• Title/Summary/Keyword: 어레이 마이크로폰

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Location Estimation Method of Steam Leak in Pipelines Using Leakage Area Analysis (누설영역 분석을 이용한 배관 증기누설 위치 추정 방법)

  • Kim, Se-Oh;Jeon, Hyeong-Seop;Son, Ki-Sung;Park, Jong Won
    • Journal of the Korean Society for Nondestructive Testing
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    • v.36 no.5
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    • pp.384-390
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    • 2016
  • It is important to have a pipeline leak-detection system that determines the presence of a leak and quickly identifies its location. Current leak detection methods use a acoustic emission sensors, microphone arrays, and camera images. Recently, many researchers have been focusing on using cameras for detecting leaks. The advantage of this method is that it can survey a wide area and monitor a pipeline over a long distance. However, conventional methods using camera monitoring are unable to target an exact leak location. In this paper, we propose a method of detecting leak locations using leak-detection results combined with multi-frame analysis. The proposed method is verified by experiment.

Characteristics of Wall Pressure over Wall with Permeable Coating (침투성 코팅 처리된 벽면 주위의 벽 압력 특성)

  • Song, Woo-Seog;Shin, Seung-Yeol;Lee, Seung-Bae
    • Transactions of the Korean Society of Mechanical Engineers B
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    • v.36 no.11
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    • pp.1055-1063
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    • 2012
  • Fluctuating wall pressures were measured using an array of 16 piezoelectric transducers beneath a turbulent boundary layer. The coating used in this experiment was an open-cell, urethane-type foam with a porosity of approximately 50 ppi. The ultimate objective of the coating is to provide a mechanical filter to reduce the wall pressure fluctuations. The boundary layer on the flat plate was measured by using a hot wire probe, and the CPM method was used to determine the skin friction coefficient. The wall pressure autospectra and streamwise wavenumber-frequency spectra were compared to assess the attenuation of the wall pressure field by the coating. The coating is shown to attenuate the convective wall pressure energy. However, the relatively rough surface of the coating in this investigation resulted in a higher mean wall shear stress, thicker boundary layer, and higher low-frequency wall pressure spectral levels compared to a smooth wall.

Noise Cancellation using Microphone Array in Digital Hearing Aids (디지털 보청기에서 마이크로폰 어레이를 이용한 잡음제거)

  • Bang, Dong-Hyeouck;Kil, Se-Kee;Kang, Hyun-Deok;Yoon, Gwang-Sub;Lee, Sang-Min
    • The Transactions of The Korean Institute of Electrical Engineers
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    • v.58 no.4
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    • pp.857-866
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    • 2009
  • In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.

Improvement Method and Experiment Analysis of Sniper Distance Estimation Using Linear Microphone Array (선형마이크로폰 어레이를 이용한 저격수 거리추정 개선방법과 실험 분석)

  • Jung, Seungwoo
    • Journal of the Korea Institute of Military Science and Technology
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    • v.21 no.4
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    • pp.447-455
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    • 2018
  • If a hidden enemy is shooting, there is a threat against soldiers in recent conflicts. This paper aims to improve the localization of a muzzle using microphone array. Gunshot noise can provide information about the location of muzzle with two signals, the muzzle blast from the gun barrel and the projectile sound from the bullet. Two signals arrive to the microphone array with different arrival time and angle. If the arrival angles of the two signals are estimated, distance between sniper location and the microphone array can be calculated by using geometric principles. This method was established in 2003 by Pare. But this method has a limitation that it cannot calculate the distance when the arrival angles of the two signals are same. Also it has an error when the angle difference of arrival is small. In order to overcome this limitation, a new method is proposed that uses the change of characteristic of the projectile sound with respect to vertical distance from the trajectory. The proposed method estimates the distance correctly when the arrival angle of two signals are same, and when the angle difference between two signals is increased, the estimation error increases with respect to the angle. Therefore these two methods can be selected according to the angle difference between two signals to estimate the distance of the muzzle. Below the threshold of the angle difference, the proposed method can be used to estimate distance with smaller error than the existing method. This was demonstrated by shooting tests using actual sniper rifles.

An Adaptive Microphone Array with Linear Phase Response (선형 위상 특성을 갖는 적응 마이크로폰 어레이)

  • Kang, Hong-Gu;Youn, Dae-Hui;Cha, Il-Hwan
    • The Journal of the Acoustical Society of Korea
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    • v.11 no.3
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    • pp.53-60
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    • 1992
  • Many adaptive beamforming methods have been studied for interference cancellation and speech signal enhancement in telephone conference and auditorium. Main aspect of adaptive beamforming methods for speech signal processing is different from radar, sonar and seismic signal processing because desire output signal should be apt to the human ear. Considering that phase of speech is quite insensible to the human ear, Sondhi proposed a nonlinear constrained optimization technique whose constraint was on the magnitude transfer function from the source to the output. In real environment the phase response of the speech signal affects the human auditorium system. So it is desirable to design linear phase system. In this paper, linear phase beamformer is proposed and sample processing algorithm is also proposed for real time consideration Simulation results show that the proposed algorithm yields more consistent beam patterns and deep nulls to the noise direction than Sondhi's.

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Experimental contribution analysis of external aeroacoustic noise sources to interior noise of automobile (자동차 외부 공기음향 소음원들의 실험적 실내 기여도 분석 기술 개발)

  • Lee, Myung Han;Ih, Kang Duck;Hwang, Seongil;Kim, Yong-Joe
    • The Journal of the Acoustical Society of Korea
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    • v.37 no.5
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    • pp.300-308
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    • 2018
  • The contribution analysis of various external aeroacoustic noise sources to interior noise is important, enabling to design an automobile with a low interior noise level. With a new technique, the CD (Cholesky Decomposition), it is proposed to decompose an overall interior noise spectrum into multiple spectra, each representing the contribution of a specific noise source to the interior noise. In order to validate this method, three kinds of experiments were conducted. Furthermore, it is proposed to improve the CD-based contribution analysis method to be integrated with existing exterior microphone arrays in the wind tunnel. This method was validated with an experiment with two speakers.

Estimation of Moving Target Trajectory using Optimal Smoothing Filter based on Beamforming Data (최적 스무딩 필터를 이용한 빔형성 정보 기반 이동 목표물 궤적 추정)

  • Jeong, Junho;Kim, Gyeonghun;Go, Yeong-Ju;Lee, Jaehyung;Kim, Seungkeun;Choi, Jong-Soo;Ha, Jae-Hyoun
    • Journal of the Korean Society for Aeronautical & Space Sciences
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    • v.43 no.12
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    • pp.1062-1070
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    • 2015
  • This paper presents an application of an optimal smoothing filter for moving target tracking problem based on measured noise source. In order to measure distance and velocity for the moving target, a beamforming method is applied to use the noise source by using microphone array. Also a Kalman filter and an optimal smoothing algorithm are adopted to improve accuracy of trajectory estimation by using a Singer target model. The simulation is conducted with a missile dynamics to verify performance of the optimal smoothing filter, and a model rocket is used for experiment environment to compare the trajectory estimation results between the beamforming, the Kalman filter, and the smoother. The Kalman filter results show better tracking performance than the beamforming technique, and the estimation results of the optimal smoother outperform the Kalman filter in terms of trajectory accuracy in the experiment results.