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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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A Study of Synthesizing the Sound of Gayageum with FM Algorithm
Kim, Jae-Yong ; Kwon, Min-Do ; Jang, Ho-Keun ; Woo, Jong-Sik ; Park, Ju-Sung ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 5~12
In this paper, the 3 operator FM synthesis method has been devised on the basis of the analysis of Gayageum sounds. The parameters for the proposed synthesis algorithm has been implemented by C language. It is subjectively and objectively confirmed that the sounds synthesized by C language are very similar to the original sounds. The synthesis algorithm is also implemented by DSP, and the sounds from this experiment are close to the original ones.
Design and Evaluations of Underwater Hydrophone with Self Noise Suppressing Structures; - Part Ⅱ. Influence of Acostic Damping Layer Properties -
Im, Jong-In ; Roh, Young-Rae ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 13~17
This paper investigates the influence of material properties of the acoustic damping layer in the low noise hydrophone designed in the previous paper. For increase of the insensitivity of the hydrophone to external noises, acoustic impedance and damping coefficients are selected and the effects of the selected material property on the hydrophone response to the external noises are simulated with finite element method (FEM). The results show that the damping coefficients are not influential to the structural vibration decoupling from the sensing element. On the other hand, the optimum acoustic impedance of compliant layer is estimated which is smaller than 1 Mrayl or larger than 4 Mrayl. However polymer materials, which are in general use for acoustic window and damping layers, is not appropriate for the compliant materials of this hydrophone. Therefore development of new composite materials, i.e. ceramic-polymer composite or metal-ceramic composites etc., is required for the development of effective self noise suppressing underwater hydrophones.
Energy-Dependent Preemphasis for Speech Signal Preprocessing
Kim, Dong-Jun ; Park, Sang-Hui ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 18~25
This study describes a modified preemphasis formula, what we call energy-dependent preemphasis(EDP). This uses the normalized short-term energy of speech signal, with the assumption that the source characteristics of the glottal pulses and the radiation characteristics of the lips are approximately proportional to the energy of speech signal. Using this method, speech analyses, such as AR spectrum estimation, formant detection, are performed for nonstationary starting parts of 5 Korean single vowels. The results are compared with the conventional two preemphasis methods. We found that the proposed preemphasis gave enhanced spectral shapes and more accurate formant frequencies and avoided overlapping phenomenon of adjacent two formants.
Analysis of the Sound Field in an Enclosure on the Speaker System Using the Finite Element Method
Lee, Woo-Seop ; Kim, Jung-Rag ; Chang, Ho-Gyeong ; Kim, Ye-Hyun ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 26~32
In this study, the interior acoustic field of a speaker system with rectangular closed cavity and its effect on the impedance and the sound pressure reponses are investigated by applying the finite element method to the rectangular enclosure. Based on a good agreement with the theoretical and the experimental, it is founded that FEM is useful method for the development of a speaker system generating the diverse mode.
Analysis of Speech Signals According to the Various Emotional Contents
Jo, Cheol-Woo ; Jo, Eun-Kyung ; Min, Kyung-Hwan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 33~37
This paper describes experimental results from emotional speech materials, which is analysed by various signal processing methods. Speech materials with emotional informations are collected from actors. Analysis is focused to the variations of pitch informations and durations. From the analysed results we can observe the characteristics of emotional speech. The materials from this experiment provides valuable resources for analysing emotional speech.
A Study on Least Mean Fourth (LMF) and Least Mean Squares-Fourth (LMSF) Blind Equalization Algorithm
Yoon, Tae-Sung ; Byun, Youn-Shik ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 38~44
In this study, wer derived LMF-Sato, LMSF-Sato complex blind equalization algorithms for complex data. And then, the convergence rates, the convergence characteristics at the steady state and the stability of the proposed LMF and LMSF blind equalization algorithms are compared with those of LMS-Sato blind equalization algorithm. In simulations with 16-QAM data, LMF-Sato and LMSF-Sato algorithms showed better performance comparing with LMS-Sato algorithm generally. When the initial estimation errors of the weights of the equalizer are large, LMF-Sato algorithm showed ill characteristic in stability. However, LMSF-Sato algorithm has good covergence characteristics and preserves robustness.
