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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 19, Issue 8 - Nov 2000
Volume 19, Issue 7 - Oct 2000
Volume 19, Issue 6 - Aug 2000
Volume 19, Issue 5 - Jul 2000
Volume 19, Issue 4 - May 2000
Volume 19, Issue 3 - Apr 2000
Volume 19, Issue 2 - Feb 2000
Volume 19, Issue 1 - Jan 2000
Volume 19, Issue 3E - 00 2000
Volume 19, Issue 2E - 00 2000
Volume 19, Issue 1E - 00 2000
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A Study on Speech Recognition in a Running Automobile
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 3~8
In this paper, we studied design and implementation of a robust speech recognition system in noisy car environment. The reference pattern used in the system is DMS(Dynamic Multi-Section). Two separate acoustic models, which are selected automatically depending on the noisy car environment for the speech in a car moving at below 80km/h and over 80km/h are proposed. PLP(Perceptual Linear Predictive) of order 13 is used for the feature vector and OSDP (One-Stage Dynamic Programming) is used for decoding. The system also has the function of editing the phone-book for voice dialing. The system yields a recognition rate of 89.75% for male speakers in SI (speaker independent) mode in a car running on a cemented express way at over 80km/h with a vocabulary of 33 words. The system also yields a recognition rate of 92.29% for male speakers in SI mode in a car running on a paved express way at over 80km/h.
Performance Improvement of Voice Dialing System using Post-Processing
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 9~12
Voice dialing system can recognize the speaker's command and dial the destinate phone number automatically. Such a system is useful for wireless handsets and portable communication devices. As a personal voice dialing system, all the commands are used to train the HMM for speech recognition based on owner-selected phrases. Its implementation requires much less memory space and computation resource compared to a speaker-independent system. Since only two or three training utterances per command are used in this system, it is difficult to estimate exact state duration distribution to improve the recognition performance. Therefore a post-processor is presented to improve the performance. Experiments which use the database collected through the telephone line showed that the proposed post-processor improves the recognition system performance.
Utilization of Syllabic Nuclei Location in Korean Speech Segmentation into Phonemic Units
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 13~19
The blind segmentation method, which segments input speech data into recognition unit without any prior knowledge, plays an important role in continuous speech recognition system and corpus generation. As no prior knowledge is required, this method is rather simple to implement, but in general, it suffers from bad performance when compared to the knowledge-based segmentation method. In this paper, we introduce a method to improve the performance of a blind segmentation of Korean continuous speech by postprocessing the segment boundaries obtained from the blind segmentation. In the preprocessing stage, the candidate boundaries are extracted by a clustering technique based on the GLR(generalized likelihood ratio) distance measure. In the postprocessing stage, the final phoneme boundaries are selected from the candidates by utilizing a simple a priori knowledge on the syllabic structure of Korean, i.e., the maximum number of phonemes between any consecutive nuclei is limited. The experimental result was rather promising : the proposed method yields 25% reduction of insertion error rate compared that of the blind segmentation alone.
A Study on Adaptive Model Updating and a Priori Threshold Decision for Speaker Verification System
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 20~26
In speaker verification system the HMM(hidden Markov model) parameter updating using small amount of data and the priori threshold decision are crucial factor for dealing with long-term variability in people voices. In the paper we present the speaker model updating technique which can be adaptable to the session-to-intra speaker variability and the priori threshold determining technique. The proposed technique decreases verification error rates which the session-to-session intra-speaker variability can bring by adapting new speech data to speaker model parameter through Baum Welch re-estimation. And in this study the proposed priori threshold determining technique is decided by a hybrid score measurement which combines the world model based technique and the cohen model based technique together. The results show that the proposed technique can lead a better performance and the difference of performance is small between the posteriori threshold decision based approach and the proposed priori threshold decision based approach.
