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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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Experimental Results of Performance of CFAR Detectors in Active Sonar Environment
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 3~9
In this paper, the characteristics of LFM and CW signals in active sonar environment is investigated. CA, OS and TM CFAR processors are applied to the received CW/LFM signals which are plotted in the range/doppler domain. The performances of detection are analyzed. Particularly, using the real data, we certified that the results of the experiments are identical with the theoretical performance.
Speech Synthesis Based on CVC Speech Segments Extracted from Continuous Speech
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 10~16
In this paper, we propose a concatenation-based speech synthesizer using CVC(consonant-vowel-consonant) speech segments extracted from an undesigned continuous speech corpus. Natural synthetic speech can be generated by a proper modelling of coarticulation effects between phonemes and the use of natural prosodic variations. In general, CVC synthesis unit shows smaller acoustic degradation of speech quality since concatenation points are located in the consonant region and it can properly model the coarticulation of vowels that are effected by surrounding consonants. In this paper, we analyze the characteristics and the number of required synthesis units of 4 types of speech synthesis methods that use CVC synthesis units. Furthermore, we compare the speech quality of the 4 types and propose a new synthesis method based on the most promising type in terms of speech quality and implementability. Then we implement the method using the speech corpus and synthesize various examples. The CVC speech segments that are not in the speech corpus are substituted by demonstrate speech segments. Experiments demonstrate that CVC speech segments extracted from about 100 Mbytes continuous speech corpus can produce high quality synthetic speech.
Prosodic Break Index Estimation using LDA and Tri-tone Model
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 17~22
In this paper we propose a new mixed method of LDA and tri-tone model to predict Korean prosodic break indices(PBI) for a given utterance. PBI can be used as an important cue of syntactic discontinuity in continuous speech recognition(CSR). The model consists of three steps. At the first step, PBI was predicted with the information of syllable and pause duration through the linear discriminant analysis (LDA) method. At the second step, syllable tone information was used to estimate PBI. In this step we used vector quantization (VQ) for coding the syllable tones and PBI is estimated by tri-tone model. In the last step, two PBI predictors were integrated by a weight factor. The proposed method was tested on 200 literal style spoken sentences. The experimental results showed 72% accuracy.
Analysis of the Convergence Properties of LMS and VS-LMS Algorithms for IIR Filters
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 23~32
This paper presents a stochastic convergence analysis of LMS algorithm and VS-LMS algorithm for IIR filters using equation error formulation. Under the assumption that the signal is white Gaussian, theoretical equations that characterize the mean and mean-squared behaviors of the algorithms are derived. Computer simulation results show fairly good agreements between the theoretical and the empirical behaviors of the algorithms.
Design of a 4kb/s ACELP Codec Using the Generalized AbS Principle
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 33~38
In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.
An Analysis of the Vibrational Modes for a Rectangular Plate by Using the Double Fourier Sine Series Method
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 39~44
An analysis of the frequency parameters and vibrational modes is described for a rectangular plate. Double Fourier sine series is used as a modal displacement functions of a rectangular plate and applied to the free vibration analysis of a rectangular plate under various boundary conditions. The frequency parameters obtained by the double Fourier sine series method are compared with those obtained by the theory of finite element method and Ritz method. Frequency parameters are presented for the various aspect ratios for plate. The first four modal shapes for the rectangular plate under various boundary conditions are accurately described.
Flaw Detection of Ultrasonic NDT in Heat Treated Environment Using WLMS Adaptive Filter
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 45~55
In this paper, we used the WLMS(Wavelet domain Least Mean Square) adaptive filter based on the wavelet transform to cancel grain noise. Usually, grain noise occurs in changes of the crystalline structure of metals in high temperature environment. It makes the detection of flaw difficult. The WLMS adaptive filtering algorithm establishes the faster convergence rate by orthogonalizaing the input vector of adaptive filter as compared with that of LMS adaptive filtering algorithm in time domain. We implemented the WLMS adaptive filter by using the delayed version of the primary input vector as the reference input vector and then implemented the CA-CFAR(Cell Averaging- Constant False Alarm Rate) threshold estimator. CA-CFAR threshold estimator enables to detect the flaw and back echo signals automatically. Here, we used the output signals of adaptive filter as its input signal. To Cow the statistical characteristic of ultrasonic signals corrupted by grain noise, we performed run test. The results showed that ultrasonic signals are nonstationary signal, that is, signals whose statistical properties vary with time. The performance of each filter is appreciated by the signal-to-noise ratio. After LMS adaptive filtering in time domain, SNR improves to about 2-3㏈ but after WLMS adaptive filtering in wavelet domain, SNR improves to about 4-6㏈.
Prediction of Thermo-acoustic Oscillation Characteristics in a Ducted Combustor
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 56~66
Thermoacoustic oscillation is a significant problem in cylindrical-type combustors such as common internal combustion engines, industrial furnaces, gas turbine, etc. This kind of low frequency oscillation can lead to serious consequences such as destruction of the combustor and production of strong noise. The accurate numerical simulation of thermoacoustic phenomena is a complex and challenging problem, especially when considering the chemical reaction of mixtures. As with other simulations of aerodynamics and aeroacoustics, the direct computation of thermoacoustic phenomena requires that Navier-Stokes equations be solved using accurate numerical differentiation and time-marching schemes, with non-reflecting boundary conditions. The numerical approach used here aims at qualitative analysis and efficient prediction of those problems, not at the development of an accurate scheme. The numerical prediction developed in this work is shown to be reasonably matched with experimental result.
