• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1986.04a
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• pp.1-3
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• 1986

We are concerned with transmitting numerical source data of {0, 1, 2, ..., 2k-1} through a channel coding system. The rate 1/v dual-k convolutional code with the orthogonal MFSK modulation and the Viterbl decoding is employed for the implementation of the channel coding system. The mean square error of the dual-k convolutional code is evaluated for the numerical source transmitted over an additive white Gaussian noise channel with Rayleigh fading.

• Communications for Statistical Applications and Methods
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• v.3 no.2
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• pp.43-51
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• 1996

균형일원변량모형에서 분산성분비율의 점추정에 관한 문제가 고려되어진다. 분산성분비율에 대한 점추정량의 종류를 살펴보고 추정량의 평균자승오차(MSE)를 서로 비교하여 본다. 분산성분비율에 대한 새로운 추정량이 제 안되며, 제안된 추정량을 사용하면 모의실험을 통하여 Das (1992)가 고려한 여러가지 형태의 추정량들보다 급내상관계수 ${\rho}$의 값이 대략 0.2 < ${\rho}$ < 0.7인 경우에 MSE 효율성이 높아짐을 밝혔다.

• The Journal of the Acoustical Society of Korea
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• v.38 no.3
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• pp.283-289
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• 2019

RLS (Recursive-least-squares) algorithm is known to have good convergence and excellent error level after convergence. However, there is a disadvantage that numerical instability is included in the algorithm due to inverse matrix calculation. In this paper, we propose an algorithm with no matrix inversion to avoid the instability aforementioned. The proposed algorithm still keeps the same convergence performance. In the proposed algorithm, we adopt an averaged gradient-based step size as a self-adjusted step size. In addition, a variable forgetting factor is introduced to provide superior performance for time-varying channel estimation. Through simulations, we compare performance with conventional RLS and show its equivalency. It also shows the merit of the variable forgetting factor in time-varying channels.

• KSCE Journal of Civil and Environmental Engineering Research
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• v.32 no.1B
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• pp.9-20
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• 2012

Recently to overcome limitations of conceptual, hydrological and physics based models for flood stage forecasting, multiple linear regression model as one of data-driven models have been widely adopted for forecasting flood streamflow(stage). The objectives of this study are to compare performance of different multiple linear regression models according to regression coefficient estimation methods and determine most effective multiple linear regression flood stage forecasting models. To do this, the time scale was determined through the autocorrelation analysis of input data and different flood stage forecasting models developed using regression coefficient estimation methods such as LS(least square), WLS(weighted least square), SPW(stepwise) was applied to flood events in Jungrang stream. To evaluate performance of established models, fours statistical indices were used, namely; Root mean square error(RMSE), Nash Sutcliffe efficiency coefficient (NSEC), mean absolute error (MAE), adjusted coefficient of determination($R^{*2}$). The results show that the flood stage forecasting model using SPW(stepwise) parameter estimation can carry out the river flood stage prediction better in comparison with others, and the flood stage forecasting model using LS(least square) parameter estimation is also found to be slightly better than the flood stage forecasting model using WLS(weighted least square) parameter estimation.

• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.570-575
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• 2004

In the underwater acoustic systems. we can receive signals and retrieve information about a target by using a beamforming method. The most important thing in the beamforming is finding the way to optimize the mainlobe beamwidth and the sidelobe level to the desired value. One of the prominent results of beamforming method. which has been studied. is Philip's weighting function method(1) . Philip's method adaptively adjusts its weights of array to meet the desired mainlobe beamwidth and sidelobe level. It is very similar to the design method in adaptive filter. However. this method cannot easily bring us to the desired sidelobe level due to complementary relation between mainlobe beamwidth and sidelobe level. In this paper, we propose a new algorithm using partial constrained adaptation. This method makes us circumvent the above problem and meet the specification of design easily. The proposed algorithm presents a Pattern synthesis that designer can easily control the mainlobe beamwidth and the sidelobe level to the desired value while calculation time to converge is decreasing.

• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.271-275
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• 2018

There are several methods for estimating the time delay between incoming signals to two sensors. Among them, the GCC-PHAT (Generalized Cross Correlation-Phase Transform) method, which estimates the relative delay from the signal whitening and the cross-correlation between the different signal inputs to the two sensors, is a traditionally well known method for achieving stable performance. In this paper, we have identified a part of GCC-PHAT that can improve the periodicity. Also, we apply the auto-correlation method that is widely used as a method to improve the periodicity. Comparing the proposed method with the GCC-PHAT method, we show that the proposed method improves the mean square error performance by 5 dB ~ 15 dB at the SNR above 0 dB for white Gaussian signal source and also show that the method improves the mean square error performance up to 15 dB at the SNR above 2 dB for the color signal source.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.430-433
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• 2003

This paper presents a system identification (SI) scheme in time domain using measured acceleration data. The error function is defined as the time integral of the least square errors between the measured acceleration and the calculated acceleration by a mathematical model. Damping parameters as well as stiffness properties of a structure are considered as system parameters. The structural damping is modeled by the Rayleigh damping in SI. The regularization technique is applied to alleviate the ill-posed characteristics of inverse problems. The validity of the proposed method is demonstrated by an experimental study on a shear building model.

• Journal of the Korean Institute of Intelligent Systems
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• v.10 no.6
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• pp.575-583
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• 2000

본 논문에서 임의의 데이터가 입력되면 기준 영상 중에서 가장 유사도가 큰 영상을 찾아 국부 승리자로 선택하고, 그 국부 승리자 중에서 전체 승리자를 선택하여 최종 출력값을 얻는 계층적 FGNN(Fuzzy Genetic Neural Network)을 제안하고, 이에 하이브리드 퍼지 소속함수와 유전자 알고리즘을 적용하였다. 하이브리드 퍼지 소속함수는 입력 값을 0~1 사이의 값으로 함으로써 시스템의 속도를 빠르게 하고 유전자 알고리즘을 입력값을 일정한 오차 이내로 하여 최적의 영상을 얻도록 하였다. 위의 계층적 FGNN 알고리즘을 회로 설계 및 검증하였다. 또한 제안한 FGNN을 이용하여 영상에 포함된 잡음을 제거하고, 이와 유사한 구조를 가진 FDNN(Fuzzy Decision Neural Network) 성능보다 FGNN의 성능이 우수함을 여러 가지 영상을 통하여 확인하였다. 또한 모의 실험 결과 영상에 대한 평균자승오차(MSE : Mean Square Error)를 비교하였으며, 그 결과 하이브리드 퍼지 함수와 유전자 알고리즘을 적용한 FGNN이 메디안 필터, OC, CO, FDNN 등에 비해 우수함을 확인하였다. FGNN 알고리즘을 Top-Down 방식으로 VHDL(VHSIC Hardware description Language)을 이용하여 코딩(Coding)하고, Synopsys 툴을 이용하여 하드웨어를 설계하였다. 이 알고리즘의 하드웨어는 총 5개의 블록으로 가지고 있고 각각의 블록은 파이프라인 형태로 구성하고, 이는 Synopsys 툴을 이용하여 동작 및 성능을 검증하였다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.4
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• pp.98-105
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• 1989

Various models incorporating Human Visual System (HVS) with the Hadamard transform (HT) represented by Walsh functions are considered. Using the exact frequency components of HT basis functions, the optimum modulation transfer function (MTF) which has a higher peak frequency than DCT schemes is obtimum modulation transfer function (MTF) which has a higher peak frequency than DCT schemes is obtained analytically and visually. The main criterion, for error measurement, is errors at the block boundaries which is an important factor in transform coding. The scheme which has no inverse HVS is proposed. It causes some degradation of image data but it is insignigicant. Crossing area of 4 blocks is equalized by the HVS weighting coefficients. The HVS weighted coding results in perceptually higher quality images compared with the unweighted scheme.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.19-26
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• 2009

In this paper, a minimum Euclidian distance criterion between error PDF and Dirac delta function is introduced and a constant modulus type blind equalizer algorithm based on the criterion is proposed. The proposed algorithm using constant modulus error in place of actual error term of the criterion has superior convergence and steady state MSE performance, and the error signal of the proposed algorithm exhibits more concentrated density function in blind equalization environments. Simulation results indicate that the proposed method can be a reliable candidate for blind equalizer algorithms for multipoint communications.