• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.22-33
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• 2005

In this paper, an efficient signal interference control technique to improve the high convergence speed of LMS algorithms is introduced in the adaptive transversal filter of DS/SS. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, is analyzed to prove theoretically the improvement of high convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Also, an increase in the stop-size parameter ${\mu}$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling which is downed the rate of convergence of the adaptive equalizer. Increasing the steady-state value of the average squared error, proposed algorithm also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• Proceedings of the KSMRM Conference
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• 1997.10a
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• pp.41-41
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• 1997

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.533-535
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• 2018

As the use frequency of electronic communication equipment increases due to the influence of the 4th industrial revolution, the importance of image and signal processing is increasing. However, due to noise caused by various causes, the reliability of the equipment is degraded and malfunctions are caused. In this paper, we propose an algorithm to remove AWGN in most environments. The existing methods show relatively poor performance due to the smoothing phenomenon at the boundary of the image. To overcome this problem, we proposed a filter algorithm that adapts to the boundary region using the Sobel edge detection to remove the noise. And using the PSNR compared with traditional methods, such as to demonstrate the performance of the proposed algorithm.

• Journal of IKEEE
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• v.14 no.1
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• pp.33-39
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• 2010

In this paper, a 5Gb/s receiver with an adaptive equalizer for serial link interfaces is proposed. For effective gain control, a least-mean-square (LMS) algorithm was implemented with two internal signals of slicers instead of output node of an equalizing filter. The scheme does not affect on a bandwidth of the equalizing filter. It also can be implemented without passive filter and it saves chip area and power consumption since two internal signals of slicers have a similar DC magnitude. The proposed adaptive equalizer can compensate up to 25dB and operate in various environments, which are 15m shield-twisted pair (STP) cable for DisplayPort and FR-4 traces for backplane. This work is implemented in $0.18-{\mu}m$ 1-poly 4-metal CMOS technology and occupies $200{\times}300{\mu}m^2$. Measurement results show only 6mW small power consumption and 2Gbps operating range with fabricated chip. The equalizer is expected to satisfy up to 5Gbps operating range if stable varactor(RF) is supported by foundry process.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.11
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• pp.1545-1550
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• 2021

We propose an optimal variable step size adjustment method for the simplified subband affine projection algorithm (Simplified SAP; SSAP) in a subband structure based on a polyphase decomposition technique. The proposed method provides an optimal step size derived to minimize the mean square deviation(MSD) at the time of updating the coefficients of the subband adaptive filter. Application of the proposed optimal step size in the SSAP algorithm using colored input signals ensures fast convergence speed and small steady-state error. The results of computer simulations performed using AR(2) signals and real voices as input signals prove the validity of the proposed optimal step size for the SSAP algorithm. Also, the simulation results show that the proposed algorithm has a faster convergence rate and good steady-state error compared to the existing other adaptive algorithms.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.389-396
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• 1997

Several types of electrical artifact seen on electroencephalogram( EEG) records are described. Those are the EOG and the PVC roller pump noise, and so on. An adaptive digital filtering of the electroencephalogram( EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, whore artifacts may not be acceptable in some cafes of automatic EEG processing. Thus we studied the method for cancelling these artifacts. This proposed method does not use the reference channel, and is realized by connecting the linear predictor and the fixed FIR filter for the EOG artifact, and by cascading the linear predictor and the noise canceller for the pump artifact. The simulation results illustrate the performances of the proposed method in terms of the capability of interferences suppression. In the results we obtained about 20 dB noise reduction.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.11-18
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• 2002

In this paper, a gain-adapted speech recognition method in noise is developed in the time domain. Noise is assumed to be colored. To cope with the notable nonstationary nature of speech signals such as fricative, glides, liquids, and transition region between phones, the nonstationary autoregressive (NAR) hidden Markov model (HMM) is used. The nonstationary AR process is represented by using polynomial functions with a linear combination of M known basis functions. When only noisy signals are available, the estimation problem of noise inevitably arises. By using multiple Kalman filters, the estimation of noise model and gain contour of speech is performed. Noise estimation of the proposed method can eliminate noise from noisy speech to get an enhanced speech signal. Compared to the conventional ARHMM with noise estimation, our proposed NAR-HMM with noise estimation improves the recognition performance about 2-3%.

• 전자공학회논문지 IE
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• v.44 no.4
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• pp.1-10
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• 2007

In this thesis, we study the implementation of hands-free phone in a car, taking acoustic echo canceller, in order to remove acoustic echo effectively. Conventional coustic echo canceller used for only adaptive filtering has much difficulty to solve both echo and double-talk problem. To tackle this problem, we propose acoustic echo canceller consisting of adaptive filter using a modified NLMS, VAD to catch exact voice activity duration using two independent forgetting factors, double-talk detector to detect fast and precise double talk duration using cross-correlation between microphone signal and residual echo, and output controller using VAD and double-talk detector. The proposed hands-free phone taking acoustic echo canceller shows the performance that has not acoustic echo and guarantees full duplex.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.543-546
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• 2001

영상에 포함된 잡음은 시각적인 문제를 일으킬 뿐만 아니라, MPEG이나 H.263과 같은 영상 압축 시스템의 부호화 효율을 떨어뜨린다. 따라서 영상 압축 시스템의 입력으로 이러한 잡음이 포함된 신호가 들어갈 때, 잡음 제거 필터를 사용하여 잡음을 제거한 후 영상 압축을 하는 것이 시각적인 면에서나 압축 효율적인 면에서 매우 효과적이다. 본 논문에서는 이웃한 4개의 화소값을 참조하여 잡음의 존재 여부를 판단하고, 판단 결과를 이용하여 선택적으로 잡음을 제거하는 적응형 십자형 중간값(median) 필터를 제안한다. 제안된 방법을 이용하면 전체 영상에 걸쳐 필터를 이용하는 방법에 비해 계산량이 크게 줄고, 영상의 필터 처리후에 나타나는 뭉개짐(blurring) 현상을 줄일 수 있다. 또한 잡음이 처리된 영상을 시간방향으로 Look-up Table에 따른 IIR필터를 통과시킴으로써 시간상으로 존재하는 잡음을 제거하여 동영상의 주관적 화질을 향상시킬 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2A
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• pp.91-99
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• 2011

Several repeater systems are used to enhance the cell coverage to location such as shadow and rural areas in mobile systems. But the general RF repeater solutions are not suitable for high power outdoor environment because it has the weakness such as self oscillation problem With adoption of a adaptive digital filter technology, feedback interference cancellation repeater prevents oscillation by detecting and canceling the unwanted feedback signal between transmission and receiver antenna. In this paper, dual NLMS based interference cancellation method is proposed and the step size adaptation can be implemented by the estimation of the feedback channel Doppler frequency characteristics. The performance of the proposed algorithm is quantified via analysis and simulation for the static and multipath fading feedback channels.