• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.1
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• pp.9-15
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• 2011

Communication environments in the context of underwater channel are characterized to be bad by the characteristics of multipath. Multipaths are affected by various factors e.g. the temperature and the salinity of the ocean. In this paper, the representative sound speed profiles were calculated in the southern part of Baengnyeoung island so that the eigen-ray paths with the channel impulse responses were determined using the average sound speed profile of last decade. The performance of underwater communication was analyzed using the BPSK modulation and time reversal method. The significant differences of results were shown according to the change of season and carrier frequency by using computer simulation. In addition, improved performance is obtained using preprocess channel impulse response for the better comparison of two cases of summer and autumn.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.8
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• pp.823-831
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• 2008

Since it needs the light computational load and small database, sound source localization method using time delay of arrival(TDOA method) is applied at many research fields such as a robot auditory system, teleconferencing and so on. Researches for time delay estimation, which is the most important thing of TDOA method, had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Silverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time. Also, we define the performance index(PI) to show a similar tendency to R/D ratio, and propose the method to improve the performance of time delay estimation with PI.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.02a
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• pp.139-142
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• 2008

헤드폰 또는 이어폰으로 오디오 청취 시 흔히 음상이 머리 내부에 맺히는 현상이 발생하게 되며, 이러한 현상을 음상 내재화(Inside Head Localization; IHL)라 한다. 오디오의 음상이 머리 주변 혹은 머리 내부에 맺히게 되면 공간감이나 입체감이 떨어지게 되어 음향의 현실감을 저하시키게 되며 또한 청취에 따른 피로도가 증가하게 된다. 이러한 음상 내재화 현상을 제거하여, 헤드폰/이어폰을 통해 오디오 청취 시 음상이 머리의 외부에 맺히도록(Out of Head Localization; OHL) 하는 기술을 음상 외재화(Sound Externalization) 기술이라 한다. 룸 임펄스 응답이 방향 큐와 연계하여 생성되었을 경우 외재화가 가능하다는 실험적 사실을 바탕으로 기존의 음상 외재화 방법은 일반적인 HRTF (Head Related Transfer Function)를 이용하여 외재화 필터를 구성해왔다. 본 논문에서는 구체마이크로폰을 이용하여 녹음한 멀티채널 룸 임펄스 응답을 기반으로 모델링 된 외재화 필터를 이용한 음원 외재화 방법을 제안한다. 또한 실험 및 결과 분석을 통하여 본 알고리즘의 전방 음원 외재화 성능의 우수성을 입증하고, 외재화 알고리즘 적용 후의원 신호 음상 보존 성능을 확인한다.

• International Journal of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.161-167
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• 2021

This paper proposes a new personalized HRTF estimation method which is based on a deep neural network (DNN) model and improved elevation reproduction using a notch filter. In the previous study, a DNN model was proposed that estimates the magnitude of HRTF by using anthropometric measurements [1]. However, since this method uses zero-phase without estimating the phase, it causes the internalization (i.e., the inside-the-head localization) of sound when listening the spatial sound. We devise a method to estimate both the magnitude and phase of HRTF based on the DNN model. Personalized HRIR was estimated using the anthropometric measurements including detailed data of the head, torso, shoulders and ears as inputs for the DNN model. After that, the estimated HRIR was filtered with an appropriate notch filter to improve elevation reproduction. In order to evaluate the performance, both of the objective and subjective evaluations are conducted. For the objective evaluation, the root mean square error (RMSE) and the log spectral distance (LSD) between the reference HRTF and the estimated HRTF are measured. For subjective evaluation, the MUSHRA test and preference test are conducted. As a result, the proposed method can make listeners experience more immersive audio than the previous methods.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.208-212
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• 2009

Ground wave is an acoustic wave propagating at a sediment sound speed in the case that sediment sound speed is constant with depth, which is explained by modal dispersion effects. In this paper, the ground wave in time domain is simulated using the ray-based approach, which is possible because the modal dispersion can be explained by the guiding of energy caused by reflection and refraction in the waveguide geometry. For a Pekeris waveguide, the ground wave can be interpreted as a sequence of head waves, called a head wave sequence [Choi and Dahl, J. Acoust. Soc. Am. 119, 3660-3668 (2006)]. The ground wave is simulated by convolution of the source signal with a channel impulse response of the head wave sequence, which is compared with simulated signals obtained via a Fourier synthesis of a complex parabolic equation (PE) field.

