• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.38 no.3
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• pp.258-263
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• 2010

Sound localization algorithm usually adapts three step process: sampling sound signals, estimating time difference of arrival between microphones, estimate location of sound source. To apply this process in indoor environment, sound localization algorithm must be strong enough against reverberant and noisy condition. Additionally, calculation efficiency must be considered in implementing real-time sound localization system. To implement real-time robust sound localization system we adapt four low cost condenser microphones which reduce the cost and total calculation load. And to get TDOA(Time Differences of Arrival) of microphones we adapt GCC-PHAT(Generalized Cross Correlation-Phase Transform) which is robust algorithm to the reverberant and noise environment. The position of sound source was calculated by using iterative least square algorithm which produce highly accurate position data.

• Journal of IKEEE
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• v.14 no.1
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• pp.47-54
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• 2010

In this study, we analyzed the reduction of noise in a noise environment using 2, 3, 4 or 5 microphones in digital hearing aids. In order to be able to use this in actual digital hearing aids, we made the experiment microphone set similar to the behind-the-ear type (BTE) and then recorded the signal accordingly, with each situation. With the recorded signals, we reduced the noise in each signal by a noise reduction algorithm using multi-microphones. As a result, in the case of By comparing the SNR (Signal to Noise Ratio) and PESQ (Perceptual Evaluation of Speech) measurements, before and after the noise reduction, the results showed that the improvement in performance was highest when three or four microphones were used. Generally, when two or more microphones were used, we found that as the number of microphones increased there was an increase in performance.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.509-517
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• 2013

In this paper, we researched a method that makes a good acoustic-speech system using a digital signal processing technique with dynamic microphone as a transducer. Good acoustic-speech system should deliver the original sound input to electric signal without distortion. By measuring the frequency response of the microphone, adjustment factors are obtained by comparing measured data and standard frequency response of microphone for each frequency band. The final sound levels are obtained using the developed adjustment factors of frequency responses from the microphone and speaker to match the original sound levels using the digital signal processing technique. Then, we minimize the changes in the frequency response and level due to the variation of the distance from source to microphone, where the frequency responses were measured according to the distance changes.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.04a
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• pp.816-820
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• 2011

A method for increasing the difference of side-lobe level in spherical microphone array is presented. In array signal processing, it is known that narrow interval between sensors can increase the difference between main lobe and side-lobe of array response which eventually increase the source recognition capability. Recent commercial array being used, however, have shown certain limitation in using the number of sensors due to its costs and geometrical size of array. To overcome this problem, we have adapted MEMS sensors into spherical microphone array. To check out the improvement, two different types of spherical microphone array were designed. One array is composed with 32 regular instrument microphones and the other one is 85 MEMS sensors. Simulation and experiments were conducted on a sinusoidal noise source with two arrays. The time history data were analyzed with spherical harmonic decomposition and beamforming technique. 85 MEMS sensors array showed the improved side-lobe level suppression by more than 4 dB above the frequency content of 2 kHz compared to 32-sensor array.

• The Journal of the Acoustical Society of Korea
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• v.41 no.1
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• pp.1-8
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• 2022

This paper identifies the cause of the high frequency calibration limit of the acoustic chamber type calibrator for microphone calibration and presents a method to improve it. By using a commercial finite element analysis software, we analyzed the calibration frequency limit of the acoustic chamber type calibrator through eigen-frequency and frequency domain analysis. Based on this, we designed and fabricated an acoustic chamber type calibrator that can precisely calibrate within 1 dB from about 2 Hz to 6.4 kHz and verified its performance through experiments. The acoustic chamber type calibrator fabricated through this study has the advantage of being able to calibrate multiple microphones simultaneously in a wide frequency range, so it can be usefully used for simple calibration for multiple microphones.

• Journal of rehabilitation welfare engineering & assistive technology
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• v.2 no.1
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• pp.27-33
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• 2009

Generally, implantable microphone has been implanted in the temporal bone for implantable middle ear hearing devices (IMEHDs). In this case, the microphone's membrane can be damaged and can be generated biological noise. In order to overcome the these problems, the location of implanted microphone should be changed. As an alternative, the microphone can be implanted in the external auditory canal. However, the sound emission can be produced because of vibration transducer toward reverse direction from the tympanic membrane to the external auditory canal. In this paper, an amount of the emitted sound is measured using a probe microphone as the changing the position of microphone in the external auditory canal of a physical ear model, which is similar to acoustical and vibratory properties of the human ear. Through the measured value, the location of the microphone was assumed in the external auditory canal. According to the analysis, the microphone input sound can be decreased when microphone position become more distance from the tympanic membrane in the auditory canal. However, the external auditory canal is not appropriated to implantable microphone position, because sound emission is not completely eliminated.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.2
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• pp.268-274
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• 2016

To find the location of a random noise source present in the three-dimensional space is required at least four microphones. Using four microphones distributed in a three-dimensional space, noise source imaging technique was applied and evaluated on their performance. To compensate resolution problem which comes from both the position of the sensor array is fixed and the sampling frequency is low, up-sampling technique and interpolation function were applied. Five different interpolation methods were applied such as zero-padding, zero-order hold, first-order hold, spline function, and random signal padding. The up-sampling rate were chosen by two, four, eight times, and counting up 16 times. As a result, it was possible to more accurately estimate the position of the noise source according to the higher of the up-sampling rate. It also found that the first-order hold and the spline function's performance were slightly falling relative to other methods.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.18 no.1
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• pp.90-97
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• 2004

In this paper, the optimal position selection of error microphones and control speakers to attenuate interior noise from outside noise sources using active noise control techniques is presented. To get an optimal control characteristics of adaptive noise control systems, it is necessary to optimize the positions of microphones and speakers. New type of simulated annealing method has been proposed as searching techniques to find optimal speakers and microphones positions in which the sum of the squared pressures at microphone positions in an enclosure can be best minimized. Computer simulations and experiments have been performed to show the effectiveness of the proposed method.

• Journal of the Korea institute for structural maintenance and inspection
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• v.15 no.3
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• pp.171-177
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• 2011

The feasibility of using low cost, unpowered, unshielded dynamic microphones is investigated for cost effective contactless sensing of impact-echo signals in concrete structures. Impact-echo tests on a delaminated concrete slab specimen were conducted and the results were used to assess the damage detection capability of the low cost system. Results showed that the dynamic microphone successfully captured impact-echo signals with a contactless manner and the delaminations in concrete structures were clearly detected as good as expensive high-end air pressure sensor based non-contact impact-echo testing.