A Study on the Characteristics of the Acoustic Reflection of the Wedge Shaped Underwater Sound Absorptive Tile
Kang, Sung-Oug ; Hong, Woo-Young ; Song, Yeong-Il ; Lyu, Ki-Sang ; Baek, Chil-Ki ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 45~50
Four different wedge angle absorptive tiles were designed and made, and the magnitudes of the reflected acoustic fields by the wedge shaped underwater sound absorptive tiles were measured. The minimum magnitude was found at the angle of 30
and the maximum of it was found at the wedge angle of 120
from measured the reflected acoustic fields at the front sides of the tiles. The fact that as wedge angle of the absorptive tile increases, the reflection coefficient is increasing is verified. The measured reflected acoustic fields were not dependent on the frequency in the range of 10kHz~30kHz used in this experiment for the same wedge angle tile. The measured reflected acoustic fields at the back sides of the tiles show that they are independent from both type of the absorptive tiles and the frequencies used in the experiments. The measured values and the computed values by the numerical model for the reflected acoustic fields of the wedge shaped absorptive tiles are fairly well comparable with one another.
The Measurement of the Depth of Crack using Images of SLAM
Hwang, Ki-Hwan ; Jun, Kye-Suk ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 51~56
In this paper, we studied the configuration and depth measurement method of the crack in the interior of solid with scanning laser acoustic microscope. Precision measurement method of crack depth is required in SLAM because that system reconstructs the shadow image to the transmission coefficient. We proposed this method that used geometrical structure to the shadow area of SLAM images obtained from oblique incidence and the mode conversion of ultrasound in specimen and then experimented it. For this experiment, we fabricated various specimens which had the vertical line-crack with different depth and made the wedge as 20
for oblique incidence. Experimental results showed that the shadow area of SLAM images were proportional to the depth of crack. Measured depth error to the crack was less than 6% compared with practical crack depth.
Acoustic Echo Canceller for Stereophonic Tele-conferencing System.
Jeong, Kyu-Hwa ; Lee, Won-Cheol ; Youn, Dae-Hee ; Cha, Il-Hwan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 57~63
This paper proposes a new stereophonic acoustic echo canceller to preven the performance degradation at the instant of far-end talker's movement or change in transmission room environment in stereophonic teleconferencing system. In stereophonic acoustic echo canceller, the filter coefficients of the adaptive filters for echo cancellation do not have unique solutions and not converge to their optimum values. Therefore, the change of the far-end transmission room environment leads to degradation of ERLE(Echo Return Loss Enhanceement). Moreover, their computational complexity increases as the number of the filter coefficient increases, and their adaptive filters converge very slowly. So the real-time implementation is very difficult. To overcome these problems we propose a pre-processor consisting of an adaptive filter for making pseudo stereophonic signal, and it results in improved performance at the instant of environment change at far-end side and reduction of the total complexity
A Study on the Korean Syllable As Recognition Unit
Kim, Yu-Jin ; Kim, Hoi-Rin ; Chung, Jae-Ho ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 64~72
In this paper, study and experiments are performed for finding recognition unit fit which can be used in large vocabulary recognition system. Specifically, a phoneme that is currently used as recognition unit and a syllable in which Korean is well characterized are selected. From comparisons of recognition experiments, the study is performed whether a syllable can be considered as recognition unit of Korean recognition system. For report of an objective result of the comparison experiment, we collected speech data of a male speaker and processed them by hand-segmentation for phoneme boundary and labeling to construct speech database. And for training and recognition based on HMM, we used HTK (HMM Tool Kit) 2.0 of commercial tool from Entropic Co. to experiment in same condition. We applied two HMM model topologies, 3 emitting state of 5 state and 6 emitting state of 8 state, in Continuous HMM on training of each recognition unit. We also used 3 sets of PBW (Phonetically Balanced Words) and 1 set of POW(Phonetically Optimized Words) for training and another 1 set of PBW for recognition, that is "Speaker Dependent Medium Vocabulary Size Recognition." Experiments result reports that recognition rate is 95.65% in phoneme unit, 94.41% in syllable unit and decoding time of recognition in syllable unit is faster by 25% than in phoneme.
Design studies for mandrel type fiber-hydrophones with FEM
Im, Jong-In ; Roh, Yong-Rae ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 73~80
This paper describes structural optimization of optical fiber-wound mandrel hydrophones with Finite Element Method (FEM). The hydrophone is supposed to have operation frequency range of up to 10 kHz and show omni-directional sensitivity pattern at 5 kHz. Studied parameters are mandrel geometry, molding thickness, and material properties of constitutional parts of the hydrophone. Theoretical calculation result shows that pressure sensitivity of the hydrophone increased as either mandrel length or molding thickness gets larger. Also higher pressure sensitivity requires a mandrel or molding material with relatively low Youngs modulus or Poissons ratio. Hydrophone bandwidth increases either as the mandrel length becomes shorter or as the mandrel becomes harder. The omni-directional characteristic is improved as the mandrel length becomes shorter, at 5 kHz. With the above results, we determine the structure of an optical fiber-wound mandrel hydrophone which has the pressure sensitivity of
Rad./Pa, operation frequency range of up to 10 kHz, and shows omni-directional sensitivity pattern at 5 kHz.
