• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.44 no.5
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• pp.407-415
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• 2016

In this paper, an aviation image fusion method is proposed for creating an informative fused image through gray scale images within noise. The proposed method is based on an adaptive guided filter which adjusts regulation parameter of the filter based on peak signal noise ratio (PSNR) in order to behave as an edge-preserving filtering property. Simulation results demonstrate that the proposed method preserves the edge information of the input image and reduces the noise effect while maintaining designed PSNR.

• The Journal of the Acoustical Society of Korea
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• v.17 no.3
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• pp.52-58
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• 1998

본 논문에서는 시간 지연이 없는 서브밴드 필터 구조를 사용한 능동 소음 제어 시 스템을 제안하였다. 제안된 시스템은 기준 입력 신호와 2차 경로의 전달 함수를 컨볼루션하 여 만들어지는 filtered reference 신호가 서브밴드내에서 생성될 수 있도록, 2차 소음원과 오차 센서 사이의 전기·음향학적인 경로를 나타내는 2차 전달 함수를 각 서브밴드로 재구 성함으로써, 알고리듬 구현시 계산량을 감소시킨다. 또한 2차 경로의 전달함수가 시간에 따 라 변화하는 경우에도 능동 소음 제어 시스템의 소음 제어 성능을 유지할 수 있도록, 각 밴 드마다 두 개의 적응필터를 사용한 on-line 시스템 인지 구조를 제안하여 on-line 시스템 인 지에 필요한 계산량을 감소시켰다. 본 논문에서 제시한 능동 소음 제어 시스템의 제어 성능 과 on-line 시스템 인지 성능을 모의 실험을 통하여 검증하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.474-476
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• 2009

In the radio communication, such as echo cancellation and channel equalization, adaptive filtering is very practical. Its convergence behavior that is used for updating the weights depends on the correlation of the input signal and length of adaptive filter. Highly correlated input and long length of adaptive filter deteriorate the convergence behavior. To solve this problem, recently, subband affine projection algorithm which pre-whiten the correlation of the input and update the weights in subband structure has been presented. This paper presents convergence analysis method of multiple constrained subband affine projection algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.3
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• pp.723-728
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• 2010

This study proposed a design of a portable device for measuring heart rate using photoplethysmograph signal to minimize load of a nurse increased from insufficiency of an internal hospital nurse, and algorithm to measure reliable heart rate in PPG signals despite the existence of patient's motion artifacts. The proposed method for measuring heart rate is the method to minimize the motion interference by using the adaptive filter based on harmonic characteristic of PPG signal. To evaluate the performances of the a portable device implemented by the proposed method, we used several motion artifacts including finger and wrist movements; we then compared out results with the performance of the moving average filter. In this results, the proposed method showed a better performance than that of the moving average filter. Therefore, when nurses use the a portable device for measuring heart rate proposed in this study, it will enable to improve nurse work and to measure the reliable heart rate.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.270-277
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• 2010

In this paper, we proposed an algorithm that improves tone quality of noisy audio signals by using ICA(Independent Component Analysis) algorithm and perceptual filters. Many algorithms have been proposed to eliminate the noise from the audio signals, such as spectral subtraction method, perceptual filter, etc. The perceptual filter uses a noise that is acquired from silent ranges in the input signal. In this case, the improvement rate of tone quality decreases if the noise energy is changed by the environmental variation in a signal frame. But the proposed method estimates a noise that is changed at each frame using ICA algorithm. The estimated noise is applied to perceptual filter. To show the performance of the proposed algorithm, several tests are performed to various input signals. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR), noise-to-mask ratio (NMR) and Degradation Category Rating (DCR) test.

• The Journal of the Acoustical Society of Korea
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• v.8 no.4
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• pp.29-38
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• 1989

Detection of an underwater target by acoustic wave raises various difficulties due to unpredictable noise interference which originates from clutter, reverberation, and variations of medium characteristics with time and location. The SNR and the range resolution of conventional SONAR systems using a matched filter are generally poor, since the latter is optimum only in the additive white noise case. Furthermore, it cannot compensate for variations of the detection level which are responsible for the resultant detection errors. In this paper, the unpredictable interferences are compensated for by using an adaptive filter. It recursively estimates the channel impulse response based on the received echo signal. In the low noise environments, the estimated impulse response is close to the true one, providing a good range resolution, and a matched filter is used subsequently for the purpose of detection. It is shown through computer simulation that good performance can be achieved via the two steps of filtering. Also, the detection level remains unchanged without any additional provisions. Finally, we present the characteristics of the employed adaptive filter parameters.

• The Journal of Korean Institute of Information Technology
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• v.15 no.12
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• pp.157-164
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• 2017

Mobile ECG signal measurement is a technique to measure small signals of several mV, and many studies have been conducted to remove noise including wandering scheme. Removal of the equipotential line noise caused by shaking or movement of the electrode cable is one of the core research contents for the electrocardiogram measurement. In this study, we proposed a modified step-size of combined NLMS(normalized least squares) and DLMS(delayed least squares) adaptive filter to eliminate baseline noise from ECG signals. The proposed method mainly adjusts initial filter step-size to reduce distortion of original ECG signals characteristic after eliminating baseline noise. The modified filter step-size is scaled by filter order size and distortion minimization factor. This method is suitable for portable ECG device with a small processor and less power consumption. This technique also decreases computation time which is essential for real-time filtering. The proposed filter also increase the signal to noise ratio (SNR) compared to conventional NLMS filter.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1994-2000
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• 1993

A new lattice structure using decision feedback and augmented prediction for estimating and suppressing the narrowband interference is presented. The performance of the proposed interference canceller is compared to the conventional interference cancellation filter. The reference signal of the interference canceller is formed by using the chip decisions, which is correlated with the narrowband interference components of the received signal. The decision feedback technique reduce the distortion of the desired signal which is introduced by the interference canceller through the use of feedback chip decisions. And by linear prediction of the error signal, the residual interference component of can be eliminated, Using this unconteminated error signal to update the adaptive filter coefficients, the performance of the rejection can be improved. In the simulation, it is assumed that the processing gains are 7 and 15, signal to interference ratio is -10[dB], and 5% interference band. The results show that the BER performance of the proposed filter structure is improved by 1~3dB.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.249-252
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• 2002

본 논문에서는 음향반향제거기에서 상관계수를 이용하여 동시통화 구간을 검출하는 방법에서 검출 오류로 인해 발생되는 문제점에 대해서 다룬다. 상관계수(correlation coefficient)를 이용한 DT 검출 방법에서 동시통화 구간과 반향경로의 변화를 명확하게 구분 짓는 문턱값 설정이 어렵기 때문에 때때로 검출 오류가 발생한다. 즉, 동시통화 중간에 반향경로가 변함으로써 동시통화 구간의 끝점 검출에 실패하거나 반항경로 변화를 DT로 잘못 인식하는 경우가 발생하는데, 이럴 경우 더 이상 적응필터의 계수를 갱신을 할 수 없는 상태에 빠지기도 한다. 본 논문에서는 반향제거기에 보조필터를 사용하여 이러한 문제점을 해결하고자 한다. 이는 보조필터가 기준입력신호(reference signal)를 이용하여 변화된 반향신호 성분은 추정할 수 있지만 근단화자 신호는 추정할 수 없다는 점을 이용한다 실험을 통해 제안한 알고리즘이 검출 오류로 인해 발생되는 문제를 효율적으로 해결할 수 있음을 확인하였다.