• Transactions of the Korean Society for Noise and Vibration Engineering
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• 제18권6호
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• pp.612-618
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• 2008

Microphone array beamforming method has been recognized as an important aeroacoustic research field and become a standard technique in localizing sound sources. This method also used in flight acoustic measurement, and especially, it is very useful when measure sounds inside the wind tunnel. In measuring sound which is inside the wind tunnel by traditional beamforming method, there are some errors caused by airstream. The speed and the propagation path of the sound changes as it travel through the airstream. This makes the error which the position of sound is changed a little bit to the down stream direction. In this paper, validation test has made about the correction equation for this wind effects of previous researches. And beamforming including shear layer correction was performed about a sound source in the anechoic open-jet wind tunnel.

• Proceedings of the Acoustical Society of Korea Conference
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• 한국음향학회 1999년도 학술발표대회 논문집 제18권 1호
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• pp.332-335
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• 1999

The purpose of this study is to make clear that the relationship between the directional properties of late arriving sound and listener envelopment (LEV). Psycho-acoustical experiments are performed with an objective measure of inter-aural cross-correlation(IACC) in order In predict whether LEV is perceived equally in two kinds of sound fields with horizontal or vertical component of late arriving sound. It is found that LEV is affected by not horizontal component of late arriving sound but also vertical one.

• The Journal of the Acoustical Society of Korea
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• 제31권4호
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• pp.253-259
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• 2012

Goldingham's measurement of the velocity of sound undertaken in the early nineteenth century was the first large-scale measuring enterprise which considered various meteorological factors such as temperature, humidity, atmospheric pressure, the direction of the wind, etc. Goldingham's successful performance of measuring the velocity of sound by employing the sounds of cannons as sound source in Madras (now Chennai), a colonial region of India, for one and a half years was supported by material, institutional and social resources. As the official astronomer at the Madras Observatory, he was benefitted by the undemanding employment of accurate measuring instruments under the support of the Madras Army enabled him to gain reliable data and his reputation as professional experimentalist facilitated the acknowledgment of their trustworthiness.

• Journal of the Korean Society of Industry Convergence
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• 제26권2_1호
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• pp.225-230
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• 2023

This research addresses sound command based human tracking problems for autonomous cleaning mobile robot in a networked AI space. To solve the problem, the difference among the traveling times of the sound command to each of three microphones has been used to calculate the distance and orientation of the sound from the cleaning mobile robot, which carries the microphone array. The cross-correlation between two signals has been applied for detecting the time difference between two signals, which provides reliable and precise value of the time difference compared to the conventional methods. To generate the tracking direction to the sound command, fuzzy rules are applied and the results are used to control the cleaning mobile robot in a real-time. Finally the experiment results show that the proposed algorithm works well, even though the mobile robot knows little about the environment.

• KIISE Transactions on Computing Practices
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• 제21권7호
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• pp.443-451
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• 2015

This paper proposes a signal modelling method that can effectively detect the shock wave(SW) sound and muzzle blast(MB) sound from the gunshot of a small arm rifle. In order to localize a counter sniper in battlefield, an accurate detection of both shock wave sound and muzzle blast sound are the necessary keys in estimating the direction and the distance of the counter sniper. To verify the performance of the proposed algorithm, a real gunshot sound in a domestic military shooting range was recorded and analyzed. From the experimental results, the proposed signal modelling method was found to be superior to the comparative system more than 20% in a shock wave detection and 5% in a muzzle blast detection, respectively.

• Journal of Advanced Marine Engineering and Technology
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• 제41권1호
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• pp.83-90
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• 2017

In this paper, a sound reception system using a phase difference of whistle signals is proposed and analyzed based on spectral analysis. The proposed system receives whistle signals from four microphones installed in four different directions with 90-degree intervals between them. The proposed algorithm detects the phase of each received signal based on spectral analysis and estimates the direction of the whistle signal by obtaining the phase difference between the received signals from two adjacent microphones. Furthermore, we theoretically analyze the phase difference between two adjacent received signals according to their arrival angles and implement the proposed system using a DSP chip. In addition, the operation of the proposed algorithm are verified using the implemented system in a laboratory environment. Experimental results show that the proposed scheme can well estimate the direction of the whistle signal.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 한국정보통신학회 2012년도 춘계학술대회
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• pp.405-408
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• 2012

Recently, Smartphone takes center stage as the next generation game platform, so there's fierce competition between various companies which enter the mobile market. According to Smartphone market is wide, Smart phone App game sound is more widely. This study attempts to analyse the role of sound effects and BGM(background music) during game play or game development, and make a comparison between game play with and without Sound effects, so that to suggest the direction of development of the App game sound for Smartphone and to remind the importance of sound effects and emphasize the necessity of development of the professional game sound.

• JSTS:Journal of Semiconductor Technology and Science
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• 제14권2호
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• pp.175-183
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• 2014

In this paper, we present the design and implementation of real-time sound localization based on $0.13{\mu}m$ CMOS process. Time delay of arrival (TDOA) estimation was used to obtain the direction of the sound signal. The sound localization chip consists of four modules: data buffering, short-term energy calculation, cross correlation, and azimuth calculation. Our chip achieved real-time processing speed with full range ($360^{\circ}$) using three microphones. Additionally, we developed a dedicated sound localization circuit (DSLC) system for measuring the accuracy of the sound localization chip. The DSLC system revealed that our chip gave reasonably accurate results in an experiment that was carried out in a noisy and reverberant environment. In addition, the performance of our chip was compared with those of other chip designs.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 한국정보통신학회 2012년도 추계학술대회
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• pp.454-457
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• 2012

Recently, Smartphone takes center stage as the next generation platform, so there's fierce competition between various companies which enter the mobile market. According to Smartphone market is wide, Smart phone sound is more widely. in this paper, we analyze the necessity of sound desing tailored to the characteristics of mobile sources of sound that can be optimized for the iphone mobile production and mobile sources to meet the need for a sound design and the development direction of the sound, professional sound production suggest the need for.

• Proceedings of the KIEE Conference
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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• pp.878-881
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• 2003

To acquire an appointed speaker's clear voice signal from inspect-camera, picture-conference or hands free microphone eliminating interference noises needs to be preceded speaker's position automatically. Presumption of sound source position's basic algorithm is about measuring TDOA(Time Difference Of Arrival) from reaching same signals between two microphones. This main project uses ADF(Adaptive Delay Filter) [4] and CPS(Cross Power Spectrum) [5] which are one of the most important analysis of TDOA. From these analysis this project proposes presumption of real time sound source position and improved model NI-ADF which makes possible to presume both directions of sound source position. NI-ADF noticed that if auditory sense of humankind reaches above to some specified level in specified frequency, it will accept sound through activated nerve. NI-ADF also proposes practicable algorithm, the presumption of real time sound source position including both directions, that when microphone loads to some specified system, it will use sounds level difference from external system related to sounds of diffraction phenomenon. In accordance with the project, when existing both direction adaptation filter's algorithm measures sound source, it increases more than twice number by measuring one way. Preserving this weak point, this project proposes improved algorithm to presume real time in both directions.