• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 제14회 신호처리 합동 학술대회 논문집
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• pp.81-84
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• 2001

We propose a novel noise littering method based on adaptive windowing. To restore a noisy signal adaptive filtering methods have been widely researched and used. However, conventional adaptive filtering methods have a trade-off between noise suppression and edge preservation since they adopt fixed size filters. In this paper applying the adaptive windowing concept to adaptive filtering, we overcome the trade-off, The filter size is adaptively selected depending on signal statistics. The visual results of the signal and image restorations convincingly show the superior preservation of edge and detail and suppression of noise for the proposed adaptive windowed adaptive filter compared with conventional methods.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• 한국인터넷방송통신학회논문지
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• 제17권1호
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• pp.83-88
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• 2017

본 논문은 디지털 부호를 전송할 때 통신 채널에서 발생되는 부호간 간섭을 줄일 수 있는 MMA (Multiple Modulus Algorithm) 적응 등화 알고리즘에 adaptive modulus와 adaptive step size 개념을 적용한 H-MMA (Hybrid-MMA) 알고리즘의 등화 성능평가에 관한 것이다. 기존의 MMA 적응 등화기의 탭 계수 갱신은 등화기 출력를 이용하여 오차 신호를 얻게 되므로 2차원 QAM 신호에서 진폭과 위상을 보상할 수 있지만, 이때 스텝 크기와 modulus는 고정되므로 등화 성능이 저하된다. 이와 같은 문제점을 개선하기 위하여 등화기 출력 신호의 크기에 비례토록 adaptive modulus와 적응 스텝 크기를 변화시킬 수 있는 Hybrid adaptive 개념을 MMA에 적용하므로서 2차원 QAM 신호에서 등화 성능을 개선할 수 있다. 논문에서 제안하는 H-MMA 알고리즘의 성능을 기존의 MMA 알고리즘과 비교하기 위하여 동일한 채널에서 컴퓨터 시뮬레이션을 수행하였다. 시뮬레이션 결과 수렴 속도에서는 제안 H-MMA가 MMA보다 다소 느려지지만, 정상 상태 이후 모든 성능 지수에서 훨씬 우월하며 등화 잡음이 감소됨을 확인하였다.

• 한국인터넷방송통신학회논문지
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• 제19권5호
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• pp.107-113
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• 2019

본 논문에서는 CR-MMA (Constellation Reduction-Multi Modulus Algorithm) 등화 알고리즘에 adaptive modulus와 adaptive stepsize를 적용하므로서 등화 성능을 개선할 수 있는 Hybrid-CRMMA 알고리즘을 제안한다. CR-MMA 알고리즘은 16-QAM과 같은 nonconstant modulus 신호를 constant modulus 신호로 reduction한 후, 송신 신호의 통계치인 고정 modulus를 이용하여 오차 신호를 발생하므로 기존 MMA의 적응 등화 성능을 개선시키기 위하여 등장하였다. 제안 방식인 Hybrid-CRMMA는 CR-MMA의 constellation reduction 개념에 등화기 출력 신호의 전력에 비례토록 adaptive modulus와 오차 신호의 비선형성을 이용하여 adaptive stepsize를 적용하였으며, 이의 개선된 등화 성능을 컴퓨터 시뮬레이션을 통해 확인한다. 이를 위한 성능 지수로는 등화기 출력 신호 성상도, 수렴 특성을 나타내는 잔류 isi, 최대 찌그러짐, MSE 및 알고리즘의 외부 잡음에 대한 강인성을 나타내는 SER을 사용하였다. 컴퓨터 시뮬레이션 결과 제안 Hybrid-CRMMA가 기존 CR-MMA보다 모든 성능 지수에서 우월함을 확인하였다.

• 음성과학
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• 제9권4호
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• pp.275-287
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• 2002

In man-machine communication by speech under the noisy environment, the quality of speech may be degraded severely for the machine to recognize correctly. Especially when the corrupting noise occupies the same band as the speech, the conventional fixed filters cannot filter out the noise effectively. In recent, to resolve such a problem adaptive noise canceller (ANC) is frequently used, which is based upon adaptive filters. The Adaptive recursive filters perform better than adaptive nonrecursive filters due to the added poles, but the stability may be severely threatened. In this paper an ANC system employing the adaptive recursive filter is proposed to enhance the speech corrupted by noise. And the stability of the adaptive recursive filter is guaranteed by employing the adaptive compensator.

