• 한국인터넷방송통신학회논문지
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• 제12권1호
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• pp.195-200
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• 2012

본 논문에서는 동시통화에 강인한 특성을 갖는 알고리즘의 추적성능을 개선시킬 수 있는 새로운 방법을 제안하였다. 반향경로의 변화를 검출하는 방법과 적응필터의 계수 적응식을 변경하는 방법을 제안하였다. 오차신호 대 스케일 파라메터의 비는 반향경로가 변화한 경우와 동시통화가 발생한 경우에 지속시간이 다른 특성을 가지며 이를 이용하여 반향경로 변화를 검출한다. 제안한 적응필터계수 적응식은 잘못된 오차신호선택을 방지하도록 하여 추적 성능을 개선시킨다. 실제 음성신호와 ITU-T G.168에서 제공하는 반향경로를 사용한 실험을 통하여 제안한 방법이 기존의 방법에 비해 4dB 이상 성능을 개선시킴을 확인하였다.

• 한국멀티미디어학회:학술대회논문집
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• 한국멀티미디어학회 2000년도 춘계학술발표논문집
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• pp.449-452
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• 2000

This paper proposes a new, adaptive Dead Reckoning model, called Dynamic Dead Reckoning , for Distributed Interactive Simulation and humanoid avatar systems. The proposed model can overcome the weak points of traditional Dead Reckoning caused by a fixed threshold and strong dependency on rotation event. This paper introduces new criteria for update message filtering , named as Update lifetime. The Dynamic Dead Reckoning keeps the balance between extrapolation fidelity and filtering performance by two component models, Variable Threshold Mechanism and Rotation Event model. The experimental results show that the proposed model can lower the increment rate of update traffic to the increase of rotation frequency without any significant loss of accuracy.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1992년도 한국자동제어학술회의논문집(국내학술편); KOEX, Seoul; 19-21 Oct. 1992
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• pp.541-547
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• 1992

The phased array antenna has the ability to perform adaptive sampling by directing the radar beam without inertia in any direction. The adaptive sampling capability of the phased array antenna allows each sampling time interval to be varied for each target, depending on the acceleration of each target at any time. In this paper we design a three-dimensional adaptive tracking algorithm for the phased array radar system with a given set of measurement parameters. The tracking algorithm avoids taking unnecessarily frequent samples, while keeping the angular prediction error within a fraction of antenna beamwidth so that the probability of detection will not be degraded during a track update illuminations. In our algorithm, the target model and the sampling rate are selected depending on the target range and the target maneuver status which is determined by a maneuver detector. A detailed simulation is conducted to test the validity of our tracking algorithm for encounter geometries under various conditions of maneuver.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1997년도 한국자동제어학술회의논문집; 한국전력공사 서울연수원; 17-18 Oct. 1997
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• pp.321-324
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• 1997

In this paper, Active Noise Control(ANC) algorithm is proposed based on the estimated frequency estimator of the reference signal. The conventional feedforward ANC algorithms should measure the reference and use it to calculate the gradient of the squared error and filter coefficients. For ANC systems applied to aircrafts and passenger ships, engines from which reference signal is usually measured is so far from seats where main part of controller is placed that the scheme might be difficult to implement or very costly. Feedback ANC algorithm which doesn't need to measure the reference uses the error signal to update the filter and is sensitive to unexpected transient noise like a sneeze, clapping of hands and so on The proposed algorithm estimates frequencies of the desired signal in real time using adaptive notch filter. New frequency estimation algorithm is proposed with the improved convergence rate, threshold SNR and computational simplicity. Reference is not measured but created with the estimated frequencies. It has strong similarity to the conventional feedback control because reference is made from error signal. Enhanced error signal is used to update the controller for better performance under the measurement noise and impact noise. The proposed ANC algorithm is compared with the conventional feedback control.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2008년도 하계종합학술대회
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• pp.895-896
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• 2008

We propose a adaptive combination of affine projection algorithm (APA) filters with different update interval. Two APA filters with different update interval are adapted independently in order to keep the advantages of both component filters. This novel scheme provides improvement of performance in term of the convergence rate and the steady-state error. Experimental results show good properties of the proposed algorithm.

• 한국컴퓨터정보학회논문지
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• 제19권3호
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• pp.55-62
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• 2014

본 논문에서는 기존의 주기적 링크 상태 갱신 기법의 장점인 링크 상태 갱신(Link State Update : LSU) 메시지 수 제어 기능과 적응형 링크 상태 갱신 기법의 장점인 라우팅 성능을 가지는 하이브리드 링크 상태 갱신 기법(Hybrid LSU)을 제안한다. Hybrid LSU 기법은 네트워크 트래픽 정보를 기반으로 임계값이 변하는 적응형 링크 상태 갱신 기법을 가지고 있으며 플로우의 요구 대역폭 정보를 통해 LSU 메시지 전송 중요도를 판단하고 단위 시간 당 업데이트 비율에 따라 LSU 메시지 전송 여부를 판단하여 LSU 메시지 수를 제어할 수 있는 메커니즘을 포함한다. 성능 평가를 위해 기존에 제시된 다양한 LSU 기법과 본 논문에서 제안하는 Hybrid LSU 기법을 MCI 네트워크상에서 라우팅 블록율과 링크 당 평균 LSU 메시지의 개수 등을 성능 평가 항목으로 시뮬레이션 하였다. 그 결과 트래픽 부하가 높아질수록 라우팅 블록율 성능은 그대로 유지하면서 평균 LSU 메시지의 개수를 기존의 적응적 링크 상태 갱신 기법에 비해 10% 정도 줄일 수 있었고 이를 통해 제안하는 기법의 우수성을 확인하였다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1992년도 학술논문발표회 논문집 제11권 1호
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1993년도 한국자동제어학술회의논문집(국내학술편); Seoul National University, Seoul; 20-22 Oct. 1993
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• pp.145-149
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• 1993

This paper considers the adaptive predictive control problem of a system characterized by a multiplexed measurements and multirate sampling mechanism. Plant outputs are measured in various sampling rates through a multiplexed measurement system where a single common instrument is shared by several controllers. In general, output measurement sampling rate is assumed to be slower that input update rate. An adaptive predictive control algorithm is developed for systems with multiplexed measurements.

• 전기학회논문지
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• 제62권1호
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• pp.133-139
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• 2013

Convex combination of two adaptive filters is an efficient method to improve adaptive filter performances. In this paper, a subband convex combination method of two adaptive filters for fast convergence rate in the transient state and low steady state error is presented. The cost function of mixing parameter for a subband convex combination is defined, and from this, the coefficient update equation is derived. Steady state analysis is used to prove the stability of the subband convex combination. Some simulation examples in system identification scenario show the validity of the subband convex combination schemes.

• 대한전기학회논문지:시스템및제어부문D
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• 제53권6호
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• pp.446-452
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• 2004

Under the condition of a common narrowband target signal and interference signals from several directions, the linearly constrained minimum variance (LCMV) method using the generalized sidelobe canceller (GSC) for adaptive beamforming has been exploited successfully However, in the case of wideband signals, the length of the adaptive filter must be extended. As a result, the complexity of the beamformer increases, which makes real-time implementation difficult. In this paper, we improve the convergence characteristics of the adaptive filter using the transform domain normalized least mean square (NLMS) approach based on the subband GSC structure without the increase of complexity. Besides, the M-MAX algorithm, which is one of various selective coefficient updating methods, is employed in order to remarkably reduce the computational cost without decreasing the convergence quality. With the combination of these methods, we propose a computationally efficient wideband adaptive beamformer and verify its efficiency through a series of simulations.