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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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Adaptive Filtering Algorithms for Stereophonic Acoustic Echo Cancellers
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 3~11
The conventional stereophonic acoustic echo cancellers need two adaptive filters to estimate one channel echo signal. Since the two channel signals are strongly correlated, the ESR of the input signals is considerably increased whatever the input signals may be. This causes the slow convergence of the adaptive filter for echo cancellation. To speed up the convergence, the AP algorithm is frequently used for the stereophonic acoustic echo canceller although there isn't a fast version for 2-channel case. The AP algorithm can be approximated with the Gram-Schmidt orthogonalization and a TDL structure. We propose a two channel algorithm for stereophonic acoustic echo canceller with the approximated AP algorithm.
The Effect of the Telephone Channel to the Performance of the Speaker Verification System
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 12~20
In this paper, we compared speaker verification performance of the speech data collected in clean environment and in channel environment. For the improvement of the performance of speaker verification gathered in channel, we have studied on the efficient feature parameters in channel environment and on the preprocessing. Speech DB for experiment is consisted of Korean doublet of numbers, considering the text-prompted system. Speech features including LPCC(Linear Predictive Cepstral Coefficient), MFCC(Mel Frequency Cepstral Coefficient), PLP(Perceptually Linear Prediction), LSP(Line Spectrum Pair) are analyzed. Also, the preprocessing of filtering to remove channel noise is studied. To remove or compensate for the channel effect from the extracted features, cepstral weighting, CMS(Cepstral Mean Subtraction), RASTA(RelAtive SpecTrAl) are applied. Also by presenting the speech recognition performance on each features and the processing, we compared speech recognition performance and speaker verification performance. For the evaluation of the applied speech features and processing methods, HTK(HMM Tool Kit) 2.0 is used. Giving different threshold according to male or female speaker, we compare EER(Equal Error Rate) on the clean speech data and channel data. Our simulation results show that, removing low band and high band channel noise by applying band pass filter(150～3800Hz) in preprocessing procedure, and extracting MFCC from the filtered speech, the best speaker verification performance was achieved from the view point of EER measurement.
Improvements of an English Pronunciation Dictionary Generator Using DP-based Lexicon Pre-processing and Context-dependent Grapheme-to-phoneme MLP
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 21~27
In this paper, we propose an improved MLP-based English pronunciation dictionary generator to apply to the variable vocabulary word recognizer. The variable vocabulary word recognizer can process any words specified in Korean word lexicon dynamically determined according to the current recognition task. To extend the ability of the system to task for English words, it is necessary to build a pronunciation dictionary generator to be able to process words not included in a predefined lexicon, such as proper nouns. In order to build the English pronunciation dictionary generator, we use context-dependent grapheme-to-phoneme multi-layer perceptron(MLP) architecture for each grapheme. To train each MLP, it is necessary to obtain grapheme-to-phoneme training data from general pronunciation dictionary. To automate the process, we use dynamic programming(DP) algorithm with some distance metrics. For training and testing the grapheme-to-phoneme MLPs, we use general English pronunciation dictionary with about 110 thousand words. With 26 MLPs each having 30 to 50 hidden nodes and the exception grapheme lexicon, we obtained the word accuracy of 72.8% for the 110 thousand words superior to rule-based method showing the word accuracy of 24.0%.
The Interaction Between Stress Waves in Elastic Solids for an Ultrasonic Viscometer and Adjacent Viscous Fluids
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 28~34
The effects of the viscosity of an adjacent viscous fluid on the characteristics of the elastic waves have been studied theoretically and experimentally. Expressions for the wave speed and attenuation of the elastic waves of transverse motion, such as the torsional wave propagating in a circular cylinder and the Love wave in a layered half-space solid, have been obtained as functions of the viscosity and mass density of the fluid by exact and asymptotic analyses. The theoretical results have been compared with experimental observations, and it has been demonstrated that a device described herein can be used as a sensor for measuring the viscosity of a fluid with a known mass density.
Performance Improvement of Acoustic Echo Canceller Using Post-Processor
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 35~43
In this paper, a new robust adaptive algorithm and a post-processing method are proposed to improve the performance of AEC without computational burden. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. To reduce the residual echoes, a new post-processing method, which is co-operated with the proposed noise-robust adaptive algorithm, is proposed in this paper. The method is based on the correlation of the desired signal and the estimation error signal. The residual echoes are attenuated as proportional to the correlation normalized with the power of desired signals. The normalized correlation plays a role as Wiener filter for residual echoes. In the double-talk situation, the estimation error signals, that are residual echoes, dominantly include the near-end speaker's speech and the normalized correlation closes to 1. Therefore, the near-end speaker's speech can be transmitted without being attenuated. When the desired signals consists of only the acoustic echoes, the residual echoes are mostly attenuated and canceled by the proposed post-processor. The computation of AEC using the proposed post-processor is comparable to NLMS algorithm.
A Parallel Speech Recognition Model on Distributed Memory Multiprocessors
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 44~51
This paper presents a massively parallel computational model for the efficient integration of speech and natural language understanding. The phoneme model is based on continuous Hidden Markov Model with context dependent phonemes, and the language model is based on a knowledge base approach. To construct the knowledge base, we adopt a hierarchically-structured semantic network and a memory-based parsing technique that employs parallel marker-passing as an inference mechanism. Our parallel speech recognition algorithm is implemented in a multi-Transputer system using distributed-memory MIMD multiprocessors. Experimental results show that the parallel speech recognition system performs better in recognition accuracy than a word network-based speech recognition system. The recognition accuracy is further improved by applying code-phoneme statistics. Besides, speedup experiments demonstrate the possibility of constructing a realtime parallel speech recognition system.
