• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.05a
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• pp.769-773
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• 2001

In order to obtain the fundamental data on the underwater sound of fishes for marine ranching. The underwater sound of Red Seabream(Pagrus major) and Schlegel's black rockfish(Sebastes schlegeli) were measured, which character were summarized by FFT analysis.

• Transactions of the Korean Society of Mechanical Engineers
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• v.10 no.5
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• pp.601-610
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• 1986

The acoustic intensity in the very near field of a vibrating surface reveals information about the location of sound sources and sinks. A system model of tennis racket was developed from simultaneous measurement of excitation force, surface vibration and the near field sound pressure. The characteristics of structural dynamics were obtained by standard experimental modal analysis techniques while the sound radiation characteristics were determined by estimating the acoustic intensity. In this paper, the information about vibration behviour was obtained by acoustic intensity method and some, experiments for verification were carried out. Close correlation was found between experimentally determined acoustic intensity and vibration mode patterns of the tennis racket.

• Proceedings of the KOSOMBE Conference
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• v.1993 no.05
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• pp.113-115
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• 1993

In this paper, we proposed a speech support system of personal computer for the blind. The system is consist of hardware part and software part. The hardware part are consist of personal computer and sound card. The software part are consist of sound driver system, character table and sound output algorithm. This system can recognize inputted characters from keyboard and program produced character strings.

• Proceedings of the Korean Society of Computer Information Conference
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• 2014.01a
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• pp.1-4
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• 2014

Although physical modeling synthesis is becoming more and more efficient in rich and natural high-quality sound synthesis, its high computational complexity limits its use in portable devices. This constraint motivated research of single-instruction multiple-data many-core architectures that support the tremendous amount of computations by exploiting massive parallelism inherent in physical modeling synthesis. Since no general consensus has been reached which grain sizes of many-core processors and memories provide the most efficient operation for sound synthesis, design space exploration is conducted for seven processing element (PE) configurations. To find an optimal PE configuration, each PE configuration is evaluated in terms of execution time, area and energy efficiencies. Experimental results show that all PE configurations are satisfied with the system requirements to be implemented in portable devices.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.637-640
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• 2005

We propose a sound source localization method using the Head-Related-Transfer-Function (HRTF) to be implemented in a given platform. HRTFs contain not only the information regarding proper time delays but also phase and magnitude distortions due to diffraction and scattering by the shading object. Therefore, a set of HRTFs for any given platform provides a substantial amount of information as to the whereabouts of the source. In this study, we introduce new phase criterion in order to find the sound source location in accordance with the HRTF database empirically obtained in an anechoic chamber with the given platform. Using this criterion, we analyze the estimation performance of the proposed method in a household environment.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.698-703
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• 1994

In the actual sound environment, the random signal often shows a complex fluctuation pattern apart from a standard Gaussian distribution. In this study, an evaluation method for the sound environmnetal system is proposed in the generalized form applicable to the actual stochastic phenomena, by introducing two type information processing methods based on the regression model of expansion series type and the Fuzzy probability. The effectiveness of the proposed method are confirmed experimentally too by applying it to the observed data in the actual noise environment.

• Proceedings of the Korea Information Processing Society Conference
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• 2022.05a
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• pp.326-327
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• 2022

현행 전자출결 시스템은 강의실이 아닌 곳에서 지정된 숫자만 제공되면, 정당하지 않은 방법으로도 출석을 진행할 수 있다. 이런 맹점을 개선하기 위해 본 논문에서는 인증용 데이터를 텍스트가 아닌 음성 데이터를 이용한다. 본 논문에서 제안하는 SOUND-QR 인증 시스템은 음원을 전달할 때 특정 데이터가 포함된 음원을 잡음과 왜곡 환경에 강한 오디오로 변환하고 이러한 음원을 각 기기에서 입출력시 발생하는 특성 차이를 인증에 활용함으로써 현장에 있는 기기에서만 출결이 인정되도록 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.11a
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• pp.121-123
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• 2007

현재까지 유해한 컨텐츠(Contents)를 차단하기 위한 활발한 연구가 있었으나, 사람의 사운드(sound)와 이미지(image)를 통합한 필터링(filtering) 기법에 대한 연구는 활발히 이루어지지 않은 측면이 있다. 본 논문은 이미지(image) 데이터 중 피부색 분포 비율과 사운드(sound) 데이터 중 주파수 분석을 통한 심층적인 기법을 활용하여 현재까지 진행되고 있는 이미지 필터링(image filtering)방법에 대한 수행 결과보다 획기적으로 개선된 성능을 보이고자 한다. 즉, 사운드와 이미지의 특징 정보를 이용한 성인 컨텐츠(Adult Contents)분류 기법을 활용하는 것으로 성인 컨텐츠(Adult Contents)에서 두드러지는 특징을 보이는 사운드 패턴을 분석하여 현재까지 한정된 자원인 이미지만을 활용한 기법보다는 현저한 향상된 수행능력을 예측해 볼 수 있다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.3
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• pp.386-392
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• 2002

Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality of the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to verify the effectiveness of the proposed algorithm, we experimented in variance network settings. The results show that the proposed algorithm improves on the performance of the conventional buffer management algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.6
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• pp.1188-1193
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• 2009

Todays, many young people suffer from noise-induced hearing loss by using wearable hearing devices, such as Bluetooth headset. This paper present hearing loss reduction and more natural volumn control algorithms considering individual hearing characteristics and threshold of feeling for Bluetooth headset. Experimental results using CSR Bluetooth headset example design board(DEV-PC-1645) show that individuals maybe able to perceive without the inconvenience at the less sound intensity and the more sensitive frequency bands. As a result, we may prevent hearing loss to reduce excessive sound energy in each frequency bands.