Vector differentiation problem - LMS type

Started by Country_Chiel in comp.dsp15 years ago 2 replies
LMS

This is something I am not clear about suppose I have the quadratic function (where ' denotes transpose) J=X'AX and I wish to...

This is something I am not clear about suppose I have the quadratic function (where ' denotes transpose) J=X'AX and I wish to differentiate this (as in LMS problems) wrt X I know dJ/dX = 2AX (I think) assuming A is symmetric. But how do I diferentiate J=GAG' ie dJ/dG where G is a matrix and A is symmetric like before? Thanks


improving an lms filter

Started by mk in comp.dsp15 years ago 2 replies

hi, I have an adaptive equalizer which implements a dfe which works reasonably well but I am trying to reduce the excess MSE after convergence....

hi, I have an adaptive equalizer which implements a dfe which works reasonably well but I am trying to reduce the excess MSE after convergence. What changes can I make by throwing in some more hardware at it to get slightly lower MSE ? I am afraid I can afford only 2x~3x more multipliers only. Any hope ? I am trying to keep my convergence rate constant. Thanks.


LMS Pseudo Inverses of Complex Matrices

Started by Bob Cain in comp.dsp15 years ago 7 replies
LMS

I have an overconstrained system of equations (m equations in n variables with m> n) where the variables are complex. Using the...

I have an overconstrained system of equations (m equations in n variables with m> n) where the variables are complex. Using the Moore-Penrose pseudoinverse method of solution is giving me results that are terribly counterintuitive. Does anyone know what exactly is minimized in the least squares sense when the variables and solution coeficients are complex? Thanks, Bob -- "


Porting LMS from floating-point to fixed-point processor

Started by Daniel in comp.dsp15 years ago 35 replies

Hello Everybody, for my diploma thesis, I have to implement a Least-Mean-Square Algorithm on a fixed-point DSP (TI 6416). The LMS was...

Hello Everybody, for my diploma thesis, I have to implement a Least-Mean-Square Algorithm on a fixed-point DSP (TI 6416). The LMS was implemented on a floating-point processor(TI 6713) earlier, so I just to the code and copied it. Of course, there are a lot of float variables in the code. When I ran the program, it workes for small FIR orders (6), but the larger the order of the filter, th...


Problems after freezing FIR coefficients of Acoustic Echo Canceller

Started by johan kleuskens in comp.dsp15 years ago 12 replies

Hi, We are currently working on an acoustic echocanceller based on the well know NLMS principle. This echocanceller works fine as long as we...

Hi, We are currently working on an acoustic echocanceller based on the well know NLMS principle. This echocanceller works fine as long as we feed the echocanceller with a echo signal that is generated by an audio processing program. When using this ideal echo signal, freezing the FIR coefficients works like it should: the echo is still cancelled because the FIR tabs contain a represen...


Writing an echo canceller

Started by Duane in comp.dsp15 years ago 2 replies

Hey everyone, I started writing an echo canceller as a side project, but am already hitting a few snags.. I have a few research papers on the...

Hey everyone, I started writing an echo canceller as a side project, but am already hitting a few snags.. I have a few research papers on the floor here I've been playing with. My first problem is with the NLMS algorithm.. Dispite implementing several versions of them, I still always seem to have a problem with it going unstable.. I'm not even sure if it's unstable or not, but it can ge...


filtered - x - LMS

Started by Gie78 in comp.dsp15 years ago 1 reply
LMS

Thanks for your prompt reply. When I insert an delay into the primary path the result would be much better. OK I think this problem is known as...

Thanks for your prompt reply. When I insert an delay into the primary path the result would be much better. OK I think this problem is known as "causality problem". And the crosscorrelation between the disturbance signal and the filterd-x-signal how should these crosscorrelation look like? I think the two signals should be well correlated for a good result, or? But could you explain me why?...


filtered-x-LMS

Started by Gie78 in comp.dsp15 years ago 1 reply

First of all thanks for the explanation. I want to implement a feedforward method, is there a more detailed explanation of the correlation...

First of all thanks for the explanation. I want to implement a feedforward method, is there a more detailed explanation of the correlation problem at Kuo and Morgan "Active noise control Systems". If so on which pages or chapter can I find it? One more question do you know both books Kuo and Elliots? If so which book do you prefer? What do you think make it sense to buy both books? Thank...


Real time implementation - Adaptive filtering of evoked potentials

Started by Axey in comp.dsp14 years ago 12 replies

Hi all, we are doing a project on "Adaptive Filtering of Evoked Potentials". The thing is we wanted to implement the same in real time...

Hi all, we are doing a project on "Adaptive Filtering of Evoked Potentials". The thing is we wanted to implement the same in real time i.e get the evoked potentials data in real time and filter the noise using adaptive LMS algorithm. We are using the DSP processor from Texas Instruments "TMS320C54X" for this purpose. We are using the "Code composer" software where the C-code is conve...


LMS convergence criterion

Started by Johan in comp.dsp14 years ago 1 reply
LMS

Hi, I want to perform the convergence stability of my LMS algo. but I don't know how. I use the error, the variance of the coefficients and a...

