• The Journal of Engineering Research
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• v.8 no.1
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• pp.59-65
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• 2006

본 논문에서는 마이크로폰 어레이 시스템을 이용하여 여러 화자의 음성 정보로부터 각 화자가 위치한 방향을 추정하는 기술 개발 내용을 다룬다. 성능 향상을 위한 전처리 과정으로 비선형 증폭기를 사용하여 거리에 따른 영향을 최소화하는 과정과 잡음에 대한 강인성을 얻기 위해 음성활성 영역을 검출하는 과정을 포함한다. 등간격으로 배치된 마이크로폰 어레이 시스템의 기하학적 특성에 따른 음원의 위치와 신호의 지연시간차이와의 상관관계로부터 화자의 위치를 역으로 추정하는 알고리즘을 기본으로 하여 가능성 척도를 계산하고 이를 활용하여 가능성이 높은 것들을 클러스터링하여 가능성이 있는 후보를 선정하여 화자의 방향을 검지한다. 이 과정에서 오인식을 최소화하기 위하여 가능성이 희박한 영역에 대한 추정 억제 방법으로 부정식 추론법을 적용하였다. 2 화자의 음성 신호를 입력으로 한 실험을 통하여 제안한 방법에 의한 다화자 방향검지의 가능성을 알아보았다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.4 s.304
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• pp.111-118
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• 2005

In this paper, we employ the modulation technique for improving the characteristics of beamformer in the low frequencies and thus improving the overall noise reduction performance. In the 1-dimensional uniform linear microphone arrays, we can suppress the narrowband noise component using the delay-and-sum beamforming. But, for the wideband noise signal, the delay-and-sum beamformer does not work well for the reduction of low frequency component because the inter-element spacing is usually set to avoid spatial aliasing at high frequencies. Hence, the beamwidth is not uniform with respect to each frequency and it is usually wider at the low frequencies. In order to obtain the beamwidth independent of frequencies, subarray systems[1][2][3][4] and multi-beamforming[5] have been proposed. However these algorithms need large space and more microphones since they are based on the theory that the size of the array is proportional to the wavelength of the input signal. In the proposed beamformer, we reduce the low frequency noise by using modulation technique that does not need additional sensors or non-uniform spacing. More Precisely, the array signals are split into subbands, and the low frequency components are shifted to high frequencies by modulation and reduced by the delay-and-sum beamforming techniques with small size microphone array. Experimental results show that the proposed technique Provides better performance than the conventional ones, especially in the low frequency band.

• The Journal of the Acoustical Society of Korea
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• v.30 no.3
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• pp.115-122
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• 2011

In this paper, we present a method to estimate an accurate real-time sound source direction based on time delay of arrival by using generalized cross correlation with four cross-type microphones. In general, existing systems have two disadvantages such as system embedding limitation due to the necessity of data acquisition for signal processing from microphone input, and real-time processing difficulty because of the increased number of channels for sound direction estimation using DSP processors. To cope with these disadvantages, the system considered in this paper proposes hardware design for enhanced real-time processing using microphone array signal processing. An accurate direction estimation and its design time reduction is achieved by means of an efficient hardware design using spatial segmentation methods and verification techniques. Finally we develop a system which can be used for embedded systems using a sound codec and an FPGA chip. According to experimental results, the system gives much faster real-time processing time compared with either PC-based systems or the case with DSP processors.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2017.11a
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• pp.44-45
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• 2017

본 논문에서는 비균등 선형 마이크로폰 어레이를 활용한 convolutional neural network (CNN) 기반의 음원분리 방법을 제안한다. 우선, 주어진 어레이 배치에 따라 채널간의 시간차를 분석하고, 분석된 시간차에 따라 주파수별로 방사각과 넓이에 따라 입력 오디오 신호의 spectral magnitude를 예측한다. 그러고 나서, CNN 분류기로부터 최적의 방사각과 넓이를 선별하고 이를 통해 음원을 분리한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2015.11a
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• pp.21-22
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• 2015

현재 UHDTV (Ultra-High-Definition TV) 시대에 사용될 멀티미디어 부호화로 MPEG-H를 표준화로 진행하고 있다. 향후 방송용 오디오 콘텐츠는 채널 오디오 콘텐츠에서 진화하여 객체 오디오 콘텐츠까지도 필요하게 된다. 이에 따라, 본 논문에서는 고품질의 방송용 오디오 콘텐츠를 제작하기 위한 비균등 선형 마이크로폰 어레이 기반의 음원분리 방법을 제안한다. 제안된 방법은 주어진 어레이 배치에 따라 채널간의 시간차를 분석하고, 이에 따른 객체 오디오 생성을 위한 음원분리 기술을 적용한다. 제안된 기법의 성능을 검증하기 위하여 음원분리도를 측정하였고, MVDR (Minimum Variance Distortionless Response) 빔형성기와 성능을 비교하였다. 비교 결과, 제안된 기법이 MVDR 빔형성기에 비하여 12.8% 높은 음원분리도 수치를 나타낸 것을 확인하였다.

