• 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.

• 대한의용생체공학회:의공학회지
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• 제10권2호
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• pp.83-88
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• 1989

This paper is a study on the design of adptive filter for QRS complex detection. We propose a simple adaptive algorithm to increase capability of noise cancelation in QRS complex detection with two stage adaptive filter. At the first stage, background noise is removed and at the next stage, only spectrum of QRS complex components is passed. Two adaptive filters can afford to keep track of the changes of both noise and QRS complex. Each adaptive filter consists of prediction error filter and FIR filter The impulse response of FIR filter uses coefficients of prediction error filter. The detection rates for 105 and 108 of MIT/BIH data base were 99.3% and 97.4% respectively.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제17권12호
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• pp.3242-3265
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• 2023

Ensuring reliability of a machinery system involve the prediction of remaining useful life (RUL). In most RUL prediction approaches, noise is always considered for removal. Nevertheless, noise could be properly utilized to enhance the prediction capabilities. In this paper, we proposed a novel RUL prediction approach based on noise injection and a Kalman filter ensemble of modified bagging predictors. Firstly, we proposed a new method to insert Gaussian noises into both observation and feature spaces of an original training dataset, named GN-DAFC. Secondly, we developed a modified bagging method based on Kalman filter averaging, named KBAG. Then, we developed a new ensemble method which is a Kalman filter ensemble of KBAGs, named DKBAG. Finally, we proposed a novel RUL prediction approach GN-DAFC-DKBAG in which the optimal noise-injected training dataset was determined by a GN-DAFC-based searching strategy and then inputted to a DKBAG model. Our approach is validated on the NASA C-MAPSS dataset of aero-engines. Experimental results show that our approach achieves significantly better performance than a traditional Kalman filter ensemble of single learning models (KESLM) and the original DKBAG approaches. We also found that the optimal noise-injected data could improve the prediction performance of both KESLM and DKBAG. We further compare our approach with two advanced ensemble approaches, and the results indicate that the former also has better performance than the latters. Thus, our approach of combining optimal noise injection and DKBAG provides an effective solution for RUL estimation of machinery systems.

• 지구물리와물리탐사
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• 제3권3호
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• pp.83-87
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• 2000

송신원의 파워 증가가 제한되고 인공잡음이 존재하는 지역에서 양질의 탄성파 자료를 획득하기 위하여 근/원기준점(reference)을 이용한 탄성파 잡음예측필터를 개발하였다. 잡음예측필터에 사용된 방법은 backpropagation 알고리즘을 이용한 3층의 인공신경망(neural network)으로서, 훈련자료(training data) 및 검증자료(testing data)에 훈련된 잡음예측필터를 적용시 신호대잡음비(signal-to-noise ration)를 약 3배 정도 증가시켰다. 그러나, 일반적으로 전기, 전자탐사 자료의 질을 향상하기 위해 사용되는 스케일링(scaling)기법으로는 전혀 탄성파의 잡음을 제거할 수 없었다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -1
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• pp.260-263
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• 2000

A technique that uses linear prediction to achieve noise reduction in a voice signal which has been mixed with an ambient noise (Signal to Noise (S-N) ratio = about 0dB) is proposed. This noise reduction method which is based on the linear prediction estimates the voice spectrum while ignoring the spectrum of the noise. The performance of the noise reduction method is first examined using the transversal linear predictor filter. However, with this method there is deterioration in the tone quality of the predicted voice due to the low level of the S-N ratio. An additional processing circuit is then proposed so as to adjust the noise reduction circuit with an aim of improving the problem of tone deterioration. Next, we consider a practical application where the effects of round on errors arising from fixed-point computation has to be minimized. This minimization is achieved by using the lattice predictor filter which in comparison to the transversal type, is Down to be less sensitive to the round-off error associated with finite word length operations. Finally, we consider a practical application where noise reduction is necessary. In this noise reduction method, both the voice spectrum and the actual noise spectrum are estimated. Noise reduction is achieved by using the linear predictor filter which includes the control of the predictor filter coefficient’s update.

