• 대한전기학회논문지:시스템및제어부문D
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• 제54권2호
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• pp.104-115
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• 2005

Median filters in the signal processing have been most widely used and have demonstrated the strongest effects. This paper proposes the adaptive median filters with noise detection. The proposed basic algorithm of the filters is to judge whether or not the noises exist on the ground of The Noise Judgment Standards. Just in case the existence of the noises is verified by the algorithm, it takes the median filter. In order to judge the existence of the noises by the algorithm, this paper introduced the noise detection method by local central variance. As a result of comparing and analyzing the features and performance of the proposed filters and the existing [5]-[10] filters on the same conditions, it was verified that the former proved to be better than the latter, Observed even by naked eyes, it was similar, too. Accordingly, it's proved that the adaptive median filters by local central variance are useful in removing the impulse noise of the median filter and reinforce the edge preservation ability.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2000년도 제15차 학술회의논문집
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• pp.240-240
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• 2000

This paper presents two algorithms based on the concept of the frequency domain adaptive filter(FDAF). First the frequency domain recursive least squares(FRLS) algorithm with the overlap-save filtering technique is introduced. This minimizes the sum of exponentially weighted square errors in the frequency domain. To eliminate discrepancies between the linear convolution and the circular convolution, the overlap-save method is utilized. Second, the sliding method of data blocks is studied Co overcome processing delays and complexity roads of the FRLS algorithm. The size of the extended data block is twice as long as the filter tap length. It is possible to slide the data block variously by the adjustable hopping index. By selecting the hopping index appropriately, we can take a trade-off between the convergence rate and the computational complexity. When the input signal is highly correlated and the length of the target FIR filter is huge, the FRLS algorithm based on the block processing technique has good performances in the convergence rate and the computational complexity.

• 전자공학회논문지B
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• 제30B권6호
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• pp.28-34
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• 1993

An algorithm for MMADF(modified multiplication-free adaptive filter) which need not to multiplication arithmatic operation is proposed to improve the performance of FSE (fractionally spaced equalizer) which reduce the ISI(intersymbol interference) in signal transfer channel. The input signals are quantized using DPCM and the reference signals is processed using a first-order linear prediction filter. The convergence properties of Sign. MADF and M-MADF algorithm for updating of the coefficients of a FIR digital filter of the fractionally spaced equalizer (FSE) are investigated and compared with one another. The convergence properties are characterized by the steady state error and the convergence speed. It is shown that the convergence speed of M-MADF is almost same as Sign algorithm and is faster than MADF in the condition of same steady state error. Especially it is very useful for high correlated signals.

• 한국산학기술학회논문지
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• 제13권8호
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• pp.3647-3653
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• 2012

능동 소음제어 시스템에 사용되는 IIR 필터 구조는 구조적으로 안정성이 보장되어야 하며 이는 분모 전달 함수의 근이 단위원 내부에 존재하여야 한다. 따라서 이를 결정하는 제어 필터의 계수의 적절한 조정이 중요해 진다. 본 논문에서는 적응과정에서 불안정할 우려가 있는 IIR 필터 구조를 가지는 Filtered_U LMS 알고리즘에 안정화 알고리즘과 수렴속도 향상을 위한 퍼지논리를 이용한 수렴계수 계산 알고리즘을 제안하였다. 제안한 알고리즘이 FIR 필터 구조 알고리즘보다 계산량이 적고 수렴특성이 우수함을 시뮬레이션을 통하여 보였다.

• 한국정보처리학회논문지
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• 제6권7호
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• pp.1980-1987
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• 1999

복원 영상은 원 영상에 비해 항상 왜곡 및 잡음 요소가 첨가되는 경향이 있다. 영상 복원에서는, 변형 요소를 포함한 영상의 잡음, 또는 왜곡 정보를 교정하여 복원 영상의 품질을 향상시키고, 원 영상에 가장 근접한 값으로 표현하여야 한다. 영상 복원을 위한 공간 필터 중에서 선형 필터는 쉽게 구현될 수 있고, 가우시안 잡음 제거율이 높다는 장점이 있지만, 얼룩이나 임펄스 잡음 제거에 대해서는 좋지 않은 성능을 보이기 때문에, 이러한 단점을 보완할 수 있는 비선형 필터 알고리즘으로 본 논문에서는 적응성 다단계 최적화 필터(OAMF : optimal adaptive multistage filter)라는 영상 복원 공간 필터를 제안하였다. 적응성 다단계 최적화 필터는 영상 복원에서 필터링 시간 감소, 잡음 제거율 증가 그리고 외곽선 정보의 보존률 증가 등을 목적으로 역전파 학습 알고리즘의 가중치 학습법을 기반으로 적응성 다단계 필터(AMF)를 최적화 한 것이다. 본 논문에서 제시한 영상 복원 공간필터가 기존의 다른 필터들에 비해 임펄스 잡음 제거와 외곽선 정보 보존 기능, 가우시안 잡음 제거 능력 등이 향상됨을 시뮬레이션 결과로 입증하였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2004년도 심포지엄 논문집 정보 및 제어부문
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• pp.41-43
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• 2004

This paper is concerned with attitude estimation using low cost, small-sized accelerometers and gyroscopes. A two step extended Kalman filter is proposed, which adaptively compensates external acceleration. External acceleration is the main source of estimation error. In the proposed filter, direction of external acceleration is estimated. According to the estimated direction, the accelerometer measurement covariance matrix of the two step extended Kalman filter is adjusted. The proposed algorithm is verified through experiments.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1991년도 학술발표회 논문집
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• pp.38-41
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• 1991

RLS algorithms are required for applications to adaptive line enhancers, adaptive equalizers for voiceband telephone and HF modems, and wide-badn digital spectrum mobile raio in which their convergence time and tracking speed are significant. The fast QR RLS algorithm satisfies above the requirements. Its computational complexity is linearly proportional to the tap number of a filter, N and its performance remains numerically stable. From the result of simumulation, the fast QR RLS algorithm represented Cioffi is better than gradient based algorithm in its initial performance when being applied to an adaptive line enhancer for cancelling noise.

• 융합신호처리학회논문지
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• 제11권3호
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• pp.238-243
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• 2010

A new implementation method for MPDSAP(Maximally Polyphase Decomposed Subband Affine Projection) adaptive filter is proposed. The affine projection(AP) adaptive filter achieves fast convergence speed, however, its implementation is so expensive because of the matrix inversion for a weight-updating of adaptive filter. The maximally polyphase decomposed subband filtering allows the AP adaptive filter to avoid the matrix inversion, moreover, by using a pipelining technique, the simple subband structured AP is suitable for VLSI implementations concerning throughput, power dissipation and area. Computer simulations are presented to verify the performance of the proposed algorithm.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2005년도 제36회 하계학술대회 논문집 D
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• pp.3066-3068
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• 2005

Noise can make surrounding environments inferior and deteriorates operation efficiency, and it can bring aural damage as well as give a person psychological stress. Therefore, necessity of study about noise control is increased for better labor conditions and agreeable habitat. In this paper, implementation of active mufflers using a stable IIR adaptive filters is presented. The IIR filter structure is more effective when acoustic feedback exists, but the adaptive IIR filters could be unstable when the filter algorithm is not yet converged. A stabilizing process for adaptive IIR filter is introduced in this paper. Experiments using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• Journal of Communications and Networks
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• 제7권3호
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).