• 대한전기학회:학술대회논문집
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• 대한전기학회 1995년도 하계학술대회 논문집 B
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• pp.937-940
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• 1995

The purpose of this paper is to present a nonlinear system identification method, where an adaptive nonlinear recursive state-spare(ANRSS) filter is employed as its filter structure, and a variable step (VS) algorithm is applied as its adaptation law. To demonstrate the validity of the proposed method, some simulation results are included.

• 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.

• 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.

• ETRI Journal
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• 제34권4호
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• pp.564-571
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• 2012

This paper presents an algorithm for removing high-density impulsive noise that generates some serious distortions in edge regions of an image. Although many works have been presented to reduce edge distortions, these existing methods cannot sufficiently restore distorted edges in images with large amounts of impulsive noise. To solve this problem, this paper proposes a method using connected lines extracted from a binarized image, which segments an image into uniform and edge regions. For uniform regions, the existing simple adaptive median filter is applied to remove impulsive noise, and, for edge regions, a prediction filter and a line-weighted median filter using the connected lines are proposed. Simulation results show that the proposed method provides much better performance in restoring distorted edges than existing methods provide. When noise content is more than 20 percent, existing algorithms result in severe edge distortions, while the proposed algorithm can reconstruct edge regions similar to those of the original image.

• 한국인터넷방송통신학회논문지
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• 제16권3호
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• pp.81-87
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• 2016

본 논문에서는 기존 MMA 적응 등화 알고리즘의 안정성과 낮은 신호대 잡음비에서 robustness를 개선하기 위해 adaptive modulus와 miniumum-disturbance 기법을 적용한 AM-SCS-MMA (Adaptive Modulus-Soft Constraint Satisfaction-MMA) 알고리즘의 성능을 해석하였다. AM-SCS-MMA는 적응 등화를 비용 함수를 최소화하기 위해 adaptive modulus와 기존의 LMS 나 gradient descent algorithm 대신 deterministic optimization problem의 minimum-disturbance 기법을 적용하여 탭 계수를 갱신하므로서 채널에서 발생되는 진폭과 위상 찌그러짐에 의한 부호간 간섭을 동시에 줄이면서 등화 필터의 안정성 및 다양한 잡음에 대한 roburstness를 개선시킬 수 있다. 이의 개선 성능을 확인하기 위해 시뮬레이션을 수행하였으며 등화기 출력 성상도, 잔류 isi, MSE와 채널 추적 능력을 나타내는 EMSE (Excess MSE) 및 SER을 적용하였다. 컴퓨터 시뮬레이션의 결과 AM-SCS-MMA는 MMA보다 잔류 isi와 MSE에서는 수렴 속도는 늦지만 정상 상태 이후 잔여량이 감소되고 열악한 신호대 잡음비에서 robustness가 있었지만, 채널 추적 능력에서는 열화됨을 확인하였다.

• 전자공학회논문지S
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• 제34S권4호
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• pp.42-49
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• 1997

A conventional adaptive noise canceller (ANC) using LMS algorithm suffers from the misadjustment of adaptive filter weights due to the gradient-estimate noise by input speech signal at steady state. In this paper, an ANC is proposed which uses the combination of VSLMS (variable step size LMS) and SA (sign algorithm) to improve steady state performance and convergence rate. SA algorithm is applied in speech region to prevent the weights from perturbing by output speech of ANC and VSLMS algorithm is applied to improve convergence rate and channel tracking ability in silence region and adaptive transient region. In compute rsimulation, the performance of the proposed VSLMS-SA combination algorithm is much better than LMS algorithm and the algorithm, recently proposed by greenberg, with adaptation step-size parameter determine dby sum method in convergence rate, channel tracking and steady state performance.

• 한국멀티미디어학회:학술대회논문집
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• 한국멀티미디어학회 2003년도 춘계학술발표대회논문집
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• pp.423-426
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• 2003

Adaptive algorithm used in Acoustic Echo Canceller (AEC) needs fast convergence algorithm when reference signal is colored speech signal. Set-Membership Affine Projection (SMAP) algorithm is derived from the constraint, which is the minimum value adaptive filter coefficient error. In this paper, we test the characteristic about noise of the SMAP algorithm and proposed modified version of SMAP algorithm fur using at AEC. As the projection order increase, the convergence characteristic of the SMAP algorithm is improved where no noise space. But if the noise uncorrelated with input signal exists, the AEC shows bad performance. In this paper, we propose normalized version of adaptive constants using estimated error signal for robust to noise and show the good performance through AEC simulation.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1988년도 한국자동제어학술회의논문집(국내학술편); 한국전력공사연수원, 서울; 21-22 Oct. 1988
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• pp.172-177
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• 1988

In this paper, a new fast algorithm of FIR least squares filter with linear phase is presented. The general unknown statistics case is considered, whereby only sample records of the data are available. Taking advantage of the near-to-Toeplitz+Hankel structure of the resulting normal equation, a fast algorithm which gurantees the linear phase constraint, is developed that recursively produces the filter coefficient of linear phase FIR filter for a single block of data.

• 한국정보전자통신기술학회논문지
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• 제10권1호
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• pp.102-110
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• 2017

본 논문에서는 다채널 디지털 보청기에 적용 가능한 적응 음향 궤환 제거(Adaptive Feedback Cancellation : AFC) 알고리즘을 실시간으로 구현한다. 다채널 디지털 보청기는 일반적으로 난청 보상을 위해 FFT 필터뱅크 기반 광역 동범위압축(Wide Dynamic Range Compression) 알고리즘을 사용한다. 구현한 실시간 음향 궤환 제거 알고리즘은 다채널 디지털 보청기와 동일한 FFT 필터뱅크를 사용하여 WDRC와 함께 하나의 통합된 구조를 가짐으로써 보청기 배터리 수명에 영향을 미치는 연산량 측면에서 이득을 볼 수 있었다. 구현된 음향 궤환 제거 알고리즘은 고정 및 변화하는 음향 궤환 경로를 실시간으로 추정하여 보청기 출력 신호의 품질을 향상시킴을 확인하였다. 또한 비선형적인 입, 출력에 의해 음향 궤환 제거기가 정상적으로 작동하지 못해 출력 신호의 포화가 일어날 경우 감소 이득을 적용하여 시스템의 안정성을 높이고자 하였다. 결과적으로 다양한 실제 사용 환경에서 강건하게 동작하는 알고리즘을 구현할 수 있었다. 본 알고리즘은 추후 음질 개선 알고리즘 등 다양한 기능의 추가 구현이 용이하다.

• 대한전자공학회논문지SP
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• 제39권2호
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• pp.137-146
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• 2002

데이터 전송채널에서 부호간간섭(ISI)을 제거하기 위해 개선된 비적적응 디지털필터(IMADF)의 구조와 수렴해석이 이루어진다. 0평균, 백색잠음하에서 분할등화기(FSE)를 이용한 IMADF의 수렴특성을 해석한다. 실험결과 IMADF 알고리즘의 수렴독성이 Sign 알고리즘과는 같으나, MADF 알고리즘 보다 우수하다. 특히 입력신호의 상관관계가 높을 때 유용한 특성을 갖는다.