• 로봇학회논문지
• /
• 제3권2호
• /
• pp.131-136
• /
• 2008

This paper proposes a method to extract the personal information using a microphone array. Useful personal information, particularly customers, is age and gender. On the basis of this information, service applications for robots can satisfy users by offering services adaptive to the special needs of specific user groups that may include adults and children as well as females and males. We applied Gaussian Mixture Model (GMM) as a classifier and Mel Frequency Cepstral coefficients (MFCCs) as a voice feature. The major aim of this paper is to discover the voice source parameters of age and gender and to classify these two characteristics simultaneously. For the ubiquitous environment, voices obtained by the selected channels in a microphone array are useful to reduce background noise.

• 자동차안전학회지
• /
• 제12권4호
• /
• pp.70-77
• /
• 2020

In this study, we estimate the vehicle speed by analyzing the acoustic data recorded in a single microphone of a surveillance camera. The frequency analysis of the acoustic data corrects the Doppler effect, which is a characteristic of the moving sound source, and reflects the geometric relationship according to the location of the sound source and the microphone on the two-dimensional plane. The acoustic data is selected from the horn sound that is mainly observed in an urgent situation among various sound sources that may occur in a traffic accident, and the characteristics of the monotone source are considered. We verified the reliability of the proposed method by time domain acoustic analysis and actual vehicle evaluation. This method is effective and can be used for traffic accident analysis in the blind spot of the camera using a single microphone built into the existing surveillance camera.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
• /
• pp.102-105
• /
• 2000

This paper propose new sound localization algorithm that calculates TDOA(Time Difference Of Arrival) between the two received signals via two microphone array, The proposed Subband CPSP is a development of Previous CPSP method using subband approach. It first split the received microphone signals into three frequency bands and then calculates subband CPSP with corresponding SNR weights. This type of algorithm, Subband CPSP, can provide more accurate TDOA estimation results because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, computer simulation was conducted and it was compared with previous CPSP method. From the both simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than accuracy for TDOA estimation.

• The Journal of the Acoustical Society of Korea
• /
• 제24권3E호
• /
• pp.115-122
• /
• 2005

Two twin microphones may produce particular patterns of binaural directivity by time delays between twin microphones. The boundary element method (BEM) was used for the simulation of the sound pressure field around the head model in order to quantify the acoustic head effect. The sound pressure onto the microphone was calculated by the BEM to an incident sound pressure. Then a planar directivity pattern was formed by four sound pressure signals from four microphones. The optimal binaural directivity pattern may be achieved by adjusting time delays at each frequency while maintaining the forward beam pattern is relatively bigger than the backward beam pattern.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2011년도 춘계학술대회 논문집
• /
• pp.816-820
• /
• 2011

본 논문은 구형 마이크로폰 어레이의 부엽 레벨의 차를 증가시키기 위한 방법에 대한 연구 내용을 다루었다. 일반적인 어레이 신호처리에서 마이크로폰을 조밀하게 배치함으로써 어레이 응답에서의 주엽과 부엽 간의 차이를 늘릴 수 있고 어레이의 소음원 판별능력을 증가시킨다. 최근 사용되고 있는 상용 에레이들은 제작 단가와 어레이의 크기 때문에 센서의 수를 늘리는데 한계를 보이고 있다. 이런 문제를 극복하기 위해 본 연구에서는 MEMS 센서를 이용하여 구형 어레이에 적용하였다. 구형 마이크로폰 어레이를 이용한 시뮬레이션과 실험을 통해 정현파 소음원을 측정하였다. 실험을 위해 32 개의 일반 측정용 마이크로폰을 이용한 어레이와 85 개의 MEMS 마이크로폰을 이용한 구형 어레이를 제작하였다. 구형 조화 분해기법과 빔형성기법을 이용하여 측정 데이터를 분석하였다. 2 kHz 이상의 소음원에 대하여 MEMS 마이크로폰 어레이가 4 dB 이상의 부엽 저감 능력을 가지는 것을 확인하였다.

