• Transactions of the Korean Society of Automotive Engineers
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• v.19 no.4
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• pp.84-90
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• 2011

In active noise control filter, LMS algorithms which used for control filter, assure the convergence property, and computational burden of these algorithms are proportionate to the filter taps. The convergence speed of LMS algorithms is mainly determined by value of the convergence coefficient, so optimal selection of the value of convergence coefficient is very important. In this paper, We proposed novel adaptive fuzzy logic LMS algorithms with FIR filter structure which has better convergence speed and less computational burden than conventional LMS algorithms, for single channel active noise control with ill conditioned signal case. Computer simulations were performed to show the effectiveness of a proposed algorithms.

• 제어로봇시스템학회:학술대회논문집
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• 1992.10a
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• pp.823-828
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• 1992

A novel steepest descent adaptive filter algorithm, which uses the instantaneous stochastic gradient for the steepest descent direction, is derived from a newly devised performance index function. The performance function for the new algorithm is improved from that for the LMS in consideration that the stochastic steepest descent method is utilized to minimize the performance index iterativly. Through mathematical analysis and computer simulations, it is verified that there are substantial improvements in convergence and misadjustments even though the computational simplicity and the robustness of the LMS algorithm are hardly sacrificed. On the other hand, the new algorithm can be interpreted as a variable step size adaptive filter, and in this respect a heuristic method is proposed in order to reduce the noise caused by the step size fluctuation.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.90-94
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• 2008

It is well-blown that the convergence rate gets worse when an input signal to an adaptive filter is correlated. In this paper we propose a new adaptive filtering algorithm that makes the convergence rate much improved even for highly correlated input signals. By introducing an orthogonal constraint between successive input signal vectors we overcome the slow convergence problem of the LMS algorithm with the correlated input signal. Simulation results show that the proposed algerian yields fast convergence speed and excellent tracking capability under both time-invariant and time-varying environments, while keeping both computation and implementation simple.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.889-890
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• 2006

This paper pertains to introducing the design of adaptive filters for the cancellation of muscle noise among several types of noise sources from the ECG signal. We used EMG signals measured along with ECG at the same time to use it as the reference input to the adaptive filter for the experiments. PSD results showed that the statistical characteristics of ECG are closely correlated with those of EMG.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.36 no.11
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• pp.819-825
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• 1987

The ALE problem, which tries to recover a sinusoidal signal corrupted by noise, has been solved using FIR filters. Recently several methods have been proposed using a norch filter of IIR type. In this study, the notch filter was represented with a parameter and auxiliary signals were generated by using an adaptive observer. A simple method is proposed to estimate the parameter. This method is tested under various circumstances by changing the input frequency, S/N ratio, and the type of the noise. The simulation shows that this method gives much better results than the other known methods with respect to the input S/N ratio and converging times. This method is simple and does not require much conputation, so it can be easily implemented in real time applications.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.37.2-37
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• 2001

This paper addresses the robust adaptive output feedback tracking for nonlinear systems under disturbances whose bounds are unknown. A new algorithm is proposed for estimation of unknown bounds and adaptive control of the uncertain nonlinear systems. The State estimation is solved using K-filters, together with the construction of a bound of an error in the state estimation due to the perturbation of the disturbance. Tuning functions are used to estimate unknown system parameters without overparametrization. The proposed control algorithm ensures that the out put tracking error converges to a residual set which can be arbitrarily small, while maintaining the boundedness of all other variables. A simulation shows the effectiveness of the proposed approach

• Journal of the Korean Institute of Telematics and Electronics
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• v.24 no.3
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• pp.387-393
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• 1987

The introduction of a digiral subscriber loop transmission system necessitates an optimized line interface solution. To meet this objective an adaptive line equalizer has been developed. The equalizer can be compensated up to 42 dB line loss at 200KHz, and operated up to 3.2 Km transmission length (0.4 mm\ulcornercable)at a rate of 400Kb/s. This has been builted using a variable \ulcorner equalizer to compensate a frequency-attenuation characteristics of metallic cable, an AGC (automatic gain control) circuits with simple control algorithm, and various filters to minimize a transmission constraints over subscriber loop. The purpose of this paper is to present a short description of a design of the adaptive line equalizer with a summary of implementation results. Some design concepts and considerations which results in an implementation of the equalizer are also given.

• The Korean Journal of Applied Statistics
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• v.21 no.2
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• pp.233-245
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• 2008

In this work, a new adaptive weighted mean filter is proposed for preserving image details while effectively suppressing impulse noise. The proposed filter is based on a noise pixel detection-estimation strategy. All the pixels are first detected using an impulse noise detector. Then the detected noise pixels are replaced with the output of the weighted mean filter over adaptive working window according to the rate of corrupted neighborhood pixels, while noise-free pixels are left unaltered. We compare the proposed filter to other existing filters in the qualitative measure and quantitative measures such as PSNR and MAE as well as computation time to verify the capability of the proposed filter. Extensive simulations show that the proposed filter performs better than other filters in impulse noise suppression and detail preservation without increasing of running time.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.57-63
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• 1997

This paper proposes a new stereophonic acoustic echo canceller to preven the performance degradation at the instant of far-end talker's movement or change in transmission room environment in stereophonic teleconferencing system. In stereophonic acoustic echo canceller, the filter coefficients of the adaptive filters for echo cancellation do not have unique solutions and not converge to their optimum values. Therefore, the change of the far-end transmission room environment leads to degradation of ERLE(Echo Return Loss Enhanceement). Moreover, their computational complexity increases as the number of the filter coefficient increases, and their adaptive filters converge very slowly. So the real-time implementation is very difficult. To overcome these problems we propose a pre-processor consisting of an adaptive filter for making pseudo stereophonic signal, and it results in improved performance at the instant of environment change at far-end side and reduction of the total complexity.

• Journal of Digital Contents Society
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• v.18 no.1
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• pp.141-147
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• 2017

Scalable extension of High Efficiency Video Coding (SHVC) standard uses the up-sampled residual data from the base layer to make a residual data in the enhancement layer. This paper describes an efficient algorithm for improving coding gain by using the filtered residual signal of base layer in the Scalable extension of High Efficiency Video Coding (SHVC). The proposed adaptive filter selection mechanism uses the smoothing and sharpening filters to enhance the quality of inter-layer prediction. Based on two filters and the existing up-sampling filter, a rate-distortion (RD)-cost fuction-based competitive scheme is proposed to get better quality of video. Experimental results showed that average BD-rate gains of 1.5%, 2.1%, and 1.7% for Y, U and V components, respectively, were achieved, compared with SHVC reference software 5.0, which is based on HEVC reference model (HM) 13.