• Title/Summary/Keyword: Microphone position

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Design of the broadband and compact phase-calibrator for array microphones (어레이 마이크로폰용 광대역 소형 위상교정기의 설계)

  • Ju, Hyeong-Sick;Kim, Yang-Hann
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2004.11a
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    • pp.1032-1035
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    • 2004
  • Pressure distribution is measured by way microphones to identify noise sources in the space. For example, beam-forming method or acoustic holography use phase information to identify the source. Therefore, the phase is significant information to correctly identify the source position. However, due to the microphone characteristics and measuring systems, measured signals always have errors, which make the identification difficult. Therefore, phase calibration of microphones is needed. Duct and speaker systems are generally used as calibrators. Acoustic characteristics of the calibrator are, of course, functions of many Parameters of the system: i.e. duct size, frequency, and microphone spacing. In this paper, design parameters which effect on the performance and size of the calibrators are considered. Then the parameters would be applied to design and real product of the phase-calibrator.

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A Study Absolute Position Estimation of Sound Source (3차원 음향홀로그래픽을 이용한 음원위치 추정에 관한 연구)

  • Kim, Chun-Duk;Sim, Dong-Youn;Jang, Bee;Lee, Chai-Bong;Cha, Kyung-Hwan
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.5
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    • pp.76-82
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    • 1997
  • The paper describes simulations and experimental results using a measuring system which utilizes the acoustic holographic method in order to exactly estimate an absolute position of a sound source. The measuring surface is installed to satisfy with a far field to the sound source and is composed of linear arrayed seven microphones. A measurement is simultaneously recorded by a reference microphone setting up a neighbour sound source and the linear arrayed seven microphones which are moved to the same interval. An absolute position of sound source is estimated by the cross-spectrum method to the received sounds between a reference and the measuring microphones. Phase differences of each microphone and time delays during scanning are compensated to the reference microphone and the measuring time of the first column. An optimal interval for each microphone in the measuring surface is decided by a numerical simulation. A source signal makes use of a sinusoid, and S/N ratio is 30dB in the experiment. The optimal microphone's interval in the simulation and the experiment is decided in order to satisfy with the Nyquist space sampling condition related to the wave length of 2kHz sinusoid. Mainlobe width of a estimated 3D hologram in the case of 2kHz source signal is decreased to 87% and 30% in comparison to 500Hz and 1kHz, and then a valid of simulation results is confirmed. Therefore, we verified a utilization of the study for a sound source estimation using 3ㅇ acoustic holographic method.

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A Speaker Detection System based on Stereo Vision and Audio (스테레오 시청각 기반의 화자 검출 시스템)

  • An, Jun-Ho;Hong, Kwang-Seok
    • Journal of Internet Computing and Services
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    • v.11 no.6
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    • pp.21-29
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    • 2010
  • In this paper, we propose the system which detects the speaker, who is speaking currently, among a number of users. A proposed speaker detection system based on stereo vision and audio is mainly composed of the followings: a position estimation of speaker candidates using stereo camara and microphone, a current speaker detection, and a speaker information acquisition based on a mobile device. We use the haar-like features and the adaboost algorithm to detect the faces of speaker candidates with stereo camera, and the position of speaker candidates is estimated by a triangulation method. Next, the Time Delay Of Arrival (TDOA) is estimated by the Cross Power Spectrum Phase (CPSP) analysis to find the direction of source with two microphone. Finally we acquire the information of the speaker including his position, voice, and face by comparing the information of the stereo camera with that of two microphone. Furthermore, the proposed system includes a TCP client/server connection method for mobile service.

Flight Path Measurement of Drones Using Microphone Array and Performance Improvement Method Using Unscented Kalman Filter (마이크로폰 어레이를 이용한 드론의 비행경로 측정과 무향칼만필터를 이용한 성능 개선법에 대한 연구)

  • Lee, Jiwon;Go, Yeong-Ju;Kim, Seungkeum;Choi, Jong-Soo
    • Journal of the Korean Society for Aeronautical & Space Sciences
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    • v.46 no.12
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    • pp.975-985
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    • 2018
  • The drones have been developed for military purposes and are now used in many fields such as logistics, communications, agriculture, disaster, defense and media. As the range of use of drones increases, cases of abuse of drones are increasing. It is necessary to develop anti-drone technology to detect the position of unwanted drones using the physical phenomena that occur when the drones fly. In this paper, we estimate the DOA(direction of arrival) of the drone by using the acoustic signal generated when the drone is flying. In addition, the dynamics model of the drones was applied to the unscented kalman filter to improve the microphone array detection performance and reduce the error of the position estimation. Through simulation, the drone detection performance was predicted and verified through experiments.

