• Journal of the Korean Society for Railway
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• v.14 no.6
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• pp.477-483
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• 2011

In this paper, noise source localization of the Korean high speed train was conducted by using delay and sum beam-forming method of a microphone array. At first, the microphone array having irregular configuration was designed and the resolution of which was analyzed from parameters such as 3-dB bandwidth and maximum side-lobe level. After the demonstration, the microphone array was applied on the high speed train and noise localization of the high speed train driving at 300 km/h was performed successfully.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.04a
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• pp.816-820
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• 2011

A method for increasing the difference of side-lobe level in spherical microphone array is presented. In array signal processing, it is known that narrow interval between sensors can increase the difference between main lobe and side-lobe of array response which eventually increase the source recognition capability. Recent commercial array being used, however, have shown certain limitation in using the number of sensors due to its costs and geometrical size of array. To overcome this problem, we have adapted MEMS sensors into spherical microphone array. To check out the improvement, two different types of spherical microphone array were designed. One array is composed with 32 regular instrument microphones and the other one is 85 MEMS sensors. Simulation and experiments were conducted on a sinusoidal noise source with two arrays. The time history data were analyzed with spherical harmonic decomposition and beamforming technique. 85 MEMS sensors array showed the improved side-lobe level suppression by more than 4 dB above the frequency content of 2 kHz compared to 32-sensor array.

• Journal of KIISE
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• v.43 no.1
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• pp.13-26
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• 2016

This paper proposes a new sound localization algorithm that can improve localization performance based on a tetrahedron-shaped microphone array. Sound localization system estimates directional information of sound source based on the time delay of arrival(TDOA) information between the microphone pairs in a microphone array. In order to obtain directional information of the sound source in three dimensions, the system requires at least three microphones. If one of the microphones fails to detect proper signal level, the system cannot produce a reliable estimate. This paper proposes a tetrahedron- shaped sound localization system with a coordinate transform method by adding one microphone to the previously known triangular-shaped system providing more robust and reliable sound localization. To verify the performance of the proposed algorithm, a real time simulation was conducted, and the results were compared to the previously known triangular-shaped system. From the simulation results, the proposed tetrahedron-shaped sound localization system is superior to the triangular-shaped system by more than 46% for maximum sound source detection.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.573-578
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• 2003

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this paper, it was described to be able to enhance the speech quality through microphone array, and received the only signal of speaker. Unfortunately, as it using estimated the signal in advance, it is not matched in a real acoustic environment so it has poor performance. In this paper is proposed for Adaptive Matched Filter Microphone Array that estimated acoustic room environment from the received the signal and study of the efficiency through simulations.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.47-53
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• 1997

This paper concerns an efficient approach to forming a beam pattern of a microphone array to deal with broadband signals such as speech in a teleconference. A numerical method is proposed to find updated location of sidelobes for equalizaing the sidelobes via perturbation of array parameters such as array weight or microphone spacing. Thus the microphone array is optimized in a Dolph-Chebyshev sense such that directional or background noises incident in an array visual range are eliminated efficiently. It is shown that perturbation of microphone spacing yields an optimum pattern more appropriate for dealing with broadband signals than that of array weight. Also, a novel method is proposed to find a beam pattern which is robust with respect to sidelobe in a scanning situation. Computer simulation results are presented.

• Broadcasting and Media Magazine
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• v.19 no.1
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• pp.56-64
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• 2014

본 고에서는 증강현실 오디오 기술에 대한 연구동향을 소개한다. 특히 증강현실 오디오 실현을 위한 핵심기술이라고 할 수 있는 오디오 객체를 획득하기 위한 단일지향성 및 쌍지향성 마이크로폰을 이용한 마이크로폰 어레이 설계 기술과 오디오 채널간의 신호 강도차 분석을 통한 오디오 객체 획득 기술에 대해 설명한다.

• Journal of the Korean Society for Precision Engineering
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• v.26 no.11
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• pp.20-28
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• 2009

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.3-10
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• 2000

This paper proposes acoustic holographic measuring system to estimate an absolute position of sound source. Using the measured signals, the estimation of the position is calculated by the Cross-spectrum algorithm of the beamformed signal and a linear arrayed microphone's signals. As the results of comparing the reference microphone method with beamforming method through the measurement of sound field, the beamforming acoustic holographic method is progressed above 20 percent than that of a reference microphone method in the resolution, and the utility of the proposed system could be confirmed.

• 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.