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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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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 23, Issue 8 - Nov 2004
Volume 23, Issue 7 - Oct 2004
Volume 23, Issue 6 - Aug 2004
Volume 23, Issue 5 - Jul 2004
Volume 23, Issue 4 - May 2004
Volume 23, Issue 3 - Apr 2004
Volume 23, Issue 2 - Feb 2004
Volume 23, Issue 1 - Jan 2004
Volume 23, Issue 4E - 00 2004
Volume 23, Issue 3E - 00 2004
Volume 23, Issue 2E - 00 2004
Volume 23, Issue 1_E - 00 2004
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Study of Acoustic Streaming at Resonance by Longitudinal Ultrasonic Vibration Using Particle Imaging Velocimetry
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 340~352
Acoustic streaming induced by the microscopic longitudinal ultrasonic vibration at 28.5 ㎑ is visualized between the quiescent glass plate and ultrasonic vibrator by particle imaging velocimetry(PIV) using laser. To investigate the augmentation of air flow velocity of acoustic streaming. the velocity variations of air streaming between the stationary plate and ultrasonic vibrator are measured in real-time. It is experimentally investigated that the magnitude of the acoustic streaming dependent upon the gap between the ultrasonic vibrator and stationary p1ate results in the variations of the average velocity fields as a outcome of the bulk air flow caused by the ultrasonic vibration. In addition. maximum acoustic streaming velocity exists at resonant gap. 18mm that is one of the resonant gaps (H=18, 24, 30, 36㎜) at which resonance occurs. The variation of the local maximum turbulent intensity with axial direction appear to reveal the value of 8%∼70% dependent upon the gap between the quiescent glass plate and ultrasonic vibrator. Shearstress is also maximized at the center region of the vibrator and the vorticity is also maximum and minimum in the neighborhood of the center of the vibrator at which the local maximum turbulent intensity and shear stress exist.
A Study on Propagation of Guided Waves in a Steam Generator Tube
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 353~361
Propagation of the guided waves in a steam generator (SG) tube was investigated. Dispersion curves and the incident angles corresponding to the specific modes were calculated for the SG tube. The modes of guided wave were identified by time-frequency diagrams obtained by short time Fourier transform. Group velocities were also determined from the time-frequency diagrams obtained at the different separations of transducers. In experiment. distinct mode conversion was not observed when the guided ultrasound passed curved region of the S/G tube. The optimum mode of guided wave for the inspection of SG tube was suggested and verified by experiments.
Frequency Dependence of High-Frequency Bottom Reflection Loss Model
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 362~369
The High-frequency (30 ∼ 120 ㎑) bottom reflection loss at rough water-sediment interface is affected by the gram size distribution of the sediments. The roughness of the bottom surface is represented by "acoustical roughness. g/sub R/" The grain size of sandy sediments is g/sub R/∼O(1) and the dependence as a function of frequency. We suggest the modified bottom reflection loss model (HYBRL model , HanYang university Bottom Reflection Loss model) that include in the deviation of the reflection loss as a function of the grain size distribution and frequency dependence. And bottom reflection loss model of frequency dependence and deviation of bottom properties is verified by water tank and field experiments.
A Study on Three Dimensional Array Shape Calibration of the Bottom Mounted Array by Iterative Least Squares
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 370~375
This paper proposes an algorithm that estimates three dimensional array shape calibration about the bottom-mounted sensor array. under the assumption that the active sources are in the far-field with unknown positions. Under some assumptions. we calculate the sensor positions via an algebraic solutions of a least squares problem that the linear equations are related to the sensor positions and directions or arrival. We give examples of algorithm performance from both computer simulations and sea test. We also illustrate the performance of sensor positions estimation as a function of time delay estimation variance and the distribution of the localizing sources.
Reduction of Radiated Noise by Eigen-property Control
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 376~382
The interaction between a vibrating structure and a surrounding acoustic medium determines the acoustic power propagating into the far-field. A straightforward method to reduce the radiated power is to reduce the vibration of the structure. However it is more efficient to control the modes of the structure separately since each vibration mode of the structure has different radiation efficiency. An efficient method to reduce the sound radiation in the low frequency region is proposed by reducing the radiation efficiency of the structure. Numerical simulations are carried out for a simply-supported beam in which the feed-forward control is applied to reduce the volume velocity of each structural mode. This method is found to be very efficient in reducing low frequency sound radiation.
