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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
Editor in Chief :
Volume & Issues
Volume 14, Issue 6 - Dec 1995
Volume 14, Issue 5 - Oct 1995
Volume 14, Issue 4 - Aug 1995
Volume 14, Issue 3 - Jun 1995
Volume 14, Issue 2 - Apr 1995
Volume 14, Issue 1 - Feb 1995
Volume 14, Issue 2E - 00 1995
Volume 14, Issue 1E - 00 1995
Selecting the target year
A Study on Mean Flow Velocity Measurement by Cross Correlation of Ultrasonic Waves
Kim, Chang-Ho ; Lee, Dug-Ki ; Paik, Jong-Seung ; Jho, Moon-Jae ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 5~15
An application of the cross correlation technique by adopting ultrasonic waves for water pipe flow measuring purpose is studied. It is a non-intrusive flow metering method by determining the time of the flight of the flow turbulent noise and its non-obstructing mechanism enables to reduce process energy loss due to the flowmeter obstruction. A digital signal processor for the purpose of the real time Fourier transform was employed for the fast time calculation of the flow velocity. The overall accuracy was found as about
for flow velocities from 0.25 m/s up to 16 m/s and for the pipe inside diameters from 50mm to 248mm. The cross correlation technique can be used for the tap water utility including most common liquid flows.
Improvement of performance for the LBG algorithm by the decision of initial codevectors
Hong, Chi-Hwun ; Ch0, Che-Hwang ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 16~29
Choosing initial codevectors in the LBG algorithm controls the performance of a codebook, because it only guarantees a locally optimal codebook. In this paper, we propose the decision method of initial codevectors by a decision radius which takes for feature vectors DC, low frequency, medium frequency and high frequency terms generated by a DCT. The more the decision radius is increased in order to decide initial codevectors, the more the number of membership vectors and the standard deviation for distance among the initial codevectors are increased. To obtain improved performance for a codebook, the decision radius for DC term is required above 0.9 of the membership rate and those for low frequency, medium frequency and high frequency terms under 0.6 of it.
A study on the DSP Analysis for the CAT application
Jeon, Dong-Keun ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 30~39
In this paper, study on implementation of FFT analyzer applied to CAT, A/D conversion module, DSP module and VXIbus interface module are implemented in hardware and calculation program and control software are implemented in DSP module and VXIbus interface module, respectively. The control of the modules using PC is realized in software. The real time bandwidth of the FFT analyzing device is 100KHz. At sampling rate of 200KHz and with 2048 point FFT, the result of applying sine, triangular and rectangular wave of 20KHz to FFT analyzing device is compared with the FFT analyzed results of Hewlett-Packard 3562A dynamic output range of -40dBV- +30dBV, correct results are obtained and results of applying 10KHz, 20KHz and 50KHz input are compared and the correct values are obtained.
Convergence of the Filtered-x LMS Algorithm for Canceling Multiple Sinusoidal Acoustic Noise
Lee, Kang-Seung ; Lee, jae-Chon ; Youn, Dae-Hee ; Kang, Young-Suk ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 40~49
Application of the filtered-x LMS adaptive filter to active noise cancellation requires to estimate the transfer charactersitics between the output and the error signal of the adaptive canceler. In this paper, we derive the filtered-x adaptive noise cancellation algorithm and analyze its convergence behavior when the acoustic noise consists of multiple sinusoids. The results of the convergence analysis of the filtered-x LMS algorithm indicate that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components : Phase estimation error and estimated gain. In particular, the convergence is shown to strongly affected by the accuracy of the phase response estimate. Simulation results are presented to support the theoretical convergence analysis.
A Study on Real-time Implementing of Time-Scale Modification
Han, Dong-Chul ; Lee, Ki-Seung ; Cha, Il-Hawan ; Youn, Dae-Hee ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 50~61
A time scale modification method yielding rate-modified speech while conserving the characteristic of speech was implemented in real-time using a goneral purpose digital signal processor. Time scale modification changed pronunciation speed only, producing a time difference between the input signal and the modified signal, making it impossible to implement it in real-time. In this thesis, a system was implemented to remove the time difference between the input and modified signals. Speech signals slowed down or speeded up by a physical time scale modification method, such as adjusting the motor speed of the cassett tape recorder, was used as the input signal. Physical modification that controled only the inter speed of the cassette tape player distorted the pitch period of the original speech. In this study, a real-time system was implemented so that the pitch-distorted speech was reconstructed back to the original by fractional sampling pitch shifting using an FIR filter, and this signal was time scale modified to match the cassette tape recorder motor speed using SOLA time-scale medification. In experiments using speech signals medifiedby the proposed method, results obtained using a 16-bit resolution ADSP2101 processor and using computer simulations employing floating point operations showed about the same average frame signal-to-noise ratio of about 20 dB.
