• 전기학회논문지P
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• 제65권3호
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• pp.171-177
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• 2016

Noise problem that occurs in living environment is a big trouble in the economic, social and environmental aspects. In this paper, the filtered-X LMS algorithms, the adjoint LMS algorithms, and the robust adjoint LMS algorithms will be introduced for applications in active noise control(ANC). The filtered-X LMS algorithms is currently the most popular method for adapting a filter when the filter exits a transfer function in the error path. The adjoint LMS algorithms, that prefilter the error signals instead of divided reference signals in frequency band, is also used for adaptive filter algorithms to reduce the computational burden of multi-channel ANC systems such as the 3D space. To improve performance of the adjoint LMS ANC system, an off-line measured transfer function is connected parallel to the LMS filter. This parallel-fixed filter acts as a noise controller only when the LMS filter is abnormal condition. The superior performance of the proposed system was compared through simulation with the adjoint LMS ANC system when the adaptive filter is in normal and abnormal condition.

• 한국음향학회지
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• 제12권1호
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• pp.37-46
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• 1993

The adaptive algorithm for the Volterra filter is considered. Owing to its simplicity, the LMS algorithm for adaptive Volterra filter(AVF) is widely used as in linear adaptive filters. However, the convergence speed is unsatisfactory. For improving the convergence speed, the frequency domain LMS second order adaptive Volterra filter(FLMS-AVF) is proposed and analyzed. We show that the time and frequency domain LMS AVF's have the same steady state performance under approprate conditons. Moreover, it can be shown that this algorithm can improve the convergence speed significantly by applying self-orthogonalizing method.

• 한국자동차공학회논문집
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• 제19권4호
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• pp.84-90
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• 2011

In active noise control filter, LMS algorithms which used for control filter, assure the convergence property, and computational burden of these algorithms are proportionate to the filter taps. The convergence speed of LMS algorithms is mainly determined by value of the convergence coefficient, so optimal selection of the value of convergence coefficient is very important. In this paper, We proposed novel adaptive fuzzy logic LMS algorithms with FIR filter structure which has better convergence speed and less computational burden than conventional LMS algorithms, for single channel active noise control with ill conditioned signal case. Computer simulations were performed to show the effectiveness of a proposed algorithms.

• 전기학회논문지
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• 제60권1호
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• pp.150-155
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• 2011

In an active noise control systems, an IIR filter may cause a problem in stability beacause of its poles. For IIR filter, its poles goes sometimes out of a unit circle in a z-plane in the transition state, where the adaptive algorithm converges to the optimum value, which causes the system to diverge. Fuzzy LMS algorithm has a better convergence property than conventional LMS algorithms, but is not applicable to IIR filter because of the reasons. Stabilized adaptive algorithm could be improves stability by moving the pole of IIR filer toward the origin forcibly in the transient state, and by introducing forgetting factor to maintain the optimum convergence when it reaches to the steady state. In this paper, We proposed stabilized adaptive fuzzy LMS algorithms with IIR filter structures, for single channel active noise control with ill conditioned signal case. Computer simulations were performed to show the effectiveness of a proposed algorithm.

• 디지털융복합연구
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• 제11권7호
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• pp.135-140
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• 2013

잡음 추정과 검출 알고리즘에서는 LMS Filter를 이용하여 변화하는 잡음 환경에 빠르게 적응할 수 있도록 한다. 하지만 LMS Filter는 잡음 추정을 위한 일정 시간 동안 적응 시간이 필요하며 신호의 변화가 일어날 경우 더 많은 적응 시간이 소요되는 단점을 가지고 있다. 따라서 이를 보완하기 위하여 프레임 단위의 AELMS Filter를 이용한 잡음 제거 방법을 제안한다. 본 논문은 잡음 환경에서 입력되는 신호를 프레임 단위로 분할하고 평균과 분산을 이용한 예측 LMS Filter를 구성하여 잡음을 제거하므로 잡음 환경이 변화하더라도 빠른 적응 시간으로 잡음을 제거한다. 또한 환경 잡음과 음성 신호가 혼합되어 입력될 때 잡음을 제거하여 음성의 고유 특성을 유지하고 음성 정보 손상을 줄이기 위한 방법이다. 프레임 단위의 AELMS Filter를 이용한 잡음 제거 방법으로 잡음 제거 성능을 평가하였다. 실험 결과 변화하는 환경 잡음을 제거하여 얻은 감쇠도가 평균 6.8dB 향상되었다.

• 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.

• 대한전자공학회논문지
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• 제21권2호
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• pp.29-35
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• 1984

LMS(least mean Square) 알고리즘은 그 간편성에 의하여 적응 필터의 대표적인 알고리즘으로 되고 있다. 본 논문에서는 잘 알려진 1차원 LMS 원응 필터를 2차원 필터로 광장시킬 수 있음을 보였으며, 필터의 목산 속도 양선방안 및 2차원 적응 필터에서 발생하는 몇 가지 문제점들에 관하여 고찰하였다.

• 대한전자공학회논문지
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• 제19권5호
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• pp.12-19
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• 1982

현재까지의 적응 필터(adaptive filter)의 설계 방법을 대별하면 국부 파라미타 최적화(local parameter optimization)방법[1]과 안정성(stability)을 주한점으로 하는 방법[1-3] 의 두 가지로 된다. 그리고 이들 중에서 비교적 간단한 방법은 로칼 파라미터 최적화 방법으로서, 이것에서는 스티피스트-디샌트(steepest-descent)방법[15]을 이용하는 LMS 알고리즘을 대표적인 것으로 들 수 있다. 적응 필터의 설계에 있어서 가장 중요한 것은 수검 속도를 높이는 일이다. 본 논문은 적응 비순환 필터의 설계를 위한 고속 수검 알고리즘을 개발하는 문제에 관하여 연구한 것으로, 적응 이득(adaptation-gain)을 적절히 조정함으로써, 종래 사용되어 오던 LMS 알고리즘 및 그의 변형인 여러 알고리즘에 비하여 수검 속도를 높일 수 있으며 동시에 안정성이 높은 새로운 알고리즘을 제시하였다. 그리고 제안된 알고리즘을 이용한 적응 필터의 특성 배선 문제를 다각도로 검토하였다.

• 한국소음진동공학회논문집
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• 제22권2호
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• pp.146-155
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• 2012

The filtered-x LMS(FX-LMS) algorithm has been applied to the active noise control(ANC) system in an acoustic duct. This algorithm is designed based on the FIR(finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filtered-u LMS algorithm(FU-LMS) based on infinite impulse response(IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• 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.