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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 22, Issue 8 - Nov 2003
Volume 22, Issue 7 - Oct 2003
Volume 22, Issue 6 - Aug 2003
Volume 22, Issue 5 - Jul 2003
Volume 22, Issue 4 - May 2003
Volume 22, Issue 3 - Apr 2003
Volume 22, Issue 2 - Feb 2003
Volume 22, Issue 1 - Jan 2003
Volume 22, Issue 1E - 00 2003
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Ultrasound Transducers in Several Tens MHz Band and Their Applications
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 622~628
Recently, ultrasound transducers in several tens MHz band, which can give the spatial resolution higher than several tens micrometers, have been intensively studying for applying to medical diagnostic fields of ophthalmology and dermatology. In this paper, the background of the studies, structures and characteristics of the transducers, and images obtained by the transducers are briefly reviewed.
Directivity Characteristics Control of Ultrasonic Transducer Array Using Two-layered Piezoelectric Transducer
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 629~636
It will be very convenient if the directivity characteristics of ultrasonic transducer array are controllable by the purpose of use in the fields of sonar system or ultrasonic diagnostic system, In this paper, a control method of the directivity characteristics was suggested. The transducer array was consisted of two-layered piezoelectric vibrators. Efficiency of each vibrator is controlled in 2nd harmonic mode by electrical capacitance. Therefore, the beam width of the transducer array can be controlled by changing the capacitance. The directivity characteristics of the array were analyzed experimentally and theoretically. As the results, it is confirmed that -3 dB beam width of main lobe can be controlled in the range of 7.6°∼16.2°.
Optimal Structural Design of a Tonpilz Transducer by Means of the Finite Element Method
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 637~644
In this study, with the FEM we analyzed the variation of the resonance frequency, bandwidth, and sound pressure of the Tonpilz transducer in relation to its design variables. Through statistical multiple regression analysis of the results, we derived functional forms of the resonance frequency, bandwidth, and sound pressure in terms of the design variables. By applying the constrained optimization technique, SQP-PD, to the derived function, we determined the optimal structure of the transducer that could provide the highest sound pressure level at the resonance frequency of 30,000 Hz and having the -3 dB bandwidth more than 10%, The validity of the optimized results was confirmed through comparison of the optimal performance with that of the FEA. The optimal design method proposed could reflect all the cross-coupled effects of multiple structural variables, and could determine the detailed geometry of the transducer with great efficiency and rapidity.
Revised Beamforming Inversion Method for Ocean Acoustic Tomography
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 645~651
This paper presents a revised beamforming inversion method for ocean acoustic tomography. In the proposed inversion method, the relation between group velocity and phase velocity that are the characteristics of the waveguide is used for the inversion of perturbed sound speed profile. The group velocity and phase velocity can be expressed as a function of the travel time and arrival angle of the received signals that are analyzed by the beamforming signal processing. This paper illustrates the simulated results of inversion for the fluctuated sound speed profile of the East Korea Sea and we found the applicability of revised beamforming inversion method to range independent ocean.
Frequency Dependence of High-frequency Bottom Reflection Loss Measurements
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 652~659
High-frequency(40∼120 kHz) reflection loss measurements on the water-sandy sediment with a flat interface were conducted in a water tank for various grazing angles. The water tank(5×5×5 m) was filled with a 0.5 m-thick-flat bottom of 0.5ø-mean-grain-size sand. Reflection losses, which were experimentally obtained as a function of grazing angle and frequency, were compared with the forward loss model, APL-UW model (Mourad & Jackson, 1989). For frequencies below 60 kHz, the observed losses well agree with the reflection loss model, however, in cases for frequencies above 70 kHz, the observed losses are greater by 2∼3 dB than the model results. The model calculation, which does not fully account for the vertical scale of roughness due to grain size, produce less bottom losses compared to the observations that correspond to large roughness based on the Rayleigh parameter in the wave scattering theory. In conclusion, for the same grain-size-sediment, as frequencies increase, the grainsize becomes the scale of roughness that could be very large for the frequencies above 70 kHz. Therefore, although the sea bottom was flat, we have to consider the frequency dependence of an effect of roughness within confidential interval of grain size distribution in reflection loss model.
Analysis of the Sound Source Field Using Spatial Transformation of the Sound Pressure in a Near-field
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 660~669
This paper describes a theory to calculate sound source field from the spatial transform of sound field and the measured cross-power spectrum of sound pressure over a hologram plane close to a sound source, Calculating method is proposed to solve sound pressures from cross-power spectrums over a hologram plane, For this, Taylor series for the nonlinear equations is expanded, and it is calculated using Newton-Raphon method, Also, a wave number filter is used to reduce errors that is occurred on the backward propagation, and is performed numerical simulation of the circular piston sound source with infinite baffle in water to verify the proposed theory.
Underwater Transient Signal Detection Using Higher-order Statistics and Wavelet Analysis
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 670~679
This paper deals with application of wavelet transform, which is known to be good for time-frequency analysis, in order to detect the underwater transient signals embedded in ambient noise. A new detector of acoustic transient signals is presented. It combines two detection tools: wavelet analysis and higher-order statistics. Using both techniques, the detection of the transient signal is possible in low signal to noise ratio condition. The proposed algorithm uses the wavelet transform of a partition of the signal on frequency domain, and then higher-order statistics tests the Gaussian nature of the segments.
