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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
Editor in Chief :
Volume & Issues
Volume 13, Issue 6 - Dec 1994
Volume 13, Issue 5 - Oct 1994
Volume 13, Issue 4 - Aug 1994
Volume 13, Issue 3 - Jun 1994
Volume 13, Issue 2 - Apr 1994
Volume 13, Issue 1 - Feb 1994
Volume 13, Issue 2E - 00 1994
Volume 13, Issue 1E - 00 1994
Selecting the target year
Characteristics of Acoustic Impulse Response of Submerged Cylindrical Objects as Elements of Target-Scattered Echo
Kim, Jae-Soo ; Seong, Nak-Jin ; Lee, Sang-Young ; Kim, Kang ; Yu, Myong-Jong ; Cho, Woon-Hyun ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 5~5
Simulation of the target-scattered echo requires the understanding of scattering mechanism at the highlight points. In this paper, the basic assumption of Highlight Model is reviewed through the analyzed data obtained in the acoustic water tank experiment. The analysis shows that the scattering mechanism involves pulse elongation and frequency shift as elements of target-scattered echo, and that the internal structures affect the temporal response of the target-scattered echo significantly. The band-limited impulse response or Green's function due to the diffraction from highlight points of internal structures is not mere delta function, but acts like a filter, which causes frequency shift and is elongated in time.
The Output SINR of the Linearly Constrained Broadband Beamformer
Gwak, Byeong-Jae ; Kim, Gi-Man ; Cha, Il-Hwan ; Yun, Dae-Hui ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 14~14
In this paper, we derive expressions for the output signal-to-interference plus noise ratio(SINR) of the linearly constrained broadband beamformer in noncoherent situations using a vector approach. The incoming broadband signals are assumed to have flat spectra.
A Fast Pitch Searching Algorithm Using Correlation Characteristics in CELP Vocoder
Lee, Joo-Hun ; Bae, Myung-Jin ; Ann, Sou-Guil ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 20~20
The major drawback to the Code Excited Linear Prediction(CELP) type vocoders is their large computational requirements. In this paper, a simple method is proposed to reduce the pitch searching time in the pitch filter almost without degradation of quality. Bease upon the observational regularity of the correlation function of speech, the searching range can be restricted to the positive side in pitch search. This is done by skipping the negative side with the width which is estimated from the previous positive envelope. In addition to that, the maximum number of available lags can be limited by the threshold,
, which is set on 58 empirically. So, only the limited numbers of lags are considered in pitch search, which is less than a half of that of the full search method. By using the proposed method in pitch search, its required computations are greatly reduced. Experimental result shows 51% time reduction almost without lowering the speech quality in segmental SNR measure.
Feature Extraction from the Strange Attractor for Speaker Recognition
Kim, Tae-Sik ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 26~26
A new feature extraction technique utilizing strange attractor and artificial neural network for speaker recognition is presented. Since many signals change their characteristics over long periods of time, simple time-domain processing techniques should e capable of providing useful information of signal features. In many cases, normal time series can be viewed as a dynamical system with a low-dimensional attractor that can be reconstructed from the time series using time delay. The reconstruction of strange attractor is described. In the technique, the raw signal will be reproduced into a geometric three dimensional attractor. Classification decision for speaker recognition is based upon the processing or sets of feature vectors that are derived from the attractor. Three different methods for feature extraction will be discussed. The methods include box-counting dimension, natural measure with regular hexahedron and plank-type box. An artificial neural network is designed for training the feature data generated by the method. The recognition rates are about 82%-96% depending on the extraction method.
Design of The Loudness Ratings And Talker Echo For ISDN Telephone
Hong, Jin-Woo ; Kang, Kyeong-Ok ; Kang, Seong-Hoon ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 32~32
It is the purpose of this paper to describe the methods for establishing loudness ratings and talker echo out of transmission quality of ISDN telephone connected to fully digital network. In order to design the desirable loudness ratings and talker echo for ISDN telephone, the model system of digital speech communication for subjective tests is developed. Using this model system, opinion tests which decide the optimal CODEC input level, the range of overall loudness rating, sidetone masking rating and talker echo are performed. From the results of tests, we decided that the loudness ratings are 6 to 8dB for sending, 0 to 2dB for receiving, and 8 to 12dB for sidetone masking rating. And, the terminal coupling loss of TCLw of at least 40dB is necessary to provide echo-free telephone communications to telophone users when the overall loudness rating of ISDN telephone is normalized to 10dB.
