• Journal of the Korea Society of Computer and Information
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• v.12 no.3
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• pp.259-266
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• 2007

The parameters of electric acoustic characteristic used as standards to evaluate the performance of a small microphone are composed of sensitivity, harmonic distortion, frequency response, directivity and others. Such characteristic parameters should be designed differently depending on a purpose, so it is important to verify whether a small microphone was made appropriately for the purpose after measuring the acoustic characteristics. Therefore, a system that can measure the acoustic characteristic parameters of a small microphone using DSP, not only simultaneously but also in real-time, was implemented in this paper. To verify the implemented system, four kinds of microphones were measured and the results were compared with the data values of the acoustic characteristics of each microphone. There were a little discrepancy between them because the conditions when measuring the characteristics were not identical. But it was verified that the errors are within the error tolerance and it proved that the system can be used in place of the conventional equipment used in measuring the acoustic characteristics of small microphones.

• The Journal of the Acoustical Society of Korea
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• v.26 no.2
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• pp.61-68
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• 2007

The frequency response of a microphone, which indicates the frequency range that a microphone can output within the approved level, is one of the most significant standards used to measure the characteristics of a microphone. At present, conventional methods of measuring the frequency response are complicated and involve the use of expensive equipment. To complement the disadvantages, this paper suggests a new algorithm that can measure the frequency response of a microphone in a simple manner. The algorithm suggested in this paper generates the Optimized Aoshima's Time Stretched Pulse(OATSP) signal from a computer via a standard speaker and measures the impulse response of a microphone by convolution the inverse OATSP signal and the received by the microphone to be measured. Then, the frequency response of the microphone to be measured is calculated using the signals. The performance test for the algorithm suggested in the study was conducted through a comparative analysis of the frequency response data and the measures of frequency response of the microphone measured by the algorithm. It proved that the algorithm is suitable for measuring the frequency response of a microphone, and that despite a few errors they are all within the error tolerance.

• The Journal of the Acoustical Society of Korea
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• v.39 no.1
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• pp.47-56
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• 2020

The sound source localization technique has various application fields in the era of internet-of-things, for which the probe size becomes critical. The localization methods using the acoustic intensity vector has an advantage of downsizing the layout of the array owing to a small finite-difference error for the short distance between adjacent microphones. In this paper, the acoustic intensity vector and the Time Difference of Arrival (TDoA) method are compared in the viewpoint of the localization error in the far-field. The comparison is made according to the change of spacing between adjacent microphones of the three-dimensional microphone array arranged in a tetrahedral shape. An additional test is conducted in the reverberant field by varying the reverberation time to verify the effectiveness of the methods applied to the actual environments. For estimating the TDoA, the Generalized Cross Correlation-Phase transform (GCC-PHAT) algorithm is adopted in the computation. It is found that the mean localization error of the acoustic intensimetry is 2.9° and that of the GCC-PHAT is 7.3° for T₆₀ = 0.4 s, while the error increases as 9.9°, 13.0° for T₆₀ = 1.0 s, respectively. The data supports that a compact array employing the acoustic intensimetry can localize of the sound source in the actual environment with the moderate reflection conditions.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1993.04a
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• pp.123-127
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• 1993

소음방사의 이해 및 효과적인 소음제어를 위해서는 소음원의 특성, 음장의 공간상 방사 특성 등을 아는 것이 중요하며, 이를 위해 많은 연구가 진행되 어 왔다. 특히 다수의 마이크로폰 어레이를 이용한 음향 홀로그래피 방법에 의한 실험적 음장 예측 방법이 소개되었고 연구가 진행됨에 따라 많은 실용 가능성을 보여 주었다. 음향 홀로그래피 방법에는 측정상 제한이 필연적으로 존재할 수밖에 없는데, 이에 따른 오차가 존재하며 결국 예측음장의 신뢰도 를 떨어뜨리는 요인이 된다. 본 연구의 목적은 측정조건에 따른 오차의 요인 을 고찰하고 이를 정량적으로 표현함으로써 음향 홀로그래피 방법의 적용에 도움을 주고자 한다. 평면 음향 홀로그래피에 나타나는 오차는 둘러 싸기 오 차(wraparound error), 앨리애싱(aliasing), 창문영향(window effect)으로 나 눌 수 있는데, 오차는 측정구경의 크기와 마이크로폰 사이의 간격등의 측정 조건 뿐만 아니라 음원의 특성, 홀로그램 평면의 위치 등에 직접적인 영향을 받게 된다. 본 연구에서는 오차해석을 위한 기본 연구로써 점음원(monopole) 과 쌍극자(dipole)음장의 파수 스펙트럼을 해석적으로 구하고 이를 기본으로 평면 음향 홀로그래피 적용시 존재하는 앨리애싱에 대해 고찰하고 전산기 모의 실험 (computer simulation)을 통해 오차를 최소화하는 측정조건을 제 시하고자 한다.