• 설비공학논문집
• /
• 제4권3호
• /
• pp.163-174
• /
• 1992

This paper presents numerical and experimental methods of investigating dynamic characterisics of reed valves of hermetic reciprocating compressors. For the natural frequency, two different techniques have been tried : microphone method and strain gage method. In the microphone method, acoustic pressure signals from excited valves have been analyzed, while signals of tiny strain gauge attached on the reed valves have been utilized in the strain gage method. The empirically determined natural frequencies have been compared to the ones calculated by finite element method. Reasonably good agreements between the experimental and numerical results have been found, implying that the natural frequencies of reed valves could be obtained by FFM alone with enough accuracy.

• 한국소음진동공학회논문집
• /
• 제25권9호
• /
• pp.628-636
• /
• 2015

In this study, theoretical and experimental analyses were performed for creating and moving the zone of quiet(ZoQ) to the ear location of a sitter by using active sound control technique. As the ZoQ is actively created at the location of the error microphone basically with an active sound control system using an algorithm such as the filtered-x least mean square(FxLMS), the virtual microphone control(VMC) method was considered to move the location of the ZoQ to around the sitter`s ear. A chair system with microphones and loudspeakers on both sides was manufactured for the experiment and thus an active headrest against the swept narrowband noise as the primary noise was implemented with a real-time controller in which the VMC algorithm was embedded. After the control experiment with and without the VMC method, the location variation of the ZoQ by analyzing the error signals measured by the error and the virtual microphones. Therefore, it is observed that the FxLMS with the VMC technique can provide the re-location of the ZoQ from the error microphone location to the virtual microphone location. Also it is found that the amount of the attenuation difference between the two locations was small.

• The Journal of the Acoustical Society of Korea
• /
• 제25권1E호
• /
• pp.25-31
• /
• 2006

Two twin microphones may produce particular patterns of binaural directivity by time delays between the twin microphones. The boundary element method (BEM) was used for the simulation of the sound pressure field around the KEMAR head model in order to quantify the acoustic head effect. The sound pressure onto the microphone was calculated by the BEM to an incident sound pressure. Then a planar directivity pattern was formed by four sound pressure signals from four microphones. The optimal binaural directivity pattern may be achieved by adjusting time delays at each frequency while maintaining the forward beam pattern is relatively bigger than the backward beam pattern. The simulation results were verified by the experimental measurement.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2005년도 춘계학술대회논문집
• /
• pp.164-169
• /
• 2005

Microphone array techniques are being used widely in wind tunnel measurements for identification of the distributed aerodynamic noise sources on the model being tested. Depending on the frequencies and sound levels, conventional beamforming algorithm has limitation in separating two adjacent sources. Several modifications to the classical beamforming have been developed to enhance way resolution and reduce sidelobe levels. In this Paper the robust adaptive beamforming and the CLEAN algorithm are used to compare to the result of conventional beamforming method. It is found that the CLEAN algorithm is capable of pin-pointing locations of multiple sources nearby, while these sources are unidentifiable with robust adaptive or conventional beamforming techniques.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2000년도 춘계학술대회논문집
• /
• pp.1675-1680
• /
• 2000

Recently the uncertainty has been made rapid progress in various fields of industry but the uncertainty measurement method of acoustical test (i.e. Insertion loss, Absorption ratio, Transmission loss etc,) hasn't been established. In this study, the uncertainty of measurement method for ducted silencers is carried out according to ISO 7235. The standard uncertainty factors are composed of sound pressure level, microphone sensitivity and pistonphone calibration in this measurement. Sound pressure level is type A evaluation of uncertainty, microphone sensitivity and pistonphone calibration are type B evaluation of uncertainty. The combined standard uncertainty is calculated by two type evaluation. The expanded uncertainty is expressed by the combined standard uncertainty multiply k value which is yield the effective degree of freedom.