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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 16, Issue 6 - Dec 1997
Volume 16, Issue 5 - Oct 1997
Volume 16, Issue 4 - Aug 1997
Volume 16, Issue 3 - Jun 1997
Volume 16, Issue 2 - Apr 1997
Volume 16, Issue 1 - Feb 1997
Volume 16, Issue 3E - 00 1997
Volume 16, Issue 2E - 00 1997
Volume 16, Issue 1E - 00 1997
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Time Delay Estimation of Two Signals in Wavelet Transform Domain
Kim, Jae-Kuk ; Lee, Young-Seok ; Kim, Sung-Hwan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 5~10
In this paper, a new time delay estimation algorithm, WTD-LMSTDE was proposed. This method has great improvement in convergence rate relative to the time domain approach by decreasing the eigen value spread of input signal autocorrelation matrix. The performance of the algorithm was evaluated for the cases of time invariant time delay and time varying time delay. In the case of time invariant time delay, the estimation accuracy of WTD-LMSTDE was better than that of LMSTDE from 3.3% to 12.5% with respect to SNR. In the case of time varying time delay, the mean error power of WTD-LMSTDE in linear increased delay environment was decreased about 2.4dB compared to that of LMSTDE under noise-free condition. As a result, we showed that the performance of WTD-LMSTDE is better than of LMSTDE.
The Implementation of the Real-Time Active Noise Control System for Attenuating the Engine Noise in a Car
Kwon, Oh-Sang ; Cha, Il-Whan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 11~20
The passive noise control techniques used until now cancel the noise in terms of the characteristics of materials, which increase the mass and the dimension and have a limit that is effective only to attenuate the high frequency components of the noise. But the active noise control techniques developed in recent years have merits that they decrease the mass and the dimension and are effective to attenuating the low frequency noises. In this paper, the real-time active noise control system attenuating the engine booming noise in a car using the digital signal processing(DSP) techniques in terms of the principle of active noise control. The multiple-error filtered-x LMS(Least-Mean Square) algorithm is used as the adaptive algorithm for active noise control and is implemented using the DSP processor Motorola DSP56001 as a controller. According to the result that the experiments are performed for the engine as the RPM changes in a car, the noise attenuating performances are achieved in an overall car interior and is verified to be 20 dB higher for pure-tone and globally, 15 dB.
Vibration Analysis for the Defective Ball Bearing under Radial Loads
Kang, Byoung-Yong ; Lee, Woo-Seop ; Chang, Ho-Gyeong ; Kim, Ye-Hyun ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 21~28
In this study, comparison between Harris-theoretical values and experimental data of load-deflection characteristics in bearing was made. The experiments are conducted under the conditions of the various radial loads and speed of shaft. In the case of non-defective ball bearing, the experimental data agreed well with the Harris-theoretical values for the small steady radial load but not for the large steady radial load. For the radial load bearing, the experimental results show that the stiffness of bearing at the single and multiple defective bearing are bigger in the radial defectiion than in the axial deflection. Load-deflection characteristics for the bearing defect part make it possible to detect the presence of a defect in bearing.
Detection Range of Passive Sonar System in Range-Dependent Ocean Environment
Kim, Tae-Hak ; Kim, Jea-Soo ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 29~34
The prediction of detection range of a passive sonar system is essential to estimate the performance and to optimize the operation of a developed sonar system. In this paper, a model for the prediction of detection range in a range-dependent ocean environment based on the sonar equation is developed and tested. The prediction model calculates the transmission loss using PE propagation model, signal excess, and the detection probability at each target depth and range. The detection probability is integrated to give the estimated detection range. In order to validate the developed model, two cases are considered. One is the case when target depth is known. The other is the case when the target depth is unknown. The computational results agree well with the previously published results for the range-independent environment. Also,the developed model is applied to the range-dependent ocean environment where the warm eddy exists. The computational results are shown and discussed. The developed model can be used to find the optimal frequency of detection, as well as the optimal search depth for the given range-dependent ocean environment.
A Study on the Design and the Construction of a Korean Speech DB for Common Use
Kim, Bong-Wan ; Kim, Jong-Jin ; Kim, Sun-Tae ; Lee, Yong-Ju ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 35~41
Speech database is an indispensable part of speech research. Speech database is necessary to use in speech research and development processes, and to evaluate performances of various speech-processing systems. To use speech database for common purpose, it is necessary to design utterance list that has all the possible phonetical events in minimal number of words, and is independent of tasks. To meet those restrictions this paper extracts PBW set from large text corpus. Speech database that was constructed using PBW set for utterance list and its properties are described in this paper.
