• The Journal of the Acoustical Society of Korea
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• 제18권2E호
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• pp.10-17
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• 1999

The aim of the work described in this paper is to develop a complex underground acoustic system which detects and localizes the origin of an underground hammering sound using an array of hydrophones located about 100m underground. Three different methods for the sound localization will be presented, a time-delay method, a power-attenuation method and a hybrid method. In the time-delay method, the cross correlation of the signals received from the array of sensors is used to calculate the time delays between those signals. In the power-attenuation method, the powers of the received signals provide a measure of the distances of the source from the sensors. In the hybrid method, both informations of time-delays and power-ratios are coupled together to produce better performance of position estimation. A new acoustic imaging technique has been developed for improving the hybrid method. This new acoustic imaging method shows the multi-dimensional distribution of the normalized cost function, so as to indicate the trend of the minimizing direction toward the source location. For each method the sound localization is carried out in three dimensions underground. The distance between the true and estimated origins of the source is 28m for a search area of radius 250m.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 한국소음진동공학회 2004년도 추계학술대회논문집
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• pp.911-915
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• 2004

Recently, several experimental techniques for identifying the noise sources distributed over a moving vehicle are being developed and used in order to design a low noise vehicle. The beamforming method, which uses phase information between several microphones to localize the source position, is proved to be one of the promising techniques applicable even under complicated test environments. In this study a beamforming algorithm is developed and applied to measure the dominant noise sources on a passenger car moving at constant speed. Unlike the acoustic signals from a stationary noise source, the sound generated from a moving source is distorted due to the Doppler effects. The sound pressure are measured with an spiral array system composed of 26 microphones and a pair of photo sensors are used to measure the. vehicle speed. The information about the speed and relative position of the vehicle are used to eliminate the Doppler effects from the measured pressure signal by using a de-Dopplerization algorithm. The noise generated from a moving vehicle can be grouped in many ways, however, tire noise and the noise generated from the engine are distinguishable at the speeds being tested.

• The Journal of the Acoustical Society of Korea
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• 제39권1호
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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.

• The Journal of the Acoustical Society of Korea
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• 제37권6호
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• pp.412-421
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• 2018

The Line array sonar consisting of several hydrophones increases array gain and improves the performance for detecting the direction of the target compared to single hydrophone. However, line array sonar produces the bearing error that makes it difficult to determine the bearing of incoming source signal due to the relation between bearing angle of target and vertical angle of multipath signals. Vertical angles of multipath are varied with the geometry of receiver and target and various underwater environments, therefore it is necessary to consider the bearing error to estimate accurately the bearing of the target. In this study, acoustic modelling was performed to understand the effect of multipath signals on the target signal. The errors of bearing angle estimated from the bottom bounced signals are calculated with several environment. In addition, the expected bearing line, as a function of source-receiver range, compensated for the bearing error is predicted from the estimated bearing angle.

• The Journal of the Acoustical Society of Korea
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• 제21권8호
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• pp.686-694
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• 2002

This paper proposes an algorithm for estimating positions of array elements placed on a sea floor using acoustic signal in multipath ocean environment. The positions of array elements are estimated by using the travel times of m-sequence signal influenced by the multi-paths environment. The horizontal distance between source and receiver calculated based on the ray model. The proposed paper the algorithm is verified by both simulation data and field experiment in the Bast Sea.

• Journal of the Korea Society of Computer and Information
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• 제6권1호
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• pp.51-59
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• 2001

In this paper, performance of the double circular array antenna was analyzed, which has sharp directivity with specific direction and is compatible to shape pencil beam. The character of each array geometry which N numbers of isotropic point source is arranged in planar with type of a square, circular and double circular was compared and analyzed. As a result, we fined that double circular geometry was good performance rather than square and circular array geometry in the point of directivity. SLL and HPBW.

• 한국신재생에너지학회:학술대회논문집
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• 한국신재생에너지학회 2005년도 춘계학술대회
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• pp.127-131
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• 2005

The paper addresses modeling and analysis of a grid-connected photovoltaic generation system (PV system). PSCAD/EMIDC. an industry standard simulation tool for studying the transient behavior of electric power system and apparatus. is used to conduct all aspects of model implementation and to carry out extensive simulation study. An equivalent circuit model of a solar cell has been used for modeling solar array. A PWM voltage source inverter (VSI) and its current control scheme have been implemented. A maximum power point tracking (MPPT) technique is employed for drawing the maximum available energy from the PV array. Comprehensive simulation results are presented to examine PV array behaviors and PV system control performance in response to irradiation changes. In addition, dynamic responses of PV array and system to network fault conditions are simulated and analysed

• Journal of Sensor Science and Technology
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• 제25권3호
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• pp.189-195
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• 2016

In this paper, we newly introduce a visible light communication (VLC) repeater using the LED array in an LED lamp. The LED array is used for a light source in the repeater, which radiates light both for illumination and data transmission. A VLC repeater is made by adding some electronic circuits to the LED array including a photodetector, a demodulator, and a modulator. The repeater is installed on the ceiling of a room like a conventional LED lamp, receives the VLC signal from an arbitrary transmitter, recovers data, and radiates the signal to wide area in the room. We used a carrier frequency of 100 kHz for the uplink from a transmitter to the repeater, and 500 kHz for the downlink from the repeater to a receiver. The repeater is useful for increasing the transmission path to wide area over the obstacles that may exist between VLC transmitters and receivers.

• Phonetics and Speech Sciences
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• 제4권4호
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• pp.87-92
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• 2012

Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.

• The Journal of Korea Robotics Society
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• 제3권2호
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• pp.131-136
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• 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.