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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 21, Issue 8 - Nov 2002
Volume 21, Issue 7 - Oct 2002
Volume 21, Issue 6 - Aug 2002
Volume 21, Issue 5 - Jul 2002
Volume 21, Issue 4 - May 2002
Volume 21, Issue 3 - Apr 2002
Volume 21, Issue 2 - Feb 2002
Volume 21, Issue 1 - Jan 2002
Volume 21, Issue 1E - 00 2002
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A Method on the Learning Speed Improvement of the Online Error Backpropagation Algorithm in Speech Processing
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 430~437
Having a variety of good characteristics against other pattern recognition techniques, the multilayer perceptron (MLP) has been widely used in speech recognition and speaker recognition. But, it is known that the error backpropagation (EBP) algorithm that MLP uses in learning has the defect that requires restricts long learning time, and it restricts severely the applications like speaker recognition and speaker adaptation requiring real time processing. Because the learning data for pattern recognition contain high redundancy, in order to increase the learning speed it is very effective to use the online-based learning methods, which update the weight vector of the MLP by the pattern. A typical online EBP algorithm applies the fixed learning rate for each update of the weight vector. Though a large amount of speedup with the online EBP can be obtained by choosing the appropriate fixed rate, firing the rate leads to the problem that the algorithm cannot respond effectively to different learning phases as the phases change and the number of patterns contributing to learning decreases. To solve this problem, this paper proposes a Changing rate and Omitting patterns in Instant Learning (COIL) method to apply the variable rate and the only patterns necessary to the learning phase when the phases come to change. In this paper, experimentations are conducted for speaker verification and speech recognition, and results are presented to verify the performance of the COIL.
A New Analysis and a Reduction Method of Computational Complexity for the Lattice Transversal Joint (LTJ) Adaptive Filter
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 438~445
In this paper, the necessity of the filter coefficients compensation for the lattice transversal joint (LTJ) adaptive filter was explained in general and with ease by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed using the property that speech signal is stationary during a short time period and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%.
The Relative Position Estimate of the Moving Distributed Sources Using the Doppler Scanning Technique
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 446~454
This paper presents the Doppler Scanning technique which enables us to detect the relative positions of moving distributed sources using Doppler frequency shift estimate when the moving source consists of distributed sources with different signature frequencies. Doppler frequency shifts of characteristic frequencies of machinery noise sources such as ship's generator and propeller, with tine along CPA (Closest Point of Approach of moving source) are unique, and can be functioned with respect to each source position. Therefore, this technique can be applied to estimate the relative geometrical positions between machinery noise sources. The Extended Kalman Filter (EKF) which has a high frequency resolution with high time resolution, is adopted for improving accuracy of Doppler frequency shift estimate geometric resolution of machinery positions since machinery noise sources show in general low frequency band characteristics with limited spacial distance. The performance of the technique is examined by the numerical simulations and is verified by the experiment using loudspeaker sources on the roof of the car.
Speaker Adaptation for Voice Dialing
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 455~461
This paper presents a method that improves the performance of the personal voice dialling system in which speaker independent phoneme HMM's are used. Since the speaker independent phoneme HMM based voice dialing system uses only the phone transcription of the input sentence, the storage space could be reduced greatly. However, the performance of the system is worse than that of the system which uses the speaker dependent models due to the phone recognition errors generated when the speaker independent models are used. In order to solve this problem, a new method that jointly estimates transformation vectors for the speaker adaptation and transcriptions from training utterances is presented. The biases and transcriptions are estimated iteratively from the training data of each user with maximum likelihood approach to the stochastic matching using speaker-independent phone models. Experimental result shows that the proposed method is superior to the conventional method which used transcriptions only.
Wideband Speech Coding Algorithm with Application of Wavelet Transform
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 462~470
Wideband speech, characterized by a bandwidth of 50∼7000 ㎐, sounds more natural and intelligible, and is less tiring to listen to when compared to narrowband speech characterized by a bandwidth of 300∼3400 ㎐. Wideband speech coders, however, have not been as successful as the narrowband speech coders because of their higher bit rate. In this paper, we propose a new wideband speech coder which combines the European standard of a narrowband speech coder, i.e., GSM-EFR, and a transform coder using the discrete wavelet transform. The proposed wideband speech coder operates as follows input speech is first split into two subbands with equal bandwidth and the two subband signals are coded and decoded by each subband coder. A GSM-EFR is adopted as a lower subband coder and a subband coder with wavelet transformed speech is designed for a upper subband coder. The total bit rate of the proposed coder is 18.9kbps (12.2 kbps for lower band coder and 6.7 kbps for upper band coder), and informal listening test results have shown that the proposed coder has comparable speech quality to that of G.722 with 56 kbps.
