• Journal of the Institute of Convergence Signal Processing
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• v.19 no.2
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• pp.42-46
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• 2018

Adaptive signal processing is quite important in various signal and communication environments. In adaptive signal processing methods since the least mean square(LMS) algorithm is simple and robust, it is used everywhere. As the step is varied in the variable step(VS) LMS algorithm, the fast convergence speed and the small excess mean square error can be obtained. Various variable step LMS algorithms are researched for better performances. But in some of variable step LMS algorithms the computational complexity is quite large for better performances. The fixed step LMS algorithm with a low computational complexity merit and the variable step LMS algorithm with a fast convergence merit are combined in the proposed sporadic step algorithm. As the step is sporadically updated, the performances of the variable step LMS algorithm can be maintained in the low update rate using Fibonacci sequence. The performances of the proposed variable step LMS algorithm are proved in the adaptive equalizer.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.245-250
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• 2009

In communications and signal processing area, a number of least mean square adaptive algorithms have been used because of simplicity and robustness. However the LMS algorithm is known to have slow and non-uniform convergence. Various variable step size LMS adaptive algorithms have been introduced and researched to speed up the convergence rate. A variable step size LMS algorithm using the error difference for updating the step size is proposed. Compared with other algorithms, simulation results show that the proposed LMS algorithm has a fast convergence. The theoretical performance of the proposed algorithm is also analyzed for the steady state.

• Journal of the Institute of Convergence Signal Processing
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• v.12 no.2
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• pp.128-132
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• 2011

Least mean square adaptive algorithms have played important role in radar, sonar, speech processing, and mobile communication. In mobile communication area, the convergence rate of a LMS algorithm is quite important. However, LMS algorithms have slow and non-uniform convergence rate problem For overcoming these shortcomings, various variable step LMS adaptive algorithms have been studied in recent years. Most of these recent LMS algorithms have used complex variable step methods to get a rapid convergence. But complex variable step methods need a high computational complexity. Therefore, the main merits such as the simplicity and the robustness in a LMS algorithm can be eroded. The proposed deterministic variable step LMS algorithm is based upon a simple deterministic function for the step update so that the simplicity of the proposed algorithm is obtained and the fast convergence is still maintainable.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.12A
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• pp.1836-1842
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• 2000

이 논문은 레일레이 감쇄 채널 환경에서 IMT-2000 파일럿 심볼 구조의 W-CDMA 시스템 역방향 링크의 채널 추정에 관한 LMS 적용형 알고리즘 성능을 WMSA(Weighted Multi-Slot Averaging)(K=1,2,3), 일정 추정이득 (Constant estimation gain) 및 RLS 알고리즘 성능과 비교 분석하였다. 이 논문의 모형은 IMT-2000 3GPP 규격의 W-CDMA 채널 구조, 변조 및 파일럿 패턴을 이용하였다. 파일럿 심볼 위치의 채널추정은 LMS 알고리즘을 이용하고 데이터 심볼 위치의 채널보상은 선형 보간으로 수행하였다. 저속 도플러 주파수에서는 WMSA(K=1,2,3) 성능이 일정 추정이득, RLS 및 LMS 적응형 알고리즘 성능보다 우수하며, WMSA(K=1) 성능의 경우 일정 추정이득, RLS 및 LMS 적응형 알고리듬 성능과는 큰 차이가 없다. 그리고 LMS 알고리즘 성능은 WMSA(K=1) 성능과 매우 비슷한 결과를 얻었다. 그러나 도플러 주파수가 고속화될수록 LMS 알고리즘의 성능이 WMSA(K=1), 일정 추정이득 및 RLS 알고리즘 성능보다 우수함을 확인하였다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.19 no.5
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• pp.12-19
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• 1982

In general the design methods of adaptive filter are divided into two categories, one is based upon the local parameter optimization theory and the other is based upon stability theory. Among the various design techniques, the LMS algorithm by steepest-descent method which is based upon local parameter optimization theory is used widely. In designing the adaptive filter, the most important factor is the convergence rate of the algorithm. In this paper a new algorithm is proposed to improve the convergence rate of adaptive firter compared with the commonly used LMS algorithm. The faster convergence rate is obtained by adjusting the adaptation gain of LMS algorithm. And various aspects of improvement of the adaptive filter characteristics are discussed in detail.

