• The Journal of the Acoustical Society of Korea
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• v.13 no.4
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• pp.77-85
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• 1994

In this study an adaptive algorithm with optimum convergence factor for steepest descent method is proposed, which controls automatically the filter order to take the appropriate level. So far, fixed order filters have been used when adaptive filter is employed according to the priori knowledge or experience in various adaptive signal processing applications. But, it is so difficult to know the filter order needed in real implementations that high order filters have to be performed. As a result, redundant calculations are increased in the case of high order filters. The proposed variable length optimum convergence factor (VLOCF) algorithm takes the appropriated filter order within the given one so that the redundant calculation is decreased to get the enhancement of convergence speed and smaller convergence error during the steady state. The proposed algorithm is evaluated to prove the validity by computer simulation for system Identification.

• Transactions of the Korean Society of Mechanical Engineers B
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• v.34 no.10
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• pp.947-955
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• 2010

Most oil and gas reservoirs, which have some light hydrocarbon components, show sensitive phase behavior in response to changes in the composition of the internal fluid. When evaluating and developing plans for oil and gas fields, flash calculation, PVT analysis, and saturation-property calculation are necessary for analyzing reservoir characteristics and pipeline flows. In general, the determination of saturation properties such as dew point and bubble point is considered a difficult task because of the poor convergence of the calculation methods. In this study, several new initial-value-guessing methods and root-finding methods are proposed; parametric analysis were carried out to verify the improvement in convergence. Finally, these new ideas and methods were successfully applied to the new GUI based multi-phase behavior simulator.

• Proceedings of the Computational Structural Engineering Institute Conference
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• 2002.10a
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• pp.567-574
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• 2002

새로운 자유격자 관사를 이용한 점별 계산법을 제안한다 이동 최소 자승법을 이용한 기저의 생성과 기저의 근사적 미분을 동시에 구해내는 자유격자 근사를 유도하여, 직접 점별 계산법을 고안하였다. 기존의 자유 격자 법에서는 기저의 직접 미분을 사용하므로 높은 계산 비용이 필요하지만, 이 논문에서 제안된 방법은 기저의 생성과 동시에 기저의 근사적 미분을 구하게 된다. 또한 기존의 방법에서 필요하였던, 창 함수(window function)의 미분가능성을 연속성으로 대치할 수 있으므로, 주어진 문제에 따라 다양한 창 함수를 이용할 수 있다. 기저의 재생성과 interpolation의 수렴성을 소개하고, 수치 예제로서, Poisson 문제를 통해 이 방법의 유효함을 보인다.

• Proceedings of the Korean Nuclear Society Conference
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• 1997.05b
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• pp.495-502
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• 1997

변위형 유한요소해에 기초하고 공액근사개념 및 Loubignac의 변위장 개선방법을 국부영역에 적용하여 국부영역에서의 응력장의 정확도를 향상시킬 수 있는 방법을 제안하였다. 제안된 방법에서 계산된 국부영역의 응력장은 전체영역에 대한 응력장 개선 결과에 근접하며 유한요소 평형방정식을 잘 만족하친 있을 뿐만 아니라 수회 이내의 반복계산내에 수렴하고 있어서 계산시간이 크게 줄어들 수 있어서 국부영역에 대한 상세응력해석에 적절하게 이용될 수 있다.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.10
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• pp.1012-1015
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• 2011

The decision-directed blind equalization algorithm is often used due to its simplicity and good convergence property when the eye pattern is open. However, in a channel where the eye pattern is closed, the decision-directed algorithm is not guaranteed to converge. Hence, a modified Bussgang-type algorithm using a hyperbolic tangent function for zero-memory nonlinear(ZNL) function has been proposed and applied to avoid this problem by Filho et al. But application of this algorithm includes the calculation of hyperbolic tangent function and its derivative or a look-up table which may need a large amount of memory due to channel variations. To reduce the computational and/or hardware complexity of Filho's algorithm, in this paper, an improved method for the decision-directed algorithm is proposed. In the proposed scheme, the ZNL function and its derivative are respectively set to be the original signum function and a narrow rectangular pulse which is an approximation of Dirac delta function. It is shown that the proposed scheme, when it is combined with decision-directed algorithm, reduces the computational complexity drastically while it retains the convergence and steady-state performance of the Filho's algorithm.

