• Journal of Korea Multimedia Society
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• v.13 no.7
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• pp.960-972
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• 2010

In this paper, we present a software for placing instruments on stage based on the acoustic properties of the concert hall. In order to simulate the changes in sound depending on the positions of the instruments, we incorporated the idea of location-based reverberation effect which can be realized through the convolution of instrument sounds with the impulse responses from the respective instrument positions. And we developed a software with a real-time convolution engine which enables the user to conveniently simulate the resulting sound of various instrument placements. The software was tested with the impulse response data measured at two concert halls of the National Center for Korean Traditional Performing Arts and Korean traditional instrument sounds. Results of these experiments show that simulated reverberation effects properly represent the spatial placement of instruments on stage.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.1067-1070
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• 2002

This paper deals with the factors affecting measurement of sound insulation performance of the ceiling panels. The factors to be experimently examined are as follows: sound diffuseness of the test room by investigating sound pressure distribution, Reverberation time, and influence of speaker location on measured results. Based on the investigation of sound pressure distribution over measuring points, it can draw a conclusion that rectangular shaped rooms as test rooms have a serious problem associated with the diffuse sound field.

• Korean Journal of Air-Conditioning and Refrigeration Engineering
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• v.14 no.1
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• pp.63-72
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• 2002

The main objective of this study is to propose a practical two-microphone impedance tube method to measure the sound transmission loss for flexible sound isolation sheets without the use of the time-consuming and expensive reverberation room. This method was based on the sound decomposition theory developed by Seybert using the spectral density functions of the incident and reflected sound waves. In order to verify the validity of the experimental results, the measured sound transmission losses from the proposed method were compared with the measured data from the reverberation room method and the calculated data from the theory satisfying the mass law of sound isolation material. The resulted trends of the sound transmission losses versus frequencies for several different sound isolation sheets were almost same for each other and agreed quite well in both methods except at some low frequency region. From the experimental results, it was found that the accuracy of sound isolation capability obtained by two-microphone impedance tube method depends upon the microphone spacing, the distance from the first microphone to the test sample surface and the test sample location.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.1
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• pp.71-77
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• 2005

In this paper we implement a headphone-based 5.1 channel 3-dimensional (3D) sound generation system using HRTF (Head Related Transfer Function). Each mono sound source in the 5.1 channel signal is localized on its virtual location by binaural filtering with corresponding HRTFs, and reverberation effect is added for spatialization. To reduce the computational burden, we reduce the number of taps in the HRTF impulse response and model the early reverberation effect with several tens of impulses extracted from the whole impulse sequences. We modified the spectrum of HRTF by weighing the difference of front-back spec01m to reduce the front-back confusion caused by non-individualized HRTF DB. In informal listening test we can confirm that the implemented 3D sound system generates live and rich 3D sound compared with simple stereo or 2 channel down mixing.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.785-790
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• 1994

The active noise control which regards the acoustic power as a target function to be minimized, is analyzed to test its feasibility of which simplifies the measurement system compared with the global acoustic energy based active noise control system. In fact, it is found that the acoustic power based active noise control strategy is equally likely as good as the global acoustic energy based active noise control method if the acoustic field of interest is diffusive or very low model density one. In the intermediate model density field, we also demonstrate that the power based control gives the similar results as the energy based control in terms of global sound energy reduction for the lightly damped enclosure which might be most important system in practical application. From all the theoretical and power based control strategy is dependent on the characteristics of the acoustic field to be controlled; i.e., the model density distribution, the degree of reverberation, and on the strength of modal interaction of the control source with the primary source; i.e., the location of control source.

• Phonetics and Speech Sciences
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• v.3 no.3
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• pp.109-114
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• 2011

To estimate sound direction of arrival with a pair of microphones, a method based on Time Difference of Arrival (TDOA) estimation using the Cross Power Spectrum Phase (CPSP) function is largely used due to its simplicity and good performance. In this paper, we investigate CPSP maximum values for various SNRs and adverse environments, and propose a novel method to improve the estimation performance of sound direction of arrival. The proposed method applies a threshold to the CPSP values and increases the reliability of the estimated sound direction. Through computer simulation for various SNRs, we validate the effectiveness of the proposed method. When the threshold was set to 0.1, more than 90% of success rate of sound direction of arrival estimation has been achieved for directions of $10^{\circ}$, $40^{\circ}$, $70^{\circ}$ from the source location even with reverberation times of 0.1s.