A Study on the Frequency Characteristics of a Class IV Flextensional Transducer
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 67~73
We constructed a class Ⅳ Flextensional transducer, and analyzed the effects of the variation in its material properties and structure on the resonance frequency of the transducer. We used the FEM for the analysis. Major axis length, minor axis length, thickness and material properties of the shell turned out to have large effects, while volume of the insert material, material properties and thickness of the ceramic bar have little. Thickness of the nodal-plate has no effect on the resonance frequency. Results of the present work can be utilized to design Class W Flextensional transducers of various resonance frequency.
Effect of Different Misfired Source on Seismic Survey Quality
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 75~79
The beam patterns of source array and changes in the far-field signatures are compared and analyzed each other in order to identify the seismic capability affected by the misfired source at the multi-channel seismic source array. In the primary pulse amplitude of far-field signature, the 66% of seismic capacity are sustained if approximately 40% of source are misfired among whole gun volume. When the sources with the same distances are misfired at the 154㎐, the beam width of the long- and wide-array which is identical regardless of arraying pattern. The beam width has a tendency to narrow now from 41 to 34 according to increase the volume of misfired source at the long-array beam pattern. Therefore, the source array of small volume are suitable for the shallow seismic survey because of producing adequate beam patterns with narrow beam width.
On a Split Model for Analysis Techniques of Wideband Speech Signal
Park, Young-Ho ; Ham, Myung-Kyu ; You, Kwang-Bock ; Bae, Myung-Jin ;
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 80~84
In this paper, the split model analysis algorithm, which can generate the wideband speech signal from the spectral information of narrowband signal, is developed. The split model analysis algorithm deals with the separation of the 10/sup th/ order LPC model into five cascade-connected 2/sup nd/ order model. The use of the less complex 2/sup nd/ order models allows for the exclusion of the complicated nonlinear relationships between model parameters and all the poles of the LPC model. The relationships between the model parameters and its corresponding analog poles is proved and applied to each 2/sup nd/ order model. The wideband speech signal is obtained by changing only the sampling rate.
On the relationship between the phonetic realizations of the allophones of the Korean liquid /l/ and their prosodic status
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 85~91
The purpose of this study is to investigate phonetic realization of flap [r], one of the allophones of Korean /l/. Phonetic realization of a segment is affected by not only its neighboring segments but also its prosodic position in an utterance. This study examined how various prosodic positions affect the phonetic realization of [r]. Effects of the four prosodic positions on the phonetic realization of [r] were examined: utterance initial, Intonation Phrase initial, Accentual Phrase initial, and Accentual Medial positions. Word positional effect was also examined: word initial, medial, and final positions. Acoustic and statistical analyses showed that flap [r] was realized in a variety of phonetic forms: from sonorant(the most reduced form) to short stop(the least reduced form). It was shown that generally. word-initial position is stronger than word-medial position. It was also shown that in many cases, utterance-initial position and intonation-phrase-initial position are stronger than accentual-phrase-initial and accentual-phrase-medial positions. Sonorants were observed more often in the prosodically weaker portions. VOT duration was also shorter in accentual-phrase-initial and accentual-phrase-medial positions.
An Propagation Path Analysis for Optimal Position Selection of Microcell Base Station in the Mobile Communication System
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 92~100
In the microcell mobile communication, we propose algorithms processing operational disposition to exactly analysis propagation environments from the base station to mobile stations. Algorithms are developed by the triangle analysis method can operate variable propagation paths and reflect numbers. For simulation, we suppose that mobile stations are located in the shadow region of the line of sight and the area of the non-line of sight sloping against the line of sight area at variable angles. By analyzing the results of simulation using proposed algorithms, we can be applied to the optimal position selection of the base station in the microcell mobile communication.
Estimation of Concrete Strength Based on Artificial Intelligence Techniques
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 101~111
This paper presents concrete pattern recognition method to identify the strength of concrete by evidence accumulation with multiple parameters based on artificial intelligence techniques. At first, variance(VAR), zero-crossing(ZCR), mean frequency(MEANF), and autoregressive model coefficient(ARC) and linear cepstrum coefficient(LCC) are extracted as feature parameters from ultrasonic signal of concrete. Pattern recognition is carried out through the evidence accumulation procedure using distance measured with reference parameters. A fuzzy mapping function is designed to transform the distances for the application of the evidence accumulation method. Results(92% successful pattern recognition rate) are presented to support the feasibility of the suggested approach for concrete pattern recognition.
A Study on Variation and Determination of Gaussian function Using SNR Criteria Function for Robust Speech Recognition
The Journal of the Acoustical Society of Korea, volume 18, issue 7, 1999, Pages 112~117
In case of spectral subtraction for noise robust speech recognition system, this method often makes loss of speech signal. In this study, we propose a method that variation and determination of Gaussian function at semi-continuous HMM(Hidden Markov Model) is made on the basis of SNR criteria function, in which SNR means signal to noise ratio between estimation noise and subtracted signal per frame. For proving effectiveness of this method, we show the estimation error to be related with the magnitude of estimated noise through signal waveform. For this reason, Gaussian function is varied and determined by SNR. When we test recognition rate by computer simulation under the noise environment of driving car over the speed of 80㎞/h, the proposed Gaussian decision method by SNR turns out to get more improved recognition rate compared with the frequency subtracted and non-subtracted cases.