A study on the location of microphones in measurement considering the frequency characteristics of elevator noise in households (세대 내 승강기 소음 주파수특성을 고려한 측정 시 마이크로폰 위치에 관한 연구)

  • Min-Woo Kang;Yang-Ki Oh
    • The Journal of the Acoustical Society of Korea
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    • v.42 no.2
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    • pp.124-132
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    • 2023
  • When the building becomes high, the number of households increases and they are adjacent to the elevator. So, frequency of use of elevators will increase. Elevator noise is bound to increase in the future. However, there are currently no legal standards for elevator noise or measurement and evaluation methods that can clearly measure elevator noise in Korea. Although some methods for measuring elevator noise are presented in KS F ISO 16032, this standard is not a standard established for elevator noise. It is a standard that integrates the overall measurement method of building equipment and equipment, and the position of the microphone is selected by the experimenter during measurement. Elevator noise is characterized by a low sound pressure level as the noise in the mid-low frequency band is important. However, even today, complaints from residents about elevator noise are increasing. In this study, the position of the microphone that can most sensitively pick up the elevator noise when measuring the elevator noise was studied. According to the distance from the wall and the height from the floor, a total of 9 microphone positions were measured and analyzed. As a result of the experiment, it was confirmed that the elevator noise has a very high influence in the 63 Hz band. The measured value at the center point was identified as a factor that lowered the overall elevator noise level value.

An efficient space dividing method for the two-dimensional sound source localization (2차원 상의 음원위치 추정을 위한 효율적인 영역분할방법)

  • Kim, Hwan-Yong;Choi, Hong-Sub
    • The Journal of the Acoustical Society of Korea
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    • v.35 no.5
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    • pp.358-367
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    • 2016
  • SSL (Sound Source Localization) has been applied to several applications such as man-machine interface, video conference system, smart car and so on. But in the process of sound source localization, angle estimation error is occurred mainly due to the non-linear characteristics of the sine inverse function. So an approach was proposed to decrease the effect of this non-linear characteristics, which divides the microphone's covering space into narrow regions. In this paper, we proposed an optimal space dividing way according to the pattern of microphone array. In addition, sound source's 2-dimensional position is estimated in order to evaluate the performance of this dividing method. In the experiment, GCC-PHAT (Generalized Cross Correlation PHAse Transform) method that is known to be robust with noisy environments is adopted and triangular pattern of 3 microphones and rectangular pattern of 4 microphones are tested with 100 speech data respectively. The experimental results show that triangular pattern can't estimate the correct position due to the lower space area resolution, but performance of rectangular pattern is dramatically improved with correct estimation rate of 67 %.

Acoustic Characteristics of Watermelon According to Impact Conditions (타격조건에 따른 수박의 음파특성)

  • 최동수;최규홍;이영희;이강진;김만수
    • Journal of Biosystems Engineering
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    • v.27 no.1
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    • pp.67-76
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    • 2002
  • This study was conducted to investigate the effects of impact conditions on the acoustic characteristics of a watermelon. The study was crucial to develop a device for nondestructive internal quality evaluation of a watermelon by an acoustic impulse response method. An impact device was constructed with a pendulum to hit the watermelon, a microphone to detect the acoustic impulse responses, and a digital oscilloscope and computer to store and analyze the data. The selected samples were Guemcheon cultivar watermelons(Citrulus Vulgaris Schrad) harvested on Oct. 20,1998. Sixty watermelons were tested on flour different types of sample holders, with four kinds of ball made of different materials, at four bevels of the angular position of the pendulum and distance from the watermelon to the microphone. Since the magnitudes of frequencies obtained by hitting with the steel and rubber ball were relatively small at the bandwidths of above 500 Hz, it was shown that the steel and rubber ball were not suitable far a hitting ball in the pendulum to get informations on internal quality of the watermelon. In case of using broth of the wood and acryl ball, almost the same and good acoustic responses were shown on the wide range of frequency bandwidth. Therefore, it seemed that the acryl ball was more suitable to the test than the wood ball in considering its mechanical properties. The acoustic characteristics of the watermelon were not shown a significant difference between the types of sample holder. The amplitudes of the acoustic signals and the magnitudes of frequencies from the whole samples increased with increase of the angular position of pendulum and with decrease of the distance from the watermelon to the microphone. However, the resonance resonance of the sample were almost the same regardless of the angular positions and the distances.

Performance Enhancement of Whistle Sound Source Tracking Algorithm using Time-Scale Filter Based on Wavelet Transform

  • Moon, Serng-Bae
    • Journal of Navigation and Port Research
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    • v.28 no.2
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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.

A method to find the position of fault in a moving vehicle using microphone arrays (마이크로폰 어레이를 이용하여 차량 하부에서 발생한 결함의 위치를 찾아내는 방법)

  • Kim, Yang-Hann;Jeon, Jong-Hoon
    • Proceedings of the KSR Conference
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    • 2006.11b
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    • pp.144-151
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    • 2006
  • Sound generated from a moving vehicle often carries information on the condition of vehicle, for example, whether it has faults or not, where the fault exists. The latter is possible especially by MFAH(moving frame acoustic holography) and beamforming method. MFAH is applicable to the sound source of pure tone or narrow band noise. For the beamforming method, we have to know what kind of wave the sound source radiates, for example, plane wave or spherical wave. That is, whether the above methods are applicable depends on the characteristics of sound source. To apply these methods to the fault detection, we have to know the characteristics of wave from faults. In this research, a machine diagnosis technique based on the above holographic approaches is introduced to find the position of faults. The signal due to faults is modeled based on the fact that the faults radiate impulsive noise, and analyzed in time and frequency domain. The way how MFAH and beamforming method can be used is introduced to find the position of source.

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