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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 18, Issue 8 - Nov 1999
Volume 18, Issue 7 - Oct 1999
Volume 18, Issue 6 - Aug 1999
Volume 18, Issue 5 - Jul 1999
Volume 18, Issue 4 - May 1999
Volume 18, Issue 3 - Apr 1999
Volume 18, Issue 2 - Feb 1999
Volume 18, Issue 1 - Jan 1999
Volume 18, Issue 4E - 00 1999
Volume 18, Issue 3E - 00 1999
Volume 18, Issue 2E - 00 1999
Volume 18, Issue 1E - 00 1999
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FFT에 기반한 병렬 디지털 신호처리시스템의 성능분석
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 3~9
This paper concerns performance of a parallel digital signal processing system. The performance of the system is analyzed in terms of CPU cycles required for 1024-point FFT computation. The number of cycles is estimated in three different approaches; FFT algorithm-based, assembly level source code-based, and probability-based. The results of analysis indicate that on a bus-based system the best performance for FFT is achieved with a single board. Because in some applications like FFT, where frequent data exchanges among processors occur, the number of communication cycles increases as the number of boards. It is observed that inter-board communication degrades overall system performance for the FFT computation. Also shown is that linear increase in performance can be obtained if multiple buses are employed.
The Analysis of Spectrum on the Barkhausen Noise of Hysteresis Loops on Neutron Irradiated Material
심철무 ; 장기옥 ; 박국남 ; 조만순 ;
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 7~7
In relation to a non-destructive evaluation of irradiation damages, the changes in the hysteresis loop and Barkhausen noise amplitude and the harmonics frequency due to a neutron irradiation were measured and evaluated. The Mn-Mo-Ni low alloy steel of RPV was irradiated to a neutron fluence of 2.3 ×
n/㎠ (E ≥1 MeV) at 288℃. The saturation magnetization of neutron irradiated metal did not change. The neutron irradiation caused the coercivity to increase, whereas susceptibility to decrease. The amplitude of Barkhausen noise parameters associated with the domain wall motion were decreased by a neutron irradiation. The spectrum of Barkhausen noise is analysed with an applied frequency of 4 Hz and 8 Hz, sampling time of 50 μ sec and 20 μ sec. The harmonic frequency shows 4 Hz, 8 Hz, 12 Hz and 16 Hz reflected from an unirradiated specimen. On the contrary, the harmonic frequency disappeared on the irradiated specimen. In addition to the amplitude, the harmonic frequency of Barkhausen noise is taken into accounts as a promising tool for monitoring the irradiation induced degradation of the reactor materials such as a SA508 of PWR-RPV steel and a Zr₄ of HANARO-CNH.
A Study on Design and Fabricate of a Intermediate Frequency Band SAW Filter
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 10~15
We have studied a method to design and fabricate the Intermediate Frequency(IF) band pass filter with low shape factor which is used for CDMA base station on the 35°Y-cut X-propagation Quartz substrate. In order to fabricate a device of the low shape factor for the IF SAW filter on this substrate, we employed apodization weighted type interdigital transducer(IDT) as an input and withdrawal weighted type IDT as an output by using impulse modelling method. Also, using the Kaiser-Bessel window function, we have adopted 2200pairs and 1000pairs of input and oueut IDT respectively to minimize the effect of ripple. Furthermore, the width and the space of IDT finger are 3.6 ㎛ and 3.5 ㎛ respectively. Thus, we can have optimal results when the IDT thickness is 6000Å in consideration of the ratio of SAW's wavelength while it's aperture is 2mm for impedance matching. The fabricated SAW filter for CDMA had the property of almost 115.2MHz of a center frequency, less then 1.27MHz of bandwidth, less than 1.3 of shape factor, - l5dB of out band attenuation insertion loss and -45dB of rejection band.
Denoising Based on the Adaptive Lifting
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 13~13
This paper introduces an adaptive wavelet transform based on the lifting scheme, which is applied to signal denoising. The wavelet representation using orthogonal wavelet bases has received widespread attention. Recently the lifting scheme has been developed for the construction of biorthogonal wavelets in the spatial domain. Wavelet transforms are performed through three stages: the first stage or Lazy wavelet splits the data into two subsets, even and odd, the second stage calculates the wavelet coefficients (highpass) as the failure to interpolate or predict the odd set using the even, and the third stage updates the even set using neighboring odd points (wavelet coefficients) to compute the scaling function coefficients (lowpass). In this paper, we adaptively find some of the prediction coefficients for better representation of signals and this customizes wavelet transforms to provide an efficient framework for denoising. Special care has been given to the boundaries, where we design a set of different prediction coefficients to reduce the prediction error.
