• The Journal of the Acoustical Society of Korea
• /
• v.21 no.2E
• /
• pp.63-70
• /
• 2002

In this paper, an optimization technique for thin walled beams of vehicle body structure is proposed. Stiffness of thin walled beam structure is characterized by the thickness and typical section shape of the beam structure. Approximate functions for the section properties such as area, area moment of inertia, and torsional constant are derived by using the response surface method. The approximate functions can be used for the optimal design of the vehicle body that consists of complicated thin walled beams. A passenger car body structure is optimized to demonstrate the proposed technique.

• The Journal of the Acoustical Society of Korea
• /
• v.11 no.1
• /
• pp.67-79
• /
• 1992

For effective on-line ARMA parameter estimation, a covariance type ARMA fast transversal filter (FTF) algorithm is presented. The proposed algorithm is a covariance type implementation of ELS(Extended Least Squares) estimator and it is a fast time update recursion which is based on the fact that the correlation matrix of ARMA model satisfies the shift invariance property in each sub-block. The geometric approach is used in the derivation of the proposed algorithm. It takes small computational burden of 13N+37 MADPR(Multiplication And Division Per Recursion). Also, AR and MA orders can be independetly and arbitrarily specified.

• The Journal of the Acoustical Society of Korea
• /
• v.27 no.1E
• /
• pp.30-34
• /
• 2008

An improvement to the existing blind signal separation (BSS) method has been made in this paper. The proposed method models the inherent signal dependency observed in acoustic object to separate the real-world convolutive sound mixtures. The frequency domain approach requires solving the well known permutation problem, and the problem had been successfully solved by a vector representation of the sources whose multidimensional joint densities have a certain amount of dependency expressed by non-spherical distributions. Especially for speech signals, we observe strong dependencies across neighboring frequency bins and the decrease of those dependencies as the bins become far apart. The non-spherical joint density model proposed in this paper reflects this property of real-world speech signals. Experimental results show the improved performances over the spherical joint density representations.

• The Journal of the Acoustical Society of Korea
• /
• v.17 no.4E
• /
• pp.35-40
• /
• 1998

In this paper, an efficient quantization method of line spectrum pairs(LSP) with cascaded structure of vector quantizer and scalar quantizer is proposed. First, input LSP parameters is vector-quantized using a codebook a with a moderate number of entries. In the second stage of quantization, the components of residual vector are individually quantized by the scalar quantizer. The utilization of ordering property of LSP parameters and the inclusion of interframe prediction improve the quantizer performance and remove the stability check routine after quantization procedure. The new vector-scalar hybrid quantizer using 26 bits/frame shows a transparent quality of speech that an average spectral distortion is 1 dB and the frame proportion with above 2 dB spectral distortion is less than 2%. The performances of proposed quantization method is evaluated in the transmission errors.

• The Journal of the Acoustical Society of Korea
• /
• v.15 no.3E
• /
• pp.78-85
• /
• 1996

The set of admissible time-variations in the input signal can be separated into two categories : slow parameter changes and large parameter changes which occur infrequently. A common approach used in the tracking of slowly time-varying parameters is the exponential recursive least-squares(RLS) algorithm. There have been a variety of research works on the error analysis of the exponential RLS algorithm for the slowly time-varying parameters. In this paper, the focus has been given to the error analysis of exponential RLS algorithms for the input data with abrupt property changes. The voiced speech signal is chosen as the principal application. In order to analyze the error performance of the exponential RLS algorithm, deterministic properties of the exponential RLS algorithms is first analyzed for the case of abrupt parameter changes, the impulsive input(or error variance) synchronous to the abrupt change of parameter vectors actually enhances the convergence of the exponential RLS algorithm. The analysis has also been verified through simulations on the synthetic speech signal.

• Journal of the Korea Safety Management & Science
• /
• v.16 no.4
• /
• pp.391-396
• /
• 2014

For the purpose of finding out the acoustical characteristics of exterior materials in bridge, analytical studies are performed with boundary elements method by using the commercial program SYSNOISE. Before analysis, to figure out material property, it was conducted experiments of absorption coefficient for absorptive material. And prediction of pressure were conducted I GIRDER type (before & after installation of absorption panel ) and BOX GIRDER type (before & after installation of absorption panel) The results show that when the absorption panel is installed, environment around bridge can help reduce traffic noise. It was proved to be the effective noise reduction counter-plan for a traffic noise in the bridges.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.1
• /
• pp.30-35
• /
• 2001

The most important characteristic of Emile Bell's sound is a beating. It is modulation phenomenon which appears as a result of interference multiplication in time domain. This modulation phenomenon can be modeled as DSB-SC which suppress carrier and signals distributed both sides. The beatiog wave is observed in Laman distribution signal for polyvinyl speech signal, water vein wave, tide wave. The beating wave is caused by asymmetry Property of the bell.

• Proceedings of the Acoustical Society of Korea Conference
• /
• autumn
• /
• pp.197-200
• /
• 2000

This paper presents a time-based Pitch Determination Algorithm (PDA) for reliable estimation of pitch Period (PP) in speech signal. In proposed method, we use the Dyadic Wavelet Transform (DyWT), which detects the presence of Glottal Closure Instants (GCI) and uses the information to determine the pitch period. And, the proposed method also uses the periodicity property of DyWT to detect unsteady GCI. To evaluate the performance of the proposed methods, that of other PDAs based on DyWT are compared with what this paper proposed. The effectiveness of the proposed method is tested with real speech signals containing a transition between voiced and the unvoiced interval where the energy of voiced signal is unsteady. The result shows that the proposed method provides a good performance in estimating the both the unsteady GCI positions as well as the steady parts.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1994.06a
• /
• pp.1046-1051
• /
• 1994

It is known that SOFM has the property of effectively creating topographically the organized map of various features on input signals, SOFM can effectively be applied to the recognition of Korean phonemes. However, is isn't guaranteed that the network is sufficiently learned in SOFM algorithm. In order to solve this problem, we propose the learning algorithm combined with the conventional K-means clustering algorithm in fine-tuning stage. To evaluate the proposed algorithm, we performed speaker dependent recognition experiment using six phoneme classes. Comparing the performances of the Kohonen's algorithm with a proposed algorithm, we prove that the proposed algorithm is better than the conventional SOFM algorithm.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1994.06a
• /
• pp.904-911
• /
• 1994

Code Excited Linear Prediction (CELP) speech coders exhibit good performance at data rates below 4.8 kbps. The major drawback to CELP type coders is their many computation. In this paper, we propose a new pitch search method that preserves the quality of the CELP vocoder with reducing complexity. The basic idea is to apply the preprocessing technique beforehand grasping the autocorrelation property of speech waveform. By using the proposed method, we can get approximately 77% complexity reduction in the pitch search.