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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 19, Issue 8 - Nov 2000
Volume 19, Issue 7 - Oct 2000
Volume 19, Issue 6 - Aug 2000
Volume 19, Issue 5 - Jul 2000
Volume 19, Issue 4 - May 2000
Volume 19, Issue 3 - Apr 2000
Volume 19, Issue 2 - Feb 2000
Volume 19, Issue 1 - Jan 2000
Volume 19, Issue 3E - 00 2000
Volume 19, Issue 2E - 00 2000
Volume 19, Issue 1E - 00 2000
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A Study for Beamforming Acoustic Holographic Method Using Linear Arrayed Microphones
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 3~10
This paper proposes acoustic holographic measuring system to estimate an absolute position of sound source. Using the measured signals, the estimation of the position is calculated by the Cross-spectrum algorithm of the beamformed signal and a linear arrayed microphone's signals. As the results of comparing the reference microphone method with beamforming method through the measurement of sound field, the beamforming acoustic holographic method is progressed above 20 percent than that of a reference microphone method in the resolution, and the utility of the proposed system could be confirmed.
The Design of Chorus DSP Chip Using Psychoacoustic Model and SOLA Algorithm
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 11~19
This research deals with the implementation procedures of a chorus processing DSP for karaoke system. It is necessary to compress the chorus data to store as many choruses as we can. We apply MPEG-1 audio algorithm to compress the chorus data. And the chorus system must be accompanied with the karaoke that can change the key and the tempo. So the chorus DSP must be able to change the key and tempo of the chorus data. We apply SOLA (Synchronized Overlap and Add) to do it. We designed the chorus DSP that can compress the chorus, change the key and tempo. And we verified the chorus DSP logic using FPGA. The used FPGA are two FLEX10K100s made by ALTERA. Finally we make the ASIC chip of chorus DSP and verify its operation.
Surface Classification and Its Threshold Value Selection for the Recognition of 3-D Objects
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 20~25
This paper proposes the method of surface classification and threshold value selection for surface classification of the three-dimensional object recognition. The processings of three-dimensional image processing system consist of three steps, i.e, acquisition of range data, feature extraction and matching process. This paper proposes the method of shape feature extraction from the acquired rage data in the entire three-dimensional image processing system. In order to achieve these goals, firstly, this article proposes the surface classification method by using the distribution characteristics of sign value from range values. Also pre-existing method which uses the K-curvature and K-curvature has limitation in the practical threshold value selection. To overcome this, this article proposes the selection of threshold value for surface classification. Finally, the effectiveness of this article is demonstrated by the several experiments.
Low Rate Speech Coding Using the Harmonic Coding Combined with CELP Coding
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 26~34
In this paper, we propose a 4kbps speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding in the voiced frame and uses the vector excitation coding with the structure of analysis-by-synthesis in the unvoiced frame, respectively. But two mode coding method is not effective for transition frame mixed in voiced and unvoiced signal and a new method beyond using unvoiced/voiced mode coding is needed. Thus, we designed a time-separated transition coding method for transition frame in which a voiced/unvoiced decision algorithm separates unvoiced and voiced duration in a frame, and harmonic-harmonic excitation coding and vector-harmonic excitation coding method is selectively used depending on the previous frame U/V decision. In the decoder, the voiced excitation signals are generated efficiently through the inverse FFT of harmonic magnitudes and the unvoiced excitation signals are made by the inverse vector quantization. The reconstructed speech signal are synthesized by the Overlap/Add method.
Continuous Speech Recognition based on Parmetric Trajectory Segmental HMM
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 35~44
In this paper, we propose a new trajectory model for characterizing segmental features and their interaction based upon a general framework of hidden Markov models. Each segment, a sequence of vectors, is represented by a trajectory of observed sequences. This trajectory is obtained by applying a new design matrix which includes transitional information on contiguous frames, and is characterized as a polynomial regression function. To apply the trajectory to the segmental HMM, the frame features are replaced with the trajectory of a given segment. We also propose the likelihood of a given segment and the estimation of trajectory parameters. The obervation probability of a given segment is represented as the relation between the segment likelihood and the estimation error of the trajectories. The estimation error of a trajectory is considered as the weight of the likelihood of a given segment in a state. This weight represents the probability of how well the corresponding trajectory characterize the segment. The proposed model can be regarded as a generalization of a conventional HMM and a parametric trajectory model. The experimental results are reported on the TIMIT corpus and performance is show to improve significantly over that of the conventional HMM.
A Blind Segmentation Algorithm for Speaker Verification System
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 45~50
This paper proposes a delta energy method based on Parameter Filtering(PF), which is a speech segmentation algorithm for text dependent speaker verification system over telephone line. Our parametric filter bank adopts a variable bandwidth along with a fixed center frequency. Comparing with other methods, the proposed method turns out very robust to channel noise and background noise. Using this method, we segment an utterance into consecutive subword units, and make models using each subword nit. In terms of EER, the speaker verification system based on whole word model represents 6.1%, whereas the speaker verification system based on subword model represents 4.0%, improving about 2% in EER.
