• The Journal of the Acoustical Society of Korea
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• v.37 no.4
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• pp.206-214
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• 2018

This paper proposes an impact point estimation algorithm of the rifle using microphone sensors. The proposed algorithm resolves the time synchronization problem by expanding the existing ToA (Time of Arrival) method to TDoA (Time Difference of Arrival) method and verifies the performance of the algorithm through the actual shooting experiments. By comparing analysis of the actual and the estimated impact points by the algorithm, it is confirmed that the proposed algorithm has excellent performance by estimating the impact point accurately within the tolerance range.

• Journal of Mechanical Science and Technology
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• v.20 no.11
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• pp.1993-2001
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• 2006

The piezoelectric bimorph film, which, as an actuator, can generate more effective displacement than the usual PVDF film, is used to control the turbulent boundary-layer flow. The change of wall pressures inside the turbulent boundary layer is observed by using the multi-channel microphone array flush-mounted on the surface when actuation at the non-dimensional frequency $f_b^+$:=0.008 and 0.028 is applied to the turbulent boundary layer. The wall pressure characteristics by the actuation to produce local displacement are more dominantly influenced by the size of the actuator module than the actuation frequency. The movement of large-scale turbulent structures to the upper layer is found to be the main mechanism of the reduction in the wall- pressure energy spectrum when the 700$700{\nu}/u_{\tau}$-long bimorph film is periodically actuated at the non- dimensional frequency $f_b^+$:=0.008 and 0.028. The biomorph actuator is triggered with the time delay for the active forcing at a single frequency when a 1/8' pressure-type, pin-holed microphone sensor detects the large-amplitude pressure event by the turbulent spot. The wall-pressure energy in the late-transitional boundary layer is partially reduced near the convection wavenumber by the open-loop control based on the large amplitude event.

• Transactions of the Korean Society of Automotive Engineers
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• v.5 no.2
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• pp.162-172
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• 1997

We investigated passby noise measurement method in a small-sized semi-anechoic chamber satisfying the American based SAE J1470 Recommended Practice to facilitate the measurements. We have tired two passby noise prediction methods. One is line array microphone method in which the free space sound field is decomposed into its eigenfunctions in the spherical coordinates and rearranged according to the order of the spherical Hankel function. However, due to the characteristics of the spherical Hankel function, it is impossible to distinguish the function's characteristics according to the order in farfield. Consequently it can be applied in the transient region of the nearfield and the farfield. The other method is nearfield acoustic holography(NAH). Although measuring hologram for the several operational engine speeds by conventional scanning method is time-consuming work, we can greatly reduce the measuring time by selecting the appropriate engine speed through preexperimental knowledge. To verify this method we experimented with the outdoor passby noise measurements and the passby noise prediction in the small-sized semi-anechoic chamber for the identical passenger vehicle and obtained reasonable and acceptable results.

• Journal of Advanced Navigation Technology
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• v.10 no.4
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• pp.326-332
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• 2006

In this study, the antenna was proposed for the omni-directional characteristic in horizontal plane. Therefore we proposed $1{\times}4$ microstrip patch array on cylindrical surface for studying microstrip patch antennas. This antenna is designed for 2.45GHz ISM band and applications. This antenna can be applied to the base station of wireless microphone and access point of wireless LAN. The length and width of the patch antenna and the width of the feed line were calculated by using the theory of microstrip patch antenna, by using the both the 2.5D and 3D EM simulators the optimized antenna characteristics are obtained. From result of measured, antenna's impedance of coaxial waveguide port was 51.915-j3.688 ${\Omega}$, the return loss was -31dB and VSWR was 1.081.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.713-717
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• 2014

This paper presents a beamforming technique for locating impulsive sound source. The conventional frequency-domain beamformer is advantageous for localizing noise sources for a certain frequency band of concern, but the existence of many frequency components in the wide-band spectrum of impulsive noise makes the beamforming image less clear. In contrast to a frequency-domain beamformer, it has been reported that a time-domain beamformer can be better suited for transient signals. Although both frequency- and time-domain beamformers produce the same result for the beamforming power, which is defined as the RMS value of its output, we can use alternative directional estimators such as the peak value and crest factor to enhance the performance of a time-domain beamformer. In this study, the performance of three different directional estimators, the peak, crest factor and RMS output values, are investigated and compared with the incoherent interfering noise embedded in multiple microphone signals. The proposed formula is verified via experiments in an anechoic chamber using a uniformly spaced linear array. The results show that the peak estimation of beamformer output determines the location with better spatial resolution and a lower side lobe level than crest factor and RMS estimation in noise free condition, but it is possible to accurately estimate the direction of the impulsive sound source using crest factor estimation in noisy environment with stationary interfering noise.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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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.

• Smart Structures and Systems
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• v.29 no.6
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• pp.777-783
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• 2022

Even though several deep learning-based methods have been applied in the field of acoustic source localization, the previous works have only been conducted using the two-dimensional representation of the beamforming maps, particularly with the planar array system. While the acoustic sources are more required to be localized in a spherical microphone array system considering that we live and hear in the 3D world, the conventional 2D equirectangular map of the spherical beamforming map is highly vulnerable to the distortion that occurs when the 3D map is projected to the 2D space. In this study, a 3D deep learning approach is proposed to fulfill accurate source localization via distortion-free 3D representation. A target function is first proposed to obtain 3D source distribution maps that can represent multiple sources' positional and strength information. While the proposed target map expands the source localization task into a point-wise prediction task, a PointNet-based deep neural network is developed to precisely estimate the multiple sources' positions and strength information. While the proposed model's localization performance is evaluated, it is shown that the proposed method can achieve improved localization results from both quantitative and qualitative perspectives.

• Proceedings of the KSR Conference
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• 2011.10a
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• pp.670-677
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• 2011

In this research, noise of the Korean high-speed train running at different speed from 150 to 300 km/h was measured by using microphones and the difference of sound pressure according to its driving velocities was analyzed. In addition, sound pressure characteristics of front, pantograph, bogies and between-car sections of the Korean high-speed trains were deduced by applying delay and sum beam-forming method using a microphone array. From the research, the effects of locations of the high-speed train were investigated. The results of this research can be utilized to reduce noise of a high-speed train.

• ETRI Journal
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• v.32 no.6
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• pp.921-931
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• 2010

It has been shown that the principal subspace-based multichannel Wiener filter (MWF) provides better performance than the conventional MWF for suppressing interference in the case of a single target source. It can efficiently estimate the target speech component in the principal subspace which estimates the acoustic transfer function up to a scaling factor. However, as the input signal-to-interference ratio (SIR) becomes lower, larger errors are incurred in the estimation of the acoustic transfer function by the principal subspace method, degrading the performance in interference suppression. In order to alleviate this problem, a principal subspace modification method was proposed in previous work. The principal subspace modification reduces the estimation error of the acoustic transfer function vector at low SIRs. In this work, a frequency-band dependent interpolation technique is further employed for the principal subspace modification. The speech recognition test is also conducted using the Sphinx-4 system and demonstrates the practical usefulness of the proposed method as a front processing for the speech recognizer in a distant-talking and interferer-present environment.

• 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.