Denoising of Speech Signal Using Wavelet Transform
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 27~34
This paper deals with speech enhancement methods using the wavelet transform. A cycle-spinning scheme and undecimated wavelet transform are used for denoising of speech signals, and then their results are compared with that of the conventional wavelet transform. We apply soft-thresholding technique for removing additive background noise from noisy speech. The symlets 8-tap wavelet and pyramid algorithm are used for the wavelet transform. Performance assessments based on average SNR, cepstral distance and informal subjective listening test are carried out. Experimental results demonstrate that both cycle-spinning denoising(CSD) method and undecimated wavelet denoising(CWD) method outperform conventional wavelet denoising(UWD) method in objective performance measure as welt as subjective listening test. The two methods also show less "clicks" that usually appears in the neighborhood of signal discontinuities.
The Relativity between Vibration of Phantom and Its Break Efficiency Due to Position of Focus Induced by Piezoelectric Extracorporeal Shock Wave Lithotripter
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 35~40
In this paper, the relation between the radiated sound and the vibration due to piezoelectric ESWL(Extracorporeal Shock Wave Lithotripter) is examined and the results of the experiments are represented. Next, the relation between the focal point and the vibration of the objects is examined. The same experiments with the objects that can be broken are done and the relation between the vibration and the break efficiency of the phantom is experimentally investigated. These results show that the relativity between the power of the peak frequency and the break efficiency can be confirmed.
Determination of the Effective Elastic Constants of a Superlattice Film by Measuring SAW Velocities
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 41~45
The effective elastic constants of a single-crystal superlattice film have been determined by two methods based on the velocities of surface acoustic waves (SAW). One method uses formulas to calculate the effective elastic constants of a superlattice from the known elastic constants of the constituent layers. The calculated effective elastic constants are tested by comparing the corresponding SAW velocities calculated for thin-film/substrate systems with the corresponding SAW velocities measured by line-focus acoustic microscopy (LFAM). The other method determines the effective elastic constants of the superlattices by inverting the SAW velocity dispersion data measured by LFAM. The results of both methods applied to a TiN/NbN superlattice film are in good agreement.
The Effect of Internal Waves on Acoustic Propagation
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 46~52
Internal waves existing in the stratified ocean significantly affect acoustic propagation. In order to understand the effects of internal waves on acoustic propagation, the sound speed fluctuations due to internal waves are generated based on the Garrett-Munk spectrum which is derived from measured data in the East Sea. The acoustic propagation, where internal waves are present, is simulated numerically using a Galerkin higher order parabolic equation method(SNUPE). These results show favorable comparison to in-situ acoustic propagation data from the East Sea. To investigate the effects of acoustic propagation in random media, scintillation index is adopted and comparison between the measured and numerically simulated data is made.
Study on the Electromagnetic Wave Propagation Characteristics at 1.95GHz
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 53~59
In this paper, considering configuration of the earth and environment of country, it is classified a radio propagation environment Multipath propagation measuring system based on PN code correlation detection method is built to study a wave propagation characteristics of 1.95GHz. Considering the characteristic artificial structure of country, Cihwa plant superintendent area has been selected for basic and accurate measurement a radio propagation characteristic. The experiments are carried out with respect to the RMS delay Spread and the received power in the two kind of geographical areas, LOS (Line of Sight) and N-LOS (Non Line of Sight). The measurement result of that received power of LOS area is -1.6∼-28.8dBm, RMS delay spread is 0.023㎲∼0.22㎲ and received power of N-LOS area is 16∼36.5dBm and RMS delay spread is 0.068∼0.37㎲.