Hi, I want to perform the convergence stability of my LMS algo. but I don't know how. I use the error, the variance of the coefficients and a time (a step time to compute the variance). I know that the last one isn't very accurate. Anybody can help me. Thank you in advance. Jo This message was sent using the Comp.DSP web interface on www.DSPRelated.com


adaptive filter

Started by ldkha1979 in comp.dsp14 years ago 1 reply

hi all, i have to design an adaptive filter using Matlab. What i need is to find a symmetric FIR filter from 2 signals, desired and input. I...

hi all, i have to design an adaptive filter using Matlab. What i need is to find a symmetric FIR filter from 2 signals, desired and input. I don't know why my results cannot be converged. I'm using NLMS algorithm. Here are my code. Can anyone help me to find my wrong part, or can give me a suitable one? Thks so much function h = adp_filter(d,x,f0) % h = adp_filter(d,x,f0) % Inputs: % ...


Adaptive Equilization

Started by Anonymous in comp.dsp14 years ago 1 reply

I'm looking at the block diagram on page 247 of "Adaptive Signal Processing" from B.Widrow and S.D.Stearns. Anyway, the block diagram shows an...

I'm looking at the block diagram on page 247 of "Adaptive Signal Processing" from B.Widrow and S.D.Stearns. Anyway, the block diagram shows an LMS filter in the center, and the Quantizer on the bottom right. The output of the Quantizer goes to a subtraction element (which subtrats the LMS output from the Quantizer), then the output goes to the "Gate". First, what is the output of the Q...


Adaptive Filter - LMS Algorithm

Started by Praveen in comp.dsp14 years ago 6 replies

Hello, I am implementing the LMS Algorithm for acoustic echo canceller at a very basic level. In order to adapt the co-efficients of the...

Hello, I am implementing the LMS Algorithm for acoustic echo canceller at a very basic level. In order to adapt the co-efficients of the filter using LMS algorithm, a desired signal is required. This would minimise the error between the output signal(with echo) and the desired signal. I would like to know what is the desired signal that can be assumed here? Since the adaptive filter woul...


Least Mean Fourth

Started by Real_McCoy in comp.dsp14 years ago 1 reply
LMS

I have been looking at Widrows paper on LM4 (like LMS but minimises E(error^4) ) It looks good in that the min error is smaller and the cost...

I have been looking at Widrows paper on LM4 (like LMS but minimises E(error^4) ) It looks good in that the min error is smaller and the cost in computation is not much more - so why isn't it used more? I saw some papers on stability issues and I have a paper on Normalised LM4. McC


Paper on LMS adaptive equalization

Started by Alberto in comp.dsp14 years ago 4 replies

I am in search of the PDF of this paper : "Efficient realization of asynchronous LMS adaptive equalization" by Jan W.M. Bergmans I have...

I am in search of the PDF of this paper : "Efficient realization of asynchronous LMS adaptive equalization" by Jan W.M. Bergmans I have been unable to find it with Google, nor to find an email address for the author. Does anybody have it ? Can you send it to my email address dibene @ usa.net (remove the spaces surrounding the @ symbol) Thanks in advance Alberto


problem with reference signal in LMS algoritm

Started by ralph in comp.dsp14 years ago 16 replies

I want to use LMS algoritm to adaptive noise cancellation. The problem is that I have only primary signal and thats all what I have, I...

I want to use LMS algoritm to adaptive noise cancellation. The problem is that I have only primary signal and thats all what I have, I don't know anything aboyt reference signal, and my question is: how can I get reference signal? What should I take as reference signal? Please help me with this problem


LMS algorithm for noise cancellation in speech

Started by radha_r in comp.dsp14 years ago 7 replies

hi all, i am trying to implement the LMS algorithm that cancells the background noise from the input signal needed.i have tried it for sin wave...

hi all, i am trying to implement the LMS algorithm that cancells the background noise from the input signal needed.i have tried it for sin wave it is working fine.But when i have tried to give noise mixed input signal(wav file) it was not working properly.. the scenario i followed is first i took a wav file as the input signal and then created the noise input with rand() fn/. in matlab.Then...


Can anybody tell me sth. about Leaky NLMS

Started by Jing in comp.dsp14 years ago 3 replies

Hi, I am trying to implement leaky NLMS for adaptive beamforming, but I found that the noise reduction performance is worse than that of NLMS....

Hi, I am trying to implement leaky NLMS for adaptive beamforming, but I found that the noise reduction performance is worse than that of NLMS. The data I used are simulated data, and I assume the noise and speech sources are well seperated. Can anybody give me some suggestions about applying leaky NLMS for GSC? Is Leaky really better than that of NLMS? Thanks a lot! Best Regards, Jing ...


beamforming question/help...

Started by Anonymous in comp.dsp14 years ago 14 replies

I've got a question regarding weight calculation in a beamformer... In previous work, I was able to ADAPTIVELY determine weights...

I've got a question regarding weight calculation in a beamformer... In previous work, I was able to ADAPTIVELY determine weights using something like LMS, but I needed to generate an actual inbound signal, noise, reference, interferers and what-not... What I'd like to do now is pick a few directions in which to intentionally form beams, and other directions to intentionally form nulls. Th...


echo canceller initial convergence

Started by Anonymous in comp.dsp14 years ago 2 replies

Hi All, I am trying to implement a long tail (selectable 512/1024 taps) echo canceller thats compliant at least with tests 2 and 3 of g.168....

Hi All, I am trying to implement a long tail (selectable 512/1024 taps) echo canceller thats compliant at least with tests 2 and 3 of g.168. The implementation is using leaky NLMS, coded in C using 16 bit fixed point arithmetic (the coefficients are also stored as 16 bit, not 32). I have two concrete questions and, in addition to those, wanted to put out a "general call for advice". The ...