• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.701-708
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• 2003

This paper presents an acoustic source localization technique on 2D cavity model in flow using a phased microphone way. Investigation was performed on cavity flows of open and closed types. The source distributions on 2D cavity flow were investigated in anechoic open-jet wind tunnel. The array of microphones was placed outside the flow to measure the far field acoustic signals. The optimum sensor placement was decided by varying the relative location of the microphones to improve the spatial resolution. Pressure transducers were flush-mounted on the cavity surface to measure the near-filed pressures. It is shown that the propagated far field acoustic pressures are closely correlated to the near-field pressures. It is also shown that their spectral contents are affected by the cavity parameter L/D.

• Journal of Information Processing Systems
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• v.13 no.1
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• pp.83-94
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• 2017

Near-field source localization algorithms are very sensitive to sensor gain/phase response errors, and so it is important to calibrate the errors. We took into consideration the uniform linear array and are proposing a blind calibration algorithm that can estimate the directions-of-arrival and range parameters of incident signals and sensor gain/phase responses jointly, without the need for any reference source. They are estimated separately by using an iterative approach, but without the need for good initial guesses. The ambiguities in the estimations of 2-D electric angles and sensor gain/phase responses are also analyzed in this paper. We show that the ambiguities can be remedied by assuming that two sensor phase responses of the array have been previously calibrated. The behavior of the proposed method is illustrated through simulation experiments. The simulation results show that the convergent rate is fast and that the convergent precision is high.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.1
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• pp.71-78
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• 2009

Sound source localization systems in service robot applications estimate the direction of a human voice. Time delay information obtained from a few separate microphones is widely used for the estimation of the sound direction. Correlation is computed in order to calculate the time delay between two signals. In addition, PHAT weighting function can be applied to significantly improve the accuracy of the estimation. However, FFT and IFFT operations in the PHAT weighting function occupy more than half of the area of the sound source localization system. Thus efficient FFT and IFFT designs are essential for the IP implementation of sound source localization system. In this paper, we propose an efficient FFT/IFFT design method based on the characteristics of human voice.

• Geophysics and Geophysical Exploration
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• v.10 no.4
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• pp.285-292
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• 2007

Magnetotelluric data recorded in the middle part of the Korean Peninsula are contaminated by severe noises at dead-band frequencies. In this study, we estimated the location of noise source using a source localization method. Since conventional beamforming techniques were not adequate for the localization of electromagnetic sources, we used the matched field processing and a genetic algorithm. The solutions for the strong noise signals tend to be localized in a narrow area, whereas those for natural MT signals shows randomly distributed patterns. The strong noise sources are mainly located in the western part of Kyonggi-do.

• The Journal of the Acoustical Society of Korea
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• v.43 no.2
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• pp.178-183
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• 2024

This study introduces an enhanced sound source localization technique, bolstered by a data-driven deep learning approach, to improve the precision and accuracy of direction of arrival estimation. Focused on refining Time Difference Of Arrival (TDOA) based sound source localization, the research hinges on accurately estimating TDOA from cross-correlation functions. Accurately estimating the TDOA still remains a limitation in this research field because the measured value from actual microphones are mixed with a lot of noise. Additionally, the digitization process of acoustic signals introduces quantization errors, associated with the sampling frequency of the measurement system, that limit the precision of TDOA estimation. A deep learning-based approach is designed to overcome these limitations in TDOA accuracy and precision. To validate the method, we conduct comprehensive evaluations using both two and three-microphone array configurations. Moreover, the feasibility and real-world applicability of the suggested method are further substantiated through experiments conducted in an anechoic chamber.

• ETRI Journal
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• v.34 no.5
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• pp.684-689
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• 2012

A mode-singular-value-decomposition (SVD) maximum likelihood (ML) estimation procedure is proposed for the source localization problem under an additive measurement error model. In a practical situation, the noise variance is usually unknown. In this paper, we propose an algorithm that does not require the noise covariance matrix as a priori knowledge. In the proposed method, the weight is derived by the inverse of the noise magnitude square in the ML criterion. The performance of the proposed method outperforms that of the existing methods and approximates the Taylor-series ML and Cram$\acute{e}$r-Rao lower bound.

• Proceedings of the Korean Society of Medical Physics Conference
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• 2006.08a
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• pp.115-115
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• 2006

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.332-339
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• 2012

Beamspace transform algorithm transforms spatial-domain data - such as x, y, z dimension - into incidence-angle-domain data, which is called beamspace-domain data. The beamspace transform method is generally used in source localization and tracking, and adaptive beamforming problem. When the beamspace transform method is used in multichannel audio source separation, the inverse beamspace transform is also important because the source image have to be reconstructed. This paper studies the beamspace transform and inverse transform algorithms for multichannel audio source separation system, especially for the beamspace-domain multichannel NMF algorithm.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.10
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• pp.164-170
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• 2014

A single near-field sounde source bearing and ranging method based on a independent doublet array is proposed. In the common case of bearing estimation method, unform linear array or uniform circular array are used. It is constrained retaining aperture because of array structure to estimate the distance of the sound source. Recent using independent doublet array sound source's bearing and distance esmtimation method is proposed by wide aperture. It is limited to the case doublets are located on a straight line. In this paper, we generalize the case and estimate the localization of a sound source in the various array structure. The proposed algorithm was verified performance through simulation.