## Forums Search for: LMS

## Identification of Non-Minimum Phase zeros

inIf we have a non-minimum phase FIR system (1-2z^-1) when it is quite easy to identify this system (assuming the input is known white driving...

If we have a non-minimum phase FIR system (1-2z^-1) when it is quite easy to identify this system (assuming the input is known white driving noise) using LMS or recursive-least squares. However, if we add uncorrelated white noise at the output then the equivalent system appears to be an innovations model where the zero is found from spectral factorisation ie the LMS algorithm identifies th...

## LMS implementatiom for echo cancelation ad co-channel interference supression

inI am working on a project in which, based on the LMS adaptive algorithm, I have to perform the following: echo cancelation and...

I am working on a project in which, based on the LMS adaptive algorithm, I have to perform the following: echo cancelation and co-channel interference supression. I have tried the inplementation of the algorithm in Mathlab but for noise cancelation. For noise cancelation it was was simple: d = awgn(x, 22);// signal distorted by noise SNR is -22 dB mu = 0.08;//step size ha = adaptfilt.lms(20, ...

## Unstability in NLMS algorithm for Echo Cancellation

inHi every body, I have implemented a line echo canceller based on NLMS algorithm (TI app. note SPRA129) but sometimes the coeficient...

Hi every body, I have implemented a line echo canceller based on NLMS algorithm (TI app. note SPRA129) but sometimes the coeficient updatation falls into a bad state and echo will amplify. I changed the algorithm in a way that when I find this situation has happend I reset the coeficients to zero so it takes time again for the coeficients to converge and during this time you will hear echo. B...

## low frequency response of adaptive equalizer?

inHi, I have an LMS based adaptive equalizer which mostly works well but it doesn't tolerate low frequency inputs too well. The channel...

Hi, I have an LMS based adaptive equalizer which mostly works well but it doesn't tolerate low frequency inputs too well. The channel consists of two transformers (high pass) and cable (low pass) and when there is long periods of no-change in data (which is possible unfortunately even though the data is scrambled) and the SNR goes down during these long periods of no change. I think this is ...

## Adaptive filters

inHi I've read a book about adaptive noise reduction methods. I don't understand why the original signal (without noise) is needed in...

Hi I've read a book about adaptive noise reduction methods. I don't understand why the original signal (without noise) is needed in Wiener filtering, RLS filtering, LMS filtering... I mean, in most of the cases, you don't have any reference of the original signal when you want to reduce noise in a noisy signal! (otherwise, if you already know how the original signal is, why do you have to handl...

## Karhunen-Loeve and Transform-Domain LMS Algorithm

inDear readers, my question is on something that looks like a variation of the Karhunen-Loeve Transform. The normal Karhunen-Loeve Transform...

Dear readers, my question is on something that looks like a variation of the Karhunen-Loeve Transform. The normal Karhunen-Loeve Transform would be like this: for a signal u of length L and an adaptive FIR filter of order M, we compute the Autocorrelation Matrix and its eigenvectors. In Matlab: L=max(size(u)); ruu=xcorr(u,u)/L; M=16; Ruu=toeplitz(ruu(L:L+M-1)); % Correlation Matrix...

## Implementing NLMS for Echo cancelation in Speech

inHi I am trying to write an acoustic/line echo canceller. I choose the quite well known nlms algo. First I tried to implent it in...

Hi I am trying to write an acoustic/line echo canceller. I choose the quite well known nlms algo. First I tried to implent it in matlab. When I checked the algo with any "test input" like sin wave, zigsaw etc, I did get convergance. Then I tried it with speech, but failed. I tried to improve it by implementing anti-correlation filter ( Pre-whitening). I did saw minimal improvments in the...

## Adaptive filtering

inHi everyone I'm trying to do adaptive filtering with Matlab (LMS adaptive filtering). But with this method, it seems that I need to know the...

Hi everyone I'm trying to do adaptive filtering with Matlab (LMS adaptive filtering). But with this method, it seems that I need to know the original signal form (without noise). And I don't have information about the original signal (it's a large band signal, ofdm modulation). Do you know if it's possible to reduce noise with this method? Thanks a lot.

## expectations from a DFE ?

inHi, I've been experimenting with adaptive filters for channel equalization and I have a problem with decision feedback case. Initially I...

Hi, I've been experimenting with adaptive filters for channel equalization and I have a problem with decision feedback case. Initially I started with a T/2 fractional linear equalizer adapted with LMS and then converted this to delayed LMS with two delays. I am currently not adding any noise just trying to compensate for the ISI generated by the channel although there is quantization error a...

