• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1997년도 한국자동제어학술회의논문집; 한국전력공사 서울연수원; 17-18 Oct. 1997
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• pp.1379-1382
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• 1997

We introduce a fast implementation of the adaptive transversal filter which uses least-mean-square(LMS) algorithm. The fast Hadamard transform(FHT) is used for the implementation of the filter. By using the proposed filter we can get the significant time reduction in computatioin over the conventional time domain LMS filter at the cost of a little performance. By computer simulation, we show the comparison of the propsed Hadamard-domain filter and the time domain filter in the view of multiplication time, mean-square error and robustness for noise.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.2120-2125
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• 2005

The principal idea of vibration isolation is to filter out the response of the system over the corner frequency. The isolation objectives are to transmit the attitude control torque within the bandwidth of the attitude control system and to filter all the high frequency components coming from vibration equipment above the bandwidth. However, when a reaction wheels or control momentum gyros control spacecraft attitude, vibration inevitably occurs and degrades the performance of sensitive devices. Therefore, vibration should be controlled or isolated for missions such as Earth observing, broadcasting and telecommunication between antenna and ground stations. For space applications, technicians designing controller have to consider a periodic vibration and disturbance to ensure system performance and robustness completing various missions. In general, past research isolating vibration commonly used 6 degree order freedom isolators such as Stewart and Mallock platforms. In this study, the vibration isolation device has 3 degree order freedom, one translational and two rotational motions. The origin of the coordinate is located at the center-of-gravity of the upper plane. In this paper, adaptive notch filter finds the disturbance frequency and the reference signal in filtered-x least mean square is generated by the notch frequency. The design parameters of the notch filter are updated continuously using recursive least square algorithm. Therefore, the adaptive filtered-x least mean square algorithm is applied to the vibration suppressing experiment without reference sensor. This paper shows the experimental results of an active vibration control using an adaptive filtered-x least mean squares algorithm.

• Journal of Information Processing Systems
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• 제15권4호
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• pp.748-764
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• 2019

In order to deal with the filtering delay problem of least mean square adaptive filter noise reduction algorithm and music noise problem of spectral subtraction algorithm during the speech signal processing, we combine these two algorithms and propose one novel noise reduction method, showing a strong performance on par or even better than state of the art methods. We first use the least mean square algorithm to reduce the average intensity of noise, and then add spectral subtraction algorithm to reduce remaining noise again. Experiments prove that using the spectral subtraction again after the least mean square adaptive filter algorithm overcomes shortcomings which come from the former two algorithms. Also the novel method increases the signal-to-noise ratio of original speech data and improves the final noise reduction performance.

• 전자공학회논문지S
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• 제36S권7호
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• pp.81-89
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• 1999

통신에 의한 전송 영상은 잡음이나 번짐 또는 일그러짐 등을 항상 포함한다. 본 논문에서는 적응형 일반스텍 최적화 필터(OAGSF: optimal adaptive generalized stack filter)라는 영상복원 공간 필터를 제안하였는데, 이는 영상의 복원에서 잡음 제거율과 외곽선 정보의 보존률의 증가을 위해 신경회로맘의 역전파 학습 알고리즘의 가중치 학습 알고리즘을 기반으로 적응형 일반스택 필터(AGSF)를 최적화 시킨 것이다. 적응형 일반스택 필터는 일반스택 필터(GSF: generalized stack filter)와 적응형 다단계 메디안 필터(AMMF; adaptive multistage median filter)로 구분하고, 일반스텍 필터는 스택 필너치 기능을 보완한것이고, 적응형 다단계 메디안 필터는 메디안 필터의 외곽선 정보 보존률을 높인 것이다. 신경회로망의 역전파 학습 알고리즘에 대하여 두가지 가중치 학습 알고리즘인 최소평균절대 (LMA:Least Mean Absolute) 알고리즘과 최소평균자승(LMS: Least Mean Square) 알고리즘을 이용하여 적응형 일반스택 필터를 최적화하였다. 본 논문에서 제시한 신경회로망을 이용한 영상복원 공간필터에 대해 실험결과를 통해 제시하였다.

• 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.

• 한국전자통신학회논문지
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• 제5권2호
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• pp.145-151
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• 2010

본 논문에서는 두 개의 음으로부터 전달되어온 음장의 상태를 구하여 역 필터를 구성하는 적응필터 및 신경회로망을 사용한 음장의 역 필터링 시스템을 제안한다. 본 논문에서는 최소 2승 평균법을 사용하여 FIR 필터의 계수를 계산하여 이를 갱신함으로써 역 필터링을 구축하는 방법을 사용한다. 본 논문에서 제안한 신경회로망 및 적응필터의 기법에 의하여 비선형 왜곡이 있는 간단한 파형이 학습 가능한 것을 실험 결과로부터 확인할 수 있었다.

• 한국소음진동공학회논문집
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• 제13권4호
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• pp.239-246
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• 2003

Application of the filtered-x Least Mean Square(LMS) adaptive filter to active noise control requires to estimate the transfer characteristics between the output and the error signal of the adaptive controller. In this paper, we derive the filtered-x adaptive noise control algorithm and analyze its convergence behavior when the acoustic noise consists of multiple sinusoids. The results of the convergence analysis of the filtered-x LMS algorithm indicate that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components Phase estimation error and estimated gain. In particular, the convergence is shown to be strongly affected by the accuracy of the phase response estimate. Simulation results are presented to support the theoretical convergence analysis.

• 한국위성정보통신학회논문지
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• 제10권2호
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• pp.19-23
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• 2015

위성통신서비스의 가용율 및 시스템 throughput 향상을 위해 사용하는 리턴링크 ACM(Adaptive Coding & Modulation)의 원리를 기술하였고, LMS(Least Mean Square) 기반 적응형 필터를 이용한 채널 예측 및 단말의 전송 MODCOD(Modulation & Code rate) 결정 알고리즘의 최적화 과정을 서술하였다. 시뮬레이션 결과 LMS 알고리즘은 필터 계수가 2차이고, ${\mu}$(step size) 값이 0.00026인 경우 MMSE(Minimum Mean Square Error)가 최소임을 알 수 있다. 이때 MODCOD 결정 알고리즘을 위한 SNR 마진이 0.3dB일 경우 MODCOD 결정 오차를 최소화 할 수 있음을 확인하였다.

• 대한의용생체공학회:의공학회지
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• 제13권2호
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• pp.107-114
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• 1992

An adaptive filter for noise cancellation of ECG Is proposed. An adaptive noise canceller using the least mean squares algorithm Is used to reduce unwanted noise. An adaptive filter for nolse cancella lion minimizes the mean-square error between a primary input and a reference input. A primary input is the noisy ECG, and a reference input is a noise that Is correlated in some way with the noise in the primary input or a signal that is correlated only with ECG in the primary input.

• Journal of Power Electronics
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• 제16권1호
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• pp.310-318
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• 2016

This paper presents a new orthogonal signals generator (OSG) with DC Offset rejection for implementing a phase locked loop (PLL) in single-phase grid-connected power systems. An adaptive filter (AF) based on the least mean square (LMS) algorithm is used to constitute the OSG in this study. The DC offset in the measured grid voltage signal can be significantly rejected in the developed OSG technique. This generates two pure orthogonal signals that are free from the DC offset. As a result, the DC offset rejection performance of the presented single-phase phase locked loop (SPLL) can be enhanced. A mathematical model of the developed OSG and the principle of the adaptive filter based SPLL (AF-SPLL) are presented in detail. Finally, simulation and experimental results demonstrate the feasibility of the proposed AF-SPLL.