Estimation of Speeker Recognition Parameter using Lyapunov Dimension
Yoo, Byong-Wook ; Kim, Chang-Seok ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 42~48
This paper has apparaised ability of speaker recognition and speech recognition using correlation dimension and Lyapunov dimension. In this method, speech was regarded the cahos that the random signal is appeared in determinisitic raising system. we deduced exact correlation dimension and Lyapunov dimension with searching important orbit from AR model power spectrum when reconstruct strange attractor using Taken's embedding theory. We considered a usefulness of speech recognition and speaker recognition using correlation dimension and Lyapunov dimension that characterized reconstruction attractor. As a result of consideration, which were of use more the speaker recognition than speech recognition, and in case of speaker recognition using Lyapunov dimension were much recognition rate more than speaker recognitions using correlation dimension.
Propagation Environments of a Suburban Area
Kim, Jae-Sub ; Park, Chang-Kyun ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 49~56
In mobile communications, it is very important that we predict the propagation environments of radiation pattern, in order to decide the service area, select the best location of the best station, design the cell etc. Therefore, by analyzing the propagation prediction model that is varied according to the kind of antenna, the beam angle, the terrain and obstacles, we expect that the economic operating of communication networks, the calling quality and the service of subscriber will be enhanced. In this paper, we select the around of Seji base station in Naju-city Chonnam for modern suburban area and measure the field strength to propose the optimal propagation prediction model for suburban areas. We propose the propagation prediction model that, it is not found in the other models until now, consists of the correction coefficient with the relative differences of antenna effective height of the base station and mobile station for minimizing errors. Finally, comparing the results of the field test with the computer simulation(PPGIS : Propagation Prediction Geographic Information System) results for the Hata model, the Egri model, the Carey model and the propose model, we confirm the property of the proposed model.
Front-End Processing for Speech Recognition in the Telephone Network
Jun, Won-Suk ; Shin, Won-Ho ; Yang, Tae-Young ; Kim, Weon-Goo ; Youn, Dae-Hee ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 57~63
In this paper, we study the efficient feature vector extraction method and front-end processing to improve the performance of the speech recognition system using KT(Korea Telecommunication) database collected through various telephone channels. First of all, we compare the recognition performances of the feature vectors known to be robust to noise and environmental variation and verify the performance enhancement of the recognition system using weighted cepstral distance measure methods. The experiment result shows that the recognition rate is increasedby using both PLP(Perceptual Linear Prediction) and MFCC(Mel Frequency Cepstral Coefficient) in comparison with LPC cepstrum used in KT recognition system. In cepstral distance measure, the weighted cepstral distance measure functions such as RPS(Root Power Sums) and BPL(Band-Pass Lifter) help the recognition enhancement. The application of the spectral subtraction method decrease the recognition rate because of the effect of distortion. However, RASTA(RelAtive SpecTrAl) processing, CMS(Cepstral Mean Subtraction) and SBR(Signal Bias Removal) enhance the recognition performance. Especially, the CMS method is simple but shows high recognition enhancement. Finally, the performances of the modified methods for the real-time implementation of CMS are compared and the improved method is suggested to prevent the performance degradation.
Audio Signal Coding Using Wavelet Transform
Bae, Seok-Mo ; Kim, Do-Hyoung ; Chung, Jae-Ho ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 64~70
This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.
A Study on Direction Finding Technique for Array with Faulty Elements
Kim, Ki-Man ; Youn, Dae-Hee ; Cha, Il-Whan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 71~75
In this paper, some problems that occur from faulty elements in a direction finding system composed of the linear array are studied and the method which improves the performance is proposed. The fault element means the sensor that has no output or highly reduced gain than other normal sensors. In the case of the presence of faulty elements, the performance of the conventional the spatial spectrum subject to a constraint. The corrected spatial spectrum is obtained by this vector. The computer simulations have been performed to study the performance of the proposed method. We have compared the proposed method with the subaperture processing method of one of the previous works.