• Proceedings of the Korea Multimedia Society Conference
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• 2004.05a
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• pp.179-182
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• 2004

무선통신분야에서 LMS5(Least Mean Square) 알고리즘은 식이 간단하고 계산량이 비교적 적기 때문에 널리 사용되고 있다. 그러나 시간영역에서 처리할 경우 입력신호의 고유치 변동폭이 넓게 분포되어 수렴속도가 저하하는 문제점이 있다. 이를 해결하기 위하여 신호를 FFT(Fast Fourier Trasnform)나 DCT(Discrete Cosine Transform)로 변환하여 신호간의 상관도를 제거함으로써 시간영역에서 LMS알고리즘을 적용할 때 보다 수렴속도를 크게 향강시킬 수 있다. 본 논문에서는 수렴속도 향상을 위해 시간영역의 적응 알고리즘을 직교변환인 고속웨이브렛(wavelet)변환을 이용하여 변환영역에서 수행하며, 짧은 필터계수를 가지는 DWT(Discrete Wavelet Transform)특성에 맞는 Fast running FIR 알고리즘을 이용하여 WTLMS(Wavelet Transform LMS)적응알고리즘을 통신시스템에 적용한다. 적응 알고리즘의 성능향상을 위하여 시간에 따라 적응상수의 크기를 가변시켜 수렴 초기에는 큰 적응상수로 따른 수렴이 가능하도록 하고 점차 적응상수의 크기를 줄여서 misadjustment도 줄이는 방법의 적응 알고리즘을 제안하였다. 제안한 알고리즘을 실제로 적응잡음제거기(adaptive noise canceler)에 적용하여 컴퓨터 시뮬레이션을 하였으며, 각 알고리즘들의 계산량, 수렴속도를 이용하여 각각 비교, 분서하여 그 성능이 우수함을 입증하였다.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.3
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• pp.224-229
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• 2009

In this paper, an efficient signal interference control technique to improve the convergence speed of Adaptive transversal filter with LMS algorithm is introduced. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Increasing the eigenvalue spread has the effect of controlling down the rate of convergence of the adaptive equalizer and also increasing the steady-state value of the average squared error and also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS Algorithms.

• Journal of Korea Society of Industrial Information Systems
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• v.4 no.3
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• pp.79-88
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• 1999

One of the most popular algorithm in adaptive signal processing is the least mean square(LMS) algorithm. The majority of these papers examine the LMS algorithm with a constant step size. The choice of the step size reflects a tradeoff between misadjustment and the speed of adaptation. Subsequent works have discussed the issue of optimization of the step size or methods of varying the step size to improve performance. However there is as yet no detailed analysis of a variable step size algorithm that is capable of giving both the speed of adaptation and convergence. In this paper we propose a new variable step size algorithm where the step size adjustment is controlled by square of the prediction error. The simulation results obtained using the new algorithm about noise canceller system and system identification are described. They are compared to the results obtained for other variable step size algorithm. function.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.9
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• pp.764-774
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• 1990

In this paper, make a study on comparison of adaptive filters for predictor characteristics that transversal, lattice, and joint process lattice filter is using the LMS algorithm that is simple structure and pracotical application is easy. The theoical background and structure of each adaptive filters exhibit for practical design. Adaptive convergence condition for optimal weight vector and optimal reflection coefficient make clear, and it is also shown through computer simulation. The error signals and noise characteristics of these filters make a comparative study. In view of the results, joint process lattice filter is shown that most superior characteristic in these adaptie filters.

• Journal of Digital Contents Society
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• v.5 no.4
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• pp.295-299
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• 2004

In this paper, we compare and analyze the LMS ard RLS algorithm of IEEE802.15.3(HDR-WPAN) system. The LMS algorithm have two merits that easily embody and not complex, but convergence speed is slow. The RLS algorithm have fast convergence speed, but very complex. When equalization using LMS algorithm, it can achieve adaptive equalization after 250 sample in fading environment, but case of RLS algorithm can achieve adaptive equalization after just 50 sampls. The computer simulation proved that adaptive equalizer to fast equalization and stability of HDR-WPAN system is more effective using RLS algorithm then LMS algorithm.