• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2002년도 춘계종합학술대회
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• pp.518-522
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• 2002

This paper proposes a new application of Fuzzy logic to Variable Step Size Least Mean Square (VS-LMS) adaptive beamforming algorithm in CDMA systems. The proposed algorithm adjusts the step size of the Least Mean Square (LMS) by using the application of Fuzzy logic in which the increase or decrease of step size depends on the fuzzy inference results of the Mean Square Error (MSE). Computer simulation results show that the proposed algorithm has a better capacity of tracking compared with the conventional LMS algorithms and other variable step size LMS algorithms.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 추계종합학술대회 논문집
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• pp.1183-1186
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• 1998

Adaptive nois canceller can extract the noiseremoved spech in noisy speech signal by adapting the filter-coefficients to the background noise environment. A kind of LMS algorithm is one of the most popular adaptive algorithm for noise cancellation due to low complexity, good numerical property and the merit of easy implementation. However there is the matter of increasing misadjustment at voiced speech signal. Therefore the demanded speech signal may be extracted. In this paper, we propose a fast and noise robust wavelet packet adaptive noise canceller with NLMS-SUM method LMS combined algorithm. That is, we decompose the frequency of noisy speech signal at the base of the proposed analysis tree structure. NLMS algorithm in low frequency band can efficiently dliminate the effect of the low frequency noise and SUM method LMS algorithm at each high frequency band can remove the high frequency nosie. The proposed wavelet packet adaptive noise canceller is enhanced the more in SNR and according to Itakura-Satio(IS) distance, it is closer to the clean speech signal than any other previous adaptive noise canceller.

• Journal of Communications and Networks
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• 제9권1호
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• pp.18-27
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• 2007

The conventional normalized least mean square (NLMS) algorithm is the most widely used for adaptive identification within a non-stationary input context. The convergence of the NLMS algorithm is independent of environmental changes. However, its steady state performance is impaired during input sequences with low dynamics. In this paper, we propose a new NLMS algorithm which is, in the steady state, insensitive to the time variations of the input dynamics. The square soot (SR)-NLMS algorithm is based on a normalization of the LMS adaptive filter input by the Euclidean norm of the tap-input. The tap-input power of the SR-NLMS adaptive filter is then equal to one even during sequences with low dynamics. Therefore, the amplification of the observation noise power by the tap-input power is cancelled in the misadjustment time evolution. The harmful effect of the low dynamics input sequences, on the steady state performance of the LMS adaptive filter are then reduced. In addition, the square root normalized input is more stationary than the base input. Therefore, the robustness of LMS adaptive filter with respect to the input non stationarity is enhanced. A performance analysis of the first- and the second-order statistic behavior of the proposed SR-NLMS adaptive filter is carried out. In particular, an analytical expression of the step size ensuring stability and mean convergence is derived. In addition, the results of an experimental study demonstrating the good performance of the SR-NLMS algorithm are given. A comparison of these results with those obtained from a standard NLMS algorithm, is performed. It is shown that, within a non-stationary input context, the SR-NLMS algorithm exhibits better performance than the NLMS algorithm.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1987년도 전기.전자공학 학술대회 논문집(II)
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• pp.1379-1382
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• 1987

IN ADAPTIVE LINEAR PREDICTION, AN ADAPTIVE CAPABILITY IS BUILT INTO THE PROCESSOR SUCH THAT AS THE IMAGE STATISTICS CHANGE, THE PREDICTION FILTER COEFFICIENTS THEMSELVES CHANGE, PRODUCING A NEW FILTER MORE CLOSELY OPTIMIZED TO THE NEW SET OF IMAGES STATISTICS. THE LMS ALGORITHM MAY BE USED TO ADAPT THE COEFFICIENT OF AN ADAPTIVE PREDICTION FILTER FOR IMAGE SOURCE ENCODING. IN THIS PAPER, TWO CODING SYSTEMS USING DPCM AND LMS ALGORITHMS RESPECTIVELY FOR OBTAINING THE FIRST TRANSFORMED COEFFICIENT IN HYBRID CODING ARE COMPARED.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 추계종합학술대회 논문집(4)
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• pp.115-118
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• 2000

We present a new variable step size LMS algorithm using the correlation between reference input and error signal of adaptive filter. The proposed algorithm updates each weight of filter by different step size at same sample time. We applied this algorithm to adaptive multip]e-notch filter. Simulation results are presented to compare the performance of the proposed algorithm with the usual LMS algorithm and another variable step algorithm.

• 한국음향학회지
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• 제15권5호
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• pp.30-37
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• 1996

통신 시스템에서 LMS 알고리즘을 이용하여 설계한 적응 등화기는 구조의 간단함과 효율성에 비해 수렴 속도가 느린 단점이 있다. 본 논문에서는 이러한 단점을 보완하기 위해 웨이브렛 변환 평면에서 적응 동기화를 설계하여 채널의 왜곡정도와 필터 차수의 변화에 따른 수렴 속도를 시평면 LMS 알고리즘을 이용해 설계한 등화기와 비교하였다. 시뮬레이션결과 웨이브렛 평면에서의 적응 등화기가 기존의 등화기보다 우수한 성능 향상을 보였다.

