• 정보과학회 논문지
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• 제43권1호
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• pp.13-26
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• 2016

본 연구에서는 임의 공간에서 정사면체 형태의 마이크로폰 어레이(microphone array)를 이용하여 음원(sound source)추적 성능을 개선할 수 있는 알고리즘을 제안하였다. 음원추적 시스템은 마이크로폰 어레이의 각 마이크로폰에 도착하는 음원신호의 도착 지연시간(TDOA, Time Delay Of Arrival) 정보를 이용하여 음원의 방향성 정보를 추정한다. 임의 3차원 공간에서 음원추적을 위해서는 최소 3개 이상의 마이크로폰이 필요하다. 3개 마이크로폰으로 구성된 음원추적 시스템의 경우 만약 1개의 마이크로폰이라도 신호 오차가 발생한다면 정확한 음원 방향성 추정이 불가능하다. 본 연구에서는 이러한 문제점을 개선하기 위하여 1개의 마이크로폰을 추가한 정사면체 형태(tetrahedron shaper)의 마이크로폰 어레이를 구성하고 좌표변환 기법을 이용하여 주변 잡음이나 오류에 강인한 새로운 음원추적 알고리즘을 제안하였다. 제안 알고리즘의 성능을 입증하기 위하여 3개의 마이크로폰을 이용한 삼각형 기반 음원추적 시스템과 본 연구에서 제안한 정사면체 기반 음원추적 시스템에 대하여 실시간 비교 실험을 수행하였으며, 실험 결과 제안된 정사면체 기반의 시스템이 최대 약 16% 이상의 향상된 검출율을 보였다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 추계학술대회논문집
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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.

• 한국지능시스템학회논문지
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• 제17권2호
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• pp.208-213
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• 2007

최근 가정용 로봇 연구가 활발히 진행됨에 따라, 가정과 같은 다이내믹한 환경에서 로봇이 목소리를 포함한 음원에 반응하고 그 위치를 정확히 찾아가는 것이 매우 중요해지고 있다. 인간이 목표물에 도달하기 위해 경로를 선택할 때, 그 목표물이 소리인 경우는 현재 위치에서 음원의 방향을 추적한다. 또한 그 목표물의 위치가 지도로 주워질 경우에는 현재 위치와 목표물의 절대적 방위를 기준으로 추적한다. 본 논문에서는 이처럼 사람이 다른 사람의 목소리를 듣고 반응하거나 어떤 방향으로 가고자할 때 소리가 나는 방향이나 지도를 통해 대략 자신만의 방위를 만드는 것에 착안하여, 지능형 모바일 로봇에 음원추적 장치와 전자나침반을 장착함으로써 음원의 방향 또는 절대 방위를 기준으로 목표물을 찾아가는 알고리즘을 제시하고자 한다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2010년도 추계학술대회 논문집
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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.

• 환경영향평가
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• 제14권4호
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2003년도 추계학술대회논문집
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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.

• 한국음향학회지
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• 제28권3호
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• pp.260-267
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• 2009

원하는 복잡한 음장을 지정된 구역에 정확히 형성하는 것은 음향 어레이를 이용한 응용에 있어서 가장 어렵고도 중요한 일이다. 이를 해결하기 위해, 본 논문에서는 역 경계요소법을 원용한 음향홀로그래피 방법을 이용하여 원하는 음장의 특성을 얻기 위한 음원 어레이의 필터 계수를 설계하는 방법을 제안하였다. 음원 파악에 적용되는 음향 홀로그래피는 음장에서의 음압을 측정하여 표면에서의 음원 특성을 재구성하게 되는데, 이와 유사한 음원 설계 문제에서는 목적하는 음장 특성이 주어진 조건이 되며, 음원의 체적 속도는 이러한 음장을 얻기 위한 출력 신호가 된다. 설계 과정에 있어서 먼저 목표 음장의 특성 제한 조건을 갖는 음장 데이터를 구성하고, 음원과 공간을 경계요소법으로 모델링 한 뒤, 소요되는 음원의 정보를 역으로 유도한다. 예제로서 16개의 스피커를 갖는 어레이를 이용해 전방의 반은 평면파 전파, 나머지 반은 정숙공간을 동시에 갖도록 하는 목표 음장을 구현하였다.

• The Journal of the Acoustical Society of Korea
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• 제20권4E호
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2009년도 춘계학술대회 논문집
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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.

• 한국소음진동공학회논문집
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• 제19권9호
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• pp.907-914
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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, several 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 shows the most accurate determination of multiple sources' positions.