• 한국위성정보통신학회논문지
• /
• 제11권3호
• /
• pp.80-85
• /
• 2016

본 연구는 선음원과 점음원의 이론에 근거하여 개발된 혼타입 스피커와 라인어레이 타입 스피커에 대해 비교분석하여 실제에서 이론대로 적용되는지에 대해 연구하였다. 이론적으로 점음원은 거리가 2배 됨에 따라 6dB 감쇄하며, 선음원은 거리가 2배 됨에 따라 3dB 감쇄한다. 선음원 이론에 근거하여 개발된 라인어레이 스피커 시스템이 선음원의 이론대로 작은 음압감쇄가 일어나는지에 대하여 분석하여 사용 목적과 환경에 따른 올바른 스피커의 배열 구성이 선택되도록 하는 것이 본 연구의 목적이다. 이를 위해 점음원과 선음원의 이론을 분석하였으며, 이론을 바탕으로 설계된 혼타입 스피커와 라인어레이 스피커를 시뮬레이션으로 파라메터 값들을 분석하였다.

• 한국음향학회지
• /
• 제37권1호
• /
• pp.1-11
• /
• 2018

천해 환경에서 음파 전달은 경계면에 의해 구형 분산에서 원통형 분산으로 음파 전달 조건이 전환되는데, 이 지점을 음파 전달 조건의 변환점 (transition point)이라고 정의한다. 이론적으로 거리에 따른 전달손실을 이용하여 음파 전달 조건의 변환점을 계산할 수가 있으며, 본 논문에서는 포물선 방정식 기반 음향모델을 이용하여 Pekeris 도파관에서 송 수신기가 수층의 중심에 위치한 경우 전달손실을 모의한 후 변환점을 도출하였다. 계산된 변환점은 수층과 퇴적층의 음속비로 계산된 임계각으로 추정한 임계거리와 비교, 분석되었으며, 동일한 환경에서 수층에 음향채널이 존재하는 경우와 음원 수심 변화에 따른 변환점 변동성을 확인하였다. 최종적으로 2015년 5월, 제주도 서남쪽으로 약 65 km 떨어진 SAVEX15(Shallow Water Acoustic Variability EXperiment 2015) 실험에서 획득한 천해 환경에서의 거리에 따른 저 중주파수 음파 전달 실험의 전달손실 자료를 이용하여 실험 해역에서의 음파 전달 조건 변환점을 도출하였으며, 이를 실험해역의 해양환경과 비교를 통하여 음전달 특성을 파악하였다.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2006년도 추계학술대회논문집
• /
• pp.492-497
• /
• 2006

This study gives the results of the measurements and analysises of the reverberation times in a small room such as apartment houses. We measured the RT by changing measurement conditions, which were sound sources. sound source's positions, receiving point & height, sampling time and so on. The critical factor affecting reverberation time was sound source in unoccupied houses and the reverberation time differences between result of RT using impulsive and interrupted sound source was 0.3sec at 500Hz frequency. And the difference of RT due to sound sources affected the sound insulation such as apparent sound reduction index and sound level difference about 1dB at each frequency in unoccupied houses.

• 대한기계학회논문집A
• /
• 제29권12권
• /
• pp.1597-1604
• /
• 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.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
• /
• pp.878-881
• /
• 2003

To acquire an appointed speaker's clear voice signal from inspect-camera, picture-conference or hands free microphone eliminating interference noises needs to be preceded speaker's position automatically. Presumption of sound source position's basic algorithm is about measuring TDOA(Time Difference Of Arrival) from reaching same signals between two microphones. This main project uses ADF(Adaptive Delay Filter) [4] and CPS(Cross Power Spectrum) [5] which are one of the most important analysis of TDOA. From these analysis this project proposes presumption of real time sound source position and improved model NI-ADF which makes possible to presume both directions of sound source position. NI-ADF noticed that if auditory sense of humankind reaches above to some specified level in specified frequency, it will accept sound through activated nerve. NI-ADF also proposes practicable algorithm, the presumption of real time sound source position including both directions, that when microphone loads to some specified system, it will use sounds level difference from external system related to sounds of diffraction phenomenon. In accordance with the project, when existing both direction adaptation filter's algorithm measures sound source, it increases more than twice number by measuring one way. Preserving this weak point, this project proposes improved algorithm to presume real time in both directions.

• KIEAE Journal
• /
• 제14권1호
• /
• pp.39-48
• /
• 2014

In the previous test, the verification of the standard floor impact source was carried out comparing the physical characteristics with living impact sources. The result was appeared the validation of the standard impact source was very low because of differences of physical characteristics. This study aims to evaluate annoyance and loudness of standard impact source which is used for the measurement of floor impact sound, and to compare the annoyance and loudness of living impact sources which are produced in real life. The impact sources considered are tapping machine, tire and impact ball as standard sources, and nine real sources which were chosen from the existing researches. The result showed differences of annoyance and loudness between standard impact sources and living impact sources, which means the standard impact sources may rate the performance of floor system inappropriately. In the future, the rating method should be examined how the standard impact sources are similar with real sources in the point of rating the performance of floor system.

• The Journal of the Acoustical Society of Korea
• /
• 제15권1E호
• /
• pp.39-44
• /
• 1996

Matched field localization schemes often show a high sensitivity to acoustic variabilities due to mismatch between assumed and actual environments. In this paper, we focus on the effect of source motion or Doppler on matched field processing (MEP). to accomplish this, MFP is extended to treat a moving source problem with normal mode description of the sound field. the extension involves both the temporally nonstationary and spatially inhomogeneous nature of the sound field generated by a time-harmonic point source moving uniformly in a stratified oceanic waveguide. It is demonstrated that the impact of source motion can be significant to MEP although the velocity of a moving source is much smaller than the sound velocity of the oceanic waveguide. In addition, a criteria for minimizing the effect of Doppler on MFP is discussed.

• The Journal of the Acoustical Society of Korea
• /
• 제20권4E호
• /
• pp.52-60
• /
• 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
• /
• 한국소음진동공학회 2009년도 춘계학술대회 논문집
• /
• pp.490-495
• /
• 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
• /
• 제19권9호
• /
• pp.907-914
• /
• 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.