• Journal of KIISE
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• v.43 no.1
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• pp.13-26
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• 2016

This paper proposes a new sound localization algorithm that can improve localization performance based on a tetrahedron-shaped microphone array. Sound localization system estimates directional information of sound source based on the time delay of arrival(TDOA) information between the microphone pairs in a microphone array. In order to obtain directional information of the sound source in three dimensions, the system requires at least three microphones. If one of the microphones fails to detect proper signal level, the system cannot produce a reliable estimate. This paper proposes a tetrahedron- shaped sound localization system with a coordinate transform method by adding one microphone to the previously known triangular-shaped system providing more robust and reliable sound localization. To verify the performance of the proposed algorithm, a real time simulation was conducted, and the results were compared to the previously known triangular-shaped system. From the simulation results, the proposed tetrahedron-shaped sound localization system is superior to the triangular-shaped system by more than 46% for maximum sound source detection.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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• pp.1028-1031
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• 2004

Last target of architectural acoustics is that people wish to convey voice effectively from the space adaptively in use purpose in building. But, to how exactly through space sound source that wish to deliver from indoor can be passed does quantification sound estimation method is proposing various kinds physical parameter to estimate degree of voice definition (Speech articulation) and reverberation. Result that evaluate sound source about MLS signal and Impluse signal by pistol in this treatise could know that converge in MLS and measurement error extent about reverberation time(RT) of sound benevolent person. And value is thought there is problem showing change irregularly about sound benevolent person of D50, C80 etc. Finally, in case is spread sound field in difference of sound pressure level, when measure about change of sound pressure level, sound benevolent person could know that there is no different effect.

• Journal of Environmental Impact Assessment
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• v.14 no.4
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• pp.217-225
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• 2005

In apartment buildings, floor-impact sound has bean regarded as the major source which induces complaints from residents. It is mainly due to the use of light-weight structures. The vibration produced by impact on one part of an apartment building would travel as far as the other parts of structure with a little alleviation. As a result, the impact sound from upstairs has been regarded as a main source of noise causing discontentment among occupants. This study was carried out to measure the floor-impact sound levels and evaluate the insulation performance of floor-impact sound for nine apartment buildings. The floor-impact sound levels were measured for twenty-five On-dol floor structures and various factors which influence the floor-impact sound were analyzed.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.2
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• pp.208-213
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• 2007

As recent research on home service robot has been performed actively in these days. It becomes very important for the robot to react upon voice and sound source, and then tracks an object position in dynamic environment like a home. When people choose a path for finding a destination of objects, in case of sound, they track a direction of the sound source. Or in case as a position of the object be girded with a point on map, people track the position according to absolute orientation of the present position and the sound source position. In this paper, In this manner we had views on what people decide own direction when they react one's voice or go some directions. We suggest a algorithm that intelligent mobile robots on which we installed a sound source tracking board and a digital magnetic compass board go some object's positions by the direction of sound source and absolute orientation.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.11a
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• pp.932-937
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• 2003

In apartment buildings, floor-impact sound has bean regarded as the major source which induces complaints from residents. It is mainly due to the use of light-weight structures. The vibration produced by impact on one part of an apartment building would travel as far as the other parts of structure with a little alleviation. As a result, the impact sound from upstairs has been regarded as a main source of noise causing discontentment among accupants. This study was carried out to measure the floor-impact sound levels and evaluate the insulation performance of floor-impact sound for seven apartment buildings. The floor-impact sound levels were measured for seventeen On-dol floor structures and various factors which influence the floor-impact sound were analyzed.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.260-267
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• 2009

It is very difficult to form a desired complex sound field at a designated region precisely as an application of acoustic arrays, which is one of important objects of array systems. To solve the problem, a filter design method was suggested, which employed the concept of an inverse method using the acoustical holography based on the boundary element method. In the acoustical holography used for the source identification, the measured field data are employed to reconstruct the vibro-acoustic parameters on the source surface. In the analogous problem of source array design, the desired field data at some specific points in the sound field was set as constraints and the volume velocity at the surface points of the source plane became the source signal to satisfy the desired sound field. In the filter design, the constraints for the desired sound field are set, first. The array source and given space are modelled by the boundary elements. Then, the desired source parameters are inversely calculated in a way similar to the holographic source identification method. As a test example, a target field comprised of a quiet region and a plane wave propagation region was simultaneously realized by using the array with 16 loudspeakers.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.10a
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• pp.586-591
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• 2010

The sound field can be classified to the direct and reverberant sound field. If the sound absorption material in a room is not applied sufficiently, the reverberant sound field increases and the sound pressure in a room also increases when the sound source exists in a room. Therefore, the reverberation time should be controled in order to reduce the reverberant sound as well as sound pressure level in a room. Even though the reverberation time is specified and researched widely in architectural engineering, it is rarely performed in a marine engineering. Therefore, in this research, the reverberation time in a navel vessel is researched related to the noise reduction in a cabin.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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• pp.52-60
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• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.29 no.12 s.243
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• pp.1597-1604
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• 2005

The identification of a moving noise source is important in reducing the source power of the transport systems such as airplanes or high speed trains. However, the direct measurement using a microphone running with noise source is usually difficult due to wind noise, white the source motion distorts the frequency characteristics of the pass-by sound measured at a fixed point. In this study the relationship between the spectra of the source and the pass-by sound signal is analyzed for an acoustic source moving at a constant velocity. Spectrum of the sound signal measured at a fixed point has an integral relationship with the source spectrum. Nevertheless direct conversion of the measured spectrum to the source spectrum is ill-posed due to the singularity of the integral kernel. Alternatively a differential equation approach is proposed, where the source characteristics can be recovered by solving a differential equation relating the source signal to the distorted measurement in time domain. The parameters such as the source speed and the time origin, required beforehand, are also determined only from the frequency-phase relationship using an auxiliary measurement. With the help of the regularization method, the source signal is successfully recovered. The effects of the parameter errors to the estimated frequency characteristics of the source are investigated through numerical simulations.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.04a
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• pp.490-495
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• 2009

An approach to 3D visualization of multiple sound sources has been developed with the application of a moving array technique. Frequency-domain beamforming algorithm is used to generate a beam power map and the sound source is modeled as a point source. When a conventional delay and sum beamformer is used, it is considered that 2D distribution of sensors leads to have deficiency in spatial resolution along a measurement distance. The goal of moving an array in this study is to form 3D array aperture surrounding multiple sound sources so that the improved spatial resolution in a virtual space can be expected. Numerical simulation was made to examine source localization capabilities of various shapes of array. The 3D beam power maps of hemispherical and spherical distribution are found to have very sharp resolution. For experiments, two sound sources were placed in the middle of defined virtual space and arc-shaped line array was rotated around the sources. It is observed that spherical array show the most accurate determination of multiple sources' positions.