Experimental Study on Acoustic Characteristics of Perforated Tube and Perforated Tube Muffler
Yoon, Doo-Byung ; Kim, Yang-Hann ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 62~72
The acoustic characteristics of perforated tube muffler are studied in terms of non-dimensional wavenumber ka and admittance-ratio AZ. This study includes not only the case of perforated tubes having uniform hole distribution along the length but also the case of having non-uniform hole distributions. The acoustic hole impedance and transmission loss of perforated tube of which has various hole distributions were measured. The experimental results demonstrated that the transmission loss of perforated tube is a function of non-dimensional wave number ka and admittance-ratio AZ. The transmission loss of perforated tube muffler is predicted by the numerical method which is based on Sullivans and compared with the experimental ones.
Analysis of Simultaneous Generation Mechanism of P/S Waves with the PZT Piezoelectric Ceramics
Kim, Yeon-Bo ; Roh, Yong-Rae ; Nam, Hyo-Duk ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 73~79
Most of conventional ultrasonic transducers are constructed to generate either longitudinal or shear waves, but not both of them. We investigated the mechanism of dual mode transducers that generates both of the longitudinal and shear waves simultaneously with a single PZT element. A piezoelectric ceramic PZT has the hexagonal 6mm crystal symmetry, after poling. We studied the performance of a PZT element as a function of its rotation angle so that its efficiency is optimized to excite the two waves equally strongly. The results are verified by checking the impedance variation of the element with Finite Element Methods, and checking the wave form by pulse-echo test simulation. Validity of the theoretical calculation is verified through experiments.
A Study on the tire structure-borne sound
Chi, Chang-Heon ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 80~91
A theoretical models has been prepared which describes the noise generated by tire/road interaction for the tire structure-borne sound analysis. The model begin with a set of thin shell equations describing the motion of the belt of a radial ply tire, as drived by Bohm('mechanisms of the belted tire', Igeniur-Archiv, XXXV, 1966). Structural quantities required for these equations are derived from material properties of the tire. The rolling shape of a tire is computed from the steady-state limit of these equations. Vibrational response of the tire is treated by the full dependent shell equations. The force input at the tire/road interface is calculated on the basis of tread geometry and distribution of contact patch pressure. Radiation of noise is calculated by a simpson integral. Using the programs, the effect on noise of various tire design variations is computed and discussed. Trends which lead to quiet tire design are identified.
Vibration Analysis of Ball Bearing Fault using HFRT
Kim, Ye-Hyun ; Kang, Byoung-Yong ; kim, Dong-Il ; Chang, Ho-Gyeong ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 92~100
In this study, the bearing defects were modeled and the vibration of ball bearing faults was presented for the defective pattern. The vibration signal was measured for the single and multiple defected ball bearing at the various defect positions and rotation speed, and then the signal components using the HFRT(high frequency resonance technique) were analyzed by FFT. The experimental data analysis has shown that the frequencies generated in the single or multiple defected ball bearings appear with the characteristic defect frequency and harmonics of ball pass frequency peak. Signal processing by HFRT makes it possible not only to detect the presence of a defect but also to diagnose the defect part of the bearing.
Korean Phoneme Recognition Using Self-Organizing Feature Map
Jeon, Yong-Koo ; Yang, Jin-Woo ; Kim, Soon-Hyob ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 101~112
In order to construct a feature map-based phoneme classification system for speech recognition, two procedures are usually required. One is clustering and the other is labeling. In this paper, we present a phoneme classification system based on the Kohonen's Self-Organizing Feature Map (SOFM) for clusterer and labeler. It is known that the SOFM performs self-organizing process by which optimal local topographical mapping of the signal space and yields a reasonably high accuracy in recognition tasks. Consequently, SOFM can effectively be applied to the recognition of phonemes. Besides to improve the performance of the phoneme classification system, we propose the learning algorithm combined with the classical K-mans clustering algorithm in fine-tuning stage. In order to evaluate the performance of the proposed phoneme classification algorithm, we first use totaly 43 phonemes which construct six intra-class feature maps for six different phoneme classes. From the speaker-dependent phoneme classification tests using these six feature maps, we obtain recognition rate of
and confirm that the proposed algorithm is an efficient method for improvement of recognition performance and convergence speed.
Development of a Bone Conduction Telephone for Conductive Hearing Impaired Persons and its Performance Test
Kang, Kyeong-Ok ; Kang, Seong-Hoon ;
The Journal of the Acoustical Society of Korea, volume 14, issue 2, 1995, Pages 113~122
This paper describes characteristics of a bone conduction telephone which was developed for conductive hearing impaired persons to call without additional devices and results of its performance test. Not only the hearing impaired but also normal hearing persons can use this telephone because we developed a bone conduction vibrator with which they can perceive speech signal using functions of air conductive hearing as well as bone conductive hearing. It also has tone control function compensating hearing losses for the hearing impaired originating from their hearing characteristics, and using this function together with received volume control it has received volume range of 20dB in loudness rating, which is similar effect as what a telephone set with built-in received amplifier has. From results of articulation and intelligibility tests for 19 hearing impaired persons, we can see that if their bone-conduction hearing loss is 61dB or less, they can understand words or sentences and response well with this telephone.