• Journal of the Korean Society for Precision Engineering
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• v.20 no.5
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• pp.116-123
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• 2003

These days, the researches related with the emotional robots are actively investigated and in progress. And human language, expression, action etc. are merged in the emotional robot to understand the human emotion. However, there are so many sound sources and background noise around the robot, that the robots should be able to separate the mixture of these sound sources into the original sound sources, moreover to understand the meaning of voice of a specific person. Also they should be able to turn or move to the direction of a specific person to observe his expression or action effectively. Until now, the researches on the localization and separation of sound sources have been so theoretical and computative that real-time processing is hardly possible. In this reason for the practical emotional robot, fast computation should be realized by using simple principle. In this paper the methods for detecting the direction of sound sources by using the phase difference between peaks on spectrums, and the separating the sound sources by using fundamental frequency and its overtones of human voice, are proposed. Also by using these methods, it is shown that the effective and real-time localization and separation of sound sources in living room are possible.

• Journal of Ship and Ocean Technology
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• v.8 no.3
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• pp.18-25
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• 2004

Sound reception system is required to detect the sound and the quadrantal direction of the other ship's horn sound, to overcome the effects of enclosed wall for navigation space, functioning as a sound barrier. However, the realized systems can only provide quadrantal information of the other ship. This paper presents a new arrangement of microphones, having geometrically symmetric deployment with the same distances between sensors and the same angles between adjacent sensors with respect to the geometrical center. The sound pressures received at microphones are transformed into the related envelope signals by applying Hilbert transform. The time delays between microphones are estimated by the correlation functions between the derived envelope signals. This envelope base processing mitigates the noises related to the reflection by ship and sea surface. Then, the directional information is easily defined by using the estimated time delays. The suggested method is verified by the generated signals using boundary element method for a small ship model with sea surface wave. The estimated direction is quite similar to the true one and therefore the proposed approach can be used as an efficient sound reception system.

• Journal of Convergence for Information Technology
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• v.10 no.8
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• pp.53-59
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• 2020

This paper relates to an IoT device system that detects sound source and estimates the sound source location. More specifically, it is a system using a sound source direction detection device that can accurately detect the direction of a sound source by analyzing the difference of arrival time of a sound source signal collected from microphone sensors, and track the generation direction of a sound source using an IoT sensor. As a result of a performance test by generating a sound source, it was confirmed that it operates very accurately within 140dB of the acoustic detection area, within 1 second of response time, and within 1° of directional angle resolution. In the future, based on this design plan, we plan to commercialize it by improving the reliability by reflecting the artificial intelligence algorithm through big data analysis.

• The Journal of the Acoustical Society of Korea
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• v.34 no.5
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• pp.351-359
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• 2015

Typical underwater acoustic transducers detect only the magnitude of an acoustic pressure and they have the limitation of not being able to recognize the direction of the sound signal. Hence, the authors of this paper proposed a new vector sensor structure based on Tonpilz transducers that could detect both the magnitude and the direction of a sound pressure. In the proposed structure, the piezoceramic ring was divided into four segments, and proper combination of the output voltages of the segments in response to the external sound pressure could provide the information on the orientation of the sound source. In this paper, a Tonpilz transducer has been fabricated to have the proposed structure and its characteristics has been measured to confirm the validity of the proposed structure.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.11a
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• pp.497-502
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• 2008

In this paper, we studied on an attenuation effect of automobile exhaust noise according to the direction of secondary sound source in duct ANC system. Automobile exhaust noise was recorded at 800rpm. 3500rpm and 5000rpm of a diesel engine. Directions of loudspeaker(second sound source) can be exchanged to $30^{\circ}$, $90^{\circ}$ and $150^{\circ}$ against the primary noise flow by acrylic ducts to be made for experimentation. DSP board with TMS320C6416 chip of Texas Instrument Co used to control adaptive ANC system. This ANC system is based on the single-channel FxLMS algorithm. In experiment result, when the loud speaker direction was $150^{\circ}$, the attenuation effect showed largely. In case of $90^{\circ}$ duct, the noise was a little increased. In case of $30^{\circ}$ duct, the noise was a little increased or decreased according to the frequency range and the sound pressure(dB) of exhaust noise to comply with engine rpm.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.21 no.6
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• pp.521-528
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• 2011

