• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.135-138
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• 2001

Many researchers have developed the techniques of an efficient 3-D sound system based on the psycho-acoustics of spatial hearing with multimedia or virtual reality In this paper, we propose an idea for the improved 3-D sound system using conventional stereo headphones to obtain a better sound diffusion from the mono-sound recorded at an anechoic chamber. We use the HRTF (Head Related Transfer Function) for the sound localization and the wavelet filter bank with time delay for the sound diffusion. We investigate the effects of the 3-B sound depending on the length of time delay at lowest frequency band. Also the correlation coefficient of the signals between the left channel and the right channel is measured to identify the sound diffusion.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2013.10a
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• pp.833-835
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• 2013

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2013.10a
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• pp.836-837
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• 2013

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.846-860
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• 2011

The binaural auditory system of human has ability to differentiate the direction and the distance of the sound sources by using the information which are inter-aural intensity difference(IID), inter-aural time difference(ITD) and/or the spectral shape difference(SSD). These information is generated from the acoustical transfer of a sound source to pinna, the outer ears. We can create a virtual sound system using the information which is called Head related transfer function(HRTF). However the performance of 3D sound is not always satisfactory because of non-individual characteristics of the HRTF. In this paper, we propose the algorithm that uses human's auditory characteristics for accurate perception. To achieve this, excitation energy of HRTF, global masking threshold and loudness are applied to the proposed algorithm. Informal listening test shows that the proposed method improves the sound localization characteristics much better than conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.255-264
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• 2011

The Head Related Transfer Function (HRTF) means a process related to acoustic transmission from 3d space to the listener's ear. In other words, it contains the information that human can perceive locations of sound sources. So, we make virtual 3d sound using HRTF, despite it doesn't actually exist. But, it can deteriorate some three-dimensional effect by the confusion between front and back directions due to the non-individual HRTF depending on each listener. In this paper, we proposed the new algorithm to reduce the confusion of sound image localization using human's acoustic characteristics. The frequency spectrum and global masking threshold of 3d sounds using HRTF are used to calculate the psychoacoustical differences among each directions. And perceptible cues in each critical band are boosted to create effective 3d sound. As a result, we can make the improved 3d sound, and the performances are much better than conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.36 no.6
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• pp.419-424
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• 2017

TDoA (Time Difference-of Arrival) or DoA (Direction-of-Arrival) can be used for source localization. However, the localizing performance is dependent on relative position between source and receivers, receivers' geometric structure, sound speed, and so on. In this paper we propose a source localization method with enhanced performance that combines multiple information. The proposed method uses the time TDoA, DoA and sound speed as variables. LM (Levenberg-Marquardt) method which is one of nonlinear optimizations is applied. The performances of the proposed method was evaluated by simulation. As result of simulation, the proposed method has the lower average localizing error performance than the previous method.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.713-717
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• 2014

This paper presents a beamforming technique for locating impulsive sound source. The conventional frequency-domain beamformer is advantageous for localizing noise sources for a certain frequency band of concern, but the existence of many frequency components in the wide-band spectrum of impulsive noise makes the beamforming image less clear. In contrast to a frequency-domain beamformer, it has been reported that a time-domain beamformer can be better suited for transient signals. Although both frequency- and time-domain beamformers produce the same result for the beamforming power, which is defined as the RMS value of its output, we can use alternative directional estimators such as the peak value and crest factor to enhance the performance of a time-domain beamformer. In this study, the performance of three different directional estimators, the peak, crest factor and RMS output values, are investigated and compared with the incoherent interfering noise embedded in multiple microphone signals. The proposed formula is verified via experiments in an anechoic chamber using a uniformly spaced linear array. The results show that the peak estimation of beamformer output determines the location with better spatial resolution and a lower side lobe level than crest factor and RMS estimation in noise free condition, but it is possible to accurately estimate the direction of the impulsive sound source using crest factor estimation in noisy environment with stationary interfering noise.

• The Journal of the Acoustical Society of Korea
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• v.39 no.1
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• pp.47-56
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• 2020

The sound source localization technique has various application fields in the era of internet-of-things, for which the probe size becomes critical. The localization methods using the acoustic intensity vector has an advantage of downsizing the layout of the array owing to a small finite-difference error for the short distance between adjacent microphones. In this paper, the acoustic intensity vector and the Time Difference of Arrival (TDoA) method are compared in the viewpoint of the localization error in the far-field. The comparison is made according to the change of spacing between adjacent microphones of the three-dimensional microphone array arranged in a tetrahedral shape. An additional test is conducted in the reverberant field by varying the reverberation time to verify the effectiveness of the methods applied to the actual environments. For estimating the TDoA, the Generalized Cross Correlation-Phase transform (GCC-PHAT) algorithm is adopted in the computation. It is found that the mean localization error of the acoustic intensimetry is 2.9° and that of the GCC-PHAT is 7.3° for T₆₀ = 0.4 s, while the error increases as 9.9°, 13.0° for T₆₀ = 1.0 s, respectively. The data supports that a compact array employing the acoustic intensimetry can localize of the sound source in the actual environment with the moderate reflection conditions.

• Journal of the Korea Institute of Military Science and Technology
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• v.27 no.5
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• pp.535-543
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• 2024

Studies on estimating the underwater sound source localization using acoustic signal characteristics have mainly been conducted in shallow waters. Recently, technologies for stably and efficiently estimating the underwater sound sources localization using the underwater sound propagation characteristics of the Reliable Acoustic Path(RAP) in deep water areas are being studied. Underwater surveillance technology in deep sea areas is known to have the advantage of having low detection performance variability due to time-varying underwater environments and having a small shadow zone, making it easy to stably detect underwater sound sources and estimate location even from relatively long distance. In this study, we analyzed the sound propagation characteristics based on the actual marine environment in the deep sea of the Korean Peninsula and conducted a study to analyze the estimation performance of sound source depth using the broadband interference pattern of direct wave and sea surface reflected waves radiating from underwater sound sources.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3E
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• pp.79-87
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• 2009

This paper reviews several multi-source localization methods which estimate ITDs based on zero crossings (ZCs). Employing signal-to-noise ratio (SNR) estimation from ITD variances, these ZC-based source localization methods are more robust to diffuse noise than the cross-correlation (CC)-based one with less computational complexity. In order to take reverberant environments into account, two approaches detect intervals which dominantly contain direct-path components from sources to sensors because they may effectively provide reliable ITDDs corresponding to source directions. One accomplishes the detection by comparing the original and cepstral-prefiltering-processed envelopes, and the other searches sudden increase of acoustic energy by considering typical characteristics of acoustic reverberation. Experiments for comparison of these methods demonstrate that the approach with energy-based detection efficiently achieves multi-source localization in reverberant environments.