• Journal of Satellite, Information and Communications
• /
• v.11 no.3
• /
• pp.80-85
• /
• 2016

In this paper, by comparatively analyzing horn type speaker and line array type speaker developed based on line sound source theory and point sound source theory, we research whether theory is adaptable or not in real. Academically, point sound source is attenuated as much as 6dB in accordance with double distance and line sound source is attenuated as much as 3dB in accordance with double distance. Line array speaker system developed based on line sound source is analyzed by theory of line sound source about occurring small sound pressure attenuation and it is propose of research that array composition of right speaker is selected in accordance with use purpose and environment. For this purpose, we analyze theory of point sound source and line sound source. we analyze parameter value by simulating designed horn type speaker and line array speaker based on theory.

• The Journal of the Acoustical Society of Korea
• /
• v.37 no.1
• /
• pp.1-11
• /
• 2018

Sound propagation in shallow water changes from spherical spreading to cylindrical spreading, depending on boundary conditions, and this point is defined as a transition point of the sound propagation condition. Theoretically, the transition point can be estimated using the transmission loss as a function of source-receiver range. In this paper, the transmission loss curve in a Pekeris waveguide is predicted using a parabolic-equation based acoustic propagation model and using this transmission loss curve, the range from the source of the transition point is estimated, which is compared to the critical distance calculated using the sound speed ratio of water to sediment. In addition, the effects of the sound speed profile and source depth change on the transition point are investigated. Finally, the transition point is estimated using the transmission loss data measured during the period of the SAVEX15 (Shallow Water Acoustic Variability EXperiment 2015) conducted 65 km southwest of Jeju Island in May 2015, and it is compared to the ocean environmental parameters to understand the properties of sound propagation in the experimental area.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2006.11a
• /
• 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.

• Transactions of the Korean Society of Mechanical Engineers A
• /
• v.29 no.12 s.243
• /
• 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.

• Proceedings of the KIEE Conference
• /
• 2003.11c
• /
• 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
• /
• v.14 no.1
• /
• 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
• /
• v.15 no.1E
• /
• 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
• /
• v.20 no.4E
• /
• 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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2009.04a
• /
• 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.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.19 no.9
• /
• 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.