• Journal of information and communication convergence engineering
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• v.1 no.4
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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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.10 no.6
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• pp.308-316
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• 1985

A new implementation of Block Adaptive filter in frequency domain is presented in this paper. Block digital filtering involves the calculation of a block or finite set of filter out put from a block of input values. A fast convergence algorithm of block adaptive filter is developed using Gordar theory and compared with the performance results of Satio algorithm and BLMS algorithm. Form the result we can be shown that the convergence state of given algorithm is not only faster than BLMS algorithm but also the resulting convergence error is less than the convergence error of Satio algorithm.

• Journal of the Korea Society of Computer and Information
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• v.1 no.1
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• pp.53-64
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• 1996

The methods for designing the adaptive filter performing DSP(digital signal processing)functions In mobile communication systems are Least-squares algorithm. Fast-Kalman and adaptive lattice algorithm. Least-squares algorithm It fast convergence algorithm for signal Processor of adaptive equallizer and used for eliminating inter symbol Interference which occur inmultiple path fading channel In mobille communication systems. In this paper. we propose the method of control adjustably algebra characteristics of signal vector that is sampling at some of new data sequence and confirm the improvement of fast convergence and iterative performance speed compared to existing algorithms by computer simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.1C
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• pp.1-8
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• 2007

LMS(Least Mean Square) algorithm using steepest descent way in adaptive signal processing requires simple equation and is used widely because of the less complexity. But eigenvalues change by width of input signals in time domain, so the rate of convergence becomes low. In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and the same performance as the existing fast wavelet transform algorithm with less computational complexity. The proposed filter structure is applied to wavelet based adaptive algorithm. Simulation results show a better performance than the existing one.

• Journal of Korea Multimedia Society
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• v.12 no.4
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• pp.483-491
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• 2009

In this paper, we propose a fast motion estimation algorithm using adaptive elimination of sub-block partial coefficients. The proposed algorithm predicts an adaptive threshold for each sub-block by using relationship of an initial sum of absolute difference(SAD) and a minimum SAD at the current point, and efficiently reduces unnecessary calculation time of the conventional partial distortion elimination(PDE) algorithm with the predicted threshold. Our algorithm reduces about 60% of computations of the conventional PDE algorithm without any degradation of prediction quality compared with the con ventional full search. Additionally, the proposed algorithm can be applied to other fast motion estimation 떠gorithms. the proposed Our proposing algorithm will be useful to real-time video coding applications using MPEG-2 or MPEG-4 AVC standards.

• International Journal of Control, Automation, and Systems
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• v.6 no.3
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• pp.320-326
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• 2008

This paper studies the problem of fault estimation using adaptive fault diagnosis observer. A fast adaptive fault estimation (FAFE) approximator is proposed to improve the rapidity of fault estimation. Then based on linear matrix inequality (LMI) technique, a feasible algorithm is explored to solve the designed parameters. Furthermore, an extension to sensor fault case is investigated. Finally, simulation results are presented to illustrate the efficiency of the proposed FAFE methodology.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.1
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• pp.93-99
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• 2014

This paper deals with the FC-MMA (Fast Convergence-Multi Modulus Algorithm) which is improving the convergence characteristics of the MMA (Multiple Modulus Algorithm) adaptive equalization algorithm that is used for the minimization of the intersymbol interference which occurs in the time dispersive communication channel. In the time varying charateristics and the abnormal situation like as outage of the communication channel, the adaptive equalizer needs to adapting the new environment more rapidly. For this problem, the residual isi and the maximum distortion performance index which are meaning the convergence characteristics are widely adapted in the adaptive equalizer. The 16-QAM signal is transmitted and it was confirmed that the proposed algorithm, the FC-MMA has the fast convergence performance such as in the 1.75 times fast in residual isi and 2.5 times fast in the maximum distortion in order to reaching the steady state compare to the MMA algorithm in the same channel environment by the computer simulation.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.14 no.8
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• pp.823-829
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• 2003

For high frequency microstrip line modelling, a fast inductance extraction technique using an adaptive PEEC(partial element equivalent circuit) grid is proposed. The grid refinement technique is based on the current distribution depend on the excitation frequencies and the geometry of the microstrip lines. The adaptive ids are refined mainly in the area where heavy currents reside. This technique is applied to the inductance extraction of the microstrip lines. The results show fast convergence, and this adaptive technique is efficient to reduce computing time and the number of grids.

• Journal of Digital Contents Society
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• v.5 no.4
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• pp.295-299
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• 2004

In this paper, we compare and analyze the LMS ard RLS algorithm of IEEE802.15.3(HDR-WPAN) system. The LMS algorithm have two merits that easily embody and not complex, but convergence speed is slow. The RLS algorithm have fast convergence speed, but very complex. When equalization using LMS algorithm, it can achieve adaptive equalization after 250 sample in fading environment, but case of RLS algorithm can achieve adaptive equalization after just 50 sampls. The computer simulation proved that adaptive equalizer to fast equalization and stability of HDR-WPAN system is more effective using RLS algorithm then LMS algorithm.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.9
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• pp.77-83
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• 1997

We present a fast adaptive block matching algorithm using variable search area and subsampling to estimate motion vector more exactly. In the presented method, the block is classified into one of three motion categories: zero motion vector block, medium-motion bolck or high-motion block according to mean absolute difference of the block. By the simulation, the computation amount of the presented methoe comparable to three step search algorithm and new three step search algorithm. In the fast image sequence, the PSNR of our algorithm increased more than TSS and NTSS, because our algorithm estimated motion vector more accurately.