• Journal of the Korean Society for Precision Engineering
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• v.21 no.5
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• pp.91-98
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• 2004

Human uses level difference and time difference to get space information. Therefore this paper shows that method to presume direction of sound source by time difference and to mark presumed position. The position means direction from geometrical center of sensors to the sound source. To get the time difference of microphones input level, we will be explained about arrangement of microphones which used for the sensor to take the sound signal. It is included distance among the 3 microphones and distance between microphones and sound source. Secondly, input signals are transmitted to CPU througth digital process. CPU is used to DSP(Digital Signal Processor) for manage the signal by real time. Finally, the position of sound source is perceived by an explained algorithm in this paper.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.1
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• pp.199-205
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• 2015

Indoor positioning technology has been developed very actively for the smart phone handheld by most users. Especially, many TDOA positioning systems using acoustic signal have been studied, and it estimates the smart phone position by measuring the distance between the smart phone speaker and the microphones which is installed to receive the acoustic signal from the smart phone, and by calculating the hyperbolic equations. But there are always errors for the distance measurements, and furthermore the microphone installation error produces huge position estimation error. In this paper, the position estimation error due to the position error of acoustic sensor in the 3 dimensional TDOA positioning system, is analyzed by PDOP simulation and experiment.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.05a
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• pp.964-968
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• 2007

The measurement of floor impact sound have been standardized in KS 2810-1 and 2. The height of receiving microphones position is specified in the standard as 1.2m which is almost half height of apartment rooms as a listening position. In this study, receiving positions are investigated by measuring the distribution of sound pressure levels at 792 receiving microphone positions in the standard testing building. Standard impact sources, tapping machine and impact ball, are driven on the center position in the source room where is located at the above floor. It was found that the distribution of sound pressure levels in the receiving room indicates significant deviation at different frequencies there is more than 5dB drop at 63Hz but 2dB rise at 125Hz at a height of 1.2m when the impact ball is driven, in the other case of a generating tapping machine there is more than 2dB rise at 125Hz at a height of 1.2m due to room modes.

• Journal of Institute of Control, Robotics and Systems
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• v.20 no.2
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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.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.22-28
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• 2001

This paper describes the implementation of a robust speaker position location system using the voice signal received by microphone array. To be robust to the reverberation which is the major factor of the performance degradation, low-frequency phase restoration algorithm which eliminates the influence of reverberations using the low-frequency information of the CPSP function is proposed. The implemented real-time system consists of a general purpose DSP (TMS320C31 of Texas instruments), analog part which contains amplifiers and filters, and digital part which is composed of the external memory and 12-bit A/D converter. In the real conference room environment, the implemented system that was constructed by the proposed algorithms showed better performance than the conventional system. The error of the TDOA estimation reduced more than 15 samples.

• Journal of Institute of Control, Robotics and Systems
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• v.21 no.12
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• pp.1185-1192
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• 2015

In this paper, a study of advanced sound source localization is conducted by eliminating the noise of the sound source using the discrete wavelet transform. And experiments are conducted to evaluate the performance of the proposed system that the mobile robot follows sound source stably. In addition, we compare the position estimation performance by applying a discrete wavelet transform to improve the reliability of the sound signal. The experimental results reveal that the de-nosing filter which removes the noise component in sound source can make the performance of position estimation more precisely and help the mobile robot distinguish the objective sound source clearly.

• New & Renewable Energy
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• v.5 no.2
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• pp.25-30
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• 2009

The wind tunnel test to identify the acoustic noise source position of the wind turbine blade was conducted in KARI low speed wind tunnel. Microphone array and time-domain beamforming methodology was applied to this study. To reduce the data processing time, a modified beamforming method with a criteria between calculation time step and grid size for rotating angle in the cylinderical coordinate system was proposed. The test results shows that the data processing time to identify the noise source position was reduced to 20% compared with conventional method. And the dominant noise source of the blade moves from inboard to blade tip as the frequency increases.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.05a
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• pp.837-842
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• 2004

The positions of rotating sound sources have been localized by experiments with the Doppler effects removed. In order to do-Dopplerize the sound signals emitted from moving sources, two kinds of signal reconstruction methods were applied. One is the forward propagation method and the other is the backward propagation method. Forward propagation method analyze the source emission time based on the instantaneous distance between sensors and the assumed source position, then the signals are reconstructed with respect to the emission time. On the other hand, the backward method uses time delay to do-Dopplerize the acquired data for the received time of reference. In both techniques, the reconstructed signal data were processed using beamforming algorithm to produce power distributions at the frequency of interest. Experiments have been carried out for varying frequencies, rotating speeds and the object distances. Forward propagation method has shown better performance in locating source position than the backward propagation method.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.20 no.12
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• pp.1216-1222
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• 2010

Based on the MacDonald's analytic model for the diffracted sound field of a semi-infinite noise barrier, computer simulations were performed for various positions of error microphones for an active noise barrier system. The simulation process also included the effects of floor reflections on both sides of the barrier. The results were also compared with Niu's simulation results and showed a straight line arrangement of sensors and actuators, in the order of primary source, secondary source and error microphone is better than over the top arrangement of the error microphones.

• Journal of Oral Medicine and Pain
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• v.8 no.1
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• pp.43-60
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• 1983

In order to obtain the basic data for diagnosis of occlusion the author investigated the occlusal sounds produced by mandibular closing movement from rest position to centric occlusion by using the Korotkoff sounds microphone electrodes and physiograph for Korean 20S without TMJ problems. And the author analyzed the correlation of the occlusal sounds with height, body weight, mandibular closing movement distance, maxillary sinus size and masseter muscle activities relating to the production of occlusal sounds. The obtained results were as follows : 1. The occlusal sounds prouduced by mandibular closing movements from rest position to centric occlusion were ranged from 3.08mV to 52.00mV, their maen value 19.07mV in right side, and ranged from 2.41mV to 18.33mV, their mean value 9.19mV in left side in group of Angel's class I occlusion subjects having right habitual side of mastication. 2. In general the occlusal sounds in the habitual side of mastication were greater than those in the opposite side. 3. The correlations of the occlusal sounds with height and maxillary sinus size were not significant. 4. The correaltions of the occlusal sounds with body weight, mandibular closing movement distance and masster muscle activities were significant.