• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.5
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• pp.1017-1025
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• 2000

The human auditory system has been shown to posses remarkable abilities in the localization and tracking of sound sources. The localization is the result of processing two primary acoustics cues. These are the interaural time difference(ITD) cues and interaural intensity difference(IID) cues at the two ears. In this paper, we propose TEPILD(Time Energy Previous Integration Location Detector) model. TEPILD model is constructed with time function generator, energy function generator, previous location generator and azimuth detector. Time function generator is to process ITD and energy function generator is to process IID. Total average accuracy rate is 99.2%. These result are encouraging and show that proposed model can be applied to the sound source location detector.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5C
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• pp.309-316
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• 2011

In this paper, equalization zone of two crosstalk cancellation (CC) schemes, which are the one based on only head related transfer function (HRTF) and the other one based on interaural intensity/time difference (ITD/IID) as well as HRTF is studied. To do this, the condition numbers and ITD/IID levels of two schemes are shown.

• The Journal of the Acoustical Society of Korea
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• v.29 no.6
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• pp.381-387
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• 2010

Both timbral features and spatial features are important in the assessment of multichannel audio coding systems. The prediction model, extending the ITU-R Rec. BS. 1387-1 to multichannel audio coding systems, with the use of spatial features such as ITDDist (Interaural Time Difference Distortion), ILDDist (Interaural Level Difference Distortion), and IACCDist (InterAural Cross-correlation Coefficient Distortion) was proposed by Choi et al. In that model, ITDDistswere only computed for low frequency bands (below 1500Hz), and ILDDists were computed only for high frequency bands (over 2500Hz) according to classical duplex theory. However, in the high frequency range, information in temporal envelope is also important in spatial perception, especially in sound localization. A new model to compute the ITD distortions of temporal envelopes in high frequency components is introduced in this paper to investigate the role of such ITD on spatial perception quantitatively. The computed ITD distortions of temporal envelopes in high frequency components were highly correlated with perceived sound quality of multichannel audio sounds.

• Journal of IKEEE
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• v.23 no.4
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• pp.1360-1365
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• 2019

In this paper, we describe a sound source localization(SSL) system which can be applied to mobile robot and automatic control systems. Usually the SSL method finds the Interaural Time Difference, the Interaural Level Difference, and uses the geometrical principle of microphone array. But here we proposed another approach based on the deep neural network to obtain the horizontal directional angle(azimuth) of the sound source. We pick up the sound source signals from the two microphones attached symmetrically on both sides of the robot to imitate the human ears. Here, we use difference of spectral distributions of sounds obtained from two microphones to train the network. We train the network with the data obtained at the multiples of 10 degrees and test with several data obtained at the random degrees. The result shows quite promising validity of our approach.

• Proceedings of the Korean Information Science Society Conference
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• 1999.10b
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• pp.307-309
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• 1999

입체음향이란 음원의 위치에 따라 두 귀에 입력되는 신호를 제어함으로써 시각정보 없이 음상의 위치를 파악할 수 있는 음이다. 헤드폰을 이용하면 음장이 머리 내에 위치하게 됨으로써 거리를 파악하는 것이 매우 힘들다. 본 논문에서는 모노음을 이용하여, 2채널 헤드폰에서 재생할 수 있는 3차원 음을 만들기 위하여 Interaural Time Difference(ITD)와 Interaural Intensity Difference(IID)를 이용한 머리 전달함수(Head Related Transfer Function:HRTF)를 만든 결과와 측정 HRTF 자료인 KEMAR Data를 이용한 결과를 비교하였으며, 거리 효과를 효과적으로 구현하기 위하여 잔향효과를 추가하여 음장을 머리밖으로 꺼냄으로써, 보다 향상된 3차원 음상 제어 시스템을 제안하고 실험하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.197-200
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• 2000

본 논문에서는 공간에 위치한 음원으로부터 양 귀에 들리는 음향을 스피커를 이용하여 재현 하는 기술에 대한 문제점과 이를 효과적으로 구현할 수 있는 방법에 대해 소개한다. 좌우 대칭 위치의 스피커를 이용하여, 음상 정위를 실현할 때 직면한 문제점인 Cross-talk 제거와 스피커와 청취자 양 귀사이의 음향전달 모델 선정에 대한 문제점들을 본 논문에서 고찰한다. 이러한 고찰을 통하여 Transaural 필터의 모델 제시와 본 모델의 음향학적 특성을 소개한다. 인간 청각기관의 공간 감응 인자인 lID(Interaural Intensity Difference)와 ITD(Interaural Time Difference)의 개념을 이용한 Transaural 필터의 구현 방법을 제시한다. 제시한 구현방법과 영국 ISVR의 "Stereo Dipole"과의 비교와, 끝으로, 주파수에 따른 적합한 스피커간의 배치 문제에 관한 것도 언급한다.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.289-290
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• 2007

In order to improve the performance of a speech recognition system in the presence of noise, we propose a noise robust front-end using two-channel speech signals by separating speech from noise based on the computational auditory scene analysis (CASA). The main cues for the separation are interaural time difference (ITD) and interaural level difference (ILD) between two-channel signal. As a result, we can extract 39 cepstral coefficients are extracted from separated speech components. It is shown from speech recognition experiments that proposed front-end has outperforms the ETSI front-end with single-channel speech.

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.2
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• pp.40-43
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• 2015

In this paper, the effects of the leading and the following sounds with single frequency on sound localization are investigated. The sounds with different levels and ISIs(Inter Stimuli Intervals) were used. The width of test sound is 2ms, and those of the leading and the following sounds are 10ms. 1 kHz of the test sound is utilized. The arrival time difference in the subject's ears is set to be 0.5ms. The four kinds of level differences used for one ISI are 0, -10, -15, and -20dB interval. The leading sound is found to have more effect on sound localization than the following sound is. The effect of the leading sound is also found to be dependent on the value of ISI. When the value of the ISI is small, different effects affecting the sound localization are observed.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.459-461
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• 2004

This paper presents an implementation of real-time speaker localization using audio-visual information. Four channels of microphone signals are processed to detect vertical as well as horizontal speaker positions. At first short-time average magnitude difference function(AMDF) signals are used to determine whether the microphone signals are human voices or not. And then the orientation and distance information of the sound sources can be obtained through interaural time difference and interaual level differences. Finally visual information by a camera helps get finer tuning of the speaker orientation. Experimental results of the real-time localization system show that the performance improves to 99.6% compared to the rate of 88.8% when only the audio information is used.

• Speech Sciences
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• v.11 no.4
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• pp.29-42
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• 2004

This paper presents an implementation of real-time speaker localization using audio-visual information. Four channels of microphone signals are processed to detect vertical as well as horizontal speaker positions. At first short-time average magnitude difference function(AMDF) signals are used to determine whether the microphone signals are human voices or not. And then the orientation and distance information of the sound sources can be obtained through interaural time difference. Finally visual information by a camera helps get finer tuning of the angles to speaker. Experimental results of the real-time localization system show that the performance improves to 99.6% compared to the rate of 88.8% when only the audio information is used.