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...


Equaliser for Proakis Channels

Started by cpshah99 in comp.dsp12 years ago 1 reply
LMS

Hello I am trying to implement LMS and DFE equaliser for Proakis Channels. But I am not able to get exact BER plots. If I remove the precursor...

Hello I am trying to implement LMS and DFE equaliser for Proakis Channels. But I am not able to get exact BER plots. If I remove the precursor of Channle B and and implement the random channel e.g. [0.815 0.407 0.2] instead of [0.407 0.815 0.407], I am getting good BER but not for exact Channel B. Please help me in this respect. Thanks Chintan


NLMS Step Size Question

Started by jai....@gmail.com in comp.dsp12 years ago 2 replies

I'm running into difficulty implementing the NLMS algorithm, w.r.t. step size.. I'm using Matlab to model the algorithm, and here is the core...

I'm running into difficulty implementing the NLMS algorithm, w.r.t. step size.. I'm using Matlab to model the algorithm, and here is the core of the calculation to update the tap weights: nlms_coeff = af_parms.mu/pwr(i) * en; af_parms.state.weights = af_parms.state.weights + xn .* nlms_coeff; - pwr(i) is the calculated signal power for sample 'i' - xn is the data samples used for 'L'...


MUSIC and LMS together??

Started by scc in comp.dsp12 years ago 6 replies

Hello, sorry if I seem confused...but I'm actually confused... Could someone tell me if I need to implement a DoA...

Hello, sorry if I seem confused...but I'm actually confused... Could someone tell me if I need to implement a DoA stimation algorithm (like MUSIC or ESPIRT) with an adaptative algorithm (like LMS) on development of an intelligent beamforming system?? I mean, I don't know the DoA of the signal so, how could I use only the LMS algorithm? Is there a diference between using these algorit...


How can I know my desired signal for LMS algorithm?

Started by scc in comp.dsp12 years ago 2 replies

Hi, I've got a problem with the implementation of LMS algorithm. I'm looking for desing a Digital beamforming antenna for my...

Hi, I've got a problem with the implementation of LMS algorithm. I'm looking for desing a Digital beamforming antenna for my final project.Someone told me that it's better if I use LMS algorithm to calculate the weights of the array elements because I don't know how many users there are and those users are mobile users. My problem is that LMS algorithm needs to know the desired signal...


problem understanding LMS noise whitening alg.

Started by samwo123 in comp.dsp12 years ago

Hi. I am self-studying adaptive signal processing. I've read C.R. Johnson's Adaptive Signal Processing, and some parts of Haykin's. I am...

Hi. I am self-studying adaptive signal processing. I've read C.R. Johnson's Adaptive Signal Processing, and some parts of Haykin's. I am reading this paper "Self adaptive decision feedback equalization: Application to high order QAM signal" In the paper, adaptive whitening filter is applied to the received signal. The whitened signal is then fed to another adaptive FFE equalizer to recov...


plotting time domain signal for LMS

Started by shamma in comp.dsp12 years ago

hi My LMS algorithm is working perfectly when i plot constellation but it doesnot work when i plot the time domain signal,it gives me a...

hi My LMS algorithm is working perfectly when i plot constellation but it doesnot work when i plot the time domain signal,it gives me a constant zero signal on the output can anyone tell me where i am wrong here is my matlab code Nb=2048 temp=rand(1,Nb); for i=1:Nb if(temp(i)> 0.5) a(1,i)=1; else a(1,i)=-1; end end M1=5; M2=5; N2=12; Md=N2-M2; ph=pi-pi/8; gc=.707*ex


Basic problem about cost function of LMS noise whitening filter

Started by samwo123 in comp.dsp12 years ago 5 replies

X(z)---> H(z)-----> Y(z) I want to implement an adaptive noise whitening filter h(n). The desired property of y(n) is that E{|y(n)|^6} =...

X(z)---> H(z)-----> Y(z) I want to implement an adaptive noise whitening filter h(n). The desired property of y(n) is that E{|y(n)|^6} = SigmaD (constant). H(z) is a freq response of an all pole iir filter of length N, and H(z) = 1/(1+sum(h_i*z^-i)). h(n)'s adaptation rule is h_i(n+1) = h_i(n) - mu* d(J)/dh, where J is the cost function. On papers I have, J = E{|y(n)|^2}. The goal


LMS also simulink

Started by me_chilled in comp.dsp12 years ago 1 reply
LMS

Hi All, I am wanting to implement LMS algo in simulink without using the LMS block given in simulink library. My problem is that ,I want...

Hi All, I am wanting to implement LMS algo in simulink without using the LMS block given in simulink library. My problem is that ,I want to apply the calculated filter coefficients to the ''input'' of the LMS block constantly to make it similar to the ''desired'' . But the problem is e.g. if the ''input'' and ''desired'' becomes same the filter coeffients recduces to a filter with unit im...


Decision feedback Equaliser

Started by cpshah99 in comp.dsp12 years ago 2 replies

Hello I have implemented Decision Feedback Equaliser where I dont knw the channel at RX. I am using Proakis Channel C and I am using LMS algo...