• 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.

• Journal of KIISE
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• v.43 no.1
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• pp.13-26
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• 2016

This paper proposes a new sound localization algorithm that can improve localization performance based on a tetrahedron-shaped microphone array. Sound localization system estimates directional information of sound source based on the time delay of arrival(TDOA) information between the microphone pairs in a microphone array. In order to obtain directional information of the sound source in three dimensions, the system requires at least three microphones. If one of the microphones fails to detect proper signal level, the system cannot produce a reliable estimate. This paper proposes a tetrahedron- shaped sound localization system with a coordinate transform method by adding one microphone to the previously known triangular-shaped system providing more robust and reliable sound localization. To verify the performance of the proposed algorithm, a real time simulation was conducted, and the results were compared to the previously known triangular-shaped system. From the simulation results, the proposed tetrahedron-shaped sound localization system is superior to the triangular-shaped system by more than 46% for maximum sound source detection.

• Proceedings of the Korea Information Processing Society Conference
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• 2012.04a
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• pp.950-952
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• 2012

음원 위치추정 시스템은 일반적으로 여러 개의 마이크에서 수집된 음원의 시간 간격을 이용해 음원의 위치를 추정하는 방식을 적용한다. 본 논문에서는 감시카메라에 적합한 4개의 마이크로폰을 이용한 음원 위치추정 시스템에서 마이크로폰에 수신된 음원의 도착순서를 이용해 음원의 위치를 추정하는 알고리즘을 제안하였다. 제안한 알고리즘을 시뮬레이션 프로그램을 통해 검증한 결과, 음원 추정각도의 오차는 $2^{\circ}{\sim}11.25^{\circ}$로 확인되었으며, 이는 실제각도의 오차범위인 $0^{\circ}{\sim}22.5^{\circ}$ 내에 해당하기 때문에 추정각도의 오차가 최대로 발생하더라도 음원이 발생한 위치를 파악 할 수 있음을 의미한다.

• The Journal of the Acoustical Society of Korea
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• v.13 no.3
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• pp.41-50
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• 1994

We studied an inherent error be caused by a measuring acoustic intensity using probe which can measure simultaneously the three-dimensional acoustic intensity. This three-dimensional intensity probe was constructed with four microphones, proposed by Suzuki et al. . In the computer simulation, we analyzed the nearfield measurement error with arbitary direction and each of axis direction on the ideal point source and the plate sound source which have finite size. From the results, in case of point source, we obtained accurate measurement below about 1dB when the distance of measurement was about 2.5 times with the distance among microphones in this probe. And in the case of plate sound source, the nearfield measurement error was decreased as the length of one side became above 0.02m, we obtained accurate measurement below about 1dB when the length of one side is 0.2m. The nearfield measurement error of finite size sound is small to ignore. Therefore this probe is useful to measure nearfield intensity.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.289-292
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• 1999

다수의 마이크로폰 어레이를 사용하여 소음원에서 방사하는 음장을 예측, 가시화하고 소음원의 시, 공간적 특성을 파악하기 위하여 음향 홀로그래피 방법에 대한 연구를 수행하였다. 음향 홀로그래피 방법은 실험적으로 소음원의 특성을 규명할 수 있기 때문에 많은 연구가 활발히 진행되고 있지만, 많은 개수의 마이크로폰과 신호수집장치 등이 필요하기 때문에 그 사용에 많은 제약이 있어 왔다. 음향 홀로그래피 방법중에서 대표적인 평면 음향 홀로그래피 방법을 중심으로 마이크로폰의 개수, 간격등과 같은 측정조건과 함께 마이크로폰을 스캐닝하는 방법둥에 대한 해석을 통하여 장, 단점 및 제한성을 논하였다. 또한 이러한 측정방법에서 나타나는 오차요인을 해석하고 이를 보정하는 방법에 대한 설명과 함께 실험을 통하여 이를 확인해 보았다.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.40-50
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• 1994

In this paper, the sensitivity compensation method for three-dimensional acoustic intensity probe in the higher frequency range has been studied. The measurement error in the higher frequency range is generated from the phase mismatch between microphone's signals of the probe. If the wavelength of sound signal measured is less than those of the distance between microphones of the probe, that is, the higher frequency of the sound signal, the bigger measurement error is generated. In this study, we proposed the compensation methods for one-dimensional acoustic intensity probe with two-microphones, and the efficiency of those methods were investigated by numerical calculation of computer. It was most effective method to compensate the phase mismatch between microphone for the acoustic intensity probe was investigated for the sound estimated. and the efficiency of this method in a three-dimensional probe was investigated for the sound wave travelling in the arbitrary direction by numerical calculation of computer. In this result, the efficiency was proved that, for the measurement error of 1dB or less with the three-dimensional probe of 60mm space, the frequency should be less than 1.2kHz without the error compensation method, but the frequency increased up to 2.8kHz with the error compensation method.