A variety of noise control methods have been developed as an interest on noise issues increases. Among them, noise control methods using masking effect, a phenomenon to reduce the ability to notice the unwanted sound by proper sound, to implement a pleasant sound environment have been studied under the name of soundscape. We proposed a novel vector base amplitude panning(VBAP) based noise control method to apply to the building environment. The proposed method could improve the amenity inside the building to reproduce the sounds with excellent masking effect on the incoming path of noise using the control speakers, considering the direction of noise source. The directional masking sounds can be generated by using VBPA technique. To verify the performance of the proposed method, we carried out the subjective test for the degree of amenity according to direction of the masking sound. Subjective test results showed that it is possible to improve the amenity inside the building by controlling the direction of masking sound considering the human's auditory characteristic.

• The Journal of the Acoustical Society of Korea
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• v.30 no.3
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• pp.115-122
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• 2011

In this paper, we present a method to estimate an accurate real-time sound source direction based on time delay of arrival by using generalized cross correlation with four cross-type microphones. In general, existing systems have two disadvantages such as system embedding limitation due to the necessity of data acquisition for signal processing from microphone input, and real-time processing difficulty because of the increased number of channels for sound direction estimation using DSP processors. To cope with these disadvantages, the system considered in this paper proposes hardware design for enhanced real-time processing using microphone array signal processing. An accurate direction estimation and its design time reduction is achieved by means of an efficient hardware design using spatial segmentation methods and verification techniques. Finally we develop a system which can be used for embedded systems using a sound codec and an FPGA chip. According to experimental results, the system gives much faster real-time processing time compared with either PC-based systems or the case with DSP processors.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.447-450
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• 2005

Multi-channel artificial reverberation algorithm to control perceived direction and distance is described in this paper. In conventional algorithms using IIR filters, reverberation time is the only parameter to be controlled. Moreover, since the convolution-based conventional algorithms apply only same impulse responses, but not considering sound localization, it was not realistic enough. The new algorithm proposed in this paper utilizes early reflections segmented according to the azimuth from which direct sound comes and controls perceived direction by panning the direct sound, and controls perceived distance by adjusting Energy Decay Curve (EDC) of reverberation and gain of the direct sound. In addition, the algorithm enhances Listener Envelopment(LEV) to make late reverberation incoherent among channels.

• Journal of Information Display
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• v.10 no.2
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• pp.72-75
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• 2009

The Brillouin spectrum of 4'-n-pentyl-4-cyano-biphenyl (5CB) liquid crystals homogeneously confined in a planar liquid crystal (LC) cell was measured using a 6-pass tandem Fabry-Perot interferometer. By adopting a special right-angle scattering geometry, the sound velocity of 5CB was estimated from the Brillouin shift without knowing the refractive index. The sound velocity of the longitudinal wave propagating along the direction of the directors aligned parallel to the glass plates of the LC cell was 1784${\pm}$7 m/s at 300 K. The attenuation coefficient $\alpha$ was estimated to be approximately $1.9{\times}10^6m^{-1}$, which is about twice as large as that of the longitudinal sound wave propagating along the direction perpendicular to the directors. The present method may be very useful in the evaluation of the elastic properties of the materials used in display devices, whose refractive indices are not known.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 1996.09a
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• pp.117-124
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• 1996

When navigating in or near an area of restricted visibility it is necessary to be heard the whistle bell and/or the siren of lighthouses or ships at times. Even though we can get the brief informations about the property of sound the direction and range of a sound radiator it is not easy to get the accurate informations for decision making. generally the audio frequency is known as 16-20,000Hz but the earshot is shorten and discrimination of sound is more difficult when there is some noise. The sound pressure is 60dB at the moment when human speaks 1 meter away. Usually the noise pressure in a silent room is 40dB and 60dB on the quiet street. In this study we suggest the basic algorithm to trace the direction and range of the source radiator using the signal received through not a physical sense but the microphone sensors and a series of signal of signal processing.