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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
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Journal DOI :
The Acoustical Society of Korea
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Volume & Issues
Volume 7, Issue 4 - Aug 1988
Volume 7, Issue 2 - Apr 1988
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Preferred Dealy Time and Subjective Preference Judgment for Sound Field with Single Reflection
Kang, Seong-Hoon ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 5~12
In order to know the preferred delay time of single reflection in relation in relation to the source signal, and to investigate whether or not there is any display in preference judgment of sound field between subjects of different nationalities, tests of subjective preference for musical sound fields with single reflection were preformed. The result showed that the preferred delay times agreed with the effective duration of auto-correlation function of the source signals, when the amplitude of reflection relative to the direct sound is 0dB. No fundamental disparity in series of judgement of sound field was found even for different series of Judgment with different music motifs. The result of preference test using different passages in single music showed that the fluctuation of the effective duration autocorrelation function over all the passages of the music was small. Thus, the preferred delay time can be determined by the coherence of autocorrelation function of the source signals and the amplitued of reflection.
Properties and Performance of Generalized Wilcoxon Filters
Song, Iick-Ho ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 13~21
In order to overcome the disadvantages of linear filters in certain cases of practical interest, a class of nonlinear filters(rank filters) are constructed based on a class of robust estimates, the rank estimates. A subclass of these filters, the limited-degree extended-averaging Wilcoxon filters, is then described as an interesting example of the rank filters with desirable characteristics. The properties of these filters are discussed and the performance of these filters are analyzed for ideal edges and narrow pulses.
Numerical Evaluation of The Rayleigh Integral Using the FFT Method for Transient Sound Radiation
Jeon, Jae-Jin ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 22~30
In this paper, the sound radiation from a clamped circular plate in an infinite baffle is calculated by using the FFT technique. The radiated sound fields are obtained by two-dimensional fast Fourier transform method is the spatial domain instead of a direct numerical evaluation of Rayleigh integral for economy of the computation time. The computation time is consumed at least by 1/200 times of the direct numerical evaluation on the Rayleigh integral in acoustic fields. The FFT method can be applicable to any shaped geometry as well as circular plate. The FFT solution could be very powerful in predicting the near and far fields of complex structures.
Time- and Frequency-Domain Block LMS Adaptive Digital Filters: Part Ⅰ- Realization Structures
Lee, Jae-Chon ; Un, Chong-Kwan ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 31~53
In this work we study extensively the structures and performance characteristics of the block least mean-square (BLMS) adaptive digital filters (ADF's) that can be realized efficiently using the fast Fourier transform (FFT). The weights of a BLMS ADF realized using the FFT can be adjusted either in the time domain or in the frequency domain, leading to the time-domain BLMS(TBLMS) algorithm or the frequency-domain BLMS (FBLMS) algorithm, respectively. In Part Ⅰof the paper, we first present new results on the overlap-add realization and the number-theoretic transform realization of the FBLMS ADF's. Then, we study how we can incorporate the concept of different frequency-weighting on the error signals and the self-orthogonalization of weight adjustment in the FBLMS ADF's , and also in the TBLMS ADF's. As a result, we show that the TBLMS ADF can also be made to have the same fast convergence speed as that of the self-orthogonalizing FBLMS ADF. Next, based on the properties of the sectioning operations in weight adjustment, we discuss unconstrained FBLMS algorithms that can reduce two FFT operations both for the overlap-save and overlap-add realizations. Finally, we investigate by computer simulation the effects of different parameter values and different algorithms on the convergence behaviors of the FBLMS and TBLMS ADF's. In Part Ⅱ of the paper, we will analyze the convergence characteristics of the TBLMS and FBLMS ADF's.
