• 전기학회논문지
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• 제59권2호
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• pp.476-483
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• 2010

If the data are pre-whitened, then gradient adaptive algorithms which are simpler than higher order statistics algorithms can be used in adaptive blind signal estimation. In this paper, we propose a blind subband affine projection algorithm for multiple-input multiple-output adaptive equalization in the blind environments. All of the adaptive filters in subband affine projection equalization are decomposed to polyphase components, and the coefficients of the decomposed adaptive sub-filters are updated by defining the multiple cost functions. An infinite impulse response filter bank is designed for the data pre-whitening. Pre-whitening procedure through subband filtering can speed up the convergence rate of the algorithm without additional computation. Simulation results are presented showing the proposed algorithm's convergence rate, blind equalization and blind signal separation performances.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -2
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• pp.681-684
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• 2000

Gradient-type algorithms for adaptive IIR notch filters are very attractive in terms of both performances and computational requirements. Generally, it is quite difficult to assess their performances analytically. There have been several trials to analyze such adaptive algorithms as the sign and the plain gradient algorithms for some types of adaptive IIR notch filters, but many of them still remain unexplored. Furthermore, analysis techniques used in those trials can not be directly applied to different types of adaptive IIR notch filters. This paper presents a detailed performance analysis of the sign algorithm for a well-known adaptive IIR notch filter with constrained poles and zeros, which can not be done by just applying the related existing analysis techniques, and therefore has not been attempted yet. The steady-state estimation error and mean square error (MSE) of the algorithm are derived in closed forms. Stability bounds of the algorithm are also assessed. extensive simulations are conducted to support the analytical findings.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2000년도 추계학술대회논문집
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• pp.443-448
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• 2000

In this paper, we propose a novel adaptive feedforward controller for periodic disturbance and noise cancellation, with a frequency tracking capability. It can be added to an existing feedback control system without altering the original closed-loop characteristics, which is based on adaptive algorithm. We introduce novel algorithm "Constrained AFC(adaptive feedforward controller) algorithm" that increase the convergence region regardless of the delay in the closed loop system. In the algorithms, coefficients of the controller are adapted using the residuals of constrained structure which are defined in such a way that the coefficients become time invariant. The proposed algorithm not only estimate the magnitude and phase of the tonal disturbance and noise but also track the frequency of the tone, which changes in quasi-static manner. The frequency tracking algorithm uses the instantaneous frequency approach based on Hilbert transform. A number of computer simulations have been carried out in order to demonstrate the effectiveness of proposed method under various conditions.

• 오토저널
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• 제14권5호
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• pp.95-106
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• 1992

Active control of acoustic noise is an application area of adaptive digital signal processing with increasingly interest along the last year. This work studies the implementation of the multichannel LMS filter and the application of this algorithm for the reduction of the noise inside a vechicle cabin using a number of 'secondary sources' drived by adaptive filtering of a reference noise source. Firstly, we propose the use of an adaptive method for the time-varient optimal convergence factor. Secondly, we propose the use of adaptive delayed inverse model to estimate the elastic-acoustic transfer function presented in vechicle cabin. The original, primary source is often periodic, with a known fundamental frequency. A suitably filtered reference signal can thus be used to drive the secondary sources. An algorithm is presented for adapting the coefficients of an FIR filter feeding such a secondary source in such a way as to minimize the output of a suitably placed microphone. In this algorithm, the coefficients of adaptive filter driving an array of secondary sources can be adapted to minimize the sum of the squares of the outputs of a number of error microphones. The multichannel LMS algorithm displays that such an algorithm is considered suitable to used for the global suppression of noise in vehicle cabin.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 1999년도 춘계종합학술대회
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• pp.39-42
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• 1999

어댑티브 어레이 안테나를 이동체에 적용하는 경우 가중계수벡터를 전파 환경의 변화에 고속 적응시킬 필요가 생긴다. 4소자 등간격 선형 어레이 안테나 시스템에 대하여 일정 진폭의 포락선을 갖는 BPSK와 FSK신호에 LMS와 Kalman filter 알고리즘을 적용한 컴퓨터 시뮬레이션 결과 LMS에 비하여 Kalman filter 알고리즘이 수렴성이 빠르고 신호의 추종성이 매우 뛰어남을 확인 할 수 있었다.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제11권4호
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• pp.305-309
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• 2011