## LMS and Eigenvalue Spread

inI have been reading Haykins book again and he goes on about Eigenvalue spread. Eg for a 2nd order AR (all pole) model LMS converges more slowly...

I have been reading Haykins book again and he goes on about Eigenvalue spread. Eg for a 2nd order AR (all pole) model LMS converges more slowly if the eigenvalues of the cov matrix are dispersed eg one is 10 times the other for a 2nd order system. My question is this: Can we get a similar large eigenvalue spread for a purely FIR system (as opposed to Haykins AR mode)? Also I assume the con...

## Explanation of LMS algorithm using samples

inHai All, I am a beginner in the DSP domain . And now I am working on the Acoustic Echo Cancellation. I have gone through the...

Hai All, I am a beginner in the DSP domain . And now I am working on the Acoustic Echo Cancellation. I have gone through the theoretical details of LMS algorithm and I have analysed that and I understood that why we are getting each step. But when I go for sample by sample analysation I am having several doubts.Please help me that how to analyse that .Can any one give explanat...

## How to calculate the error signal in the LMS algorithm

inHai All, I am working with Acoustic Echo Cancellation using LMS adaptive algorithm.In that I have understood each and every...

Hai All, I am working with Acoustic Echo Cancellation using LMS adaptive algorithm.In that I have understood each and every step.But I am facing the problem with error calculation.If d[n] is the desired signal that is captured by the microphone and x[n] be the filtered far end signal (that is echo replicated signal) and e[n] be the error signal that is obtained by subtracting ...

## Acoustic Echo Cancellation using LMS algorithm with C++ coding

inHai all, I am working on Acoustic Echo Cancellation using LMS algorithm . Theoretically I went through that thorougly and now I...

Hai all, I am working on Acoustic Echo Cancellation using LMS algorithm . Theoretically I went through that thorougly and now I am implementing that using C++. I am considering the 16 bit mono with 8KHz sampling frequency.As it is considered of 16 bits amplitude(intensity) of each sample should be within the range -32768 to 32767.As we know the computational equation...

## LMS for Smart Antennas

inHello all, I have to realize a Smart Antenna system using the LMS algorithm on an uniform linear array. Let's suppose that we know already the...

Hello all, I have to realize a Smart Antenna system using the LMS algorithm on an uniform linear array. Let's suppose that we know already the angles of arrival of the desired signal and of one or more interferers. How can I calculate the LMS weights if I don't know how exactly is the desired signal? I can compute the weights knowing the desired signal but then it would be useless to calcu...

## Help: Matlab programming of BLMS and FDAF

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...

## Effects of eigenvalue spread of input signal on MBER and AMBER algorithms

inAs I understood, the eigenvalue spread of the input signal will affect the performance of LMS MMSE algorithms. But will the eigenvalue spread...

As I understood, the eigenvalue spread of the input signal will affect the performance of LMS MMSE algorithms. But will the eigenvalue spread affect the performance of LMS MBER and AMBER algorithms? Thanks.

## Eigenvalue spread and steady state error

inMay I know how the eigenvalue spread affects the steady state error of LMS-type algorithms?

May I know how the eigenvalue spread affects the steady state error of LMS-type algorithms?

## Signal constellation & steady state error

inWill the signal constellation level affect the steady state error of LMS-type algorithms?

Will the signal constellation level affect the steady state error of LMS-type algorithms?

## Plot of Autocorrelation Error using LMS in Matlab

Hi !! I hope somebody can help !! N = 8000; % Number of Samples n = 0:N-1; sigmax = 0.01 L = 12 f1 = 101.7 f2 = 142.4 f3 =...

Hi !! I hope somebody can help !! N = 8000; % Number of Samples n = 0:N-1; sigmax = 0.01 L = 12 f1 = 101.7 f2 = 142.4 f3 = 231.5 fs = 1000 K = 20 % Number of realizations cumensq = zeros(K,N) en1 = zeros(K,N) % length of filter coefficients wn = zeros(1,L); a = 0.0001; muhat = 1.00 wmat = zeros(N+1,L); % 8001 filters wmat(1,:)=wn; for kk = 1: K % ensemble process xn =...

## LMS Filter in Simulink

inHi all !! I would like to use the LMS Filter in Simulink !! Any Suggestions !! How to connect the desired input with the output and error...

Hi all !! I would like to use the LMS Filter in Simulink !! Any Suggestions !! How to connect the desired input with the output and error !! I get some errors !! Don't understand !! Also there is problem with double and unsigned integers !! when i used the "Sum Block "in simulink !! Appreciate if somebody can draw it or show it more clearly !! Thanking you Ali