A Study on Analysis of Variant Factors of Recognition Performance for Lip-reading at Dynamic Environment
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 471~477
Recently, lip-reading has been studied actively as an auxiliary method of automatic speech recognition(ASR) in noisy environments. However, almost of research results were obtained based on the database constructed in indoor condition. So, we dont know how developed lip-reading algorithms are robust to dynamic variation of image. Currently we have developed a lip-reading system based on image-transform based algorithm. This system recognize 22 words and this word recognizer achieves word recognition of up to 53.54%. In this paper we present how stable the lip-reading system is in environmental variance and what the main variant factors are about dropping off in word-recognition performance. For studying lip-reading robustness we consider spatial valiance (translation, rotation, scaling) and illumination variance. Two kinds of test data are used. One Is the simulated lip image database and the other is real dynamic database captured in car environment. As a result of our experiment, we show that the spatial variance is one of degradations factors of lip reading performance. But the most important factor of degradation is not the spatial variance. The illumination variances make severe reduction of recognition rates as much as 70%. In conclusion, robust lip reading algorithms against illumination variances should be developed for using lip reading as a complementary method of ASR.
A Beamforming Method for a Perturbed Linear Towed Array
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 478~484
Linear towed arrays (LTA) have a nonlinear shape due to tow vessel motion, ocean swells and currents. By reasons of nominally linear shape, various towed array shape estimation techniques have been developed since the perturbed shape cause the error in target detection. In this paper,, we propose the beamforming method for the perturbed LTA with simple structure. The proposed method linearizes a nonlinear phase of steering vector with position information measured by two reference sensors. It can be proved using some properties of Markov transition matrix, and iteration number of linearization process is decided by variance of cross phase difference. As a result of computer simulation in the ocean environment, beampattern of the proposed method is almost same with the ideal case in my type of array shape. In the signal-to-noise ratio (SNR) performance simlation, the DOA estimation performance of the proposed beamforming method is evaluated, and the comparison with Bartlett beamformer of the LTA shows that the proposed method can estimate. the spatial characteristic of sources more accuracy.
Broadband Interference Patterns in Shallow Water with Constant Bottom Slope
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 485~493
Broadband interference patterns are studied using ship as an acoustic source in shallow waters with varying bathymetry. Waveguide invariant index (β) indicating the pattern of constructive (or destructive) interference in range-frequency domain is derived in a waveguide with constant bottom slope based on adiabatic mode theory. Using this invariant, changes of the interference patterns resulting from the variation of bottom bathymetry are analyzed. Results of the analytic interpretation is compared with those from sea experiments and numerical simulations.
Reverberation Characterization and Suppression by Means of Low Rank Approximation
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 494~502
In this paper, the Low Rank Approximation (LRA) method to suppress the interference of signals from temporal fluctuations is applied. The reverberation signals and temporally fluctuating signals are separated from the measured data using the Ink. The Singular value decomposition (SVD) method is applied to extract the low rank and the temporally stable reverberation was extracted using the LRA. The reverberation suppression is performed on the LRA residual value obtained by removing the approximate reverberation signals. In overall, the method can be applied to the suppression of reververation in active sonar system as well as to the modeling of reverberation.
Acoustic Characteristics of Mufflers with an Extended Inlet and Outlet
The Journal of the Acoustical Society of Korea, volume 21, issue 5, 2002, Pages 503~509
Cylindrical chamber silencers with an extended inlet and outlet are extensively used in many application fields to reduce the propagated noise in ducts. The basic attenuation effectiveness in the low frequency region can be explained by the reactive wave action inside the expansion chamber associated with the geometric configurations of the inlet and outlet locations, and the area expansion of the jacket. In this study. an acoustic analysis is carried out for a concentric extended pipe inserted into a simple expansion chamber. An algebraic equation is derived by using the eigenfunction expansion and orthogonality principle in which the acoustic pressures and particle velocities defined on each subdivided surface are expressed by the separable coordinates. By using the proposed analytical method, transmission losses are predicted for several configurations of the concentric extended systems and they agree very well with experimental results.