• 한국통신학회논문지
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• 제22권10호
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• pp.2339-2347
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• 1997

The performance of the adaptive noise canceller using LMS algorithm is degraded by the gradient noise due to target speech signals. An adaptive noise canceller with speech detector was proposed to reduce this performande degradation. The speech detector utilized the adaptive prediction-error filter adapted by the NLMS algorithm. This paper discusses to enhance the performance of the adaptive noise canceller forthecorlored noise. The affine projection algorithm, which is known as faster than NLMS algorithm for correlated signals, is used to adapt the adaptive filter and the adaptive prediction error filter. When the voice signals are detected by the speech detector, coefficients of adaptive filter are adapted by the sign-error afine projection algorithm which is modified to reduce the miaslignment of adaptive filter coefficients. Otherwirse, they are adapted by affine projection algorithm. To obtain better performance, the proper step size of sign-error affine projection algorithm is discussed. As resutls of computer simulation, it is shown that the performance of the proposed ANC is better than that of conventional one.

• Journal of information and communication convergence engineering
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• 제1권4호
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• pp.189-193
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• 2003

A new algorithm is presented for adaptive notch filtering of narrow band or sine signals for embedded among broad band noise. The notch filter is implemented as a constrained infinite impulse response filter with a minimal number of parameters, Based on the recursive prediction error (RPE) method, the algorithm has the advantages of the fast convergence, accurate results and initial estimate of filter coefficient and its covariance is revealed. A convergence criterion is also developed. By using the information of the noise-to-signal power, the algorithm can self-adjust its initial filter coefficient estimate and its covariance to ensure convergence.

• 대한의용생체공학회:의공학회지
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• 제16권1호
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• pp.49-56
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• 1995

본 논문은 새로운 QRS comlex의 검출 알고리듬의 설계에 관한 연구로서 전처리 과정에 있어서 2단계의 적응 여파기를 설계하여 효과적인 잡음 제거 능력을 갖는 알고리듬을 제안 하였다. 1단계에서는 배경 잡음을 제거하고 2단계에서는 QRS complex만을 통과시킴으로, 잡음의 변동 뿐만 아니라 QRS comlex의 변화에도 적응 할 수 있게 하였다. 2단계의 적응 여파기는 각각 예측 오차 여과기와 유한 임펄스 응답을 갖는 여과기로 구성된다. 이때, 유한 임펄스 여과기의 계수는 예측 아 여과기의 계수를 이용하였다. MIT/BIT 데이타 베이스 중에서 비교적 잡음이 심한 105번과 108번을 가지고 실험을 한 결과 각각 99.3%와 97.4%의 검출율을 나타내었다.

• 한국음향학회지
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• 제31권1호
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• pp.29-38
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• 2012

본 논문에서는 음성 신호에 더해진 돌발 잡음을 제거하는 시스템을 제안한다. 제안한 돌발 잡음 제거 시스템은 중앙값 필터를 이용하여 돌발 잡음을 제거한다. 중앙값 필터는 잡음을 제거하는 과정에서 음성을 왜곡시킬 수 있기 때문에, 음성의 왜곡을 최소화하기 위하여 장구간 예측 필터를 전처리단으로 사용한다. 장구간 예측 필터로 보존된 음성 정보는 잡음이 제거된 후 다시 합성된다. 본 논문에서는 돌발 잡음이 존재하는 환경에서 음성의 정보를 보존하는데 있어 단구간 예측 필터의 문제점을 밝히고 장구간 예측 필터의 우수함을 보인다. 제안한 돌발 잡음 제거 시스템의 출력 신호는 입력 신호에 비해 음성이 존재하는 구간에서 신호 대 잡음비가 약 8dB 향상 되었으며, PESQ 점수가 약 1점 증가하였다.

• 한국정밀공학회지
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• 제15권7호
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• pp.175-181
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• 1998

This paper describes realtime visual tracking system of moving object for three dimensional target using EKF(Extended Kalman Filter). We present a new realtime visual tracking using EKF algorithm and image prediction algorithm. We demonstrate the performance of these tracking algorithm through real experiment. The experimental results show the effectiveness of the EKF algorithm and image prediction algorithm for realtime tracking and estimated state value of filter, predicting the position of moving object to minimize an image processing area, and by reducing the effect by quantization noise of image.