A Recursive Estimation Algorithm for FIR System Using Higher Order Cumulants
Kim, Hyoung-Ill ; Yang, Tae-Won ; Jeon, Bum-Ki ; Sung, Koeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 81~85
In this paper, a recursive estimation algorithm for FIR systems is proposed using the 3rd and 4th order cumulants. To obtain the Overdetermined Recursive Instrumental Variable(ORIV) method type algorithm, we transform the 3'th and 4'th order cumulant relationship to a certain matrix form which is consist of only output data. From the matrix form, we induce the proposed algorithm procedure following the ORIV method. The proposed algorithm provides improved estimation accuracy with smaller data and can be applied to a time varying system as well. In addition, it reduces the estimation error due to the additive Gaussian noise compared to conventional 2'rd order based algorithms since it only uses higher than 2'rd order cumulant. Simulation results are presented to compare the performance with other HOS-based algorithms.
Optimum Design and Simulation of SAW Filters for Personal Communication Systems
Chung, Yeong-Jee ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 86~93
A Design & Simulation Tools of Surface Acoustic Wave(SAW) Filters for Mobile Communication Systems, which is based on Optimization of Impulse Samples with Object Function of Amplitude, Ripple and Group Delay Characteristics, is developed and is also evaluated by designning and simulating the SAW IF Filter for PCS. In Optimization Process, fast calculation algorithm of Object Function is proposed. With this Design Tools, Transversal SAW IF Filters can be easily designed under limited conditions of small chip size and package size. It may be also applicable to wide Band Pass Filters in future Communication Systems such as FPLMTS.
Sonar Transmitting Beam Generation using Recursion Formula
Heo, Seong-Wook ; Sung, Koeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 94~98
The most commonly used method to generate sonar transmitting beam is extracting digital samples out of memory, which are to excite transducers of the phased array respectively. As several types of signals have been used in sonar to enhance the performance of sonar in various environments, a large amount of memory is required to store them. In this paper, we adopt recursive algorithm to synthesize every different time-delayed signal for transmitting beams with small amount of memory and simple arithmetic operations. The error due to recursive calculation is also analyzed.
DLL Design and Performance Evaluation in Indoor Wireless DS-CDMA System under the Multipath Fading Effects
Im, Sung-Jun ; Ryu, Ho-Jin ; Ryu, Heung-Gyoon ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 99~105
This paper analyzes DLL(Delay lock loop) under the multipath fading effects. The evaluated performance measures include the steady-state timing error probability density function (PDF) and the mean-time-to-lose-lock (MTLL) under multipath fading effects. The discriminator characteristic S(
) is shown to be zero at the point of timing error
that is not zero, and the MTLL decreases as the delayed signal power
and delayed time
increase. We approximate the steady-state timing error PDF linearly with these variables and evaluate the steady-state timing error PDF and MTLL. The severe multipath fading effects result lower MTLL, in this case we make MTLL larger by increasing the early-late discriminator offset
. First, we calculate the timing error point
, and present the performance of DLL under multipath fading. The timing error PDF, MTLL and the performance of DLL with
are also investigated. And we conclude that the larger
makes a higher MTLL and a better performance of DLL under multipath fading effects.
Design of A 3V CMOS Fully-Balanced Complementary Current-Mode Integrator
Lee, Geun-Ho ; Bang, Jun-Ho ; Cho, Seong-Ik ; Kim, Dong-Yong ;
The Journal of the Acoustical Society of Korea, volume 16, issue 3, 1997, Pages 106~113
A 3V CMOS continuous-time fully-balanced integrator for low-voltage analog-digital mixed-mode signal processing is designed in this paper. The basic architecture of the designed fully-balanced integrator is complementary circuit which is composed of NMOS and PMOS transistor. And this complementary circuit can extend transconductance of an integrator. So. the unity gain frequency, pole and zero of integrator are increased by the extended transconductance. The SPICE simulation and small signal analysis results show that the UGF, pole and zero of the integrator is increased larger than those of the compared integrtors. The three-pole active low-pass filter is designed as a application circuit of the fully-balanced integrator, using 0.83V CMOS processing parameter.