A Design of 16-QAM Modulator by use of Direct Digital Frequency Synthesizer
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 52~57
It is very important to design of QAM modulator of high spectral efficiency for high speed data transmission. In this paper, typical 16-QAM modulator is designed by modification design of DDFS(direct digital frequency synthesizer). DDFS generates sinusoidal waveform digitally to the frequency setting word. Phase modulation is accuratly made by control of a generated phase increment value and amplitude modulation is accomplished in the D/A converter output by control of amplitude level. For the suppression of harmonics and glitch, dual-structured DDFS is studied to improve the spurious characteristics. P-Spice is used for design and simulation in mixed mode. Also we can get the satisfactory results of designed 16-QAM modulator from the constellation output.
Performance Improvement of Korean Connected Digit Recognition Based on Acoustic Parameters
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 58~62
This paper proposes use of acoustic parameters to improve the discriminability among digit models in Korean connected digit recognition. The proposed method used the logarithmic values of energy ratio between the predetermined frequency bands as additional feature parameters, based on the acoustic-phonetic knowledge. The results of our experiment show that the proposed method reduced the error rate by 46% in comparison with the baseline system. And incorporation of channel compensation technique in the proposed method yielded error reduction of about 69%.
Room Acoustic Measurement System Using Impulse Response
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 63~67
Recently, a method of measuring impulse response is widely used for a room acoustic evaluation instead of measuring reverberation time by white noise excitation. Comparing with the traditional reverberation time measurement, this method has many advantages such as good repeatability and the ability to extract various room acoustic parameters at one measurement. In this study, the author developed a measuring system that can extract mono-aural room acoustic parameters from an impulse response measured with MLS (Maximum Length Sequence) signal excitation. These room acoustic parameters include reverberation times(EDT, RT), speech intelligibilities(C50, C80, D, U50, U80, AI) and sound strength(G). This paper introduces the configuration of the developed measuring system, test results and discussions for the measurements at several rooms.
Cavitation Suppression Effects by the Modification of the Spectral Characteristics of High Intensity Focused Ultrasound
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 68~77
The paper looked into the effects of the spectral properties (waveform) of the high intensity focused ultrasound on suppression of the ultrasonic cavitation. Three different types of ultrasound were considered in the study, which were sinusoidal (1 MHz, 5 MPa), frequency modulated (from 1 MHz to 6 MHz for 10 ㎲, 5 MPa), asymmetrically shocked (fundamental frequency 1 MHz, peak positive pressure 12 MPa, peak negative pressure -4 MPa). The temporal response of an air bubble in water initially 1 ㎛ in radius to each type of the ultrasound was predicted using Gilmore bubble dynamic model and Church's rectified gas diffusion equation. It was shown that the radially pulsating amplitude of the bubble was greatly reduced for the frequency modulated wave and was little decreased for the shock wave, compared to the case that the bubble was exposed to the sinusoidal wave. It is interesting that the bubble response to the frequency modulated wave remains similar when the frequency component of the modulated ultrasound is beyond the bubble resonant frequency 3 MHz. This implies that, although the ultrasound is modulated up to 3MHz rather than up to the present 6 MHz, it is likely to produce similar cavitation suppression effects. In practice, it means that a typical narrow band ultrasonic transducer can be taken to generate an appropriate frequency modulated ultrasound to reduce cavitation activity. The present study indicates that ultrasonic cavitation may be suppressed to some extent by a proper spectral modification of high intensity ultrasound.
Glottal Weighted Cepstrum for Robust Speech Recognition
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 78~82
This paper is a study on weighted cepstrum used broadly for robust speech recognition. Especially, we propose the weighted function of asymmetric glottal pulse shape. which is used for weighted cepstrum extracted by PLP(Perceptual Linear Predictive) based on auditory model. Also, we analyze this glottal weighted cepstrum from the glottal pulse of glottal model in connection with the cepstrum. And we obtain speech features analyzed by both the glottal model and the auditory model. The isolated-word recognition rate is adopted for the test of proposed method in the car moise and street environment. And the performance of glottal weighted cepstrum is compared with both that of weighted cepstrum extracted by LP(Linear Prediction) and that of weighted cepstrum extracted by PLP. The result of computer simulation shows that recognition rate of the proposed glottal weighted cepstrum is better than those of other weighted cepstrums.
The Analysis of Resolution on the Image Reconstnlction Algorithms for Ultrasonic Diffraction Tomography
The Journal of the Acoustical Society of Korea, volume 18, issue 5, 1999, Pages 83~90
In this paper, we studied resolution to the FBP and BFP image reconstruction algorithms for ultrasonic diffraction tomography. In order to analyze the resolution to the tomographic images which can be reconstructed from the modified FBP image reconstruction algorithm by using fixed coordinate system and BFP image reconstruction algorithm which is suitable for plane structure object, we derived ambiguity functions to these algorithms and then analyzed lateral and depth resolution through simulation respectively. Simulation results show that the lateral and depth resolution to the FBP image reconstruction algorithm and the BFP image reconstruction algorithm was determined 0.27 λ, 0.70 λ and 0.39 λ, 0.98 λ at the 3dB respectively. These results imply that modified FBP and BFP image reconstruction algorithms for diffraction tomography is useful in the tomographic image reconstruction.