• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.11 no.4
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• pp.58-64
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• 2001

Sound absorbing materials are applied to the exposed surfaces of curvet subway tunnel for the reduction of curving noise level. Before the treatment, acoustical engineering simulation is performed to predict the noise level reduction for different kinds and amounts of absorbing material. The principle of geometrical acoustics is utilized to perform the simulation efficiently and accurately. The noise bevels of the inside and outside of running car body are measured to find the noise level reduction. The average noise level reduction of 8 dB has been attained. It has been shown that the simulated results are comparable to the measured ones.

• Journal of the Korean Institute of Telematics and Electronics
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• v.10 no.5
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• pp.31-38
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• 1973

The Sabine's formula has been widlely used for calculating reverberation time and applied for actual systems. The result of Sabine's method is only same as the reverberation time of one axial wave according to the wave theory. Reverberation time is mainly dependent on the standing waves. In case of the rectangular room the frequencies of three mode covering 250Hz and several intensities at various positions of the room were measures by a spectrograph. It wart found that axial wavers and tangential waves decayed more slowly than oblique waves. The experimental results showed that the amount of axial and tangential wave in a frequency band varies depending on the position in the room. It is concluded that the results give to control reverberation times in a room.

• The Journal of the Acoustical Society of Korea
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• v.36 no.6
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• pp.419-424
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• 2017

TDoA (Time Difference-of Arrival) or DoA (Direction-of-Arrival) can be used for source localization. However, the localizing performance is dependent on relative position between source and receivers, receivers' geometric structure, sound speed, and so on. In this paper we propose a source localization method with enhanced performance that combines multiple information. The proposed method uses the time TDoA, DoA and sound speed as variables. LM (Levenberg-Marquardt) method which is one of nonlinear optimizations is applied. The performances of the proposed method was evaluated by simulation. As result of simulation, the proposed method has the lower average localizing error performance than the previous method.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.22-28
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• 2001

This paper describes the implementation of a robust speaker position location system using the voice signal received by microphone array. To be robust to the reverberation which is the major factor of the performance degradation, low-frequency phase restoration algorithm which eliminates the influence of reverberations using the low-frequency information of the CPSP function is proposed. The implemented real-time system consists of a general purpose DSP (TMS320C31 of Texas instruments), analog part which contains amplifiers and filters, and digital part which is composed of the external memory and 12-bit A/D converter. In the real conference room environment, the implemented system that was constructed by the proposed algorithms showed better performance than the conventional system. The error of the TDOA estimation reduced more than 15 samples.

• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.576-582
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• 2004

A computer code that is based on the Pseudo-spectral finite difference algorithm using staggered grid is developed for the wave propagation modeling in the time domain. The advantage of a finite difference approximation is that any geometrically complicated media can be modeled. Staggered grids are advantageous as it provides much more accuracy than using a regular grid. Pseudo-spectral methods are those that evaluate spatial derivatives by multiplying a wavenumber by the Fourier transform of a pressure wave-field and performing the inverse Fourier transform. This method is very stable and reduces memory and the number of computations. The synthetic results by this algorithm agree with the analytic solution in the infinite and half space. The time domain modeling was implemented in various models. such as half-space. Pekeris waveguide, and range dependent environment. The snapshots showing the total wave-field reveals the Propagation characteristic or the acoustic waves through the complex ocean environment.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.340-348
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• 2016

Ray-based model for the calculation of the ocean surface-generated ambient noise coherence function has the form of double integral with respect to a range and a bearing angle. While the theoretical consideration about its numerical implementations was partly given in the past work of authors, the numerical results on the ocean environment have not been presented yet. In this paper, we perform numerical experiments for shallow and deep water environments. It is shown that the coherence function depends on the ocean sediment sound speed, and is more sensitive to the ocean sediment sound speed in the shallow water than in the deep water. Similar trend is also observed for varying the orientation of hydrophone pair. In addition, a post-processing technique is proposed in order to plot the noise intensity for the noise receiving angle. This procedure will supplement the weakness of the ray-based model about the output data type compared to the semi-analytic model of Harrison.

• 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.33 no.6
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• pp.358-365
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• 2014

When a ship propeller having wing type sections rotates at high speed underwater, local pressure on the blade decreases and various types of the cavitation inevitably occur where the local pressure falls below the vapor pressure. Fundamentally characteristics of the cavitation are determined by the shapes of the blade section and their operating conditions. Underwater noise radiated from a ship propeller is directly connected to the occurrence of the cavitation. In order to design low noise propeller, it is preferentially demanded to figure out key features: how the cavity is generated, developed and collapsed and how the effect of viscosity works in the process. In this study, we first perform inviscid analysis of the partial cavity generated on two dimensional hydrofoil. Secondly, viscous analysis using FLUENT with different turbulence and cavitation models are presented. Results from both approaches are also compared and estimated.

• The Journal of the Acoustical Society of Korea
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• v.35 no.6
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• pp.452-465
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• 2016

In this study, the structure of a concave annular array transducer was optimized to generate high intensity focused ultrasound for medical therapeutic application. The transducer has a phased array structure composed of several concentric channels that have 40 mm as the radius of curvature. We derived theoretical equations to analyze the sound field of the transducer and verified the validity of the equations by comparing the results calculated by the equations with those from finite element analyses. We also checked the possibility of dynamic focusing at around the geometric focal point. Further, the level of a grating lobe occurring at an unwanted position in the transducer sound field was confirmed to be reducible through the relation between the number of channels and the frequency of the transducer. Hence, the structure of the transducer was optimized to place the main lobe within a specific range from the zenith while systematically reducing the level of the maximum sidelobe including the grating lobe. The designed structure showed the performance better than that targeted at all the focal points.

• The Journal of the Acoustical Society of Korea
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• v.27 no.6
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• pp.319-324
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• 2008

In this paper, we propose a novel approach to improve the performance of speech/music classification for the selectable mode vocoder (SMV) of 3GPP2 using the discriminative weight training which is based on the minimum classification error (MCE) algorithm. We first present an effective analysis of the features and the classification method adopted in the conventional SMV. And then proposed the speech/music decision rule is expressed as the geometric mean of optimally weighted features which are selected from the SMV. The performance of the proposed algorithm is evaluated under various conditions and yields better results compared with the conventional scheme of the SMV.