• 유체기계공업학회:학술대회논문집
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• 유체기계공업학회 2002년도 유체기계 연구개발 발표회 논문집
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• pp.123-128
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• 2002

Over the range(OTR) has been applied for cooking and ventilation functions especially in northern Amarica. Because flow rate and operating rpm of the double sided sirocco fan for ventilation are much higher and than the microwave oven system, the major noise source is the sirocco fm. Recently, the quiet fan development is one of very important issues for amenity. In this study, the noise source identification using multi-microphone array system was carefully carried out and numerical simulations for understanding the aerodynamic and aeroacoustic of the fan were peformed. The sound level of tonal noise is predicted with a good accuracy but that of the broadband shows some discrepancy. In order to reduce the broadband noise, the inlet region of sirocco fan have to be modified that the secondary flow should be suppressed. Based on these results, new low noise fan and OTR is now developing.

• Smart Structures and Systems
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• 제17권1호
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• pp.1-15
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• 2016

Rayleigh wave velocity along a straight survey line on a concrete plate is measured in order to compare different non-destructive data acquisition techniques. Results from a rolling non-contact data acquisition system using air-coupled microphones are compared to conventional stationary accelerometer results. The results show a good match between the two acquisition techniques. Rolling measurements were found to provide a fast and reliable alternative to stationary system for stiffness determination. However, the non-contact approach is shown to be sensitive to unevenness of the measured surface. Measures to overcome this disadvantage are discussed and demonstrated using both forward and reverse rolling measurements.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 추계학술대회논문집
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• pp.911-915
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• 2004

Recently, several experimental techniques for identifying the noise sources distributed over a moving vehicle are being developed and used in order to design a low noise vehicle. The beamforming method, which uses phase information between several microphones to localize the source position, is proved to be one of the promising techniques applicable even under complicated test environments. In this study a beamforming algorithm is developed and applied to measure the dominant noise sources on a passenger car moving at constant speed. Unlike the acoustic signals from a stationary noise source, the sound generated from a moving source is distorted due to the Doppler effects. The sound pressure are measured with an spiral array system composed of 26 microphones and a pair of photo sensors are used to measure the. vehicle speed. The information about the speed and relative position of the vehicle are used to eliminate the Doppler effects from the measured pressure signal by using a de-Dopplerization algorithm. The noise generated from a moving vehicle can be grouped in many ways, however, tire noise and the noise generated from the engine are distinguishable at the speeds being tested.

• 한국음향학회지
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• 제20권1호
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• pp.36-42
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• 2001

본 연구는 폐쇄된 사무 공간상에서 2개의 마이크로폰을 이용하여 임의의 위치에서 발생한 음성 및 음향의 방향성 (방향각)을 추적하는 새로운 알고리듬을 제안한다. 본 논문에서 제안한 Subband CPSP (Cross Power Spectrum Phase) 알고리듬은 기존의 CPSP 알고리듬을 개선한 것으로서, 마이크로폰에 수신된 2개의 입력 신호에 대해 서브밴드 필터 뱅크를 이 용하여 대역 분할하고 각 서브밴드 대역에서 구해지는 대역별 CPSP 결과의 평균값을 제공한다. 이러한 주파수 대역 분할방식은 잡음의 영향을 각 대역으로 한정 분산시켜 사무 공간내 잡음의 영향을 각 대역으로 한정하여 음원의 방향각 계산시 발생하는 오차를 최소화할 수 있는 보다 강인하고 정확한 음원 추적 시스템을 가능하게 한다. 제안된 알고리듬의 성능을 입증하기 위해 기존의 CPSP 와 Subband CPSP 알고리듬의 실시간 음원 추적 실험을 수행하였으며, 실험 결과 제안된 Subband CPSP가 CPSP에 비해 평균 5％ 이상의 성능 향상을 가져옴을 확인할 수 있었다.

