• The Transactions of the Korean Institute of Electrical Engineers D
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• v.49 no.11
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• pp.610-616
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• 2000

This paper presents an active noise control algorithm via Walsh transform domain controller learned by genetic algorithm. Typical active noise control algorithms such as the filtered-x lms algorithm are based on the gradient algorithm. Gradient algorithm have two major problems; local minima and eigenvalue ratio. To solve these problems, we propose a combined algorithm which consist of genetic learning algorithm and discrete Walsh transform called Walsh Transform Domain Genetic Algorithm(WTDGA). Analyses and computer simulations on the effect of Walsh transform to the genetic algorithm are performed. The results show that WTDGA increase convergence speed and reduce steady state errors.

• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3213-3215
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• 2000

뇌자도 신호의 측정은 뇌에서 발생하는 자장 성분을 정밀하게 측정할 수 있으나, 신호의 크기가 매우 작기 때문에 노이즈에 매우 민감하게 동작하며 이러한 노이즈 성분의 발생원인은 외부 환경에 의하여 발생하거나 시스템 내부에서 발생하는 두가지로 나눌 수 있다. 따라서 뇌자도 신호를 측정하는데 있어서 가장 중요한 작업은 신호에 존재하는 노이즈 성분을 제거하는 것이다. 특히 뇌자도 측정 시스템에서는 외부 노이즈 성분을 제거하기 위하여 레퍼런스 채널이 존재한다. 따라서 본 논문에서는 청각 자극 신호에 의한 뇌자도 신호를 측정하고 측정한 데이터를 사용하여 레퍼런스 채널과 입력신호에 대하여 LMS 알고리즘을 이용한 적응 필터를 모델링 하였다. 그리고, 구현한 적응 필터를 이용하여 뇌자도 신호의 평균값, 표준편차의 통계적 결과를 비교하여 모델링한 적응 필터 방법의 유용성을 확인하였다.

• Proceedings of the KIEE Conference
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• 1995.07b
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• pp.904-906
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• 1995

Based on the detection of the size of datas of multipath channel characterization, we propose a new algorithm. called the impulse size based adaptive median filter(ISMF), for ghost cancellation system. The ISMF consists of two levels. The first one is the impulse noise size detection level and the second one is the adaptive median filtering level to remove the impulse noise detected. Test results confirm that the proposed ISMF removes impulse noise due to multipath channel characterization while preserving signal as well as ghosts so that the LMS algorithm performs effectively.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.9
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• pp.62-70
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• 1995

In some practical applications of the LMS Algorithm the coefficient adaptation can be performed only after some fixed delay. The resulting algorithm is known as the Delayed Least Mean Square (DLMS) algorithm in the literature. There exist analyses for this algorithm, but most of them are based on the unrealistic independence assumption between successive input vectors. Inthis paper we consider the DLMS algorithm with decreasing step size .mu.(n)=n/a, a>0 and prove the almost-sure convergence ofthe weight vector W(n) to the Wiener solution W$_{opt}$ as n .rarw. .inf. under the mixing unput condition and the satisfaction of the law of large numbers. Computer simulations for decision-directed adaptive equalizer with decoding delay are performed to demonstrate the functioning of the proposed algorithm.m.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.343-344
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• 2007

In this paper, we study the signed-DLMS adaptive algorithm of wireless repeater for solving shadow region due to propagation between base station and mobile station. The the signed-DLMS algorithm reduces interference signals from multipath and solves the oscillation problem of repeater by estimation and cancellation. To efficiently reject interference signal, the signed-DLMS adaptive algorithm is applied. The computational complexities of the signed-DLMS are reduced verse standard LMS algorithm. Wireless ICS repeater based on signed-DLMS reduces the cost and is able to increase channel capacities.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.119-120
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• 2007

The operation of Interference cancellation algorithm for wireless repeater cancellation depends on either existing correlation properties between desired signal and reference signal or not At the time, due to the correlation properties at the ICS system, adaptive algorithms without considering system delay do not function properly. Thus, this system should be oscillated. In this paper, to solve these problems, we use the delayed least mean square algorithm. For the best performance of ICS, the system delays must be estimated. To efficiently estimate the delay of ICS, we use relations between bandwidth and correlation properties of the received signal.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.6
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• pp.446-452
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• 2004

Under the condition of a common narrowband target signal and interference signals from several directions, the linearly constrained minimum variance (LCMV) method using the generalized sidelobe canceller (GSC) for adaptive beamforming has been exploited successfully However, in the case of wideband signals, the length of the adaptive filter must be extended. As a result, the complexity of the beamformer increases, which makes real-time implementation difficult. In this paper, we improve the convergence characteristics of the adaptive filter using the transform domain normalized least mean square (NLMS) approach based on the subband GSC structure without the increase of complexity. Besides, the M-MAX algorithm, which is one of various selective coefficient updating methods, is employed in order to remarkably reduce the computational cost without decreasing the convergence quality. With the combination of these methods, we propose a computationally efficient wideband adaptive beamformer and verify its efficiency through a series of simulations.

• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.2302-2304
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• 2001

In the double-talk situation where both the near-end and far-end signal present, the performance of echo cancellation using the conventional LMS algorithm is degraded easily since it freezes the adaptation in this situation. Recently CLMS and ECLMS algorithm were proposed to solve this problem. These algorithms could be used to adapt the filter's parameters continuously even in the double-talk situation. In this paper, we compare and analyze their performance. The computer simulation was performed and the results showed as that both algorithms were robust and kept the desirable performance even in the double-talk situation.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.149-152
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• 2000

통신기술의 발달과 정보화 사회로 빠르게 변화되면서, 유선ㆍ무선, 핸즈프리, 원거리 화상회의 등의 다양한 방식의 통신이 이루어지고 있다. 음성통신의 어려운 문제 중에 하나는 주위의 소음이다. 소음은 상황에 따라서 다양하고 복잡하여 그 특성을 분석하기가 어렵다. 소음의 특성과 반향 등을 분석하기 위해서는 수 천 개의 적응필터 탭이 필요하게 된다. 따라서 실시간 소음제거를 위해서는 계산량이 많아 어려움이 따르므로 계산량 감소를 위해 FFT연산에 근거한 주파수 영역의 FDAF 적응필터를 이용하게 되었다. 하지만 계산량은 상당히 감소되었지만, 적응필터의 차수가 증가하면서 시간지연과 하드웨어적으로 복잡하게 되어 블록의 차수를 줄일 수 있는 MDF를 비교 검토하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.433-438
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• 1999

This paper describes Active Noise Cancellation/Control(ANC) method that removes the information of the unnecessary noise and doesn't remove the informations of the necessary noise(warning sound, operating sound etc.) for the induced noise of the mechanical system. In this paper, the noise source Is axial fan, and the Feedback Active Noise control method that can effectively control BPF generated from the axial fan is used, and the Filtered-X LMS algorithm for adaptive algorithms is used. The experiments are executed for two case(propagating noise in the duct, emission noise for exterior free field). The part to be removed is BPF noise, and the band-pass filter not to effect to the other frequencies is used. Also, to investigate the effect of the noise reduction for human, we are compared with the results that are controlled for using Loudness before and after. As a results, we are certified that the BPF is decreased only and frequencies outside of BPF are not affected, and we acquire the reduction effects of 6.7 dB Loudness Level, then the frequency to be removed is controlled. Therefore, we can be certified that sound pressure as well as loudness can be effectively decreased for human sound quality