• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.969-974
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• 2007

In this paper, a beamforming technique is introduced to measure the acoustic impedance at both normal and oblique incidence in a free field. The acoustic impedance is obtained by separating incident and reflected signals using the adaptive nulling method which is one of the various beamforming algorithms. To obtain better results, pressure vector commonly used in array signal processing is replaced with the transfer function vector between each microphone and the white Gaussian noise is suppressed by a wavelet shrinkage technique. The experiments conducted in a semi-anechoic room show that the proposed method is efficient and accurate in measuring the acoustic impedance of sound absorbing materials under a free field condition.

• New & Renewable Energy
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• v.5 no.2
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• pp.25-30
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• 2009

The wind tunnel test to identify the acoustic noise source position of the wind turbine blade was conducted in KARI low speed wind tunnel. Microphone array and time-domain beamforming methodology was applied to this study. To reduce the data processing time, a modified beamforming method with a criteria between calculation time step and grid size for rotating angle in the cylinderical coordinate system was proposed. The test results shows that the data processing time to identify the noise source position was reduced to 20% compared with conventional method. And the dominant noise source of the blade moves from inboard to blade tip as the frequency increases.

• 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 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 the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.949-958
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• 2018

A voice recognition control system is a system for controlling a peripheral device by recognizing a voice. Recently, a voice recognition control system have been applied not only to smart devices but also to various environments ranging from IoT(: Internet of Things), robots, and vehicles. In such a voice recognition control system, the recognition rate is lowered due to the ambient noise in addition to the voice of the user. In this paper, we propose a dual channel acoustic beamforming hardware architecture for MEMS(: Microelectromechanical Systems) microphones to eliminate ambient noise in addition to user's voice. And the proposed hardware architecture is designed as ASIC(: Application-Specific Integrated Circuit) using TowerJazz $0.18{\mu}m$ CMOS(: Complementary Metal-Oxide Semiconductor) technology. The designed dual channel acoustic beamforming ASIC has a die size of $48mm^2$, and the directivity index of the user's voice were measured to be 4.233㏈.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.200-204
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• 2005

The use of beamforming method and de-Dopplerization technique was applied in studying the rotating sound sources. Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. Two main issues of the signal reconstruction in time domain are addressed herein: First, to remove Doppler effect from the measured data and to restore the original emission data of the moving source. The difference of the time domain beamforming from the frequency domain beamforming was mentioned. Also, the time domain beamforming method is deployed in the test and the comparisons were made to the frequency domain results. The time domain signal reconstruction was numerically simulated prior to the application. To validate the de-Dopplerization Performance, the rotating Point sources were examined and localized by the use of a phased array of microphone. The application of prop-rotor was conducted in a hovering condition. The results of reconstructing time signals of rotating sources and its locations were shown in the power distribution maps. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies of interest.

• Journal of the Korean Society for Railway
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• v.15 no.3
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• pp.231-236
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• 2012

In this paper, aerodynamic noise characteristics of high-speed trains were deduced from the beamforming method. First, pass-by noise of high-speed trains was measured by a microphone array system. This measurement suggested that the majority of the aerodynamic noise produced came from the bogie area, the pantograph and its cover, and inter-coach gaps. Then, beampower outputs of a position in high-speed trains were obtained from the beamforming method. By Fourier transform, sound characteristics of the position in the frequency domain were deduced from the beamforming power spectrum. From this study, aerodynamic noise characteristics from the major sources of high-speed trains were drawn.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.04a
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• pp.128-134
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• 2014

A Experimental study on frequency characteristics of the microphone array covered with Kevlar in closed test section wind tunnel. Microphones that are flush mounted in a closed test section wall of wind tunnel are subject to very high flow noise resulting from the turbulence in the wall boundary layer. At this time the microphones measure the strong hydrodynamic fluctuations generated by the flow. The phenomena are referred to a microphone self-noise and a method for reducing this has studied. In this paper the array that covered with acoustically transparent Kevlar sheet was designed and made to reduce the flow-induced self-noise. For the validation frequency characteristics of the Kevlar, the microphone array was installed on the wall and test was performed for white noise and sine wave of several frequencies using loudspeaker. In addition, the paper compared the signals as a tension of Kevlar. The results were presented that tend to decrease the sound pressure level at high frequency above 3500Hz according to existence of Kevlar.

• 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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• 2004.05a
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• pp.837-842
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• 2004

The positions of rotating sound sources have been localized by experiments with the Doppler effects removed. In order to do-Dopplerize the sound signals emitted from moving sources, two kinds of signal reconstruction methods were applied. One is the forward propagation method and the other is the backward propagation method. Forward propagation method analyze the source emission time based on the instantaneous distance between sensors and the assumed source position, then the signals are reconstructed with respect to the emission time. On the other hand, the backward method uses time delay to do-Dopplerize the acquired data for the received time of reference. In both techniques, the reconstructed signal data were processed using beamforming algorithm to produce power distributions at the frequency of interest. Experiments have been carried out for varying frequencies, rotating speeds and the object distances. Forward propagation method has shown better performance in locating source position than the backward propagation method.