• The Journal of the Acoustical Society of Korea
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• v.24 no.3
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• pp.166-170
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• 2005

This paper describes the experimental study for the position estimation method of underwater sound source using the Nearfield Acoustic Holography. The result confirms that it can be used in the identification of underwater noise sources. The sound sources in the experimental work consists of 2 spherical projectors and the near-Held sound pressure is measured in the hologram plane. From the cross-power spectra of the measured data, the complex sound pressures on the hologram plane is derived and its spatial transformation gives sound fields in a source region. The obtained sound fields in a source region showed that the position of each sound source and their relative source strength are exactly estimated. In conclusion, this technique can be applied for estimation of each source position and its relative strength contribution for the underwater multiple sound sources.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.12 no.1
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• pp.78-82
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• 2019

This paper addressed the spectrum of ultrasonic waves produced by arc and/or coronal discharge inside the switchboard. Portable ultrasound sensors are useful for detecting discharge phenomena, such as coronal means in electrical systems. However, a typical handheld ultrasound detector has a disadvantage of determining the type of problem by listening to the sound characteristics and predicting the results, as a result of the determination of whether a discharge is present. Therefore, a new method of analysis is required to distinguish ultrasonic characteristics. In this paper, we published an ultrasound analysis case study to visualize the sound of ultrasonic waves measured with ultrasonic sensors. From the results of the experiment, it was possible to detect coronal discharge and serial arc discharge without interference by the ultrasonic detection system.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.15 no.1
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• pp.31-36
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• 2004

Background and Objectives : It has been shown that the epilaryngeal tube in the human airway is responsible for vocal ring, or the singer's formant. In previous study, authors showed that in trained tenors, besides the conventional singer's formant in the region of ,5500Hz, another energy peak was observed in the region of 8,000Hz. This peak was interpreted as the second resonance of the epilarynx tube. Singers in other voice categories who produce vocal ring are assumed to have the same peak, but no measurements have as yet been made. Materials and Methods : Fifteen tenors, fourteen baritones, seven sopranos and five mezzo sopranos attending the music college, department of vocal music who could reliably produce the head and chest registers were chosen for this study. Each subject was asked to produce an/ah/sound for at least three seconds for the head register sound(tenors ; G4, barions ; E4 sopranos ; F5 and mezzosopranos ; C5) and for the chest register sound (tenors ; C3, baritones ; D3, sopranos ; D4 and Mezzosoprano ; A3). The sound data was analyzed using the Fast Fourier Transform (FFT)-based power spectrum, Long term average(LTA) power spectrum using the FFT algorithm of the Computerized Speech Lab (CSL, Kay elemetrics, Model 4300B, USA). Statistical analysis was performed using the Mann-Whitney test of the Statistical Package for Social sciences(SPSS). Results : For head register sounds, a significant increase was seen in the 2,200-3,400Hz region(p<0.05) and the Similar to the head register sounds, there was a significant increase in energy in the four trained singer group compared with the untrained group in the 2,200-3,100Hz region(p<0.05), the 7,800-8,400Hz region(p<0.05) for the chest register sounds. Conclusions : When good vocal production was made for the head and chest registers, an energy peak was observed near 2,500Hz, a frequency already known as the "singer's formant', in all subjects in the study group. Another region of increased energy was observed around 8,000Hz that had not been noticed previously. The authors believe this region to be the second singer's formant.

• Journal of Korea Multimedia Society
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• v.22 no.11
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• pp.1233-1241
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• 2019

The goal of the sound-based mechanical fault diagnosis technique is to automatically find abnormal signals in the machine using acoustic emission. Conventional methods of using mathematical models have been found to be inaccurate due to the complexity of industrial mechanical systems and the existence of nonlinear factors such as noise. Therefore, any fault diagnosis issue can be treated as a pattern recognition problem. We propose an automatic fault diagnosis method using discrete wavelet transform and power spectrum density using multi pattern recognition. First, we perform DWT-based filtering analysis for noise cancelling and effective feature extraction. Next, the power spectral density(PSD) is performed on each subband of the DWT in order to effectively extract feature vectors of sound. Finally, each PSD data is extracted with the features of the classifier using multi pattern recognition. The results show that the proposed method can not only be used effectively to detect faults as well as apply to various automatic diagnosis system based on sound.