Optimal Design of Ladder Type SAW Filters
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 16~24
Design of SAW ladder filters has been performed by a rather trial and error method, that is, by modifying the design variables step by step until designed performance of the filter satisfies given specifications. In this work, optimal design method has been developed that automatically determines the detailed pattern of the SAW ladder filter to meet the specification once desired performance is given. As a first step for the development, the analysis tool for the SAW ladder filter has been produced by means of the Smith equivalent circuit analysis technique, and its validity has been verified through comparison of its calculation result with experimental data. With the analysis tool, we have investigated the performance variation of the filter with the change of its design factors, and the result has led to the optimal design algorithm. Validity and efficiency of the algorithm has been checked through test design of several SAW ladder filter samples on the market.
A New Calculation Method for the Radiation Impedance of Transducer with Regular Square Vibrating Surface
김무준 ; 하강열 ;
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 20~20
Although the radiation impedance of a transducer with a regular square surface has been studied by many researchers, the formulas are still very complicated, which results in long computation time and low accuracy. In this paper, we propose a new algorithm for the calculation of acoustic radiation impedance in which the regular square vibrating surface of a transducer is divided into small elements and duplicate calculations are eliminated in the process of calculating mutual effects of the elements. Using this algorithm, shorter computation time and higher accuracy of results can be obtained. As a demonstration, the self and the mutual radiation impedance of transducers with a regular square surface are calculated and the accuracy of the results is evaluated.
Directivity Characteristics of Wide-Band One-Shot Beam Formed with Gaussian Weighting
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 25~31
This paper proposes a new wide-band beamforming algorithm which has Gaussian weighting at nested linear array for acoustic measurement. The beamforming is performed one-shot by using all sensors simultaneously, not octave-by-octave. Gaussian weighting of frequency-dependent function is applied to each sensor before time-delay compensation so as to control the frequency receiving band of each sensor. As the results of the simulations, it is confirmed that the proposed algorithm can form the one-shot beams having uniform directivity index and also it can be applied to the broad-band acoustic measurement.
Automatic Detection of Korean Accentual Phrase Boundaries
이기영 ; 송민석 ;
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 27~27
Recent linguistic researches have brought into focus the relations between prosodic structures and syntactic, semantic or phonological structures. Most of them prove that prosodic information is available for understanding syntactic, semantic and discourse structures. But this result has not been integrated yet into recent Korean speech recognition or understanding systems. This study, as a part of integrating prosodic information into the speech recognition system, proposes an automatic detection technique of Korean accentual phrase boundaries by using one-stage DP, and the normalized pitch pattern. For making the normalized pitch pattern, this study proposes a method of modified normalization for Korean spoken language. For the experiment, this study employs 192 sentential speech data of 12 men's voice spoken in standard Korean, in which 720 accentual phrases are included, and 74.4% of the accentual phrase boundaries are correctly detected while 14.7% are the false detection rate.
A Study on the Evaluation of Subjective Response about Rail Noise in Urban Area
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 32~37
Rail noise and vibration is one of the major problem for the residents living nearby railway tracks. The rail noise may inflict a considerable damage to physiological, spiritual, mental and discomfort to neighboring peoples. Therefore, residents living nearby railway were appeal many troubles. Nevertheless, in our country, adequate guidelines for the rail noise are not yet established because of the lack of basic data and insufficient research works. In this point, this study attempts to surveys the influence of rail noise in urban area using questionnaire. This study also present a basic data in establishing effective plans for rail noise in the future.
Effects of Drilling Degrees of Freedom in the Finite Element Modeling of P- and SV-wave Scattering Problems
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 37~37
This paper deals with a hybrid finite element method for wave scattering problems in infinite domains. Scattering of waves involving complex geometries, in conjunction with infinite domains is modeled by introducing a mathematical boundary within which a finite element representation is employed. On the mathematical boundary, the finite element representation is matched with a known analytical solution in the infinite domain in terms of fields and their derivatives. The derivative continuity is implemented by using a slope constraint. Drilling degrees of freedom at each node of the finite element model are introduced to make the numerical model more sensitive to the transverse component of the elastodynamic field. To verify the effects of drilling degrees freedom and slope constraints individually, reflection of normally incident P and SV waves on a traction free half spaces is considered. For the P-wave incidence, the results indicate that the use of slope constraint is more effective because it suppresses artificial reflection at the mathematical boundary. For the SV-wave case, the use of drilling degrees freedom is more effective by reducing numerical error at irregular frequencies.
Speech Enhancement using Spectral Subtraction and Two Channel Beamfomer
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 38~44
In this paper, a new spectral subtraction technique with two microphone inputs is proposed. In conventional spectral subtraction using a single microphone, the averaged noise spectrum is subtracted from the observed short-time input spectrum. This results in reduction of mean value of noise spectrum only, the component varying around the mean value remaining intact. In the method proposed in this paper, the short-time noise spectrum excluding the speech component is estimated by introducing the blocking matrix used in Griffiths-Jim-type adaptive beamformer with two microphone inputs, combined with the spectral compensation technique. A simulation was conducted to verify the effectiveness of the method.
