• The Journal of the Acoustical Society of Korea
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• v.6 no.2
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• pp.38-47
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• 1987

As Korean language can be phonemically classified according to the characteristic and structure of its pronunciation, Korean syllables can be divided into the phonemes such as consonant and vowel. The divided phonemes are analyzed by using the method of partial autocorrelation, and the order of partial autocorelation coefficient is 15. In analysis, it is shown that each characteristic of the same consonants, vowels, and end consonant in syllables in similar. The experiments is carried out by dividing 675 syllables into consonants, vowels, and end consonants. The recognition rate of consonants, vowels, end-consonants, and syllables are $85.0(\%)$, $90.7(\%)$, $85.5(\%)$and $72.1(\%)$ respectively. In conclusion, it is shown that Korean syllables, divided by the phonemes, are analyzed and recognized with minimum data and short processing time. Furthermore, it is shown that Korean syllables, words and sentences are recognized in the same way.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.83-91
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• 2001

This paper proposes the Focused beamforming to estimate the location of target residing near to the observation platform in the underwater environment. The Focused beamforming technique provides the location of target by the coherent summation of a series of incident spherical waveforms considering distinct propagation delay times at the sensor array. But due to the movement of the observation platform and the variation of the underwater environment, the shape of the sensor array is no longer to be linear but it becomes distorted as the platform moves. Thus the Focused beamforming should be peformed regarding to the geometric shape variation at each time. To estimate the target location, the artificial image plane comprised of cells is constructed, and the delays are calculated from each cell where the target could be proximity to sensors for the coherent summation. After the coherent combining, the beam pattern can be obtained through the Focused beamforming on the image plane. Futhermore to compensate the variation of the shape of the sensor array, the paper utilizes the Nth-order polynomial approximation to estimate the shape of the sensor array obeying the water pulley modeling. Simulation results show the performance of the Focused beamforming for different frequency bands of the radiated signal.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.243-252
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• 2016

In this paper, the incoherent reverberation model based on coupled normal mode method is presented. In the range dependent environment, one way coupled normal mode method is used to calculate the pressure from a source to a scatterer patch and the pressure from a scatterer patch to a receiver. For the computational efficiency, the sound propagation from a source/receiver to the scatterer patch is assumed to occur only in the 2D plane where a source/receiver and scatterer patch are located. For the model verification, problems of the reverberation modeling workshop I and II sponsored by the US office of Naval Research are calculated and the results are compared with the incoherent reverberation model results based on the ray method.

• Journal of the Korean Society for Nondestructive Testing
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• v.19 no.2
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• pp.100-109
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• 1999

A method is proposed to characterize single-fiber composite interphases from the frequency-domain characteristics of scattered ultrasonic waves, and its feasibility is investigated theoretically. It has been shown that the locations and magnitudes of the peaks and valleys in the frequency domain are affected significantly by the interphase properties, which may indicate the effectiveness of the proposed method. Although the frequency-domain behavior is basically associated with the resonance of the fiber-interphase system, it is not dominantly affected by the scatterer's resonance unlike that in the case of acoustic wave scattering. Therefore, the conventional acoustic resonant scattering theory is not directly applicable to the characterization of composite interphases. In order to solve the inverse problem of predicting the interphase properties from the frequency-domain characteristics of the ultrasonic scattered waves, an artificial neural network has been constructed. This approach has demonstrated reasonable accuracy in most cases considered in this study.

• Journal of the Korean Society for Nondestructive Testing
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• v.16 no.3
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• pp.155-161
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• 1996

For the condition monitoring of the journal bearing in rotating machinery, a system for their diagnosis by acoustic emission(AE) was developed. AE has been used to detect abnormal conditions in the bearing system. It was found from the field application study as well as the laboratory experiment using a simulated journal bearing system that AE RMS voltage was the most efficient parameter for the purpose of current study. Based on the above results, algorithms and judgement criteria for the diagnosis system was established. The system is composed of four parts as follows: the sensing part including AE sensor and preamplifier, the signal processing part for RMS-to-DC conversion to measure AE ms voltage, the interface part for transferring RMS voltage data into PC using A/D converter, and the software part including the graphic display of bearing conditions and the diagnosis program.

• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.65-70
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• 2000

It is blown that the loudness is one of the most important metrics in assessing the sound quality and a calculation method for loudness has been standardized for steady sounds. In this study, a new loudness model is suggested for dealing with the transient sound for a unified analysis of various practical sounds. A signal processing technique is introduced for this purpose, which is required for the band subdivision and the prediction of band-level change of transient sounds. In addition, models for the post-masking and the temporal integration are adopted in the analysis of the loudness of transient sounds. In order to solve the problem of the conventional loudness model in the pure-tone signal processing, a critical band filter is employed in the analysis, which consists of 47 critical filters having a filter spacing of a half of the critical bandwidth. For testing the effectiveness of the present model, the predicted responses are compared with the experimental data and it is observed that they are in good agreements.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.46-53
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• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.410-418
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• 2012

Parametric stereo(PS) and MPEG surround(MPS) are major spatial audio coding(SAC) tools. In this paper, the problem of the inter-channel correlation(ICC) synthesis in the conventional SAC is analyzed. Conventional methods assume that ambient components mixed to two output channels are anti-phased, while the primary components are assumed to be in-phased. This assumption can cause excessive ambient mixing for a negative-valued ICC. As a remedy to this problem, we propose a new ICC synthesis method based on an assumption that the primary components are anti-phased each other for a negative ICC. The proposed method is also applied to the approximation which works in practice. The performance of the proposed method was evaluated by computer simulations and the subjective listening tests verified that the proposed method is effective in not only headphones but also loudspeakers playback.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.24-32
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• 1997

This study aims to describe, at one aspect, the relativity between syntactic structure and prosodic phrasing, and at the other, to establish a suitable phrasing pattern to produce more natural synthetic speech. To get meaningful results, all the word boundaries in the prosodic database were statistically analyzed, and assigned by the proper boundary type. The resulting 10 types of prosodic boundaries were classified into 3 types according to the strength of the breaks, which are zero, minor, and major break respectively. We have found out that the durational information was a main cue to determine the major prosodic boundary. Using the bigram and trigram of syntactic information, we predicted major and minor classification of boundary types. With brigram model, we obtained the correct major break prediction rates of 4.60%, 38.2%, the insertion error rates of 22.8%, 8.4% on each Test-I and Test-II text database respectively. With trigram mode, we also obtained the correct major break prediction rates of 58.3%, 42.8%, the insertion error rates of 30.8%, 42.8%, the insertion error rates of 30.8%, 11.8% on Test-I and Test-II text database respectively.

• The Journal of the Acoustical Society of Korea
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• v.37 no.2
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• pp.92-98
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• 2018

This paper is a study on the generation mechanism of propeller singing based on the cavitation tunnel test, underwater impact test, finite element analysis and computational flow analysis for the model propeller. A wire screen mesh, a propeller and a rudder were installed to simulate ship stern flow, and occurrence and disappearance of propeller singing phenomenon were measured by hydrophone and accelerometer. The natural frequencies of propeller blades were predicted through finite element analysis and verified by contact and non-contact impact tests. The flow velocity and effective angle of attack for each section of the propeller blades were calculated using RANS (Reynolds Averaged Navier-Stokes) equation-based computational fluid analysis. Using the high resolution analysis based on detached eddy simulation, the vortex shedding frequency calculation was performed. The numerical predicted vortex shedding frequency was confirmed to be consistent with the singing frequency and blade natural frequency measured by the model test.