• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.81-84
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• 2001

We propose a novel noise littering method based on adaptive windowing. To restore a noisy signal adaptive filtering methods have been widely researched and used. However, conventional adaptive filtering methods have a trade-off between noise suppression and edge preservation since they adopt fixed size filters. In this paper applying the adaptive windowing concept to adaptive filtering, we overcome the trade-off, The filter size is adaptively selected depending on signal statistics. The visual results of the signal and image restorations convincingly show the superior preservation of edge and detail and suppression of noise for the proposed adaptive windowed adaptive filter compared with conventional methods.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.1
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• pp.83-88
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• 2017

This paper related with the performance evaluation of H-MMA (Hybrid-MMA) which is applying the adaptive modulus and adaptive step size concept to MMA adaptive equalization algorithm in order to reduce the intersymbol interference that is occurred in communication channel for digital code transmission. In the conventional MMA adaptive equalizer, the coefficient is updated by using the equalizer output and possible to compensation of amplitude and phase in 2nd dimensional QAM signal, the equalization performance were degraded due to fixed modulus and step size. For the overcomming the abovemensioned problem, it is possible to improving the equalization performance in the 2nd dimensional QAM signal by applying the adaptive modulus and adaptive step size propotional to equalizer output signal to the conventional MMA algorithm. The computer simulation was performed in the same channel for the compare the performance of MMA and proposed H-MMA which is proposed in this paper. As a result of simulation, the proposed H-MMA has slower convergence time in order to arriving the steady state than MMA. But after the steady state, H-MMA has more superior to the MMA in every performance index and the equalization noise was reduced.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.19 no.5
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• pp.107-113
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• 2019

This paper proposes the Hybrid-CRMMA adaptive equalization algorithm that is possible to improves the performance of CR-MMA based on adaptive modulus and adaptive stepsize. The 16-QAM nonconstant modulus signal is reduced to 4-QAM constant modulus signal, and the error signal were obtained based on the fixed statistic modulus of transmitted signal. It is possible to improving the currently MMA adaptive equalization performance. The proposed Hybrid-CRMMA composed of adaptive modulus which is propotional to the power of equalizer output and adaptive stepsize which is function of the nonlinearties of error signal, and its improved equalization performance were confirmed by computer simulation. For this purpose, the output signal constellation, the residual isi and maximum distortion and MSE that is for the convergence characteristics, the SER that is meaning the robustness of external noise of algorithm were used. As a result of computer simulation, it was confirmed that the proposed Hybrid-CRMMA has more superior performance in every index compared to currently CR-MMA.

• Speech Sciences
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• v.9 no.4
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• pp.275-287
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• 2002

In man-machine communication by speech under the noisy environment, the quality of speech may be degraded severely for the machine to recognize correctly. Especially when the corrupting noise occupies the same band as the speech, the conventional fixed filters cannot filter out the noise effectively. In recent, to resolve such a problem adaptive noise canceller (ANC) is frequently used, which is based upon adaptive filters. The Adaptive recursive filters perform better than adaptive nonrecursive filters due to the added poles, but the stability may be severely threatened. In this paper an ANC system employing the adaptive recursive filter is proposed to enhance the speech corrupted by noise. And the stability of the adaptive recursive filter is guaranteed by employing the adaptive compensator.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2004.10a
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• pp.211-216
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• 2004

In this thesis, we present the Takagi-Sugeno-Kang (TSK) fuzzy model based adaptive controller and adaptive identification for a general class of uncertain nonlinear dynamic systems. We use an estimated model for the unknown plant model and use this model for designing the controller. The hybrid adaptive control combined direct and indirect adaptive control based on TSK fuzzy model is constructed. The direct adaptive law can be showed by ignoring the identification errors and fails to achieve parameter convergence. Thus, we propose an TSK fuzzy model based hybrid adaptive (HA) law combined of the tracking error and the model ins error to adjust the parameters. Using a Lyapunov synthesis approach, the proposed hybrid adaptive control is proved. The hybrid adaptive law (HA) is better than the direct adaptive (DA) method without identifying the model ins error in terms of faster and improved tracking and parameter convergence. In order to show the applicability of the proposed method, it is applied to the inverted pendulum system and the performance is verified by some simulation results.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.21 no.4
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• pp.75-80
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• 2021

This paper relates with the Hybrid-DSE-MMA (Hybrid-Dithered Signed Error-MMA) that is possible to improving the equalization performance by using the adaptive modulus and adaptive stepsize in DSE-MMA adaptive equalizer. The DSE-MMA possible to improve the robustness performance to external noise of SE-MMA by using the sign after adding the dither signal for get the error signal in order to update the tap coefficient. But it has a drawback of performance degradation in convergence speed and residual isi by using the fixed modulus and fixed stepsize. In this paper, it was confirmed that this equalization performance degradation was improved by applying the adaptive modulus and stepsize in DSE-MMA propotional to the output power of equalizer by computer simulation. In order to compare the improved equalization performance to currently DSE-MMA, the recovered signal constellation that is the output of the equalizer, residual isi, Maximum Distortion, MSE and the SER were used as a performance index. As a result of computer simulation, the Hybrid-DSE-MMA improve the equalization performance in every index, but gives slower convergence speed compared to DSE-MMA.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.2
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• pp.170-177
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• 1986

The adaptive array antenna with the LMS algorithm has the advantage in that it can perform train't because of its slower convergencerate. In this paper, the \ulcornerLMS algorithm is applied to the adaptive array so that the convergence rate can be improved, and the performance of he adaptive array by the \ulcornerLMS algorithm is compared to, that of the LMS adaptive array. It is shown that the adaptive array by the \ulcornerLMS algorithm is superior to the LMS adaptive array in the narrow frequency band.

• 제어로봇시스템학회:학술대회논문집
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• 1993.10b
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• pp.378-384
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• 1993

In this paper, an intelligent adaptive controller is proposed for the process with unmodelled dynamics. The intelligent adaptive controller consists of the numeric adaptive controller and the intelligent tuning part. The continuous scheme is used for the numeric adaptive controller to avoid the problems occurred in the discrete time schemes. The adaptive controller is adopted to the process with time delay. It is an implicit adaptive algorithm based on GMV using the emulator. The tuning part changes the design parameters in the control algorithm. It is a multilayer neural network trained by robustness analysis data. The proposed method can improve the robustness of the adaptive control system because the design parameters are tuned according to the operating points of the process. Through the simulation, robustnesses are shown for intelligent adaptive controller. Finally, the proposed algorithms are implemented on the electric furnace temperature control system. The effectiveness of the proposed algorithm is shown from experiments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.