• 한국지능시스템학회:학술대회논문집
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• 한국퍼지및지능시스템학회 2004년도 추계학술대회 학술발표 논문집 제14권 제2호
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• pp.211-216
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• 2004

In this thesis, we present the Takagi-Sugeno-Kang (TSK) fuzzy model based adaptive controller and adaptive identification for a general class of uncertain nonlinear dynamic systems. We use an estimated model for the unknown plant model and use this model for designing the controller. The hybrid adaptive control combined direct and indirect adaptive control based on TSK fuzzy model is constructed. The direct adaptive law can be showed by ignoring the identification errors and fails to achieve parameter convergence. Thus, we propose an TSK fuzzy model based hybrid adaptive (HA) law combined of the tracking error and the model ins error to adjust the parameters. Using a Lyapunov synthesis approach, the proposed hybrid adaptive control is proved. The hybrid adaptive law (HA) is better than the direct adaptive (DA) method without identifying the model ins error in terms of faster and improved tracking and parameter convergence. In order to show the applicability of the proposed method, it is applied to the inverted pendulum system and the performance is verified by some simulation results.

• 한국인터넷방송통신학회논문지
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• 제21권4호
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• pp.75-80
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• 2021

본 논문은 DSE-MMA (Dithered Sign Error-MMA) 적응 등화기에서 adaptive modulus와 adaptive stepsize를 이용하여 등화 성능을 개선할 수 있는 Hybrid-DSE-MMA에 관한 것이다. DSE-MMA는 탭 계수 갱신을 위한 오차 신호를 얻을 때 dither 신호를 부가한 후 이의 부호만을 이용하므로 SE-MMA의 외부 잡음에 대한 robustness 성능을 개선시키지만, 고정 modulus와 stepsize를 이용하므로 수렴 속도와 잔류 isi 성능이 저하되는 단점이 있다. 논문에서는 등화기 출력 신호의 전력에 비례하는 adaptive modulus와 adaptive stepsize를 DSE-MMA에 적용하므로서 저하되는 등화 성능을 개선할 수 있음을 컴퓨터 시뮬레이션을 통해 확인한다. 개선된 등화 성능을 기존 DSE-MMA와 비교하기 위하여 수신측에서의 등화기 출력 신호인 복원된 신호 성상도, 잔류 isi, MD (Maximum Distortion), MSE 및 외부 잡음에 대한 알고리즘의 강인성을 알 수 있는 SER 성능을 사용하였다. 컴퓨터 시뮬레이션 결과 Hybrid-DSE-MMA 알고리즘은 모든 잔류 isi와 MD, MSE 및 SER등의 모든 성능 지수에서 DSE-MMA 보다 개선 되지만 수렴 속도가 늦어짐을 알 수 있었다.

• 대한전자공학회논문지
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• 제23권2호
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• pp.170-177
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• 1986

The adaptive array antenna with the LMS algorithm has the advantage in that it can perform train't because of its slower convergencerate. In this paper, the \ulcornerLMS algorithm is applied to the adaptive array so that the convergence rate can be improved, and the performance of he adaptive array by the \ulcornerLMS algorithm is compared to, that of the LMS adaptive array. It is shown that the adaptive array by the \ulcornerLMS algorithm is superior to the LMS adaptive array in the narrow frequency band.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1993년도 한국자동제어학술회의논문집(국제학술편); Seoul National University, Seoul; 20-22 Oct. 1993
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• pp.378-384
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• 1993

In this paper, an intelligent adaptive controller is proposed for the process with unmodelled dynamics. The intelligent adaptive controller consists of the numeric adaptive controller and the intelligent tuning part. The continuous scheme is used for the numeric adaptive controller to avoid the problems occurred in the discrete time schemes. The adaptive controller is adopted to the process with time delay. It is an implicit adaptive algorithm based on GMV using the emulator. The tuning part changes the design parameters in the control algorithm. It is a multilayer neural network trained by robustness analysis data. The proposed method can improve the robustness of the adaptive control system because the design parameters are tuned according to the operating points of the process. Through the simulation, robustnesses are shown for intelligent adaptive controller. Finally, the proposed algorithms are implemented on the electric furnace temperature control system. The effectiveness of the proposed algorithm is shown from experiments.

• 한국통신학회논문지
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• 제21권5호
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.