Design of VCO(Voltage Controlled Oscillator) for mobile communication with a built-in voltage regulator
Cho, Hyon-mook ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 76~84
In this paper, one of the core components of a mobile communication system, VCO(Voltage Controlled Oscillator) IC is designed. The VCO IC was designed, have realized as LC turned oscillator using varicap. LC sinusoidal tuned oscillator generally requires external inductors and thus remainding circuit is implemneted in monolithic IC. The circuit is fabricated using an 15 mask IC process and has a die size of 1150um
780um. The tests showed that VCO was operated at frequencies in the regions between 880MHz-915MHz in the control voltage range of 1V to 3V at 5V supply voltage and as the power supply was varied from 4.5V to 5.5V, the frequency varied 425KHz/V. The VCO IC has frequency shift of 1.97MHz/T, carrier level of -7dBm and power consumption of 16.7mA. Also it has phase noise of -80dBc/Hz, offset at 50KHz and harmonic response of center frequency is -41dBm. For the future development of the transceiver 1 chip, the previously mentioned external devices need to be incorporated into Si MMIC.
Analysis of Impedance of Multilayer Structure using Cepstrum Technique
Shin, Jin-Seob ; Jun, Kye-Suk ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 85~89
In this paper, the imdedance for each layer using triple cepstrum signal processing for reflected ultrasonic signal from the multilayer structure has been analyzed. The reflection coefficient can be obtained from the amplitude and the polarity of the peaks in the triple cepstrum, and then the impedance of each layer has been reconstructed by the reflection coefficient. In this experiment, four types of multilayers consisting of different metal layers were manufactured. The reflected signals from the multilayer structure have been detected by pulse-echo method. The impedances have been reconstructed by triple cepstrum technique. The experimental results have been in good agreement with the theoretical results.
Frequency Measurement of Hwang-Jong Tone in Korean Traditional Music
Pang, Hee-Suk ; Sung, Koeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 90~93
In Korean traditional music, the frequency of Hwang-Jong tone is not known precisely, to say nothing of the temperament. So we measured the frequency of Hwang-Jong tone in solo pieces for the woodwind instruments such as daegum, piri and danso. According to the results of the measurement, the frequency of Hwang-Jong tone is about 328Hz, which is closer to the frequency of E
(330Hz) than that of E
(311Hz). This result is different from the existing theory that the frequency of Hwang-Jong tone is close to that of E
An analysis of the Sound Radiation Characteristics of the King Song-Dok Bell Using Cylindrical Acoustic Holography
Kim, Yang-Hann ; Kim, Sea-Moon ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 94~100
In order to investigate the radiation of sound from the King Song-Duk bell, we measured the sound pressure around the bell at every 30
using a microphone line array which was composed of 30 microphones separated by 15cm;total number of measurement point was 360. The sound field was estimated by using cylindrical acoustic holography. The spectrum of measured sound pressure demonstrates that it is almost like white noise in the very beginning, but in 10 seconds two close frequency components(64.06Hz, 64.38Hz) remain and make a famous beating. This beating sound is often believed to make unique sound of the King Song-Duk bell. The mode shapes of that frequency components are the same except that one is rotated by 45
from the other. This phenomenon occurs at the other pairs of components are the same except ones in very high frequency range where the mode shapes are rather complex.
Active Noise Control of Short Duct using Zero Acoustic Impedance Boundary
Cha, Kyung-Hwan ; Lee, Chai-Bong ; Kim, Chun-Duck ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 101~105
The active noise control method that was developed for long duct has some problems to be applied for short duct. To overcome this problem, we apply the SISO(Single Input Single Output) algorithm for the active noise control of short duct using zero acoustic impedance boundary. The SISO algorithm can input noise signal and error signal with one microphone simultaneously. The real-time controller was implemented using TMS320E25 DSP(Digital Signal Processing) chip and it's performance was evaluated by experiment. As a result, we obtain total 4.7dBA noise reduction for 0.80m short duct.
On a Pitch Alteration Technique in Time-Frequency Hybrid Domain for High Quality Prosody Control of Speech Signal
Lee, Sang-Hyo ; Bae, Myung-Jin ;
The Journal of the Acoustical Society of Korea, volume 16, issue 4, 1997, Pages 106~109
In the area of the speech synthesis techniques, the waveform coding methods maintain the intelligibility and naturalness of synthetic speech. In order to apply the waveform coding techniques to synthesis by rule, however, we must be able to alter the pitches for prosody control of synthetic speech. In this paper, we propose a new pitch alteration technique in time-frequency hybrid domain, that compensates phase distortion of the cepstral pitch alteration method with time scaling method in the time domain. This method can remove some phase spectrum distortion which is occurred in conjunction point between the waveforms in continued frames. Also, we can obtain little magnitude spectrum distortion below 1.18% for pitch alteration of 200%.