• 한국통신학회논문지
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• 제29권11C호
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• pp.1573-1580
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• 2004

본 논문은 실제 환경에서 고품질 음성 신호 취득을 위해, 적응 마이크로폰 어레이 시스템을 위한 적응 모드 컨트롤러를 제안한다. 적응 어레이 알고리즘의 올바른 동작을 위하여, 제안된 적응 모드 컨트롤러는 시간 축의 정보뿐만 아니라 공간 축의 정보를 함께 사용한다. 제안된 적응 모드 컨트롤러는 초기화 단계와 수행 단계의 두 단계로 나뉘어 동작되는데, 초기화 단계에서는 음원 위치 추정 기술이 사용되며, 수행 단계는 신호의 상관 관계 특성에 의해 동작한다. 적응 어레이 알고리즘으로는 적응 차단 행렬을 이용한 Generalized Sidelobe Canceller가 사용되었다. 제안한 적응 모드 컨트롤러는 적응 차단 행렬이 수렴되지 않은 경우에도 사용 가능하며, 기존의 전력비 방법에 비해 안정적인 성능을 나타낸다. 본 논문은 제안한 시스템을 실제 환경에서 평가하였으며, 2m 거리에 위치한 화자에 대해 13dB SINR 향상을 얻었다.

• 소음진동
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• 제11권1호
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• pp.57-67
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• 2001

Proposed in this paper is a method of measurement of the flow rate in a pipe. The sound waves which are propagated within a pipe are characterized by that the wavenumber in the axial direction is changed according to the flow rate, and these characteristics are used in the present method of measurement of the flow rate. The amount of change in wavenumber of sound waves according to the flow rate can be obtained from the relationship among acoustic pressure signals within a pipe, which are measured by using a microphone array. The flow rate can be obtained by using the amount of change in wavenumber of sound waves and the relational equation of the flow rate. With respect to errors that can occur during the measurement of the flow rate, the types of errors and the method of correction of those errors are presented. This method of measurement of the flow rate has application limitation conditions due to the sensor interval, assumption of sound waves as plane waves, etc. The numerical simulation and experiments for measuring the flow rate of air in a pipe are performed in order to verify the applicability of this method of measurement of the flow rate. The experimental results are shown to be similar to those of the numerical simulation. And the flow rate measured is shown to be consistent with the actual value within 5% error bound.

• The Journal of the Acoustical Society of Korea
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• 제23권1E호
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• pp.3-9
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• 2004

An important inverse problem in the field of acoustics is that of reconstructing the strengths of a number of sources given a model of transmission paths from the sources to a number of sensors at which measurements are made. In dealing with this kind of the acoustical inverse problem, strengths of the discretised source distribution can be simply deduced from the measured pressure field data and the inversion of corresponding matrix of frequency response functions. However, deducing :he solution of such problems is not straightforward due to the practical difficulty caused by their inherent ill-conditioned behaviour. Therefore, in order to overcome this difficulty associated with the ill-conditioning, the problem is replaced by a nearby well-conditioned problem whose solution approximates the required solution. In this paper a microphone array are identified for which the inverse problem is optimally conditioned, which can be robust to contaminating errors. This involves sampling both source and field in a manner which results in the discrete pressures and source strengths constituting a discrete Fourier transform pair.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2006년도 추계학술대회논문집
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• pp.763-766
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• 2006

The principal of a Maglev train is that floats on a magnetic field and is propelled by a linear induction motor. One of advantages is that it generates less noise compared to the wheel-on-rail train, because there are no wheels running along the rail. However, noises due to aero-dynamic disturbance and electrical system such as VVVF inverter and SLIM still occur. In this study, the Maglev cabin noises are measured during running and zero speed conditions. Pass-by noise measurements are performed to obtain outside noise during the operation of the train on the test tract. Data include a single microphone measurement as well as microphone array measurements. The array data are useful for sound source localization and more detailed noise reduction planning.

• 공학논문집
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• 제8권1호
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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.

• 한국항해항만학회지
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• 제28권2호
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• pp.135-140
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• 2004

A purpose of developing a sound source tracking system in this paper is to reduce the noise efficiently from the received signal by microphone array and measure the signal's time delay between the microphones. I have applied the wavelet analysis algorithm to the system and calculated the sound source's relative position For the performance evaluation, I have compared with the results of utilizing the digital filtering methods based on the FIR LPF using Kaiser window function and the inverse Chebyshev IIR LPF. As a result, I have confirmed the fact that 'time-scale' filter using inverse discrete wavelet transform was suitable for this system.

• 한국정밀공학회:학술대회논문집
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• 한국정밀공학회 2013년도 춘계학술대회 논문집
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• pp.457-458
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• 2013

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2013년도 춘계학술대회 논문집
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• pp.149-150
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• 2013

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2012년도 추계학술대회 논문집
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• pp.801-802
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• 2012

• 제어로봇시스템학회논문지
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• 제20권2호
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• pp.200-210
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• 2014

An active virtual impedance algorithm is newly proposed to track a sound source and to avoid obstacles while a mobile robot is following the sound source. The tracking velocity of a mobile robot to the sound source is determined by virtual repulsive and attraction forces to avoid obstacles and to follow the sound source, respectively. Active virtual impedance is defined as a function of distances and relative velocities to the sound source and obstacles from the mobile robot, which is used to generate the tracking velocity of the mobile robot. Conventional virtual impedance methods have fixed coefficients for the relative distances and velocities. However, in this research the coefficients are dynamically adjusted to elaborate the obstacle avoidance performance in multiple obstacle environments. The relative distances and velocities are obtained using a microphone array consisting of three microphones in a row. The geometrical relationships of the microphones are utilized to estimate the relative position and orientation of the sound source against the mobile robot which carries the microphone array. Effectiveness of the proposed algorithm has been demonstrated by real experiments.