A sparse reconfigurable adaptive filter (SRAF) based on a photonic switch determines the appropriate time delays and weight values for an optical switch implementation of tapped-delay-line (TDL) systems. It is well known that the choice of switch delays is significantly important for efficiently implementing the SRAF. If the same values exist as calculating the sum of weight magnitudes for implementing the connection constraint required by the SRAF, conventional connection algorithm based on sequentially selection the maximum elements might not work perfectly. In an effort to increase the effectiveness of system identification, an upgraded connection algorithm used progressive calculation to obtain the better solution is considered in this paper. The performance of the proposed connection constraint algorithm is illustrated by computer simulation for a system identification application.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2000년도 하계학술대회 논문집 D
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• pp.2429-2432
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• 2000

Adaptive filter algorithm has been used for plant identifier and noise cancellation. This algorithm has been researched for performance enhancement of filtering. The design and development of a reliable system has been becoming a key issue in industry field because the reliability of a system is considered as an important factor to perform the system's function successfully. And the computing with reliability and fault tolerance is a important factor in the case of aviation and nuclear plant. This paper presents design of reliable adaptive filter with fault tolerance. Generally, redundancy is used for reliability. In this case it needs computing or circuit for voting mechanism or computing for fault detection or switching part. But this presented Filter is not in need of computing for voting mechanism, or fault detection. Therefore it has simple computing, and practicality for application. And in this paper, reliability of adaptive filter is analyzed. The effectiveness of the proposed adaptive filter is demonstrated to the case studies of plant identifier and noise cancellation by using DSP.

• 한국전자통신학회논문지
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• 제4권3호
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• pp.224-229
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• 2009

본 논문은 LMS 알고리즘을 이용하여 적응횡단선 필터의 수렴 속도를 향상시키기 위한 효율적인 신호간섭 제어기법을 제안한다. 수신 데이터를 재사용하여 심볼 시간 주기에 계수들을 곱함으로써 적응되는 제안된 알고리즘의 수렴특성이 수렴 속도의 향상을 이론적으로 증명하기 위해 분석한다, 스텝-크기 매개변수 ${\mu}$가 증가됨에 따라 LMS 알고리즘의 수렴 속도가 제어된다. 고유치확산을 증가시킴에 따라 적응 등화기의 수렴속도를 천천히 제어하고 평균 자승 에러의 안정-상태 값을 증가시키는 효과를 나타내며 데이터-재사용 LMS 알고리즘이 적응횡단선 필터의 수렴속도를 (B+1)배만큼 증가시켜 신호간섭제어의 우수성을 입증한다.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2009년도 춘계학술대회
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• pp.883-887
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• 2009

본 논문은 AR(P) 입력에 대해 MPDSAP 알고리즘의 적응과정의 MSE의 통계적 분석을 제시한다. 부밴드 구조에서 인접투사 알고리즘은 적응필터에 다위상 분해와 노블아이덴티티를 적용함으로써 NLMS 알고리즘으로 변환된다. 또한, P차의 Autoregressive(AR) 입력은 정규직교 분해필터에 의해 사전에 백색화 될 수 있다. 부밴드 구조에서 AR(P) 입력의 사전-백색화(pre-whitening)는 SAP 적응필터의 MSE 행동의 통계적 해석을 위한 간단하고 유효한 근사화를 제공한다.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2001년도 ICCAS
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• pp.109.5-109
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• 2001

The separation of scintillation phenomena concurrent with rain attenuation phenomena can be accomplished by filtering. Based on the analysis of satellite signal fading during rain, scintillation and rain attenuation phenomena are examined and extracting from raw data by using adaptive IIR high-pass filter and adaptive IIR low-pass filter. Adaptive IIR filter are designed by using the algorithm of Least Mean p-Power (LMP) Error Criterion which have been modified by Quantizing Gradient technique. This algorithm reduces amount of multiplication computational equal to the length of input data. It is prove here that the convergence speed, variance, bias independence on p values. For this application, p=1 is chosen. The procedure of application ...