A Study on the Digital Signal Processing for Removing the Bottle-neck Effect
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 45~52
In this thesis, a packer is proposed and designed for removing the bottle-neck effect and easy signal processing using a new algorithm with the operation frequency of 54MHz. To verify the performance of the proposed packer, DCT coefficient encoding block with ROM table using a combinational logic is designed and its output data are used the input data of the packer. Circuits in this thesis are designed by using VHDL code and its modeling and simulation are performed by SYNOPSYS tool using 0.65㎛
A Study on the Korean Broadcasting Speech Recognition
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 53~60
This paper is a study on the korean broadcasting speech recognition. Here we present the methods for the large vocabuary continuous speech recognition. Our main concerns are the language modeling and the search algorithm. The used acoustic model is the uni-phone semi-continuous hidden markov model and the used linguistic model is the N-gram model. The search algorithm consist of three phases in order to utilize all available acoustic and linguistic information. First, we use the forward Viterbi beam search to find word end frames and to estimate related scores. Second, we use the backword Viterbi beam search to find word begin frames and to estimate related scores. Finally, we use A/sup */ search to combine the above two results with the N-grams language model and to get recognition results. Using these methods maximum 96.0% word recognition rate and 99.2% syllable recognition rate are achieved for the speaker-independent continuous speech recognition problem with about 12,000 vocabulary size.
The Comparison of Speaker Adaptation Methods
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 61~66
In this paper, we proposed various speaker adaptation methods and studied the performance of these methods. Methods which were studied in this paper are MAPE(Maximum A Posteriori Probability Estimation), Linear Spectral Estimating, Multi-Layer Perceptron and ARTMAP. In order to evaluate the performance of these methods, we used Korean isolated digits as the experimental data, the hybrid speaker adaptation method, which unified MAPE, linear spectral estimating and output probability of SCHMM, showed the better recognition result than those which performed other methods. And the method using ARTMAP showed the similar result to above hybrid method.
Design of BIST Circuits for Test Algorithms Using VHDL
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 67~71
In this paper, we design circuits embedded in memory chip which perform memory testing algorithms using BIST scheme to reduce testing time and cost for testing. In order to implement circuits for MSCAN, Marching and checkerboard test algorithms, which have widely used in memory testing, we survey structure of the BIST circuits and describe each block of BIST circuits by using VHDL. Thereafter, We verify behavior of each VHDL coding block and extract BIST circuits for target testing algorithms by CAD tool for simulation and synthesis. Extracted circuits have very low area overhead.
A Wavelet Approach to Broadcast Video Traffic Modeling
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 72~77
In this paper, we propose a wavelet VQ approach to modeling VBR broadcast video traffic. The proposed method decomposes video traffic into two parts via wavelet transformation, and models each part separately. The first part, which is modeled by an AR(1) process, serves to capture the long-term trend of the traffic; the second part, classified via vector quantization, addresses the short-term behavior of the traffic. Compared with other VBR video models, our model has three advantages. First, it allows the separate modeling of long- and short-term behavior of the video traffic; second, it preserves the periodic coding structure in traffic data; and third, it provides an unified approach for the frameand slice-level traffic modeling. We demonstrate the validity of our model by statistical measurements and network performance simulation.
Performance Improvement of Mel-Cepstrum Through Optimzing Filter Banks
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 78~85
In this paper we propose a method to improve the performance of the mel-cepstrum that is widely used in speech recognition. Typically, the met-cepstrum is obtained by critical band filters that have fixed center spacing and bandwidth. However different filter characteristics produce a different mel-cepstrum, resulting in a different performance. In this paper we analyze triangular-shaped and rectangular-shaped filters. By changing the characteristics of filters such as center frequency and bandwidth, we analyze the performance of the met-cepstrum. Then utilizing the simplex method, we propose a method to optimize the critical band filters. Using the dynamic time warping, we performed speaker independent recognition experiments with Korean digit words pronounced by 10 males and 10 females. Experiments show that the rectangular-shaped filters show good performance and the mel-cepstrum obtained by the optimized filters shows better performance than filters that have fixed center spacing and bandwidth.
An Enhanced Text-Prompt Speaker Recognition Using DTW
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 86~91
This paper presents the text-prompt method to overcome the weakness of text-dependent and text-independent speaker recognition. Enhanced dynamic time warping for speaker recognition algorithm is applied. For the real-time processing, we use a simple algorithm for end-point detection without increasing computational complexity. The test shows that the weighted-cepstrum is most proper for speaker recognition among various speech parameters. As the experimental results of the proposed algorithm for three prompt words, the speaker identification error rate is 0.02%, and when the threshold is set properly, false rejection rate is 1.89%, false acceptance rate is 0.77% and verification total error rate is 0.97% for speaker verification.
A Fault Simulator for IDDQ Testing
The Journal of the Acoustical Society of Korea, volume 18, issue 1, 1999, Pages 92~96
As CMOS technologies have been rapidly developed, bridging faults have been relatively increased. IDDQ testing is a current testing methodology which can enhance reliability of the circuit since it efficiently detects bridging faults that are difficult to detect by functional testing. In this paper we consider internal bridging faults occurred in each gate of logic circuits under test and finally develop a fault simulator for IDDQ testing to detect assumed bridging faults.