• IEMEK Journal of Embedded Systems and Applications
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• v.6 no.3
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• pp.174-182
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• 2011

Conventional home audio system (e.g. AV Receiver, CD Player etc) has a various functionality of audio play, channel mixing, but the remote controller of these audio players is too complex, difficult for user to manage them effectively. Users want to use these functionalities with more easy, comprehensible way. In this study, "intelligent real-sound presentation technology" that support high quality, realistic audio and the "design of complex information and controller of real sound using intelligent real sound play and control interface" will be introduced. So user can actively, realistically enjoy and play real sound based on user's preference, emotion and circumstance, instead of user's passive service.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10a
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• pp.4-6
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• 2003

A holistic representation, such as sparse ceding or independent component analysis (ICA), was successfully applied to explain early auditory processing and sound classification. In contrast, Part-based representation is an alternative way of understanding object recognition in brain. In this paper. we employ the non-negative matrix factorization (NMF)［1］which learns parts-based representation for sound classification. Feature extraction methods from spectrogram using NMF are explained. Experimental results show that NMF-based features improve the performance of sound classification over ICA-based features.

• Journal of KIISE
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• v.43 no.1
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• pp.13-26
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• 2016

This paper proposes a new sound localization algorithm that can improve localization performance based on a tetrahedron-shaped microphone array. Sound localization system estimates directional information of sound source based on the time delay of arrival(TDOA) information between the microphone pairs in a microphone array. In order to obtain directional information of the sound source in three dimensions, the system requires at least three microphones. If one of the microphones fails to detect proper signal level, the system cannot produce a reliable estimate. This paper proposes a tetrahedron- shaped sound localization system with a coordinate transform method by adding one microphone to the previously known triangular-shaped system providing more robust and reliable sound localization. To verify the performance of the proposed algorithm, a real time simulation was conducted, and the results were compared to the previously known triangular-shaped system. From the simulation results, the proposed tetrahedron-shaped sound localization system is superior to the triangular-shaped system by more than 46% for maximum sound source detection.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.53 no.1
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• pp.19-40
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• 2017

A scientific and objective sound exposure criterion for underwater sound damage on fish has been required since there has been many disputes between an underwater sound maker and a fish damage receiver. The existing criteria are still incomplete scientifically owing to a degree of variability of underwater sounds, diversity of fish hearing sensitivity and damage types, etc. This study reviews existing studies on a hearing mechanism of fish species, manmade underwater sound characteristics and sound exposure assessment parameters, and recent sound exposure criteria. A governing equation for damage coverage estimation and damage coverage dependency on sound source level, ambient noise and transmission loss are also reviewed and interpreted based on sound exposure environments. The foreign and Korean (National Environmental Dispute Medication Commission) criteria are reviewed and compared based on scientific aspects. In addition, the deficit and limit of Korean criteria are presented. The objective of this study is to give a direction for related researches and legislation of sound exposure criteria on fish.

• Journal of the Korea Society of Computer and Information
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• v.27 no.11
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• pp.81-87
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• 2022

In this paper, we propose diffraction-based sound control method to improve sound immersion in a virtual environment. The proposed technique can express the wave and flow of sound in a physical environment and a pattern similar to diffraction in real-time. Our approach determines whether there is an obstacle from the location of the sound source and then calculates the position of the new sound reflected and diffracted by the obstacle. Based on ray tracing, it determines whether or not it collides with an obstacle, and predicts the sound level of the agent behind the obstacle by using the vector reflected and refraction by the collision. In this process, the sound attenuation according to the distance/material is modeled by attenuating the size of the sound according to the number of reflected/refracted rays. As a result, the diffraction pattern expressed in the physics-based approach was expressed in real time, and it shows that the diffraction pattern also changes as the position of the obstacle is changed, thereby showing the result of naturally spreading the size of the sound. The proposed method restores the diffusion and diffraction characteristics of sound expressed in real life almost similarly.

• The Journal of the Acoustical Society of Korea
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• v.19 no.1E
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• pp.13-21
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• 2000

In this paper, we present the description and results of experimental study of the sound power measurements based on International Standards ISO 3741 and 3745. The sound power emitted by a calibrated reference sound source was measured in a reverberation room and a free field over a reflecting plane, using the precision methods of International Standards ISO 3741 direct method and ISO 3745, respectively. The sound power measurements carried out in this study give accurate estimation and also show that both methods for determining the sound power levels of a sound source in a reverberation room and a free field over a reflecting plane, according to the ISO 3741 and 3745, respectively, have proved equally good.