• The Journal of the Acoustical Society of Korea
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• v.38 no.5
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• pp.494-510
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• 2019

Underwater acoustic communication channel parameters consist of impulse response, delay spreading, scattering function, coherence bandwidth, frequency selective fading, coherence time and time variant magnitude fading statistics on which communication system modem and channel coding are designed. These parameters are influenced by sound velocity profile, platform motion and sea surface roughness in given acoustical oceanography condition. In this paper, channel model based on phasor, channel simulator, measurement and analysis method of channel parameters are given in a fixed source-to-receiver system and the parameters are analyzed using shallow water experimental data. For two different source-to-receiver ranges of 300 m and 600 m, the parameters are characterized by three multipaths such as a direct, a surface reflection path with time variant scattering and a bottom reflection path. The results present a channel modelling method of a fixed source source-to-receiver system, channel parameters measurement and analysis methods and a system design and performance assessment method in shallow water.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.670-672
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• 2014

In this study, we investigate active control of noise transmitted through a window of enclosures minimizing the acoustic power. To reduce noise of the enclosures, passive methods with absorbing material are generally used. The passive methods, however, are limited use due to the vantilation windows. In this case, these windows are path of noise leakage. Feedforward active noise control technology is applied to minimize the sound power from the enclosure. The feedforward controller is implemented with FIR filter based on the transfer functions calculated numerically. The controller reflects the delay due to FIR filter. The noise transmitted through the window is actively controlled, and the reduction of the power is obtained by 15dB.

• The Journal of the Acoustical Society of Korea
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• v.23 no.7
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• pp.521-531
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• 2004

In this paper we developed the Physical modeling of the Sanio Gayageum using the improved digital waveguide theory. The frequency characteristics of the Gayageum body is implemented by an inverse filtering and the impulse response of the body. We obtained the synthesis sounds of the unit sound for the Gayageum using the simulation of the straight-line fits by the changes of the fundamental frequencies depending on the Amok location. Finally. we could obtain the virtual Sanio Gayageum sounds similar to the actual Gayageum by tuning the Amok positions.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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• pp.772-777
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• 2004

Opera house acoustics were subjectively evaluated in order to investigate the effect of performance stage views on the audience's perception of the seat acoustics in an opera house. Nine seats from an existing opera house were selected for the auditory and/or visual experiments according to seating area distribution and acoustical parameters such as RT and $1-IACC_{E3}$. The recorded music, convolved from the impulse response, was presented with and without visual images of the stage. Subjects were asked to assess the auditory/visual descriptors and overall impression of the music at each seat. The results showed that good visual input helps produce a favorable impression of the acoustics, but a limited view degrades acoustical impression. The acoustical parameters in the tested seats were also investigated to find the relationship between the acoustical parameters and the visual/sound impression.

• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.349-355
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• 2018

A large number of parameters have been defined in order to describe and to evaluate the acoustical propeTies in rooms. Reverberation time is an impoTant characteristic of conceT halls, theaters and studios, etc. Over the years, a number of different methods for measuring the reverberation time have been developed, the most common being: the interrupted noise method, the integrated impulse response method, and the method of recording the room response to an impulsive source. However, the reverberation time can be changed by the measurement method, sound source and microphone. Therefore, it is difficult to accurately measure the reverberation time in a room. In this paper, it will be analyzed the interpretation method of the reverberation time and discussed the limitations.