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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
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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
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Admittance and Free Wavenumber in the Cylinderical Shell by Point Excitation
Jo, Heung-Kuk ; Lee, Chai-Bong ; Kim, Jeong-Kuk ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 5~13
This paper shows newly developed equations of cylindrical shell motion, which solutions are obtained as a set of linear equation. Each linear equation is derived along each axis of cylindrical coordinates. The admittance and the free wavenumber are obtained under assumption of point excition on a cylindrical shell. Their results are shown in figures. In this results, this paper shows a possibility that a vibration and a noise generated in a cylindrical shell can be formulated as a mathematical model.
A Study on the Design of Digital Sound Processor for Music using Equal Power Density Envelope Generator and Transform Coder
Koo, Jae-Ul ; Pang, Hyo-Chang ; Kim, Jong-Han ; Kim, Won-Hoo ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 14~27
This paper presents the digital music sound DSP by using ADPCM and Perceptual Transform Corder in MPEG to compress sound data and minimize the quantization noise for musical instrument. these method are utilized to develop algorithm of equal power density envelope. And these results are applied to examine the specific characteristics of musical instrument and determine the compression method. The design of new RISC DSP which generates 32 voices of musical instrument simultaneously and the coding of 200 musical instrument sound data in 1MByte memory shows that these algorithm is very useful to regenerate musical sound by using the minimum size of memory.
A Double Loop Control Model Using Leaky Delay LMS Algorithm for Active Noise Control
Kwon, Ki-Ryong ; Park, Nam-Chun ; Lee, Kuhn-Il ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 28~36
In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.
ARMA System identification Using GTLS method and Recursive GTLS Algorithm
Kim, Jae-In ; Kim, Jin-Young ; Rhee, Tae-Won ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 37~48
This paper presents an sstimation of ARMA coefficients of noisy ARMA system using generalized total least square (GTLS) method. GTLS problem for ARMA system is defined as minimizing the errors between the noisy output vectors and estimated noisy-free output. The GTLS problem is solved in closed form by eigen-problem and the perturbation analysis of GTLS is presented. Also its recursive solution (recursive GTLS) is proposed using the power method and the covariance formula of the projected output error vector into the input vector space. The simulation results show that GTLS ARMA coefficients estimator is an unbiased estimator and that recursive GTLS achieves fast convergence.
Design and Implementation of Simple Text-to-Speech System using Phoneme Units
Park, Ae-Hee ; Yang, Jin-Woo ; Kim, Soon-Hyob ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 49~60
This paper is a study on the design and implementation of the Korean Text-to-Speech system which is used for a small and simple system. In this paper, a parameter synthesis method is chosen for speech syntheiss method, we use PARCOR(PARtial autoCORrelation) coefficient which is one of the LPC analysis. And we use phoneme for synthesis unit which is the basic unit for speech synthesis. We use PARCOR, pitch, amplitude as synthesis parameter of voice, we use residual signal, PARCOR coefficients as synthesis parameter of unvoice. In this paper, we could obtain the 60% intelligibility by using the residual signal as excitation signal of unvoiced sound. The result of synthesis experiment, synthesis of a word unit is available. The controlling of phoneme duration is necessary for synthesizing of a sentence unit. For setting up the synthesis system, PC 486, a 70[Hz]-4.5[KHz] band pass filter for speech input/output, amplifier, and TMS320C30 DSP board was used.
Recognition of Restricted Continuous Korean Speech Using Perceptual Model
Kim, Seon-Il ; Hong, Ki-Won ; Lee, Haing-Sei ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 61~70
In this paper, the PLP cepstrum which is close to human perceptual characteristics was extracted through the spread time area to get the temperal feature. Phonemes were recognized by artificial neural network similar to the learning method of human. The phoneme strings were matched by Markov models which well suited for sequence. Phoneme recognition for the continuous Korean speech had been done using speech blocks in which speech frames were gathered with unequal numbers. We parameterized the blocks using 7th order PLPs, PTP, zero crossing rate and energy, which neural network used as inputs. The 100 data composed of 10 Korean sentences which were taken from the speech two men pronounced five times for each sentence were used for the the recognition. As a result, maximum recognition rate of 94.4% was obtained. The sentence was recognized using Markov models generated by the phoneme strings recognized from earlier results the recognition for the 200 data which two men sounded 10 times for each sentence had been carried out. The sentence recognition rate of 92.5% was obtained
Feature Vector Extraction and Automatic Classification for Transient SONAR Signals using Wavelet Theory and Neural Networks
Yang, Seung-Chul ; Nam, Sang-Won ; Jung, Yong-Min ; Cho, Yong-Soo ; Oh, Won-Tcheon ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 71~81
In this paper, feature vector extraction methods and classification algorithms for the automatic classification of transient signals in underwater are discussed. A feature vector extraction method using wavelet transform, which shows good performance with small number of coefficients, is proposed and compared with the existing classical methods. For the automatic classification, artificial neural networks such as multilayer perceptron (MLP), radial basis function (RBF), and MLP-Class are utilized, where those neural networks as well as extracted feature vectors are combined to improve the performance and reliability of the proposed algorithm. It is confirmed by computer simulation with Traco's standard transient data set I and simulated data that the proposed feature vector extraction method and classification algorithm perform well, assuming that the energy of a given transient signal is sufficiently larger than that of a ambient noise, that there are the finite number of noise sources, and that there does not exist noise sources more than two simultaneously.