• ETRI Journal
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• 제32권6호
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• pp.921-931
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• 2010

It has been shown that the principal subspace-based multichannel Wiener filter (MWF) provides better performance than the conventional MWF for suppressing interference in the case of a single target source. It can efficiently estimate the target speech component in the principal subspace which estimates the acoustic transfer function up to a scaling factor. However, as the input signal-to-interference ratio (SIR) becomes lower, larger errors are incurred in the estimation of the acoustic transfer function by the principal subspace method, degrading the performance in interference suppression. In order to alleviate this problem, a principal subspace modification method was proposed in previous work. The principal subspace modification reduces the estimation error of the acoustic transfer function vector at low SIRs. In this work, a frequency-band dependent interpolation technique is further employed for the principal subspace modification. The speech recognition test is also conducted using the Sphinx-4 system and demonstrates the practical usefulness of the proposed method as a front processing for the speech recognizer in a distant-talking and interferer-present environment.

• 제어로봇시스템학회논문지
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• 제21권12호
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• pp.1185-1192
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• 2015

In this paper, a study of advanced sound source localization is conducted by eliminating the noise of the sound source using the discrete wavelet transform. And experiments are conducted to evaluate the performance of the proposed system that the mobile robot follows sound source stably. In addition, we compare the position estimation performance by applying a discrete wavelet transform to improve the reliability of the sound signal. The experimental results reveal that the de-nosing filter which removes the noise component in sound source can make the performance of position estimation more precisely and help the mobile robot distinguish the objective sound source clearly.

• 신재생에너지
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• 제5권2호
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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.

• 대한임베디드공학회논문지
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• 제8권4호
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• pp.219-225
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• 2013

This paper presents a method using Maximum-Likelihood Linear Regression (MLLR) adaptation to improve recognition performance of Limabeam algorithm for speech recognition using microphone array. From our investigation on Limabeam algorithm, we can see that the performance of filtering optimization depends strongly on the supporting optimal state sequence and this sequence is created by using Viterbi algorithm trained with HMM model. So we propose an approach using MLLR adaptation for the recognition of speech uttered in a new environment to obtain better optimal state sequence that support for the filtering parameters' optimal step. Experimental results show that the system embedded with MLLR adaptation presents the word correct recognition rate 2% higher than that of original calibrate Limabeam and also present 7% higher than that of Delay and Sum algorithm. The best recognition accuracy of 89.4% is obtained when we use 4 microphones with 5 utterances for adaptation.

• 한국철도학회:학술대회논문집
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• 한국철도학회 2007년도 춘계학술대회 논문집
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• pp.1590-1597
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• 2007

This paper deals with the noise source localization of the Korean High Speed Train (KTX) at the speed of 300 km/h. Using Microfonarray system and beamforming technology typical pass-by noise sources and their frequency characteristics are investigated. It is primarily aimed at investigating the location and characteristics of the high speed train emission. The results from the microphone array tests are also analyzed in relation to the remarks from analytic studies and experimental investigations on the high speed train that have been done with the intention of understanding the interior noise mechanism. The acoustical image shows the low frequency noise sources mainly at the position of the under part of the train at high speeds and the related source mechanism are discussed.

• Journal of Ship and Ocean Technology
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• 제8권3호
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• pp.18-25
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• 2004

Sound reception system is required to detect the sound and the quadrantal direction of the other ship's horn sound, to overcome the effects of enclosed wall for navigation space, functioning as a sound barrier. However, the realized systems can only provide quadrantal information of the other ship. This paper presents a new arrangement of microphones, having geometrically symmetric deployment with the same distances between sensors and the same angles between adjacent sensors with respect to the geometrical center. The sound pressures received at microphones are transformed into the related envelope signals by applying Hilbert transform. The time delays between microphones are estimated by the correlation functions between the derived envelope signals. This envelope base processing mitigates the noises related to the reflection by ship and sea surface. Then, the directional information is easily defined by using the estimated time delays. The suggested method is verified by the generated signals using boundary element method for a small ship model with sea surface wave. The estimated direction is quite similar to the true one and therefore the proposed approach can be used as an efficient sound reception system.