A Study on Speech Recognition using DMS Model
An, Tae-Ock ; Byun, Yong-Kyu ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 41~41
This paper proposes a DMS(Dynamic Multi-Section) model based on the information of the similar features in word pattern. This model represents each word as a time series of several sections and each section implies duration time information and typical feature vectors.The procedure to make a model in the word pattern is that typical feature vector and duration time information are reflected in the distance, when matching between word pattern and model is repeated. As the result of it, the accumulated distance by matching is to be minimized.
On a Reduction of Pitch Searching Time by Preliminary Pitch in the CELP Vocoder
Kim, Dae-Sik ; Bae, Myung-Jin ; Kim, Jong-Jae ; Byun, Kyung-Jin ; Han, Ki-Chun ; Yoo, Hah-Young ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 51~51
Code Excited Linear Prediction(CELP) vocoder exhibits good performance at data rates below 4.8 kbps. The major drawback to CELP type coders is their large amount of computation. In this paper, we propose a new pitch search method that preserves the quality of the CELP vocoder with reduced complexity. The basic idea is to restrict the pitch searching range by estimating the preliminary pitches. Applying the proposed method to the CELP vocoder, we can get approximately 87% complexity reduction in the pitch search.
A Real-Time Implementation of Isolated Word Recognition System Based on a Hardware-Efficient Viterbi Scorer
Cho, Yun-Seok ; Kim, Jin-Yul ; Oh, Kwang-Sok ; Lee, Hwang-Soo ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 58~58
Hidden Markov Model (HMM)-based algorithms have been used successfully in many speech recognition systems, especially large vocabulary systems. Although general purpose processors can be employed for the system, they inevitably suffer from the computational complexity and enormous data. Therefore, it is essential for real-time speech recognition to develop specialized hardware to accelerate the recognition steps.This paper concerns with a real-time implementation of an isolated word recognition system based on HMM. The speech recognition system consists of a host computer (PC), a DSP board, and a prototype Viterbi scoring board.The DSP board extracts feature vectors of speech signal. The Viterbi scoring board has been implemented using three field-programmable gate array chips. It employs a hardware-efficient Viterbi scoring architecture and performs the Viterbi algorithm for HMM-based speech recognition. At the clock rate of 10 MHz, the system can update about 100,000 states within a single frame of 10ms.
Paper Title : Speech Parameter Estimation and Enhancement Using the EM Algorithm
Lee, Ki-Yong ; Kang, Young-Tae ; Lee, Byung-Gook ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 68~68
In many applications of signal processing, we have to deal with densities which are highly non-Gaussian or which may have Gaussian shape in the middle but have potent deviations in the tails. To fight against these deviations, we consider a finite mixture distribution for the speech excitation. We utilize the EM algorithm for the estimation of speech parameters and their enhancement. Robust Kalman filtering is used in the enhancement process, and a detection/estimation technique is used for parameter estimation. Experimental results show that the proposed algorithm performs better in adverse SNR input conditions.
Effects of Depth-varying Compressional Wave Attenuation on Sound Propagation on a Sandy Bottom in Shallow Water
Na, Young-Nam ; Shim, Tae-Bo ; Jurng, Moon-Sub ; Choi, Jin-Hyuk ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 76~76
The characteristics of bottom sediment may be able to vary within a few meters of depth in shallow water. Since bottom attenuation coefficient as well as sound velocity in the bottom layer is determined by the composition and characteristics of sediment itself, it is reasonable to assume that the bottom attenuation coefficient is accordingly variable with depth. In this study, we use a parabolic equation scheme to examine the effects of depth-varying compressional wave attenuation on acoustic wave propagation in the low frequency ranging from 100 to 805 Hz. The sea floor under consideration is sandy bottom where the water and the sediment depths are 40 meters and 10 meters, respectively. Depending on the assumption that attenuation coefficient is constant or depth-varying, the propagation loss difference is as large as 10dB within 15 km. The predicted propagation loss is very much comparable to the measured one when we employ a depth-varying attenuation coefficient.
Theory of Acoustic Propagation in 3 Dimensional Wedge Domain
Seong, Woo-Jae ;
The Journal of the Acoustical Society of Korea, volume 13, issue 2E, 1994, Pages 83~83
Three components contribute to the acoustic field propagating in a wedge or over a ridge : a direct path arrival, an image component due to reflection from the boundaries and a component diffracted by the apex. All three contributions are included in a new, exact solution of the Helmholtz equation for the three-dimensional time harmonic field from a point source in a wedge(or over a ridge) formed by two intersecting, pressure-release plane boundaries. The solution is obtained by applying three integral transforms, and consists of and infinite sum of uncoupled normal nodes. The mode coefficients are given by a finite integral involving a Gegenbauer polynomial in the integrand, which may be computed relatively efficiently. Results of the theory for propagation over a 90 degree ridge is discussed.