• The Journal of the Acoustical Society of Korea
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• 제17권4E호
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• pp.22-27
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• 1998

A modified perturbation method is proposed to optimize the beam pattern of thinned microphone arrays. Both microphone spacing and array weight are iteratively adjusted via successive perturbation to achieve an optimum beam pattern in a Dolph Chebyshev sense. To improve the sidelobe performance, an alternative perturbation with respect to microphone spacing and array weight is implemented. Also, a linear space-tapering is employed in the perturbation process. It is demonstrated that the proposed approaches successfully yield sidelobe performances comparable to that of a normal array. Computer simulation results are presented.

• The Journal of the Acoustical Society of Korea
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• 제24권2E호
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• pp.38-45
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• 2005

The noise reduction of the In-The-Ear (ITE) hearing aid (HA) can be achieved by arrays of microphones. Each of the right and the left ears was assumed to have two HA microphones. These arrays of HA microphones produce particular patterns of directivity by some time delay between two microphones. The directivity pattern geometrically increase the S/N ratio. The boundary element method (BEM) was used for the three dimensional simulation of the HA directivity pattern with the two pairs' microphone arrays. The separation between two microphones was fixed to 10 mm. The time delay between the two microphones was calculated to produce the most narrow directivity pattern in the fore front of the head. The variation of the time delay was examined in accordance with input frequencies. This numerical analysis may be then applied for the calculation of the time delay parameter of the digital hearing aid DSP chip.

• 한국철도학회:학술대회논문집
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• 한국철도학회 2011년도 정기총회 및 추계학술대회 논문집
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• pp.2269-2275
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• 2011

This paper deals with the design of a microphone array to measure location and spectral characteristics of railway noise sources. A microphone array estimates the direction of a noise source assumed as a point source using the delayed-sum beamforming method and its performance is determined in terms of resolution and side-lobe level. A 48 channel microphone array was already developed to measure noise sources of KTX trains and a new array with 96 microphones has been designed to enhance the performance. This paper simulates the performance the microphone arrays according to the configuration of microphones and verifies it through on-site tests.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2009년도 추계학술대회 논문집
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• pp.463-464
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• 2009

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2010년도 춘계학술대회 논문집
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• pp.742-743
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• 2010

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2003년도 추계학술대회논문집
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• pp.735-741
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• 2003

A simplified view of array design and application process was introduced. Array design is critical to achieve a successful phased array measurements. A planar microphone array is designed to produce optimum performance and also to fit economic requirement in integrating data acquisition system. Certain performance characteristics are of primary concern when designing arrays. These characteristics include array resolution, spatial aliasing and array sidelobe suppression. Every array has its directional pattern that shows such characteristics. Assuming that a monopole source is located in center, beam-patterns have been simulated varying measurement conditions such as number of sensors. array aperture size, distance between array and source, frequency of interest and so on. Sensor correction was conducted on very channel using magnitudes and phased of FRF with respect to a reference microphone channel. Then with a spiral type array, measurements have been made with two point sources of same frequency in order to investigate array resolving abilities. It is observed that higher frequency source achieves better resolution than lower one does.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2011년도 춘계학술대회 논문집
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• pp.816-820
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• 2011

본 논문은 구형 마이크로폰 어레이의 부엽 레벨의 차를 증가시키기 위한 방법에 대한 연구 내용을 다루었다. 일반적인 어레이 신호처리에서 마이크로폰을 조밀하게 배치함으로써 어레이 응답에서의 주엽과 부엽 간의 차이를 늘릴 수 있고 어레이의 소음원 판별능력을 증가시킨다. 최근 사용되고 있는 상용 에레이들은 제작 단가와 어레이의 크기 때문에 센서의 수를 늘리는데 한계를 보이고 있다. 이런 문제를 극복하기 위해 본 연구에서는 MEMS 센서를 이용하여 구형 어레이에 적용하였다. 구형 마이크로폰 어레이를 이용한 시뮬레이션과 실험을 통해 정현파 소음원을 측정하였다. 실험을 위해 32 개의 일반 측정용 마이크로폰을 이용한 어레이와 85 개의 MEMS 마이크로폰을 이용한 구형 어레이를 제작하였다. 구형 조화 분해기법과 빔형성기법을 이용하여 측정 데이터를 분석하였다. 2 kHz 이상의 소음원에 대하여 MEMS 마이크로폰 어레이가 4 dB 이상의 부엽 저감 능력을 가지는 것을 확인하였다.

• 대한음성학회지:말소리
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• 제66호
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• pp.73-86
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• 2008

Beamforming is one of the spatial filtering techniques which extract only desired signals from noisy environments using microphone arrays. Fixed beamforming is a simple concept and easy to implement. However, it does not show good performance in real noisy conditions. As an adaptive beamforming, Frost algorithm can be a good candidate. It uses the concept of the linearly constrained minimum variance (LCMV) algorithm. The difference between the Frost and the LCMV algorithm is the error correction scheme which is very effective feature in the aspect of performance. In this paper, as quadrature mirror filtering (QMF)-based filterbank is utilized as the pre-processing of the Frost beamformning, the filter length and the learning rate of each band is optimized to improve the performance. The performance is measured by the signal-to-noise ratio (SNR) and the Bark's scale spectral distortion (BSD).

• 한국철도학회:학술대회논문집
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• 한국철도학회 2006년도 추계학술대회 논문집
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• pp.144-151
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• 2006

Sound generated from a moving vehicle often carries information on the condition of vehicle, for example, whether it has faults or not, where the fault exists. The latter is possible especially by MFAH(moving frame acoustic holography) and beamforming method. MFAH is applicable to the sound source of pure tone or narrow band noise. For the beamforming method, we have to know what kind of wave the sound source radiates, for example, plane wave or spherical wave. That is, whether the above methods are applicable depends on the characteristics of sound source. To apply these methods to the fault detection, we have to know the characteristics of wave from faults. In this research, a machine diagnosis technique based on the above holographic approaches is introduced to find the position of faults. The signal due to faults is modeled based on the fact that the faults radiate impulsive noise, and analyzed in time and frequency domain. The way how MFAH and beamforming method can be used is introduced to find the position of source.

• 한국멀티미디어학회:학술대회논문집
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• 한국멀티미디어학회 2003년도 춘계학술발표대회논문집
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• pp.446-449
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• 2003

The conventional GSC is inappropriate in real situation when the target signal is present. The steering vector error cancels the target signal and the target signal misadjusts the weight of the adaptive filter. To prevent the target signal cancellation, the robust GSC using the constrained adaptive filters was already proposed. However, the adaptive weight misadjustment is not settled in robust GSC. This Paper proposes a revised robust sidelobe canceller with adaptive compensator. To compensate the influence of target signal, the adaptive compensator is used in cascade. In computer simulation, we show the performance improvement by comparing the robust GSC with the proposed GSC.