of BLMS, can anybody let mw know what kind of performance I should expect for FDAF compared with BLMS. Should the MSEs similar? From my experimental results, they are very different, so I am wondering if my implementation is not correct. Also, can anybody know where I can download ...


Help: Matlab programming of BLMS and FDAF

Started by Minal in comp.dsp13 years ago

hi, I am trying to implement frequency domain adaptive filter FDAF and want to compare its performance with that of BLMS, can anybody let me...

hi, I am trying to implement frequency domain adaptive filter FDAF and want to compare its performance with that of BLMS, can anybody let me know what kind of performance I should expect for BLMS and FDAF. Should the MSEs similar to standard LMS? Is there any source for FDAF matlab code? I have referred S.S.Shaynks paper for theory , for matlab code I want to know how the serial data is arra...


Time constant for NLMS algorithm

Started by Richard_K in comp.dsp13 years ago 1 reply

It is given by "Adaptive Filter Theory" that the average time constant for LMS algorithm is 1/(2*Step Size*average of the eigenvalue). However,...

It is given by "Adaptive Filter Theory" that the average time constant for LMS algorithm is 1/(2*Step Size*average of the eigenvalue). However, the time constant for NLMS algorithm is not given. Can anyone please tell me the average time constant expression for NLMS algorithm? Many thanks.


Adaptative filter

Started by Arthur Herbert in comp.dsp13 years ago 10 replies

Hi all, I am to design a special adaptive filter in which the error signal is hihgly correlated with the input signal (in fact, the latter is...

Hi all, I am to design a special adaptive filter in which the error signal is hihgly correlated with the input signal (in fact, the latter is just a delayed version of the error...) Usual resolution of the Wiener equations is therefore not obvious. Of course, since the orthogonality principle is not respected anymore, even the simplest algorithm (stochastic gradient) can not be applied. ...


how define desired input in LMS block when i use TF blocks

Started by payam214 in comp.dsp12 years ago

Dear all I have a problem with my project i want to control the noise with using the adaptive filter(LMS)in matlab simulation block i get a good...

Dear all I have a problem with my project i want to control the noise with using the adaptive filter(LMS)in matlab simulation block i get a good result with my transfer function when i used C= -H/G in the fix controller but after i substitute the fix controller with adaptive controller then my results didn't converge my questions is: 1-how can i define the desired in put when i use the transfe...


adaptive beamforming

Started by charles1984 in comp.dsp12 years ago

I'm trying to simulate a adaptive beamformer. I refered to the block diagram of GSC in "adaptive filter theory" of S.Haykin,P.257.I'm a...

I'm trying to simulate a adaptive beamformer. I refered to the block diagram of GSC in "adaptive filter theory" of S.Haykin,P.257.I'm a little confused with the constraint matrix and quiescent weight vector. I'm wondering about how to built the signal-blocking matrix 'Ca' and how to determine the constraint matrix numerically. Any help is appreciated!


Mathlab as experimental audio DSP workstation?

Started by Anonymous in comp.dsp12 years ago 11 replies

Could Mathlab make sense using it as a evaluation platform for audio DSP? This means: Streaming high quality audio using a firewire audio...

Could Mathlab make sense using it as a evaluation platform for audio DSP? This means: Streaming high quality audio using a firewire audio interface for example through a experimental adaptive filter coded in mathlab. And doing measurements and visualizing results in vaious ways (and fast). And doing simple GUI's for controlling that conveniently.


Coherence

Started by Robert Adams in comp.dsp12 years ago 3 replies

Suppose I take a signal X(n) and apply it to an unknown system that consists of a linear filter in parallel with some sort of...

Suppose I take a signal X(n) and apply it to an unknown system that consists of a linear filter in parallel with some sort of non-linear processing, to create an output signal Y(n). If the non-linear portion of the unknown system were set to zero, then given X and Y, I could design an adaptive filter that, when fed with X, would eventually converge and produce Y. However, if the non-linea...


Energy Conservation Relation in Adaptive Filtering

Started by Manolis C. Tsakiris in comp.dsp12 years ago 3 replies

Hello, it seems that every data normalized adaptive filter or with error non-linearity obeys a kind of an energy conservation relation....

Hello, it seems that every data normalized adaptive filter or with error non-linearity obeys a kind of an energy conservation relation. This relation states that the norm of weight-error vector after the update plus the norm of the apriori estimation error equals the norm of the weight-error vector before the update plus the norm of the aposteriori estimation error. No cross-terms appear! Due t...


Adaptive System Identification using Linear Chirp

Started by Manolis C. Tsakiris in comp.dsp11 years ago 3 replies

Hello dear dsp fellows, i want to identify an unknown acoustical transfer function using an adaptive LMS system identification set-up. Both the...

Hello dear dsp fellows, i want to identify an unknown acoustical transfer function using an adaptive LMS system identification set-up. Both the plant (the unknown acoustical transfer function ) and the model (coefficients of the adaptive filter) will be fed with the same excitation signal. The difference of their outputs will form the error, which will be fed to the LMS update formula. Now, as ...


Root Least Square for DFE equaliser

Started by cpshah99 in comp.dsp11 years ago

Hello People. I have implemented adaptive DFE using RLS algorithm given in Adaptive Filter Theory by Simon Hykin. Now what I did is that I...

Hello People. I have implemented adaptive DFE using RLS algorithm given in Adaptive Filter Theory by Simon Hykin. Now what I did is that I implemented both the filter i.e. feedforward and feedback combined. w=zeros(31,1); where 1st 16 corrospond to FF and rest 15 to FB. I_hat=w'*u; where u is column vector which contains vectors for FF and FB filter. And my receiver works fine. But...


Estimating FIR Coefficient Bounds for Adaptive Filter

Started by Anonymous in comp.dsp11 years ago 1 reply

In order to provide adequate word size in an integer-arithmetic adaptive FIR filter, I want to know the bounds on the filter coefficients. I...

In order to provide adequate word size in an integer-arithmetic adaptive FIR filter, I want to know the bounds on the filter coefficients. I could probably place some bounds on parameters of the desired response, but the desired response is generally unknown a priori and should be considered arbitrary. Since the desired response is produced by a pysical system, I imagine that some properti...


Question about multiple linear regression model in adaptive filer

Started by fl in comp.dsp11 years ago 2 replies

Hi, On "Adaptive filter theory" by Simon Haykin, it gave the multiple linear regression model as: d(n) = A^H Um(n)+v(n) A and U are...

Hi, On "Adaptive filter theory" by Simon Haykin, it gave the multiple linear regression model as: d(n) = A^H Um(n)+v(n) A and U are vectors. I don't understand the reason for this model why it put noise v(n) just before the desired signal d(n). For normal FIR filtering, the noise is just before the filter input. This multiple linear regression model is not the same for the filter pr...


How to initialize RLS in practice?

Started by zqchen in comp.dsp11 years ago 2 replies

As S. Haykin's Adaptive Filter Theory suggests, we've to choose different deltas for the initialization of the recursive least squares algorithm...

As S. Haykin's Adaptive Filter Theory suggests, we've to choose different deltas for the initialization of the recursive least squares algorithm for low, medium and high SNR. Then we've to know the SNR first. But what if we cannot get the SNR before we initialize RLS? And in the condition of interference, we've to estimate the SINR first? thanks.