• 대한전자공학회논문지TC
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• 제46권3호
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• pp.19-25
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• 2009

본 논문에서는 이동통신채널에서 발생하는 간섭현상을 제거하기 위한 적응형 채널추정(adaptive channel estimate) 알고리듬을 제안하였다. 기존 LMS 알고리듬은 입출력사이 오차를 줄이기 위해 사용하는 첫 기준신호의 선택에 따라 수렴속도와 오차정확도에 많은 영향을 받는다. 본 논문에서 제안한 적응형 채널추정 알고리듬은 간섭신호와 유사한 기준신호를 정하기 위해 LMS 알고리듬을 수행하기 전에 병렬의 컨볼루션 연산을 수행한다 컨볼루션 연산을 통해 출력된 신호는 채널의 지연시간과 진폭특성을 가지고 있어 간섭신호와 유사한 특성을 가진다. 제안된 알고리듬의 성능평가는 이동통신환경과 유사한 Jake's 모델에 Doppler 주파수는 130 Hz, Random한 5개의 경로가 존재하는 Rayliegh 다중경로 채널환경에서 실험하였다. 모의실험결과 기존 LMS 알고리듬은 데이터 150개를 반복 수행함으로써 약 -40 dB의 제곱오차수렴을 보였고 제안한 적응형 채널추정 알고리듬은 데이터 200개를 반복 수행함으로써 약 -80 dB의 제곱오차수렴을 보였다. 데이터의 반복연산에 따른 수렴속도는 다소 증가하였으나 제곱오차정확도는 약 40 dB의 우수한 개선특성을 보였다.

• 한국통신학회논문지
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• 제28권3C호
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• pp.264-270
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• 2003

Application of the Filtered-X LMS adaptive filter to active noise control requires to estimate the transfer characteristics between the output and the error signal of the adaptive canceler. In this paper, we derive an adaptive control algorithm and analyze its convergence behavior when the acoustic noise is assumed to consist of multiple sinusoids. The results of the convergence analysis of the Filtered-X LMS algorithm indicate that the effects of parameter estimation inaccuracy on the convergence behavior of the algorithm are characterize by two distinct components : Phase estimation error and estimated magnitude. In particular, the convergence of the Filtered-X LMS algorithm is shown to be strongly affected by the accuracy of the phase response estimate. Simulation results of the algorithm are presented which support the theoretical convergence analysis.

• 디지털산업정보학회논문지
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• 제10권3호
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• pp.197-205
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• 2014

In applications of adaptive noise control or active noise control, the presence of a transfer function in the secondary path following the adaptive controller and the error path, been shown to generally degrade the performance of the Least Mean Square (LMS) algorithm. Thus, the convergence rate is lowered, the residual power is increased, and the algorithm can become unstable. In general, in order to solve these problems, the filtered-x LMS (FX-LMS) type algorithms can be used. But these algorithms have slow convergence speed and weakness in the environment that the secondary path and error path are varied. Therefore, I present the new algorithm called the "Bi-directional Filtered-x (BFX) LMS" algorithm with nearly equal computation complexity. Through experimental study, the proposed BFX-LMS algorithm has better convergence speed and better performance than the conventional FX-LMS algorithm, especially when the secondary path or error path is varied and the impulsive disturbance is flow in.

• 한국전자파학회논문지
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• 제19권9호
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• pp.1058-1064
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• 2008

본 논문에서는 RF(Radio Frequency) 중계기에서 그룹화 CMA(Constant Modulus Algorithm)와 LMS(Least Mean Square) 알고리즘을 이용하여 적응 필터를 적용시킨 새로운 혼합 간섭 제거기를 제안한다. 송신 안테나에서 수신안테나로 궤환되는 신호는 수신 시스템의 성능을 저하시킨다. 제안한 간섭 제거기는 그룹화 CMA 알고리즘 간섭 제거 기법을 적용시키기 때문에 기존 구조보다 나은 채널 적응 성능과 낮은 MSE(Mean Square Error)을 가진다. 이 구조는 기존 비선형 간섭 제거기에 비해 같은 MSE(Mean Square Error)에 대한 반복수와 하드웨어 복잡도를 줄여준다. 즉, 제안한 알고리즘은 LMS 알고리즘에 비해 평균 자승 에러가 적응 상수에 따라 2.5 dB 또는 4 dB 정도 낮은 값을 보였다. 또한, VSS(Variable Step Size)-LMS 알고리즘에 비해 수렴 속도가 빠르고, 비슷한 평균 자승 에러를 가진다.