• The Bulletin of The Korean Astronomical Society
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• v.46 no.1
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• pp.54.4-55
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• 2021

The Seoul National University AGN Monitoring Project (SAMP) is a welldesigned long-term AGN reverberation mapping project. SAMP focuses on the luminous AGNs out to z~0.5 with relative long time lags between AGN continuum and broad emission lines and aims to probe the high-end of the AGN broad line region (BLR) size-luminosity (R-L) relation. The pilot observations started in October 2015 for 100 AGNs to confirm the variability and the H and [O III] emission line strengths. Based on the initial variability test, 48 quasars has been continued spectroscopic monitoring since Feb. 2016 with Lick 3m and MDM 2.4m telescopes with a cadence of ~20 days. Supporting photometric monitoring in B and V band was conducted at multiple facilities including the MDM 1.3m, LOAO, and DOAO telescopes with a cadence of ~10 days. By the time of Feb. 2021, we have obtained five years spectroscopic and photometric data. More than 30 AGNs shows significant variability in five-year baseline and 16 of them show well detected lags between B-band and H. Here, we report some examples of SAMP light curves and lag detections using the first five-year data as well as the location of our 16 targets in the AGN BLR R-L relation. These measurements are consistent with the existing R-L relation and located at the high-end. With the coming data, SAMP are hopefully to report more AGNs with well detected lags. Our results demonstrate the general feasibility and potential of long-term reverberation project with medium cadence for luminous AGNs.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.452-459
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• 2021

Estimating the mutual time delay between two acoustic sensors is used in various fields such as tracking and estimating the location of a target in room acoustics and sonar. In the time delay estimation methods, there are a non-parametric method, such as Generalized Cross Correlation (GCC), and a parametric method based on system identification. In this paper, we propose a time delay estimation method based on the parametric method. In particular, we propose a method that considers the noise in each receiving acoustic sensor. Simulation confirms that the proposed algorithm is superior to the existing generalized cross-correlation and adaptive eigenvalue analysis methods in white noise and reverberation environments.

• The Journal of the Acoustical Society of Korea
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• v.41 no.6
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• pp.610-620
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• 2022

An acoustic vector sensor can simultaneously receive vector quantities, such as particle velocity and acceleration, as well as acoustic pressure at one location, and thus it can be used as a single input multiple output receiver in underwater acoustic communication systems. On the other hand, vector signals received by a single vector sensor have different channel characteristics due to the azimuth angle between the source and receiver and the difference in propagation angle of multipath in each component, producing different communication performances. In this paper, we propose a channel parameter-based weighting method to improve the performance of an acoustic communication system using a single vector sensor. To verify the proposed method, we used communication data collected from the experiment conducted during the KOREX-17 (Korea Reverberation Experiment). For communication demodulation, block-based time reversal technique which is robust against time-varying channels were utilized. Finally, the communication results showed that the effectiveness of the channel parameter-based weighting method for the underwater communication system using a single vector sensor was verified.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.115-128
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• 2012

This paper investigates the effects of diffuser on the acoustical parameters in music hall with consideration of the result of scattering coefficient measurement. A scale model hall of 600 seats with orchestra shell was used for experiments. The materials of 1:50 scale model was chosen through absorption coefficient measurement based on ISO 354. The model was matched to the computer simulation model in terms of reverberation time. In order to evaluate the effect of diffuser location, the measurements were accomplished with and without diffusers according to 7 configurations by diffuser-installed region; sidewall, balcony front, ceiling and so on. The following acoustical parameters were extracted from each measurement case; Reverberation time (RT), Early decay time (EDT), Clarity (C80), Center time (Ts), Sound strength (G) and Temporal diffusion (TD) from the auto-correlation function (ACF) of impulse responses. As a result, the absorption power and diffusion power were increased with number of diffusers. Accordingly RT, EDT and G were decreased by diffuser and the redirection of reflections was occurred briskly. Averaged TD was 6.05 to 6.30 by measurement cases. RT was found to be the most related factor to diffusion power (R = 0.94). The correlation between TD and EDT was high (R = 0.73). In addition, the effects of diffuser-installed location were discussed in terms of acoustical parameter variation.