• Proceedings of the KIEE Conference
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• 2011.07a
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• pp.2017-2018
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• 2011

본 논문에서는 QMF로 설계된 기본 필터뱅크를 이용하여 필터뱅크 시스템을 설계하고, 설계된 시스템을 이용하여 전력 외란 신호를 분해한다. 분해된 신호는 적응 예측기로 처리하여 전력 신호 사건의 트리거링점을 검출한다. 적응 필터의 수렴성능을 조절하여 순간적인 외란들을 효과적으로 검출 할 수 있다. 또한, 전력 신호에 포함된 백색잡음을 적응 필터를 이용 제거 할 수 있음을 보인다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.9
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• pp.764-774
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• 1990

In this paper, make a study on comparison of adaptive filters for predictor characteristics that transversal, lattice, and joint process lattice filter is using the LMS algorithm that is simple structure and pracotical application is easy. The theoical background and structure of each adaptive filters exhibit for practical design. Adaptive convergence condition for optimal weight vector and optimal reflection coefficient make clear, and it is also shown through computer simulation. The error signals and noise characteristics of these filters make a comparative study. In view of the results, joint process lattice filter is shown that most superior characteristic in these adaptie filters.

• Journal of the Korean Institute of Telematics and Electronics
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• v.19 no.5
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• pp.12-19
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• 1982

In general the design methods of adaptive filter are divided into two categories, one is based upon the local parameter optimization theory and the other is based upon stability theory. Among the various design techniques, the LMS algorithm by steepest-descent method which is based upon local parameter optimization theory is used widely. In designing the adaptive filter, the most important factor is the convergence rate of the algorithm. In this paper a new algorithm is proposed to improve the convergence rate of adaptive firter compared with the commonly used LMS algorithm. The faster convergence rate is obtained by adjusting the adaptation gain of LMS algorithm. And various aspects of improvement of the adaptive filter characteristics are discussed in detail.

• Proceedings of the KIEE Conference
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• 2002.07d
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• pp.2586-2588
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• 2002

본 논문에서는 $\Sigma\Delta$ 변조된 입력신호를 갖는 적응필터의 수렴특성을 연구하여 향상 방안을 제시하였다. 하드웨어적인 측면에서 효율적인 해상도를 내는 $\Sigma\Delta$ 변조기는 중저주파 대역의 신호를 처리하는데 널리 사용되고 있다. $\Sigma\Delta$ 변조신호는 항상 $\pm1$의 값만을 갖기 때문에, 사인 알고리즘을 사용하는 적응필터와 효율적으로 결합될 수 있다. 하지만, PCM 신호에 대비하여 $\Sigma\Delta$ 변조 신호의 상대적인 길이가 길어 이를 처리하는 적응필터의 길이가 증가하고, 아울러 사인 알고리즘 자체가 갖는 수렴속도의 문제점 때문에 이러한 결합은 불안정한 수렴 특성을 보이게 된다. 본 연구에서는 $\Sigma\Delta$ 변조된 입력신호에 대하여 웨이블릿 변환을 적용한 변환영역 적응필터를 설계하였으며, 수렴속도가 향상됨을 시스템 식별의 응용예를 통하여 검증하였다.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.238-247
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• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.2
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• pp.415-418
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• 2006

적응 횡단 필터는 상관 전 처리 기법으로서 실시간 간섭 신호 제거가 가능한 시간 영역 신호처리 기법을 사용한다. 적응 횡단 필터는 협대역 간섭 신호에 좋은 성능을 나타내며, 구현이 용이하고 높은 효율성을 갖는다. 적응 횡단 필터의 구성은 입력 샘플 신호의 지연 탭을 위한 FIR 필터와 전파 간섭 신호의 크기와 주파수를 결정하는 가중치 생성부로 나눌 수 있다. 본 논문에서는 가중치 생성부에 적용 되는 알고리즘 중 상대적으로 연산량이 적은 LMS와 NLMS를 적용한 적응 횡단 필터를 설계하고, GPS 수신기에 적용 하였다. 실제 측정치를 이용한 다양한 실험에 의한 항법 성능 평가를 통하여 NLMS가 LMS보다 좋은 성능을 나타냄을 확인하였다.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.2
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• pp.88-95
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• 1999

Stack filters are a class of digital nonlinear filters with excellent properties for signal restoration. Unfortunately, present algorithms for designing stack filters with large window size are limited in applications by their computational overhead and serial nature. In this paper, new, highly-parallel algorithm is developed for determining a stack filter which minimizes the mean absolute error criterion. It retains the iterative nature of the present adaptive algorithm, but significantly reduces the number of required to converge to an optima filter. A proof is also give that the proposed algorithm converges to an optimal stack filter.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.3
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• pp.231-238
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• 2004

This paper presents the high speed noise reduction processing system using the modified discrete fourier transform(MDFT) on the frequency domain. The proposed filter uses the linear prediction coefficients of the adaptive line enhance(ALE) method based on the Sign algorithm The signals with a random noise tracking performance are examined through computer simulations. It is confirmed that the fast adaptive digital filter is realized by the high speed adaptive noise reduction(HANR) algorithm with rapid convergence on the frequency domain(FD).

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.4
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• pp.295-303
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• 2008

In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.4
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• pp.986-991
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• 2014

The PPG signal used in mobile-healthcare and telemedicine system is including the various motion artifact that is signal generated from patient's movements. Recently, although the various methods to remove motion artifacts have been suggested, the performances of these methods are still not satisfactory. Therefore, this s study suggested the novel method based on the Kalman filter and adaptive filter to remove motion artifacts, and we used various motion artifacts to analyze the performance of the proposed method. In the results of experiments, the signal-to-noise ratio of proposed method showed good performace that was 4.8 times of moving average filter.