Implementation of ray tracing simulator for extracting sound field parameters
Lee, Deok-Su ; Seong, Goeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 82~89
A sound field simulator is constructed to obtain the sound field paramaters such as the magnitudes and directions of early reflections with moderate efforts. The proposed simulator is based on the hybrid ray tracing method that traces rays reached the listener position and convert them to image sound sources. By this approach, we can obtain the directional impulse response relatively easily with minimum casts. Simulation experiment results of several performace places are reported to how the versatility of the proposed simulator system.
An Autoregressive Parameter Estimation from Noisy Speech Using the Adaptive Predictor
Koo, Bon-Eung ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 90~96
A new method for autoregressive parameter estimation from noisy observation sequence is presented. This method, termed the AP method, is a result of an attempt to make use of the adaptive predictor which is a simple and reliable way of parameter estimation. It is shown theoretically that, for noisy input, the parameter vector computed from the prediction sequence is closer to that of the original sequence than the noisy input sequence is, under the spectral distortion criterion. Simulation results with the Kalman filter as a noise reduction filter and real speech data supported the theory. Roughly speaking, the performance of the parameter set obtained by the AP method is better than noisy one but worse than the EM iteration results. When the simplicity is considered, it could provide a useful alternative to more complicated parameter estimation methods in some applications.
A study on the Recognition of Continuous Digits using Syntactic Analysis and One-Stage DP
Ann, Tae-Ock ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 97~104
This paper is a study on the recognition of continuous digits for the implementation of a voice dialing system, and proposes an method of speech recognition using syntactic analysis and One-Stage DP. In order to perform the speech recognition, first of all, we make DMS model by section division algorithm and let continuous digits data be recognized through the proposed One-Stage DP method using syntactic analysis. In this study, 7 continuous digits of 21 kinds which is pronounced by 8 male speakers two or three times, are used. The speaker dependent and speaker independent recognition are performed with the above data by way of the conventional One-Stage DP and the proposed One-Stage DP using syntactic analysis under the condition of laboratory environment. From the recognition experiments, it is shown that the proposed method was better than the established method. And, the recognition accuracy of speaker dependence and independence by the proposed One-Stage DP using syntactic analysis was about 91.7% and 89.7%.
A Study on the Interference between CDMA Base Station and Analog FM Mobile Station in the Mobile Communication
Kim, In-Hwan ; Park, Chang-Gyun ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 105~113
For the case of using analog FM system and CDMA system together in the mobile communications, we experimentally calculate the appropriate frequency offset to improve the frequency efficiency by analizing the interference between both systems. As the results of the analysis, we obtained that frequency offset is 840KHz(Path Loss : 80dB) and 720KHz(Path Loss : 120dB) for the Path Loss when a output of CDMA Base Station is 0.91 W ERP. Therefore, to minimize the frequency offset, a output of Base Station will be controlled corresponding to Cover Area.
Insertion Loss by Noise Barrier on the Discontinuous Ground
Kim, Ye-Hyun ; Kim, Dong-Ill ; Jang, Ho-Kyeong ;
The Journal of the Acoustical Society of Korea, volume 14, issue 3, 1995, Pages 114~121
Outdoor experimental study is presented the insertion loss caused by barrier considering discontinuous ground condition. Measurements ware made in 1/3 octave band over the frequency range 315 Hz~3150 Hz with the various geometry of the source, receiver and barriers. The frequency range of the interference pattern depends on the phase difference between path from the edge of barrier to receiver, and hence on the acoustical properties of the ground on the receiver side of the barrier. The insertion loss by barrier, in addition to diffraction, is shown to be dependent on the ground characteristic.