Performance Improvement of Rapid Speaker Adaptation Using Bias Compensation and Mean of Dimensional Eigenvoice Models
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 383~389
In this paper. we propose the bias compensation methods and the eigenvoice method using the mean of dimensional eigenvoice to improve the performance of rapid speaker adaptation based on eigenvoice under mismatch between training and test environment. Experimental results for vocabulary-independent word recognition task (using PBW 452 DB) show that the proposed methods yield improvements for small adaptation data. We obtained about 22∼30% relative improvement by the bias compensation methods as amount of adaptation data varied from 1 to 50, and obtained 41% relative improvement in error rate by the eigenvoice method using the mean of dimensional eigenvoice with only single adaptation word.
A Design and Implementation of the VoiceXML Multiple-View Editor Using MVC Framework
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 390~399
In this paper, we design and implement a multiple-view VoiceXML editor to improve editing efficiency of the VoiceXML. The VoiceXML multiple-view Editor uses a MVC framework to support multiple views and paradigm. Our multiple-view editor consists of Model. View and Controller using MVC framework. A model, core data structure. is constructed of abstract syntax tree and abstract grammar. A view. user interface. is formalized in unparsing rules and unparser. A controller. to control model and view. is made of command interpreter and tree handler. The VoiceXML multiple-view editor overcomes a drawbacks of existing XML editors by showing document structure and context concurrently. as well as document flows. Our VoiceXML multiple-view editor. which MVC framework has been applied, provides various editing views concurrently to users. Thereby. it supports efficient and convenient editing environments for voice-web documents to users and it guarantees transparency of editors. as various views have a same consistent model.
Voice Personality Transformation Using an Optimum Classification and Transformation
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 400~409
In this paper. a voice personality transformation method is proposed. which makes one person's voice sound like another person's voice. To transform the voice personality. vocal tract transfer function is used as a transformation parameter. Comparing with previous methods. the proposed method makes transformed speech closer to target speaker's voice in both subjective and objective points of view. Conversion between vocal tract transfer functions is implemented by classification of entire vector space followed by linear transformation for each cluster. LPC cepstrum is used as a feature parameter. A joint classification and transformation method is proposed, where optimum clusters and transformation matrices are simultaneously estimated in the sense of a minimum mean square error criterion. To evaluate the performance of the proposed method. transformation rules are generated from 150 sentences uttered by three male and on female speakers. These rules are then applied to another 150 sentences uttered by the same speakers. and objective evaluation and subjective listening tests are performed.
Bayesian Fusion of Confidence Measures for Confidence Scoring
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 410~419
In this paper. we propose a method of confidence measure fusion under Bayesian framework for speech recognition. Centralized and distributed schemes are considered for confidence measure fusion. Centralized fusion is feature level fusion which combines the values of individual confidence scores and makes a final decision. In contrast. distributed fusion is decision level fusion which combines the individual decision makings made by each individual confidence measuring method. Optimal Bayesian fusion rules for centralized and distributed cases are presented. In isolated word Out-of-Vocabulary (OOV) rejection experiments. centralized Bayesian fusion shows over 13% relative equal error rate (EER) reduction compared with the individual confidence measure methods. In contrast. the distributed Bayesian fusion shows no significant performance increase.
Real-Time Implementation of the EHSX Speech Coder Using a Floating Point DSP
The Journal of the Acoustical Society of Korea, volume 23, issue 5, 2004, Pages 420~427
This paper presents real time implementation of 4kbps EHSX (Enhanced Harmonic Stochastic Excitation) speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding for voiced frames and used the vector excitation coding with the structure of analysis-by-synthesis for unvoiced frames, respectively. For transition frames mixed with voiced and unvoiced signal, we use the time-separated transition coding. In this paper. we present the optimization methods of implementation speech coder on the EMS320C6701/sup (R)/ DSP. To reduce the complex for real-time implementation. we perform the optimization method in algorithm by replacing the complex sinusoidal synthesis method with IFFT. and we apply fully pipelines hand assembly coding after converting it from floating source to fixed source. To generate a more efficient code. we also make use or the available EMS320C6701/sup (R)/ resources such as Fastest67x library and memory organization.