• Journal of Broadcast Engineering
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• v.23 no.4
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• pp.519-525
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• 2018

This paper deals with noise suppression of multi-channel data captured by microphone array using multi-channel Wiener filter. Multi-channel Wiener filter does not rely on information about the direction of the target speech and can be partitioned into an MVDR (Minimum Variance Distortionless Response) spatial filter and a single channel spectral filter. The acoustic transfer function between the single speech source and microphones can be estimated by subspace decomposition of multi-channel Wiener filter. The errors are incurred in the estimation of the acoustic transfer function due to the errors in the estimation of correlation matrices, which in turn results in speech distortion in the MVDR filter. To alleviate the speech distortion in the MVDR filter, diagonal loading is applied. In the experiments, database with seven microphones was used and MFCC distance was measured to demonstrate the effectiveness of the diagonal loading.

• The Journal of the Acoustical Society of Korea
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• v.34 no.1
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• pp.60-65
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• 2015

In this paper, we used the sound of gunshots recorded by multiple microphones to increase the accuracy of the calculation of the distance between sniper and the microphone array. This method is crucial for achieving military objectives. Gunshots are comprised of the explosion of driving gas from the muzzle and the supersonic shock wave from the flying bullet. The original distance calculation method compares the time difference of arrival and angle of incidence to estimate the sniper's location. The disadvantage of this method is that when the angles of incidence coincide the margin of error increases, to solve this problem we suggest a new method using the characteristic changes of the shock wave with the increase of perpendicular distance between the microphone and the trajectory of the bullet. This theory is verified by experiments.

• Journal of the Korean Society for Railway
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• v.15 no.1
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• pp.23-28
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• 2012

In this paper, noise of Korean high-speed trains (KTX) running at different speed from 150 to 300km/h was measured by a microphone array system. From the measurement, relation between maximum sound pressure levels and train moving speeds of KTX was drawn and a regression coefficient from the relation was also derived. Moreover, increases of SPL with speeds of KTX were analyzed in the frequency domain. From the analysis, sound characteristics of passing-by noise of KTX were provided. Then, dominant noise source areas were obtained from the measurements and propagation patterns of KTX in vertical direction were also investigated. Finally, noise sources of KTX were identified from inspection of noise maps.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.609-612
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• 2000

본 논문에서는 음원에 관한 정보가 미지의 상황에서 마이크로폰 어레이를 사용하여 두 음원신호를 분리하는 ,시스템을 제안한다 이 시스템은 두 단계로 구성되어 있으며, 첫 번째 단계에서는 파워가 큰 제 1음원의 DOA(Direction Of Arrival)를 추정하고, AMUSE(Algorithm for Multiple Unknown Signals Extraction)법을 사용한 Blind Deconvolution에 의해 음원신호의 분리를 행한다 두 번째 단계에서는 파워가 낮은 제 2음원의 강조신호를 사용하여 DSA(Delay and Sum Array)법에 의해 제 2음원의 DOA를 추정하고,AMUSE법의 출력신호와 두 음원의 DOA를 이용하여 ANF(Adaptive Notch Filter)를 구성하고, 두 음원신호의 재 분리를 행한다. 그리고, 시뮬레이션을 통해 제안한 방법의 유효성을 검토한 결과 두 음원 신호가 분리 가능한 것이 확인되었다.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.92-96
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• 2007

Microphone array beamforming method has been recognized as an important aeroacoustic research field and become a standard technique in localizing sound sources. This method also used in flight acoustic measurement, and especially, it is very useful when measure sounds inside the wind tunnel. In measuring sound which is inside the wind tunnel by traditional beamforming method, there are some errors caused by airstream. The speed and the propagation path of the sound changes as it travel through the airstream. This makes the error which the position of sound is changed a little bit to the down stream direction. In this paper, validation test has made about the correction equation for this wind effects of previous researches. And beamforming including shear layer correction was performed about a sound source in the anechoic open-jet windtunnel.