Hello I have implemented Decision Feedback Equaliser where I dont knw the channel at RX. I am using Proakis Channel C and I am using LMS algo to train my filters. Now my problem is: 1. I am not getting any gain over uncoded performance: I think because DFE creates burst of errors because of errors in feedback path. And not a single code can correct this burst error unless we use interleaver....


tracking adaptive algorithm

Started by sopapo in comp.dsp12 years ago 13 replies

Hi everyone, I'm working on my phd thesis in a field related with the active noise control and i'm looking for an algorithm with good...

Hi everyone, I'm working on my phd thesis in a field related with the active noise control and i'm looking for an algorithm with good tracking capability, to work with a time-varying input signal (non stastionary). I have made several test with FXLMS, N-FXLMS, Leaky-FXLMS without much succes, I'm going in wrong direction with this algorithms? Variable-step-size-LMS could improve that...


Adaptive Algorithm for Zero Forcing Equalizer

Started by despite in comp.dsp12 years ago 7 replies

Hello, Could you please give me an example of adaptive algorithm that can be used for a zero forcing equalizer. I am a bit confused. If I would...

Hello, Could you please give me an example of adaptive algorithm that can be used for a zero forcing equalizer. I am a bit confused. If I would like to use peak distortion as cost function, could I use LMS algorithm to make my equalizer adaptive??. -If yes, how? (Because the error fed to LMS algorithm is difference between equalizer output and desired signal. Then no difference remains with ...


step size selection for LMS algorithm

Started by cpshah99 in comp.dsp12 years ago 15 replies

Hi All I have implemented Adaptive Decision Feedback Equaliser using LMS Algorithm. Now I am using 4-QAM Modulation scheme. In the training...

Hi All I have implemented Adaptive Decision Feedback Equaliser using LMS Algorithm. Now I am using 4-QAM Modulation scheme. In the training mode, step size for feedforward filter is 0.01 and for feedback filter is 0.003. And in decision directed mode the step size for feedforward filter is 0.001 and for feedback filter is 0.003. I have got these values after playing thru different values. ...


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 ...


LMS and SIMD with adsp21161

Started by hbarcellos in comp.dsp11 years ago

Hello all, I'm a Brazilian student of Telecomunication Engineering... and I'm working with a research about Active Noise Control ! So, I'm...

Hello all, I'm a Brazilian student of Telecomunication Engineering... and I'm working with a research about Active Noise Control ! So, I'm using a FIR Filter with LMS, both with SIMD. The FIR Filter is OK, I followed the example of Analog Devices (site) but the LMS I wrotte by my own, and I'm not so sure if it's correct !! Somebody have an example of FIR using LMS with SIMD, please (211...


Please, I need a help with FIR/LMS using SIMD adsp 21161

Started by hbarcellos in comp.dsp11 years ago 3 replies

Hello all, I work with ADSP21161N... my research is about Active Noise Control... I'll try explain my problem... I've a file, with the...

Hello all, I work with ADSP21161N... my research is about Active Noise Control... I'll try explain my problem... I've a file, with the samples (160 samples - 48Khz of sample frequency) of a 300hz signal, I use that file and circular buffer to generate the signal, and than I send that signal to the output... My project is similiar a example SIMD_FIR in analog device site... I'm using a FIR...


Need help on LMS based equalizer for GSM

Started by richard_zhang in comp.dsp11 years ago 4 replies

Hi all, I am doing GSM software defined radio and trying to use normalized LMS algorithm to implement an equalizer, since LMS has low...

Hi all, I am doing GSM software defined radio and trying to use normalized LMS algorithm to implement an equalizer, since LMS has low complexity. But now I find that the BER is not improved at all after adding the equalizer block. I just simply take the normalized LMS algorithm and use it, and don't have deep investigation on it. My questions are: 1) typically, how much the BER performanc...


write data in file .dat using assemble

Started by hbarcellos in comp.dsp11 years ago 1 reply

hello all, just a question; I have a FIR filter and LMS algorithm to update the weights... after the algorithm converge I need copy the values...

hello all, just a question; I have a FIR filter and LMS algorithm to update the weights... after the algorithm converge I need copy the values of weights in a file, like a .dat or .txt... I'm going use the files in tests with matlab... Is there some way to do that in assembler (my algorithm is all in assembler)... ?? I'm using ADSP-21161 EZ-KIT, and Visual DSP 4.5... my vector of weights,...


LMS algorithm implementation

Started by sivadasankottayi in comp.dsp11 years ago 7 replies

I am currently doing research in Active Noise Control. I am quite new in this field. I have acquired some knowledge in Adaptive filtering. I...

I am currently doing research in Active Noise Control. I am quite new in this field. I have acquired some knowledge in Adaptive filtering. I also read about a few daptive algorithms. To start with I am interested to simulate a feed forward Active Noise Control System incorporating simple LMS algorithm. Anybody who has experience in this field may please help me how to simulate an adaptiv...


adaptive filters; qmf sub-band versus one long filter

Started by Robert Adams in comp.dsp11 years ago 2 replies

I have been experimenting lately with adaptive filters running in the "system identification" mode. There are 2 approaches that are used; 1)...

I have been experimenting lately with adaptive filters running in the "system identification" mode. There are 2 approaches that are used; 1) A simple nlms fir filter approach, filter length = K 2) sub-band approach, where you break the signal into N critically- sampled bands, do N nlms filters of length K/N each, followed by a reconstruction filterbank. The second approach has ma...