• International journal of advanced smart convergence
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• v.5 no.4
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• pp.21-25
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• 2016

Recent development in multi-channel surround is emerging in various formats to provide better stereoscopic and sound effects to consumers in recent broadcasting. The ability sound localize the sound sources in space is most considerable design factor on multi-channel surround system for human earing perception model. However, this paper propose the change of the sound localization according to the spacing of the speakers, which is not covered in the existing research focus on sound system design. Presently the sound system uses the position and number of the speakers to localize the sound. In the multi-channel surround environment, the proposed design uses the sound localization is caused by the directional characteristics of the speaker, the distance between the speakers and the distance between the listener and the speaker according to the directivity is required. The proposed design is simulated using virtual measurement with MATLAB simulation environment and performances are measured.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.253-256
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• 2000

This paper describes a DSP implementation of a real-time 3D sound localization algorithm with the use of a low power embedded DSP. A distinctive feature of this implementation is that the audible frequency band is divided into three, in accordance with the sound reflection and diffraction phenomena through different media from a certain sound source to human ears, and then in each subband a specific implementation procedure of the 3D sound localization is devised so as to operate real-time at a low frequency of 50MHz on a 16bit fixed-point DSP. Thus out DSP implementation can provide a listener with 3D sound effects through a headphone at low cost and low power consumption.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.264-267
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• 2000

This paper describes a real-time 3D sound localization algorithm to be implemented with the use of a Bow power embedded DSP. This algorithm first divides the audible frequency band into three, on the basis of the analysis of the sound reflection and diffraction effects through different media from a certain sound source to human ears, and then in each subband a specific procedure is devised fur the 3D sound localization so as to operate real-time on a low power embedded DSP This algorithm aims at providing a listener with the 3D sound effects through a headphone at low cost and low power consumption.

• Journal of Information Technology Applications and Management
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• v.22 no.3
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• pp.105-114
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• 2015

This paper presents a fast Time Difference of Arrival (TDOA) estimation for sound source localization. TDOA is the time difference between the arrival times of a signal at two sensors. We propose a partial cross correlation method to increase the speed of TDOA estimation for sound source localization. We do this by predicting which part of the cross correlation function contains the required TDOA value with the help of the signal energies, and then we compute the cross correlation function in that direction only. Experiments show approximately 50% reduction in the cross correlation computation time thereby increasing the speed of TDOA computation. This makes it very relevant for real world surveillance.

• Journal of Information Technology Applications and Management
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• v.24 no.1
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• pp.157-167
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• 2017

This paper proposes a simple sound source localization (SSL) method based on signal energies comparison and partial cross correlation for TDOA computation. Many sound source localization methods include multiple TDOA computations in order to eliminate front-back confusion. Multiple TDOA computations however increase the methods' computation times which need to be as minimal as possible for real-time applications. Our aim in this paper is to achieve the same results of localization using fewer computations. Using three microphones, we first compare signal energies to predict which quadrant the sound source is in, and then we use partial cross correlation to estimate the TDOA value before computing the azimuth value. Also, we apply a threshold value to reinforce our prediction method. Our experimental results show that the proposed method has less computation time; spending approximately 30% less time than previous three microphone methods.

• Proceedings of the Korea Information Processing Society Conference
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• 2012.04a
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• pp.961-963
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• 2012

The sound source localization system is used to some area such as robotic system, object localization system, guarding system and medicine. So time delay estimation and angle estimation of sound direction are studied until now. These days time delay estimation is described in LabVIEW which is used to create innovative computer-based product and deploy measurement and control systems. In this paper, the development of signal monitoring platform is presented for sound source localization. This platform is designed in virtual instrument program and implemented in two stages. In first stage, data acquisition system is proposed and designed to analyze time delay estimation using cross correlation. In second stage, data obtaining system which is applied and designed to monitor analog signal processing is proposed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8C
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• pp.1142-1148
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• 2004

A binaural filtering method using HRTF DB is generally used to make the headphone-based 3D sound. But it can make some confusion between front and back directions or between up and down directions due to the non-individual HRTF depending on each listener. To reduce the confusion of sound image localization, we propose a new method to boost the spectral cue by modifying HRTF spectra with spectrum difference between front and back directions. Informal listening tests show that the proposed method improves the front-back sound localization characteristics much better than the conventional methods

• The Journal of Korea Robotics Society
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• v.3 no.3
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• pp.237-244
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• 2008

Nowadays, research on human-robot interaction has been getting increasing attention. In the research field of human-robot interaction, speech signal processing in particular is the source of much interest. In this paper, we report a speaker localization system with six microphones for a humanoid robot called MAHRU from KIST and propose a time delay of arrival (TDOA)-based feature matrix with its algorithm based on the minimum sum of absolute errors (MSAE) for sound source localization. The TDOA-based feature matrix is defined as a simple database matrix calculated from pairs of microphones installed on a humanoid robot. The proposed method, using the TDOA-based feature matrix and its algorithm based on MSAE, effortlessly localizes a sound source without any requirement for calculating approximate nonlinear equations. To verify the solid performance of our speaker localization system for a humanoid robot, we present various experimental results for the speech sources at all directions within 5 m distance and the height divided into three parts.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.415-416
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• 2007

HRTF DB, including the information of the sounds which is arrived to our ears, is generally used to make a 3D sound. But it can decline some three-dimensional effects by the confusion between front and back directions due to the non-individual HRTF depending on each listener. In this paper, we propose a new method to use psychoacoustic theory that reduces the confusion of sound image localization. And we make use of an excitation energy by the sense of hearing. This method is brought HRTF spectrum characteristics into relief to draw out the energy ratio about the bark band and control low frequency band. Informal listening tests show that the proposed method improves the front-back sound localization characteristics much better than the conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.16-25
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• 1997

A new method to model the HRTF's(Head-Related Transfer Function), which could give improvement of the sound localization accuracy using the spatial effects by the reflected sound wave transfer characteristics, is proposed. When using the HRTF model having reflected sound wave transfer characteristics, the accuracy of sound localization was quite improved up to about 23%, compared with using the direct wave transfer characteristics only. Furthermore, it is verified that the spatial impression could be a factor to enhance the ability of sound localization.