• The Journal of the Acoustical Society of Korea
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• v.10 no.6
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• pp.71-76
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• 1991

적응 어레이 시스템에서 최적해에 도달하기 위한 계수들의 수렴 속도는 공분산 행렬의 고유치 분포율에 의해 지배된다. 본 논문에서는 우선 GSC의 고유치 분포율을 구하였다. 또한 conherent 상황에 서 GSC의 출력 파워 식을 유도하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.1
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• pp.59-66
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• 1996

In this paper, we present a new effective algorithm for element failure compensation of Generalized Sidelobe Canceller (GSC). While the GSC is well formulated, little works have been done on array element compensation in the presence of faulty elements. Element failure changes the problem of a linearly equally spaced array into that of an unequally spaced array. Typical research approaches have been directed at using search techniques to optimize unequally spaced arrays. The proposed algorithm matches the linear constraint conditions and the general shape of the desired beam pattern at the expense of an increase of beam-width in the overall main lobe. Numerical results are included to demonstrate the capability of compensation for various situations.

• ETRI Journal
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• v.38 no.2
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• pp.366-375
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• 2016

This paper describes an algorithm to suppress composite noise in a two-microphone speech enhancement system for robust hands-free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal-dominant time-frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech-dominant TFBs are identified among the previously detected nonstationary signal-dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin-wise output signal-to-noise ratio is obtained with these power estimates and a Wiener post-filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post-filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.389-394
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• 2016

MIMO (Multiple-Input-Multiple-Output) in underwater acoustic communication has distortion of received signal because of ISI (Inter-Symbol Interference) and crosstalk among transmitters. Time-reversal mirror was used for compensating of signal distortion, but it has a limit in eliminating crosstalk effectively. This paper proposes a time-reversal mirror based on GSC (Generalized Sidelobe Canceller) for removing crosstalk. The FAF05 (The Focused Acoustic Forecasting 05) experimental data has been used to verify the suggested method by comparison with the conventional time-reversal for communication performance, and it is demonstrated that the suggested method produces better communication performance results than conventional time-reversal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.54-60
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• 2005

In this Paper, an adaptive microphone array system with self-delay estimator is proposed. By showing that the adaptive blocking matrix (ABM) of the generalized sidelobe canceller (GSC) can estimate the relative time delay between each sensor, the proposed system utilizes the ABM not only for blocking target components in the blocked signal path, but also for estimating the relative time delay. Therefore, the proposed system requires only the GSC structure while maintaining the system performance similar to the conventional system using an additional time delay estimator as a preprocessor. Simulation results show that the performance of the proposed system is identical to the conventional system that uses an additional time delay estimation module.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.26 no.3
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• pp.317-325
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• 2015

In an interference environment, adaptive sidelobe blanking(adaptive SLB: ASB) algorithm effectively cancels the high-duty cycle jammer and blocks the sidelobe signals without the auxiliary antenna. The adaptive SLB for the linearly constrained minimum variance (LCMV) is proposed in this paper. In the proposed scheme, the interference covariance matrix is modified to satisfy the direction constraints of LCMV and the normalized output can be obtained to block sidelobe signals. As the LCMV can be represented as a generalized sidelobe canceller(GSC) form, which is the general framework of various adaptive beamforming(ABF) algorithms, the proposed adaptive SLB can be applied to various ABF methods. The performance of the proposed method is verified through simulation and analysis.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.37-40
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• 2001

본 논문에서는 수렴속도 향상을 위해 시간영역의 적응알고리즘을 직교변환인 아다말(Hadamard)변환을 이용하여 적응알고리즘을 변환영역에서 수행하며, 변환영역에서 수렴성능 향상을 위해 가변스텝 사이즈를 갖는 적응알고리즘을 제안한다. 제안한 알고리즘을 일반적인 부엽제거기(Generalized Sidelobe Canceller GSC)에 적용하여 컴퓨터 시뮬레이션을 하였으며, 각 알고리즘들의 계산량, 수렴 성능을 이용하여 각각 비교, 분석하여 제안한 알고리즘의 성능이 우수함을 입증하였다.

• Speech Sciences
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• v.13 no.1
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• pp.117-128
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• 2006

The two-channel Generalized Sidelobe Canceller (GSC) scheme suffers from the presence of leakage signal in the reference channel. The leakage signal is caused by the dissimilar impulse responses between microphones, and different paths from speech source to microphones. Such leakage is detrimental to speech enhancement of the GSC since the desired reference signal becomes corrupted. In order to suppress the signal leakage, two matrix injection methods are proposed. In the first method, a simple gain compensation matrix is used. In the second, a projection matrix for reducing the error between the actual and the ideal primary and reference signals, is used. This paper describes the performance degradation resulting from leakage, and proposes effective methods to resolve the problem. Representative experiments were conducted to demonstrate the effectiveness of the proposed methods on recorded speech and noise in an actual automobile environment.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.1 no.1
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• pp.21-27
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• 2008

In this paper, a PC-based real-time microphone array system with small aperture is presented. The microphone array system is based on the generalized sidelobe canceler (GSC) but it employs a new adaptation mode controller (AMC). The performance of the proposed system was evaluated in the Multimedia Room modeled on an office situation. Evaluation experiments show that the proposed system can perform with stable noise suppression.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.28 no.7
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• pp.576-579
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• 2017

Due to the existence of flash lobe caused by the radome structure, a radar return in general contains clutter component added to target echo. Unlike jammer, clutter is apt to simultaneously coexist with target echo, which hinders a monopulse radar from estimating the angle difference between target echo direction and radar steering direction. In this paper, we propose a clutter estimation algorithm applicable to generalized sidelobe canceller(GSC) based monopulse radar. We show that clutter estimation can be achieved via iterative process of the orthogonal projections. Through simulations, the proposed GSC-based system combined with the clutter estimation can outperform the conventional monopulse system in terms of target angle estimation.