A Study on Acoustical Properties of Soprano′s Singing
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 60~64
This paper studies the relation between the Fundamental Frequency (F0) and the formants of simple vowels in the Korean language sung by sopranos. It is hewn that, in soprano singing, the F0 of a vowel affects its formants. For this reason the formants of simple vowels sung by sopranos must be considered in all over the soprano singing range. We recorded the five simple vowel sounds /a/, /e/, /i/, /o/, and /u/ sung by five professional sopranos from A3 (220.0Hz) to A5 (880.0Hz) in the major scale and compared the formants of the sung vowels with those of spoken vowels. We observed that F1 and F2 of sung vowels were stable in low F0 (lower than B4) but in high F0 (higher than B4), F1 and F2 lost their stabilities. In the case of /a/, /o/, and /u/, the slope of the F1-F2 graph was about 2.6, and those of the F0-F2 and F0-Fl graphs were 2.2-2.5 and 0.7-1.0, respectively. And as the F0 increases, the F1 and F2 of sung vowels /a/, /e/, /i/, /o/, and /u/ were almost the same. At A5, the Fl and F2 of five sung vowels had the same values. This results suggest that the relation between the F0 and the formants be used to synthesize soprano's singing vowels.
Calculation Model of Time Varying Loudness by Using the Critical-banded Filters
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 65~70
It is blown that the loudness is one of the most important metrics in assessing the sound quality and a calculation method for loudness has been standardized for steady sounds. In this study, a new loudness model is suggested for dealing with the transient sound for a unified analysis of various practical sounds. A signal processing technique is introduced for this purpose, which is required for the band subdivision and the prediction of band-level change of transient sounds. In addition, models for the post-masking and the temporal integration are adopted in the analysis of the loudness of transient sounds. In order to solve the problem of the conventional loudness model in the pure-tone signal processing, a critical band filter is employed in the analysis, which consists of 47 critical filters having a filter spacing of a half of the critical bandwidth. For testing the effectiveness of the present model, the predicted responses are compared with the experimental data and it is observed that they are in good agreements.
Mass-estimation Algorithm by Vibration Response Measurement of Dynamic Balance
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 71~78
Quickness and precision are the two most important requirements for an industrial scale used in production lines. In this paper, a new approach, "Mass-estimation algorithm by vibration-response measurement of dynamic balance", is presented to improve some of drawbacks in conventional scales. The system, consisted of velocity and displacement sensors, spring scale, analog-digital converter and microcomputer, is based on full utilization of dynamic mass measurement of velocity and displacement via microcomputer-assisted real time monitoring. The resulting system, when combined with appropriate mass estimation algorithm software, has shown its effectiveness in terms of two desirable characteristics required.
A Study on the Pole-Q Reduction of Chebyshev Function Using Trade-off
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 79~83
When passband ripple α/sub p/ and stopband attenuation α/sub s/ at the w/sub s/ where the stopband begins are specified in filter design, △α/sub s/ usually exceeds the specification by △α/sub s/ due to the necessity that the order n of the filter function be an integer. In this paper, we apply a trade-off method to remove the excess stopband attenuation △α/sub s/ for reducing the value of pole-Q and improving the characteristics of the Chebyshev filter function. We also apply the trade-off method of pole-Q reduction to the modified Chebyshev function, and then the 4 types of function have been analyzed to compare in frequency and time domain characteristics. The trade-off method reduces the pole-Q which influences the filter characteristics to maximum 49.6% without increase of the order n. Thus implies that they have the improved characteristics such as the reduced passband ripple and flatter delay characteristics as compared Chebyshev filter function before trade-off. And the unit step response shows shorter delay time and settling time in time domain performance.
RLSLTDE Algorithm for Bearing Estimation of the Underwater Acoustic Signal
The Journal of the Acoustical Society of Korea, volume 19, issue 5, 2000, Pages 84~90
The bearing detection of radiated target noise is very important at underwater acoustic measurement and passive detection. It differs the arrival tines of received signal at each sensor. Therefore, the bearing can be obtained from the time delay. This paper proposes a new algorithm using the RLSL adaptive filter for TDE. The proposed method is particularly attractive when there is a limitation of priori information about the received signal spectra and when the delay is subject to variation. As the simulation results, it is shown that the proposed algorithm has better convergence characteristics and TDE speed, and so that the usefulness of proposed algorithm is confirmed.