• Journal of Korea Multimedia Society
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• v.16 no.11
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• pp.1272-1280
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• 2013

In this paper, we propose an audio surveillance system which can detect and localize dangerous sounds in real-time. The location information about dangerous sounds can render a PTZ camera to be directed so as to catch a snapshot image about the dangerous sound source area and send it to clients instantly. The proposed audio surveillance system firstly detects foreground sounds based on adaptive Gaussian mixture background sound model, and classifies it into one of pre-trained classes of foreground dangerous sounds. For detected dangerous sounds, a sound source localization algorithm based on Dual delay-line algorithm is applied to localize the sound sources. Finally, the proposed system renders a PTZ camera to be oriented towards the dangerous sound source region, and take a snapshot against over the sound source region. Experiment results show that the proposed system can detect foreground dangerous sounds stably and classifies the detected foreground dangerous sounds into correct classes with a precision of 79% while the sound source localization can estimate orientation of the sound source with acceptably small error.

• The Journal of the Acoustical Society of Korea
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• v.41 no.6
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• pp.698-704
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• 2022

In this paper, the radiated aerodynamic noise generated from sound sources of a nozzle inflow is quantitatively investigated and compared with experimental results of externally radiated noise. A high-resolution unsteady compressible Large Eddy Simulation (LES) technique is used to accurately predict the internal and external flow of three types of nozzle shape. Through using the vortex sound source for sound sources, the geometry of nozzle neck is identified as most significant aerodynamic noise sources. For validation of quantitative analysis, the vortex sound source intensity of internal nozzle flow is compared with results of external radiated noise of calculation and experiment.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.2
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• pp.24-29
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• 2009

Sound source localization system is used in a robot, a video conference and CCTV(Closed-circuit television) systems. In this Sound source localization systems are applied to human and they can receive a number of sound data frames during speaking. In this paper, we propose methods which is reducing angle estimation error by selecting sound data frame which can more precisely compute the angles from inputted sound data frame. After selected data converted to angle, the error of sound source localization recognition system can be reduced by applying to medium filter. By the experiment using proposed system it is shown that the average error of angle estimation in sound source recognition system can be reduced up to 31 %.

• KIEAE Journal
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• v.9 no.4
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• pp.23-28
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• 2009

Standard sound source currently used in heavy-weight floor impact sounds that cause many social problems has excessive low-frequency energy within a range from 63 Hz to 125 Hz, and is difficult to evaluate and measure. To solve these problems, studies are widely performed using a new impact source, the impact ball. In this study, the sound fields in a receiving room were compared and analyzed according to the current impact source, the bang machine, and the impact ball. And deviation of sound pressure level according to the impact source positions were compared. In case of impact ball, the sound pressure level was lower at 63 Hz and below and higher at 125 Hz and above. The same trend was observed at the low-frequency range on the horizontal and vertical planes, regardless of the type of the impact source, which showed the influence of the room mode. There was a problem with the variations in the sound pressure level according to the size or shape of the receiving room. And it also shows that change of source positions may effect the single number rating scheme.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.6 s.111
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• pp.565-573
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• 2006

Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. The purpose of this research is development of beamforming technique can be applied to the rotor noise source identification. For the do-Dopplerization and reconstruction of emitted sound wave, Forward Propagation Method is applied to the time domain beamforming technique. And validation test were performed using rotating sound source constructed by bended pipe and horn driver. In the validation test using sinusoidal sound wave, sufficient performance of signal processing can be seen, and the effect of measuring duration for accuracy was compared. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies and collective pitch angle, in hover condition.

• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.60-67
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• 2014

The purpose of this study is to suggest a new concept of the absolute sound level system in order to basically solve all the irrational aspects from the mastering stage to the output stage during the process of producing a sound source. Through this absolute sound level system, all the input absolute sound levels of sound sources become standardized based on 60 dB(S), and then sound source producers can implement a sound level they intend by using the differential sound level tag. Besides, by matching the output absolute sound level with the input absolute sound level, a target sound level can be implemented as a listener intends. With this system introduced, sound source producers are able to focus on the completion of a sound source itself without an unnecessary competition for sound levels, and only by inputting the differential reduction tag, they can implement a sound level balance as intended. At the same time, listeners are able to appreciate all kinds of sound sources fit for the standard of absolute s ound levels they want to listen to, without a process of trial and error, no matter what kinds of listening environments and sound systems they have.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.660-669
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• 2003

This paper describes a theory to calculate sound source field from the spatial transform of sound field and the measured cross-power spectrum of sound pressure over a hologram plane close to a sound source, Calculating method is proposed to solve sound pressures from cross-power spectrums over a hologram plane, For this, Taylor series for the nonlinear equations is expanded, and it is calculated using Newton-Raphon method, Also, a wave number filter is used to reduce errors that is occurred on the backward propagation, and is performed numerical simulation of the circular piston sound source with infinite baffle in water to verify the proposed theory.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.751-755
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• 2005

We propose a sound source localization method using the Head-Related-Transfer-Function (HRTF) to be implemented in a robot platform. In conventional localization methods, the location of a sound source is estimated from the time delays of wave fronts arriving in each microphone standing in an array formation in free-field. In case of a human head this corresponds to Interaural-Time-Delay (ITD) which is simply the time delay of incoming sound waves between the two ears. Although ITD is an excellent sound cue in stimulating a lateral perception on the horizontal plane, confusion is often raised when tracking the sound location from ITD alone because each sound source and its mirror image about the interaural axis share the same ITD. On the other hand, HRTFs associated with a dummy head microphone system or a robot platform with several microphones contain not only the information regarding proper time delays but also phase and magnitude distortions due to diffraction and scattering by the shading object such as the head and body of the platform. As a result, a set of HRTFs for any given platform provides a substantial amount of information as to the whereabouts of the source once proper analysis can be performed. In this study, we introduce new phase and magnitude criteria to be satisfied by a set of output signals from the microphones in order to find the sound source location in accordance with the HRTF database empirically obtained in an anechoic chamber with the given platform. The suggested method is verified through an experiment in a household environment and compared against the conventional method in performance.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.600-605
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• 2000

The aim of this study is to compare the sound insulation performances of window depending on the measurement methods through the field test and analyze the factors that affect the sound insulation performances. Four measurement methods which are specified in the Koran Standard 2235 and the International Standard 140-5 were selected for the study; the outdoor sound source method which is classified the l000mm method and the 10mm method, the indoor sound source method, and the ISO method. The result of this study is that the sound insulation performance of the windows was the best when measured according to the ISO method and the worst the indoor sound source method. Through the study it was found that the main factors affecting the sound insulation performance of the windows were the correction of the specimen's area and the equivalent sound absorption area of the receiving room.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.05a
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• pp.1203-1209
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• 2001

Identification of noise sources, their locations and strengths, have been taken great attention. The method that can identify noise sources normally assumes that noise sources are located at a free field. However, the sound in a reverberant field consists of that coming directly from the source plus sound reflected or scattered by the walls or objects in the field. In contrast to the exterior sound field, reflections are added to sound field. Therefore, we have to consider the reverberation effect on the source identification method. The main objective of this paper is to identify noise source in the reverberant field. At fist, we try to identify noise sources in a rigid wall enclosure using the spherical beamforming method. In many case of practical interest, the wall has an admittance so that complex reflection process occurred. In this paper, we assumed the complex reverberant field in the enclosure to be the sum of plane waves with random incidence and magnitude. Then the effects of reverberant field at interior source identification have been studied theoretically as well as experimentally