• Proceedings of the KIEE Conference
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• 2003.11c
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• pp.878-881
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• 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.

• Journal of Soil and Groundwater Environment
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• v.21 no.6
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• pp.128-134
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• 2016

Acoustic emission spectra was analyzed to investigate the distribution of sound pressure in a 36 kHz sonoreactor. The sound pressure of fundamental frequency (f: 36 kHz), harmonics (2f: 72 kHz, 3f: 108 kHz, 4f: 144 kHz, 5f: 180 kHz, 6f: 216 kHz), and subharmonics (1.5f: 54 kHz, 2.5f: 90 kHz, 3.5f: 126 kHz, 4.5f: 162 kHz, 5.5f: 198 kHz, 6.5f; 234 kHz) was measured at every 5 cm from the ultrasonic transducer using a hydrophone and a spectrum analyzer. It was revealed that the input power of ultrasound, the application of mechanical mixing, and the concentration of SDS affected the sound pressure distributions of the fundamental frequency and total detected frequencies frequencies significantly. Moreover a linear relationship was found between the average total sound pressure and the degree of sonochemical oxidation while there was no significant linear relationship between the average sound pressure of fundamental frequency and the degree of sonochemical oxidation.

• Journal of Biosystems Engineering
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• v.24 no.3
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• pp.253-258
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• 1999

This study was conducted to develop an automatic egg inspection system for detecting creaked eggs based on acoustic impulse response. This system includes a sound generator, a sound sensor with signal conditioner, and a computer. The sound generator that hit the sharp of the dull edges of an egg was constructed with a ceramic ball pendulum attached to a rotary type solenoid. The signal conditioner included a pre-amplifier and a digital signal processing (DSP) board. The parameters for distinguishing cracked and normal eggs were the area, the geometric centroid and the resonance frequency of power spectrum of the acoustic signal generated. An algorithm for on-line detection of the continuous transferring eggs was developed. The performance tests resulted with 91% success rate to separate cracked and normal eggs at the rate of 1 second per an egg.

• Korean Journal of Air-Conditioning and Refrigeration Engineering
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• v.26 no.2
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• pp.91-96
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• 2014

An acoustical model for detecting the leak location in a buried gas pipeline has been developed. This model is divided into an experimental model for sound diagnosis, and a theoretical model for sound prediction, which is based on the transfer matrix method, representing the sound pressure and the volume velocity as state variables. The power spectrum is measured by attaching only one microphone to the closed end pipe. It has been shown that the response magnitude of acoustic pressure signals calculated by the acoustical model depends upon the thickness and diameter of a pinhole. The validity for the acoustical model has been verified through a comparison between the measured and calculated results.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1995.04a
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• pp.196-201
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• 1995

The broadband and discrete sources of sound in small cooling fans of propeller type and centrifugal type were investigated to understand the turbulent vortex structures from many bladed fans using ANSI test plenum for small air-moving devices (AMDs). The noise measurement method uses the plenum as a test apparatus to determine the acoustic source spectral density function at each operating conditions similar to real engineering applications based on acoustic similarity laws. The characteristics of fans including the head rise vs. volumetric flow rate performance were measured using a performance test facility. The sound power spectrum is decomposed into two non-dimensional functions: an acoustic source spectral distribution function F(St,.phi.) and an acoustic system response function G(He,.phi.) where St, He, and .phi. are the Strouhal number, the Helmholtz number, and the volumetric flow rate coefficient, respectively. The autospectra of radiated noise measurements for the fan operating at several volumetric flow rates,.phi., are analyzed using acoustical similarity. The rotating stall in the small propeller fan with a bell-mouth guided is mainly due to a leading edge separation. It creates a blockage in the passage and the reduction in the flow rate. The sound power levels with respect to the rotational speeds were measured to reveal the mechanisms of stall and/or surge for different loading conditions and geometries, for example, fans installed with a impinging plate. Lee and Meecham (1993) studied the effect of the large-scale motions like impinging normally on a flat plate using Large-Eddy Simulation(LES) and Lighthill's analogy.[ASME Winter Annual Meeting 1993, 93-WA/NCA-22]. The dipole and quadrupole sources in the fans tested are shown closely related to the vortex structures involved using cross-correlations of the hot-wire and microphone signals.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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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.