A Study on the Automatic Speech Control System Using DMS model on Real-Time Windows Environment
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 51~56
Is this paper, we studied on the automatic speech control system in real-time windows environment using voice recognition. The applied reference pattern is the variable DMS model which is proposed to fasten execution speed and the one-stage DP algorithm using this model is used for recognition algorithm. The recognition vocabulary set is composed of control command words which are frequently used in windows environment. In this paper, an automatic speech period detection algorithm which is for on-line voice processing in windows environment is implemented. The variable DMS model which applies variable number of section in consideration of duration of the input signal is proposed. Sometimes, unnecessary recognition target word are generated. therefore model is reconstructed in on-line to handle this efficiently. The Perceptual Linear Predictive analysis method which generate feature vector from extracted feature of voice is applied. According to the experiment result, but recognition speech is fastened in the proposed model because of small loud of calculation. The multi-speaker-independent recognition rate and the multi-speaker-dependent recognition rate is 99.08% and 99.39% respectively. In the noisy environment the recognition rate is 96.25%.
Implementation of Real Time Multi-User Communication System with MPEG-4 CELP
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 57~62
In recent, the innovative improvement of a internet and computing environment make users desire the capability of processing information in real time. In this paper we implement a PC-to-PC real time multi-user communication system on the internet environment using the efficient algorithm for a real time processing and the MFEG-4 CELP codec which can be used for a low bit-rate coding from 6 to 24kbps. The implemented system produces a compressed bit-streams with the MPEG-4 CEU Mode-I 18200bps mode. There is 5 frames for a package and 1 frame has 160 samples. We can use this system to communicate with 4 users simultaneously in real time. The system is designed and examined on the Windows operating system.
Design of Digital Peaking Filters Using Q-Compensation
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 63~71
A new type of second-order digital peaking filters for professional-quality digital audio system is proposed whose frequency response can be elaborately controlled throughout the composite structure of a standard band-pass filter and a 0-dB bypass gain. The proposed method for designing the peaking filter uses the Q-compensation technique to prevent the Q-distortion caused by the variation of the gain factor and is reduced into a compact form which is proper to the real-time implementation. Methods are examined for computing its coefficients, which are exact and very straightforward to compute with small amount of the system resources.
Performance Improvement of Towed Array Shape Estimation Using Interpolation
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 72~76
A calibration technique is proposed to improve the performance of 2-D towed array shape estimation using the Kalman filter. In the case of using displacement sensors, 2-D hydrophone positions estimated by the Kalman filter are calculated by assuming that the adjacent hydrophones are horizontally equi-spaced so that maximum distance is equal to the array length. The assumption causes errors in estimating hydrophone positions. The proposed technique using linear model approximation or spline interpolation can reduce the errors by exploiting the fact that the whole length of array is preserved whatever the array shape is. The numerical experiments show that the proposed method is very effective.
Acoustic Property of Sandy Sediment in the Korea Strait Using Sediment Sound Velocimeter
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 77~85
Laboratory determinations of acoustic and physical properties in Korea Strait sediment were carried out. Sediment sound velocimeter(SSV) was employed to measure the sound velocity of sandy sediment. Distribution patterns of the acoustic and physical properties are controlled by sediment texture. The study area is divided into three provinces(mid-shelf, shelf margin and enough) based on the acoustic and physical properties. This classification matches well with the previous result based on the systems tracks and depositional systems. We suggest a geoacoustic model of the Korea Strait that replacing the old model of Briggs and Fisher.
Modeling of Piano Sound Using Method of Line-Segment Approximation and Curve Fitting
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 86~91
In this paper, we will discuss the characteristics of the magnitude and the phase of the piano sound in frequency domain by using the FFT(Fast Fourier Transform). The method deciding the parameters representing those sounds through the mathematical model is described. We used the curve fitting method for the modeling of the harmonic part of the sound including the fundamental frequency in order to minimize the errors between original sounds and modeled sounds. furthermore, we used the line segment approximation method for the modeling of the noise part around fundamental frequency. We also applied the same method for the phase model and could get the modeled sound to be similar to the original sound using the parameters. Therefore the high compression ratio comparing the modeled sound to the original sound is achieved.
Sensitivity Analysis of a Mandrel Type Fiber Optic Acoustic Sensor Using an Analytical Method
The Journal of the Acoustical Society of Korea, volume 19, issue 3, 2000, Pages 92~99
In this paper, theoretical acoustic sensitivity was derived to describe acousto-optic transduction property of the mandrel type fiber optic acoustic sensor with respect to external acoustic field. The acoustic sensitivity was analyzed in relation to both material properties and geometrical influence factors of the constitutional parts of the sensor, analytically. Validity of the theoretical results were verified through comparison with the finite element analysis results. The variation trends of the sensitivity of the sensor in relation to the studied parameters showed good agreement for the two analysis methods. According to the results, it is considered more economical to design the basic structure of the sensor with the analytic equations developed in this paper, and then to carry out further detailed analysis with the finite element method for specific points of design interest.