• Journal of Internet of Things and Convergence
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• v.6 no.4
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• pp.49-57
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• 2020

In this paper, I provided an approximation function Li*2,10(x) using logarithm integral for the counting function π*2,10(x) of consecutive deca primes. Several personal computers and Mathematica were used to validate the approximation function Li*2,10(x). I found the real value of π*2,10(x) and approximate value of Li*2,10(x) for various x ≤ 1011. By the result of theses calculations, most of the error rates are margins of error of 0.005%. Also, I proved that the sum C2,10(∞) of reciprocals of all primes with difference 10 between primes is finite. To find C2,10(∞), I computed the sum C2,10(x) of reciprocals of all consecutive deca primes for various x ≤ 1011 and I estimate that C2,10(∞) probably lies in the range C2,10(∞)=0.4176±2.1×10-3.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9B
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• pp.1785-1794
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• 1999

The affine projection algorithm has known t require less computational complexity than RLS but have much faster convergence than NLMS for speech-like input signals. But the affine projection algorithm is still much more computationally demanding than the LMS algorithm because it requires the matrix inversion. In this paper, we show that the affine projection algorithm can be realized with the Gram-Schmidt orthogonalizaion of input vectors. Using the derived relation, we propose an approximate but much more efficient implementation of the affine projection algorithm. Simulation results show that the proposed algorithm has the convergence speed that is comparable to the affine projection algorithm with only a slight extra calculation complexity beyond that of NLMS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.11
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• pp.1714-1725
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• 1993

This paper analyzes the properties of such algorithm that corresponds to the nonlinear adaptive algorithm with additional update terms, parameterized by the scalar factors a1, a2, a3 and Presents its structure. The analysis of convergence leads to eigenvalues of the transition matrix for the mean weight vector. Regions in which the algorithm becomes stable are demonstrated. The time constant is derived and the computational complexities of MLMS algorithms are compared with those of the conventional LMS, sign, LFG, and QFG algorithms. The properties of convergence in the mean square are analyzed and the expressions of the mean square recursion and the excess mean square error are derived. The necessary condition for the CFG algorithm to be stable is attained. In the computer simulation applied to the system identification the CFG algorithm has the more computation complexities but the faster convergence speed than LMS, LFG and QFG algorithms.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.5
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• pp.59-63
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• 2011

We present a lattice transversal joint (LTJ) adaptive filter with fixed reflection coefficients to achieve fast convergence with low complexity. The reflection coefficients of the filter are given by the statistics of speech signals, and the proposed order of the lattice predictor is one. Experimental results confirm that as compared to the adaptive transversal filter, the proposed adaptive filter achieves fast convergence with a negligible increase in complexity. The proposed adaptive filter converges around six times faster than the adaptive transversal filter in case of the band-limited voiced signal from the ITU-T G.168 standard.

• Journal of Korea Society of Industrial Information Systems
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• v.7 no.4
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• pp.17-21
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• 2002

Least mean square(LMS) algorithm one of the most popular algorithm in adaptive signal processing because of the simplicity and the small computation. But the convergence speed of time domain adaptive algorithm is slow when the spread width of eigen values is wide. Moreover we have to choose the step size well for convergency. in this paper, ie use adaptive algorithm of wavelet transform. And we propose a new wavelet based adaptive algorithm of wavelet transform. And we propose a high speed wavelet based adaptive algorithm with variable step size, which is linear to absolute value of error signal. We applied this algorithm to adaptive noise canceler. Simulation results are presented to compare the performance of the proposed algorithm with the usual algorithms.