Delay Characteristics and Sound Quality of Space Based Digital Waveguide Model
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 680~686
Digital waveguide model is a general method that is used in physical modeling of musical instruments. Wave motion is analyzed by time or by space in digital waveguide model. Because sampling is made via time, it is general that musical instrument model is described by wave motion of time. In this paper, we synthesized the musical instrument sound by adding instrument body model to the spatial based string model. In this way, we could improve sound quality and process musical instrument model's tone control variables effectively. We explained about delay error that happens in string and body in space based sampling and showed method to process fractional delay using FD (Fractional Delay)filter. Finally, we explained the relation between tone quality and number of delays. And we also compared the result with time base digital waveguide model.
Audio Enhancement Algorithm Using Adaptive Perceptual Filter
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 687~693
In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.
Perceptual-phonemic Contrasts of Single-word Intelligibility for Testing Korean Dysarthric Speech
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 694~702
The word intelligibility test for dysarthric speakers was designed to examine phonetic contrasts that are likely (1) to be sensitive to intelligibility impairment and (2) to contribute significantly to speech intelligibility. These phonetically contrasting word pairs were tested and proved to be reliable and to be valid, The results showed that in Korean dysarthric patients, the percentage of error in final position contrast was higher than in any other position. Unlike the results of previous studies, the initial-position contrasts were crucial in predicting the overall intelligibility among Korean patients.
Pruning Methodology for Reducing the Size of Speech DB for Corpus-based TTS Systems
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 703~710
Because of their human-like synthesized speech quality, recently Corpus-Based Text-To-Speech(CB-TTS) have been actively studied worldwide. However, due to their large size speech database (DB), their application is very restricted. In this paper we propose and evaluate three DB reduction algorithms to which are designed to solve the above drawback. The first method is based on a K-means clustering approach, which selects k-representatives among multiple instances. The second method is keeping only those unit instances that are selected during synthesis, using a domain-restricted text as input to the synthesizer. The third method is a kind of hybrid approach of the above two methods and is using a large text as input in the system. After synthesizing the given sentences, the used unit instances and their occurrence information is extracted. As next step a modified K-means clustering is applied, which takes into account also the occurrence information of the selected unit instances, Finally we compare three pruning methods by evaluating the synthesized speech quality for the similar DB reduction rate, Based on perceptual listening tests, we concluded that the last method shows the best performance among three algorithms. More than this, the results show that the last method is able to reduce DB size without speech quality looses.
Gaussian Density Selection Method of CDHMM in Speaker Recognition
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 711~716
This paper proposes the method to select the number of optimal mixtures in each state in Continuous Density HMM (Hidden Markov Models), Previously, researchers used the same number of mixture components in each state of HMM regardless spectral characteristic of speaker, To model each speaker as accurately as possible, we propose to use a different number of mixture components for each state, Selection of mixture components considered the probability value of mixture by each state that affects much parameter estimation of continuous density HMM, Also, we use PCA (principal component analysis) to reduce the correlation and obtain the system' stability when it is reduced the number of mixture components, We experiment it when the proposed method used average 10% small mixture components than the conventional HMM, When experiment result is only applied selection of mixture components, the proposed method could get the similar performance, When we used principal component analysis, the feature vector of the 16 order could get the performance decrease of average 0,35% and the 25 order performance improvement of average 0.65%.
Wavelet-based Pitch Detector for 2.4 kbps Harmonic-CELP Coder
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 717~726
This paper presents the methods that design the Wavelet-based pitch detector for 2,4 kbps Harmonic-CELP Coder, and that achieve the effective waveform interpolation by decision window shape of the transition region, Waveform interpolation coder operates by encoding one pitch-period-sized segment, a prototype segment, of speech for each frame, generate the smooth waveform interpolation between the prototype segments for voiced frame, But, harmonic synthesis of the prototype waveforms between previous frame and current frame occur not only waveform errors but also discontinuity at frame boundary on that case of pitch halving or doubling, In addtion, in transition region since waveform interpolation coder synthesizes the excitation waveform by using overlap-add with triangularity window, therefore, Harmonic-CELP fail to model the instantaneous increasing speech and synthesis waveform linearly increases, First of all, in order to detect the precise pitch period, we use the hybrid 1st pitch detector, and increse the precision by using 2nd ACF-pitch detector, Next, in order to modify excitation window, we detect the onset, offset of frame by GCI, As the result, pitch doubling is removed and pitch error rate is decreased 5.4% in comparison with ACF, and is decreased 2,66% in comparison with wavelet detector, MOS test improve 0.13 at transition region.
A Study on Regression Class Generation of MLLR Adaptation Using State Level Sharing
The Journal of the Acoustical Society of Korea, volume 22, issue 8, 2003, Pages 727~739
In this paper, we propose a generation method of regression classes for adaptation in the HM-Net (Hidden Markov Network) system. The MLLR (Maximum Likelihood Linear Regression) adaptation approach is applied to the HM-Net speech recognition system for expressing the characteristics of speaker effectively and the use of HM-Net in various tasks. For the state level sharing, the context domain state splitting of PDT-SSS (Phonetic Decision Tree-based Successive State Splitting) algorithm, which has the contextual and time domain clustering, is adopted. In each state of contextual domain, the desired phoneme classes are determined by splitting the context information (classes) including target speaker's speech data. The number of adaptation parameters, such as means and variances, is autonomously controlled by contextual domain state splitting of PDT-SSS, depending on the context information and the amount of adaptation utterances from a new speaker. The experiments are performed to verify the effectiveness of the proposed method on the KLE (The center for Korean Language Engineering) 452 data and YNU (Yeungnam Dniv) 200 data. The experimental results show that the accuracies of phone, word, and sentence recognition system increased by 34∼37%, 9%, and 20%, respectively, Compared with performance according to the length of adaptation utterances, the performance are also significantly improved even in short adaptation utterances. Therefore, we can argue that the proposed regression class method is well applied to HM-Net speech recognition system employing MLLR speaker adaptation.