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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 22, Issue 8 - Nov 2003
Volume 22, Issue 7 - Oct 2003
Volume 22, Issue 6 - Aug 2003
Volume 22, Issue 5 - Jul 2003
Volume 22, Issue 4 - May 2003
Volume 22, Issue 3 - Apr 2003
Volume 22, Issue 2 - Feb 2003
Volume 22, Issue 1 - Jan 2003
Volume 22, Issue 1E - 00 2003
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An Implementation of Acoustic Echo Canceller Using Adaptive Filtering in Modulated Lapped Transform Domain
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 425~433
Acoustic Echo Canceller (AEC) is a signal processing system for removing unwanted echo signals in teleconference and hands-free communication. Least mean square (LMS) algorithm is one of the adaptive echo cancellation algorithms and it has been most attractive because of its simplicity and robustness. However, the convergence properties of the LMS algorithm degrade with highly correlated input signals such as speech. For this reason, transform-domain adaptive filtering algorithm was introduced to decorrelate the colored input samples by using the orthogonal transform matrix such as DCT, DFT and then LMS adaptive filtering process is applied. In this paper, we propose a MLT domain adaptive echo canceller base on the MLT (Modulated lapped Transform) orthogonal transform matrix. The proposed algorithm achieves high decorrelation efficiency and fast convergence speed via modulated lapped transform of size 2NXN instead of NXN unitary transform such as DCT, DFT, Hadamad and it is applied to the acoustical echo cancellation system. Form the computer simulation with both synthesis and real speech, the proposed MLT domain adaptive echo canceller shows approximately twice faster convergence speed and 20∼30 ㏈ ERLE improvements over the DCT frequency domain acoustic echo cancellation system.
A Study on the Automatic Lexical Acquisition for Multi-lingustic Speech Recognition
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 434~442
Software internationalization, the process of making software easier to localize for specific languages, has deep implications when applied to speech technology, where the goal of the task lies in the very essence of the particular language. A greatdeal of work and fine-tuning has gone into language processing software based on ASCII or a single language, say English, thus making a port to different languages difficult. The inherent identity of a language manifests itself in its lexicon, where its character set, phoneme set, pronunciation rules are revealed. We propose a decomposition of the lexicon building process, into four discrete and sequential steps. For preprocessing to build a lexical model, we translate from specific language code to unicode. (step 1) Transliterating code points from Unicode. (step 2) Phonetically standardizing rules. (step 3) Implementing grapheme to phoneme rules. (step 4) Implementing phonological processes.
Design of a Variable Bit Rate Speech Coder Based on One-dimensional SPIHT
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 443~451
Since a codebook-based CELP coder models its excitation signal according to one of several bit rates pre-assigned to codebooks and synthesizes speech signal using codebooks, it can not support encoding of speech signal at an arbitrary bit rate in one encoder. The proposed variable bit rate speech coder encodes the excitation signal based on the bit rate assigned to a present frame of speech using one-dimensional SPIHT and wavelet transform. Also it does't need to model excitation signal (or codebook) to some types as CELP coder, and can encode excitation signal at various bit rates without exact pitch information according to user requirement. As a result, since the coder doesn't have a codebook structure, it has relatively low coder complexity and provides equal or better speech quality compared to G.729 and G.723.1 coder.
The Reduction or computation in MLLR Framework using PCA or ICA for Speaker Adaptation
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 452~456
We discuss how to reduce the number of inverse matrix and its dimensions requested in MLLR framework for speaker adaptation. To find a smaller set of variables with less redundancy, we adapt PCA (principal component analysis) and ICA (independent component analysis) that would give as good a representation as possible. The amount of additional computation when PCA or ICA is applied is as small as it can be disregarded. 10 components for ICA and 12 components for PCA represent similar performance with 36 components for ordinary MLLR framework. If dimension of SI model parameter is n, the amount of computation of inverse matrix in MLLR is proportioned to O(n⁴). So, compared with ordinary MLLR, the amount of total computation requested in speaker adaptation is reduced by about 1/81 in MLLR with PCA and 1/167 in MLLR with ICA.
A Speaker Pruning Method for Reducing Calculation Costs of Speaker Identification System
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 457~462
In this paper, we propose a speaker pruning method for real-time processing and improving performance of speaker identification system based on GMM(Gaussian Mixture Model). Conventional speaker identification methods, such as ML (Maximum Likelihood), WMR(weighting Model Rank), and MWMR(Modified WMR) we that frame likelihoods are calculated using the whole frames of each input speech and all of the speaker models and then a speaker having the biggest accumulated likelihood is selected. However, in these methods, calculation cost and processing time become larger as the increase of the number of input frames and speakers. To solve this problem in the proposed method, only a part of speaker models that have higher likelihood are selected using only a part of input frames, and identified speaker is decided from evaluating the selected speaker models. In this method, fm can be applied for improving the identification performance in speaker identification even the number of speakers is changed. In several experiments, the proposed method showed a reduction of 65% on calculation cost and an increase of 2% on identification rate than conventional methods. These results means that the proposed method can be applied effectively for a real-time processing and for improvement of performance in speaker identification.
