• The Journal of the Acoustical Society of Korea
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• v.6 no.2
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• pp.15-18
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• 1987

Multiplication free adaptive digital filtering algorithms are discussed. The proposed. The proposed algorithm uses delta modulation digital filter and the relevant filter weights are updated using the SIGN algorithms to realize an adaptive digital filter without multiplication operations. It is shown that the resulting algorithm can be implemented using simple up/down counting operations. The convergence characteristics of the proposed adaptive digital filtering algorithm and .others are investigated for a system identification problem.

• The Journal of Natural Sciences
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• v.8 no.2
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• pp.69-75
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• 1996

This paper presents a VLSI architecture for the efficient implementation of the real time multi-dimensional digital filter. The computational primitive for the filter is obtained from the state space representation of the multi-dimensional general order filter. The computational primitive is used for the data path of the processor.

• Economic and Environmental Geology
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• v.20 no.2
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• pp.119-123
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• 1987

Hankel transform of order 0 can be represented by a combination of Abel transform and Fourier transform. This Abel-Fourier method can offer computational advantages over a conventional digital linear filter method through the uses of a rapid Abel transform with shift variant recursive filter and of a fast Fourier transform. The Abel-Fourier method, however, is generally less accurate than a well-designed digital linear filter method. In geoelectrical applications, the digital linear filter method seems to be more flexible than the Abel-Fourier method.

• Transactions of the Korean Society of Automotive Engineers
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• v.1 no.3
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• pp.55-62
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• 1993

Autoregressive moving average(ARMA) model which is a time domain parametric modeling method is implemented for modeling and reproducing characteristics of exhaust noise of an automobile in various RPM range. Experiments have been carried out using 9 set of exhaust noise signals measured at 1,000-3,000 RPM range. Characteristics of sampled signals were estimated using ARMA modeling and Akaike's FPE(final prediction error) criterion to define exact model structure and for model validation. The digital filter consisted of the esitmated ARMA(70,1) model parameters was programed to reproduce exhaust noise. The spectral analysis of reproduced noise is very close to original. The results show that our approaching technique for reproducing acoustic characteristics is valid and feasible to apply in the field of noise quality control.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.3
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• pp.55-59
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• 1984

This paper describes a method of Hangout recognition using 2 - D digital filtering. The 170 patterns classified by the positions of the initial sound (consonant), middle sound (vowel) and terminal sound (consonant) of the 1,659 characters were established and models formed by using 2 - D digital filtering for each patterns were obtained. Based on these models we proposed an algorithm that can recognize KOREAN combinational characters by separating patterns from them with superpostion principles. As a result of simulation, 100% of recognition rate is obtained in the case of the print letter.

• The Journal of the Acoustical Society of Korea
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• v.7 no.2
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• pp.39-50
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• 1988

It is proposed to use a pair of frequency-domain adaptive digital filters to estimate the magnitude squared coherence (MSC) functions of two signals. Such a method requires less computations than the LMS-MSC algorithm in which the least mean square (LMS) algorithm is applied in the time domain to compute the coefficients of a pair of adaptive digital filters. The frequency-domain adaptive digital filtering algorithms considered in this paper include the constrained frequency domain LMS (CFLMS) and the unconstrained frequency domain LMS (UFLMS) algorithms. The performance of the proposed methods are compared with those of the LMS-MSC algorithm.

• Proceedings of the Korea Institutes of Information Security and Cryptology Conference
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• 1992.11a
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• pp.55-70
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• 1992

기존의 음성 정보 보호 방식의 단점인 비화의 감소, 키 수의 제한, 상관 관계를 이용한 제3자의 해독 둥의 문제점을 해결할 수 있는 호핑 필터를 이용한 이차원 진폭 스크램블링은 현대 아날로그 음성 신호에 있어서 강력한 비화 방식이다. 본 논문에서는 KAISER WINDOW FIR 필터를 이용하여 호핑 필터를 구성하였으며, 이차원진폭 스크램블링 알고리즘의 최대 단점인 동기(synchronization)문제를 해결하기 위하여 variable delay를 이용한 알고리즘을 제안하였다. 또한, 시뮬레이션을 통하여 디지탈 신호에도 응용·고찰하였다.

• Proceedings of the Korean Institute of IIIuminating and Electrical Installation Engineers Conference
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• 1991.10a
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• pp.69-74
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• 1991

본 논문은 슬라이딩 모드 제어 기법을 이용한 직류 전동기의 디지탈 속도 제어기에 관한 연구이다. 또한 연속 시스템에 적용하던 슬라이딩 모드 제어기법을 이산 시스템에 사용할 수 있도록 해석하고, 실험하였다. 슬라이딩 모드 제어기에서 발생되는 채터링(CHATTERING) 현상을 감소시키기 위해 스위칭 주파수를 높이고, 샘플링 주기를 최대한 줄였으며 제어기 출력에 필터를 도입하였다. 또한, 스위칭 지연과 외란에 대한 보상도 고려하였으며 슬라이딩 모드 제어기를 마이크로 프로세서로 구성하였다. 전력 소비를 줄이기 위해 LIMITED UNIPOLAR 모드에서 직류 전동기를 구동하였다.

• The Journal of the Acoustical Society of Korea
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• v.7 no.4
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• pp.98-109
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• 1988

This paper introduces an adaptive nonlinear filtering algorithm that uses the sequential regression(SER) method to update the second order Volterra filter coefficients in a recursive way. Conventionally, the SER method has been used to invert large matrices which result from direct application of Wiener filter theory to the Volterra filter. However, the algorithm proposed in this paper uses the SER approach to update the least squares solution which is derived for Gaussian input signals. In such an algorithm, the size of the matrix to be inverted is smaller than that of conventional approaches, and hence the proposed method is computationally simpler than conventional nonlinear system identification techniques. Simulation results are presented to demonstrate the performance of the proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.12 no.4
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• pp.373-385
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• 1987

The scaled WDF should be optimized from the point of view of dynamic range. Transformer scaling method is attempted to maximize the dynamic range by equalizing the gains in the different nodes of the WDF. In this paper, the dual network and different frequencies are used to study the effects of different topology and of sampling frequency. Comparing with the unscaled WDF, the most significant observation is that the SNR is improved in the range of 7 to 35 dB in the WDF scaled by the trasnsformer scaling method. In addition, the SNR shows a continuous drop with the increase of sampling frequency, but scaling effects seem to decrease the rate at which the SNR drop.