• Journal of the Institute of Convergence Signal Processing
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• v.7 no.2
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• pp.53-59
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• 2006

Dynamic neural networks have been applied to diverse fields requiring temporal signal processing such as system identification and signal prediction. This paper proposes the gamma neural network(GAM), which uses gamma memory kernel in the hidden layer of feedforward multilayered network, to improve dynamics of networks and then describes nonlinear adaptive prediction using the proposed network as an adaptive filter. The proposed network is evaluated in nonlinear signal prediction and compared with feedforword(FNN) and recurrent neural networks(RNN) for the relative comparison of prediction performance. Simulation results show that the GAM network performs better with respect to the convergence speed and prediction accuracy, indicating that it can be a more effective prediction model than conventional multilayered networks in nonlinear prediction for nonstationary signals.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.28-34
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• 2001

In stereo acoustic echo canceller scheme, coefficients of adaptive filter converge very slowly or misconverge to real acoustic echo path in receiving room. This is due to cross-correlation in stereo signals. In this paper, a new preprocess algorithm is proposed to improve the performance of stereo AEC(acoustic echo canceller) without computational burden. The proposed algorithm reduces cross-correlation using Gram-Schmidt orthogonality principles and nonlinear filtering. Computer simulations demonstrate that this algorithm performs well compared to conventional ones. When the acoustic path of transmitting room is changed, stereo AEC using proposed algorithm is well performed.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1687-1693
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• 2013

This paper addresses an adaptive postprocessing technique to remove blocking effects of the highly compressed images. The proposed technique removes blocking effects selectively by applying filters with different strength according to block analysis based on the directional activity. One-dimensional filters which are used to remove grid noises accomplish the adaptive filtering to the signal itself as well as to the directionality of the block. Moreover, we propose a detection method of the staircase noises and corner outliers and a two-dimensional directional filter to remove them. Experimental results for various images and bitrates show that the proposed method outperforms the conventional methods in PSNR for the objective performance and GBIM for the subjective quality evaluation.

• 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 the Institute of Convergence Signal Processing
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• v.11 no.3
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• pp.189-196
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• 2010

In this paper, we proposed the adaptive local sigma filter reducing noises generated by an image sensor. The small noises generated by the image sensor are amplified by increased an analog gain and an exposure time of the image sensor together with information. And the goal of this work was the system design that is reduce the these amplified noises. Edge data are extracted by Flatness Index Map algorithm. We made the threshold adaptively changeable by the luminance average in this algorithm that extracts the edge data not in high luminance, but just low luminance. The Local Sigma Filter performed only about the edge pixel that were extracted by Flatness Index Map algorithm. To verify the performance of the designed filter, we made the Window test program. The hardware was designed with HDL language. We verified the hardware performance of Local Sigma Filter system using FPGA Demonstration board and HD image sensor, $1280{\times}720$ image size and 30 frames per second.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.35-43
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• 1999

In this paper, a new robust adaptive algorithm and a post-processing method are proposed to improve the performance of AEC without computational burden. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. To reduce the residual echoes, a new post-processing method, which is co-operated with the proposed noise-robust adaptive algorithm, is proposed in this paper. The method is based on the correlation of the desired signal and the estimation error signal. The residual echoes are attenuated as proportional to the correlation normalized with the power of desired signals. The normalized correlation plays a role as Wiener filter for residual echoes. In the double-talk situation, the estimation error signals, that are residual echoes, dominantly include the near-end speaker's speech and the normalized correlation closes to 1. Therefore, the near-end speaker's speech can be transmitted without being attenuated. When the desired signals consists of only the acoustic echoes, the residual echoes are mostly attenuated and canceled by the proposed post-processor. The computation of AEC using the proposed post-processor is comparable to NLMS algorithm.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.101-104
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• 2004

본 논문에서는 보상기를 채용하여 안정성을 확보한 적응순환필터인 ACHARF(Adaptive Compensated Hyperstable Adaptive Recursive Filter)를 사용하여 잡음제거를 통한 음성의 음질개선을 DSP 프로세서를 통하여 구현하였다. 실험에서는 TI사의 최신 DSP 프로세서인 TMS320C6413와 스테레오 오디오 코덱인 TLV320AIC23을 탑재한 Evaluation board를 사용하였다. 2개의 입력마이크를 이용하여 음성신호와 기준 잡음신호를 별도로 수집하여 알고리즘을 수행하였으며, 실험 결과로 음질개선 효과를 확인할 수 있었다. 본 연구를 통해서 시스템의 성능개선의 핵심은 입력으로 들어오는 음성신호와의 상관도가 가능한 적은 잡음신호를 수집하는 방법이라 생각되며 앞으로 이에 대한 연구가 필요하겠다.

• The Journal of Korean Institute of Information Technology
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• v.16 no.11
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• pp.97-103
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• 2018

In this paper, the estimation method for the power signal harmonics is proposed by using the time-varying optimal finite impulse response (FIR) filter. To estimate the magnitude and phase-angle of the harmonic components, the time-varying optimal FIR filter is designed for the state space representation of the noisy power signal which the magnitude and phase is considered as a stochastic process. Since the time-varying optimal FIR filter used in the proposed method does not use any priori information of the initial condition and has FIR structure, the proposed method could overcome the demerits of Kalman filter based method such as poor estimation and divergence problem. Due to the FIR structure, the proposed method is more robust against to the model uncertainty than the Kalman filter. Moreover, the proposed method gives more general solution than the time-invariant optimal FIR filter based harmonic estimation method. To verify the performance and robustness of the proposed method, the proposed method is compared with time-varying Kalman filter based method through simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.3A
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• pp.328-336
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• 2006

A novel pre-equalization method for terrestrial DTV transmitters is presented. A pre-equalizer has been used in transmitters to correct group delay and amplitude distortions caused by a channel filter. In the proposed pre-equalizer, an equation-error adaptive IIR filtering scheme is adopted unlike the conventional pre-equalization using FIR filtering schemes. The pole-zero modelling property of IIR filters improves the signal-to-noise ratio and may deal with diverse linear distortions existing in DTV transmitters as well as the channel filter distortion. Simulation results show that the proposed IIR pre-equalizer performs better than the FIR pre-equalizer in terms of the residual mean-square error.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.205-208
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• 2002

최근 영상 회의 시스템에서 화자 위치 추정 및 음질 향상 기술이 연구되고 있다. 이 시스템에서는 마이크로폰 어레이를 이용하여, 화자의 위치를 파악하여 화자의 방향으로 카메라를 자동으로 조정해 주고 그 방향으로부터 입사되는 신호만을 수신할 수 있도록 한다. 이를 위해 마이크로폰 어레이가 연구되어져 왔다. 덜 연구에서는 시간에 따라 변화하는 음향 환경에 적응하는 적응 정합 필터 마이크로폰 어레이를 제안하고, 실험을 통해 그 성능을 고찰하였다.