APMP (Asia Pacific Metrology Programme) Regional Intercomparison Results of Acoustic Calibrators
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 463~471
The results of the APMP (Asia Pacific Metrology Programme) regional intercomparison of acoustic calibrators were reviewed and analyzed. The artefacts used in intercomparison are a sound level calibrator and a pistonphone. The microphones used to measure the output pressure level are 1" and 1/2" standard microphones (LS1P, LS2P) as well as 1" and 1/2" reference microphones (WS1P/F, WS2P/F). The results obtained using standard microphones are satisfactory, but those obtained by the reference microphones, even though E/sub n/ values are within ±1.0, showed great deviations. Such results had come from the inaccurate calibration of reference microphones. By using the correct calibration results which were obtained by the recently established international standards, the new results were very similar to those of the foreign standard institutes.
Bi-static Low-frequency Reverberation Model in Shallow Water
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 472~481
Low-frequency hi-static reverberation model (LHYREV-B, Low-frequency Hanyang univ. Reverberation model-Bistatic) based on the parabolic approximation for shallow water environment is presented. In this paper bistatic reverberation level is computed using the angle-independent scattering strength function and the wave-based acoustic model. The signal simulated by the LHYREV-B model is compared with the observed signals and it is shown that the LHYREV-B model provides a closer fit to the observed signals.
Signal Processing for Improvement of Resolution and Direction Ambiguity of Source in Line Array
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 482~488
Line may receiver contains an ambiguity on conjugate bearings, because of lacking aperture in transverse direction. To solve the left/right bearing ambiguity of line-array receiver we used line-array with cardioid beam. In addition, the synthetic aperture method adopts coherent processing of sub-aperture signals at successive time intervals in the beam domain. We presented performances about division of left/right bearing ambiguity according to array synthetic number of times in this paper.
Left/Right Bearing Discrimination with Adaptive Cardioid Beamforming
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 489~495
Single towed line array receiver contains an ambiguity on conjugate bearings because of lacking aperture in transverse direction. To solve the left/right bearing ambiguity of line array receiver this paper proposed using single line array with fixed cardioid beam. Fixed cardioid beam has problem about back beam gain exists for steering beam inherent. Back beam is makes form on direction that is different from actually source so that reduced the performance of left/right bearing discrimination. In this paper, line way with adaptive cardioid beam for resolve problem of back beam gain is proposed. So the proposed method has more improved left/right bearing discrimination than fixed cardioid beam, Simulation results show the performance of the proposed method.
Analysis of Error Tolerance in Sonar Array by the Genetic Algorithm
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 496~504
In this paper, the error tolerance of each array element to ensure a given specified error level for the array pattern is analyzed using the Genetic Algorithm. In the conventional deterministic method for synthesis of sonar way problems the computational resource required in the simulation grows rapidly as the number of way elements increases. To alleviate this numerical inefficiency, the Monte-Carlo method may be considered as an alternative technique for array syntheses. However, it is difficult to apply the method to the synthesis of array patterns because of its relatively lower accuracy in spite of moderate computational complexity. A new analysis method for estimating error tolerances in sonar arrays is Proposed since the Genetic Algorithm has significant promise to efficiently solve way synthesis problems. Through several numerical tests in linear and planar arrays, it is demonstrated that the proposed method can provide accurate results for error tolerances of sonar arrays.
Vibration Analysis of Automobile Tire Due to Road Impact
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 505~511
As the technique of automobile industry is being advanced, the advancement of vehicle ride is being required. In order to achieve this purpose, the study on the vibration which are produced by moving vehicle is carried out actively. In order to analysis, the tire vibration characteristics for passing over a cleat, the tire is modeled with 7-DOFs (degree of freedom). The model is verified against simulations and experiments. The effects of proposed tire design parameters such as the tire tread rubber, tread ring, apex are considered. According to the results of analysis, the tire design parameters that can reduce the tire and wheel vibration quantity are conducted.
A Study on Resonance Durability Analysis of Vehicle Suspension System
The Journal of the Acoustical Society of Korea, volume 22, issue 6, 2003, Pages 512~518
In this paper, resonance durability analysis is performed for the fatigue life assessment considering vibration effect of a vehicle system. In the resonance durability analysis, the frequency response and the dynamic load on frequency domain are used. Multi-body dynamic analysis, finite element analysis, and fatigue life prediction method are applied for the virtual durability assessment. To obtain the frequency response and the dynamic load history, the computer simulations running over typical pothole and Belgian road are carried out by utilizing vehicle dynamic model. The durability estimations on the rear suspension system of the passenger car are performed by using the resonance durability analysis technique and compared with the quasi-static durability analysis. The study shows that the fatigue life considering resonant frequency of vehicle system can be effectively estimated in early design stage.