Time- and Frequency-Domain Block LMS Adaptive Digital Filters: Part Ⅱ - Performance Analysis
Lee, Jae-Chon ; Un, Chong-Kwan ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 54~76
In Part Ⅰ of the paper, we have developed various block least mean-square (BLMS) adaptive digital filters (ADF's) based on a unified matrix treatment. In Part Ⅱ we analyze the convergence behaviors of the self-orthogonalizing frequency-domain BLMS (FBLMS) ADF and the unconstrained FBLMS (UFBLMS) ADF both for the overlap-save and overlap-add sectioning methods. We first show that, unlike the FBLMS ADF with a constant convergence factor, the convergence behavior of the self-orthogonalizing FBLMS ADF is governed by the same autocorrelation matrix as that of the UFBLMS ADF. We then show that the optimum solution of the UFBLMS ADF is the same as that of the constrained FBLMS ADF when the filter length is sufficiently long. The mean of the weight vector of the UFBLMS ADF is also shown to converge to the optimum Wiener weight vector under a proper condition. However, the steady-state mean-squared error(MSE) of the UFBLMS ADF turns out to be slightly worse than that of the constrained algorithm if the same convergence constant is used in both cases. On the other hand, when the filter length is not sufficiently long, while the constrained FBLMS ADF yields poor performance, the performance of the UFBLMS ADF can be improved to some extent by utilizing its extended filter-length capability. As for the self-orthogonalizing FBLMS ADF, we study how we can approximate the autocorrelation matrix by a diagonal matrix in the frequency domain. We also analyze the steady-state MSE's of the self-orthogonalizing FBLMS ADF's with and without the constant. Finally, we present various simulation results to verify our analytical results.
Adaptive Noise Cancelling 법에 의한 기계이상진단 소프트웨어 개발 (제 1 보 : Cepstrum 해석)
Oh, Jae-Eung ; Kim, Jong-Kwan ; Park, Soo-Hong ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 77~85
Many kinds of conditioning monitoring technique have been studied, so this study has inverstigated the possibility of checking the trend in the fault diagnosis of ball bearing, one of the elements of rotating machine, by applying the cepstral analyisis method using the adaptive noise cancelling (ANC) method. And computer simulation is conducted in order to verify the usefulness of ANC. The optimal adaptation gain in adaptive filter is estimated, the performance of ANC according to the change of the signal to noise ratio and convergence of least mean square algorithm is considered by simulation. It is verified that cepstral analysis using ANC method is more effective than the conventional cepstral analysis method in bearing fault diagnosis.
Mathematical Modeling of Two-Dimensional Diffraction Analysis in Anisotropic Media
Chung, Yeong-Jee ; Jin, ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 86~97
An Efficient Adaptive Digital Filtering Algorithm for Identification of Second Order Volterra Systems
Hwang, Y.S. ; Mathews, V.J. ; Cha, I.W. ; Youn, D.H. ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 98~109
This paper introduces an adaptive nonlinear filtering algorithm that uses the sequential regression(SER) method to update the second order Volterra filter coefficients in a recursive way. Conventionally, the SER method has been used to invert large matrices which result from direct application of Wiener filter theory to the Volterra filter. However, the algorithm proposed in this paper uses the SER approach to update the least squares solution which is derived for Gaussian input signals. In such an algorithm, the size of the matrix to be inverted is smaller than that of conventional approaches, and hence the proposed method is computationally simpler than conventional nonlinear system identification techniques. Simulation results are presented to demonstrate the performance of the proposed algorithm.
The Effects of Thermal Front on Sound Propagation in Shallow Seas of Korea
Na, Jung-Yul ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 110~116
The thermal front over the shallow coastal seas of Korea during the winter season provides very unique acoustic media such that wave equation is easily separable and the solutions turn out to be very simple and well known. In steady of using the WKB method to solve the radial equation the mode technique have been applied to obtain the solution. The radial propagation is rather weakly influenced by the presence of the thermal front that causes the horizontal variations of the sound speed. The physical description of the sound propagation is also presented in terms of ray tracing
Scattering Sound by a Flexinble Cylindrical Cavity
Kim, Yu-Man ; Lee, Byung-Ho ;
The Journal of the Acoustical Society of Korea, volume 7, issue 4, 1988, Pages 117~126
The pressure waves scattered by an infinite cylindrical cavity filled with air in a h0mogeneous medium have been calculated for the incident plane pressure waves. For ka = 1/2, 1, 2, 4, 10 and 20, the scattered pressure waves are plotted., where k is the wave number and a is the radius of the cylindrical hole. As an indicator of the directivity of the scattering pattern, we have defined the angle at which the magnitude of the scattered pressure wave decreases by a half(6 dB) with respect to that of the forward peak scattered pressure wave. This angle depends strongly on the values of ka and the distance r, and the angle can be used for the detection of the location and the size of the cavity in a homogeneous medium.