• Journal of the Korea Society of Computer and Information
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• v.28 no.9
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• pp.55-62
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• 2023

In this paper, we present a method that utilizes geometry-based sound generation techniques to efficiently handle multiple sound sources, wind turbulence, and temperature-dependent interactions. Recently, a method based on reduced raytracing has been proposed to update the sound position and efficiently calculate sound propagation and diffraction without recursive reflection/refraction of many rays, but this approach only considers the propagation characteristics of sound and does not consider the interaction of multiple sound sources, wind currents, and temperature. These limitations make it difficult to create sound scenes in a variety of virtual environments because they only generate static sounds. In this paper, we propose a method for efficiently constructing a sound map in a situation where multiple sounds are placed, and a method for efficiently controlling the movement of an agent through it. In addition, we propose a method for controlling sound propagation by considering wind currents and temperature. The method proposed in this paper can be utilized in various fields such as metaverse environment design and crowd simulation, as well as games that can improve content immersion based on sound.

• International journal of advanced smart convergence
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• v.6 no.2
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• pp.51-58
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• 2017

Recently, rapid advances of Ethernet and IP technology have brought many changes in the sound industry. In addition, due to AoIP-based audio transmission technology, various problems of the acoustic system (sound quality deterioration due to long distance transmission, complicated wiring) have improved dramatically. However, when many distributed audio systems are connected with AoIP equipment, if there is a problem in the equipment, it is impossible to operate the connected system. AoIP equipment only can transmit audio signals but cannot adjust the system for acoustic environment. In this paper, AoIP equipment is to be installed with sound equipment on a one-to-one basis, so that various existing problems can be solved and adjustment of sound quality (reverberation, echo, delay and EQ) can be possible by AoIP-based OAC (On-site Audio Center) with built-in DSP function. As a result, uncompressed real-time transmission by distributed transmission/receipt module in OAC (On-site Audio Center) and high quality sound by adjustment of sound quality with built-in DSP can be acquired. It is expected that OAC based sound system will be the industry standard in ubiquitous environment.

• Proceedings of the Korean Society of Computer Information Conference
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• 2021.01a
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• pp.269-271
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• 2021

본 논문에서는 실제 현실에서 표현되는 소리의 특징인 파동(Sound wave)과 흐름(Sound flow) 그리고 회절(Diffraction of sound)을 가상환경에서 실시간으로 표현할 수 있는 방법을 제안한다. 우리의 접근 방식은 소리가 재생되는 위치로부터 장애물 여부를 판단하고, 장애물이 존재할 시, 장애물로 인해 반사와 회절된 새로운 소리 위치를 계산한다. 이 과정에서 레이트레이싱 기반으로 장애물과의 충돌 여부를 판단하고, 충돌에 의해 굴절된 벡터를 이용하여 장애물 너머에서 들리는 소리의 크기를 계산하며, 충돌된 레이의 개수에 따라 소리의 크기를 감쇠시킨다. 본 논문에서 제안하는 방법을 이용한 소리의 회절은 물리 기반 접근법에서 나타나는 회절 형태를 실시간으로 표현했으며, 장애물에 따라서 회절 패턴이 변경되고, 이에 따라 소리의 크기가 자연스럽게 조절되는 결과를 보여준다. 이 같은 실험은 실제 현실에서 나타나는 소리의 퍼짐과 같은 특징을 거의 유사하게 복원해냈다.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.237-240
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• 2000

We have proposed a simple method based on IIR filters for realizing sound image localization. How-ever the nonlinearity of phase characteristics of the IIR filters, which are used for sound image localization, cause decrease of the localization accuracy. In this paper we investigate the influence of phase characteristics on the sound localization. Head-related transfer functions (HRTFs) of a dummy-head are approximated by the IIR filter. We carried out sound image localization experiment with 2-loudspeaker reproduction using the approximated HRTFs. Then the errors which obtained from experiments were compared with the theoretical values which were estimated from the phase shifts of the IIR filters. As a result there was little influence of the nonlinear phase characteristics of the IIR fitters in the localization on the horizontal plane.