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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 20, Issue 8 - Nov 2001
Volume 20, Issue 7 - Oct 2001
Volume 20, Issue 6 - Aug 2001
Volume 20, Issue 5 - Jul 2001
Volume 20, Issue 4 - May 2001
Volume 20, Issue 3 - Apr 2001
Volume 20, Issue 2 - Feb 2001
Volume 20, Issue 1 - Jan 2001
Volume 20, Issue 4E - 00 2001
Volume 20, Issue 3E - 00 2001
Volume 20, Issue 2E - 00 2001
Volume 20, Issue 1E - 00 2001
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Standardization of the Peelee
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 3~8
The pitch of Korean musical scales and intervals were calculated with the method of a One Third More and Less, that described at the AcHacKueBum. In this study, the standardized Peelee have been made and its sound frequencies were measured. The measured sound frequencies uniformly played agree well with the Korean musical scales.
Inverse Reconstruction of Sectional Area in Nonuniform Ducts by Using the Acoustical Measurement
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 9~16
This paper deals with the inverse reconstruction of sectional area in nonuniform ducts by using the acoustical measurement. There have been many theoretical and experimental studies on the duct area reconstruction. In this research, the method using the impulse response function and area reconstruction algorithm was employed because of its mathematical and experimental simplicity. Based on the study results on the drawback of conventional impulse excitation method, a new measurement method is proposed, that uses the random noise source and the discrete inverse Fourier transform. It is found that the reconstruction errors of the present method is smaller than the conventional method. A random error analysis is performed in order to investigate the causes of reconstruction error and to clarify the applicable data range for area reconstruction.
A Reverberation Cancellation Method Using the Escalator Algorithm in Active Sonar
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 17~25
Traditional adaptive noise cancelling methods rely their performance on various interfering parameters, such as convergence speed, tracking ability, numerical stability, relative frequency characteristics between target and reverberation signals, and activity of the target. In this paper, an adaptive noise cancelling method is suggested, which Provides a successful tradeoff mon these factors. It is designed to work on the transform domain, adopts the Gram-Schmidt orthogonalization process, and is implemented by the escalator algorithm. The transform domain approach supports a tradeoff between the convergence speed and numerical cost. The proposed method is verified by applying a real-data collected in the shallow waters off the east coasts of korea. It is shown that it has a good reverberation-rejection capability even for the target signal with adjacent frequency components to those of the reverberation, and its performance is invariant for the activity of the target.
Implementation of a Robust Speaker Recognition System in Noisy Environment Using AR HMM with Duration-term
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 26~33
Though speaker recognition based on conventional AR HMM shows good performance, its lack of modeling the environmental noise makes its performance degraded in case of practical noisy environment. In this paper, a robust speaker recognition system based on AR HMM is proposed, where noise is considered in the observation signal model for practical noisy environment and duration-term is considered to increase performance. Experimental results, using the digits database from 100 speakers (77 males and 23 females) under white noise and car noise, show improved performance.
Real-time Implementation of the AMR Speech Coder Using
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 34~39
An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.
Implementation of a Multimodal Controller Combining Speech and Lip Information
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 40~45
In this paper, we implemented a multimodal system combining speech and lip information, and evaluated its performance. We designed speech recognizer using speech information and lip recognizer using image information. Both recognizers were based on HMM recognition engine. As a combining method we adopted the late integration method in which weighting ratio for speech and lip is 8:2. By the way, Our constructed multi-modal recognition system was ported on DARC system. That is, our system was used to control Comdio of DARC. The interrace between DARC and our system was done with TCP/IP socked. The experimental results of controlling Comdio showed that lip recognition can be used for an auxiliary means of speech recognizer by improving the rate of the recognition. Also, we expect that multi-model system will be successfully applied to o traffic information system and CNS (Car Navigation System).
Recognition Time Reduction Technique for the Time-synchronous Viterbi Beam Search
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 46~50
This paper proposes a new recognition time reduction algorithm Score-Cache technique, which is applicable to the HMM-base speech recognition system. Score-Cache is a very unique technique that has no other performance degradation and still reduces a lot of search time. Other search reduction techniques have trade-offs with the recognition rate. This technique can be applied to the continuous speech recognition system as well as the isolated word speech recognition system. W9 can get high degree of recognition time reduction by only replacing the score calculating function, not changing my architecture of the system. This technique also can be used with other recognition time reduction algorithms which give more time reduction. We could get 54% of time reduction at best.
A Study on Combining Bimodal Sensors for Robust Speech Recognition
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 51~56
Recent researches have been focusing on jointly using lip motions and speech for reliable speech recognitions in noisy environments. To this end, this paper proposes the method of combining the visual speech recognizer and the conventional speech recognizer with each output properly weighted. In particular, we propose the method of autonomously determining the weights, depending on the amounts of noise in the speech. The correlations between adjacent speech samples and the residual errors of the LPC analysis are used for this determination. Simulation results show that the speech recognizer combined in this way provides the recognition performance of 83 ％ even in severely noisy environments.
Design and Construction of the Acoustic Horn for Magnetostrictive Ultrasonic Transducer
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 57~65
In this paper, we designed the acoustic horn for magnetostrictive ultrasonic transducers in a theoretical manner, and validity of the analysis was verified through comparison with the results of finite element analysis. Results of the two analysis methods showed good agreement with each other. The theoretical method can fairly quickly determine the horn length that satisfies given frequency specification, but also has the drawback that it is applicable only to the frequency range over the cut-off frequency. According to the results, the catenoidal horn can provide larger amplification than the exponential horn. It was also found that it is more desirable for the region having the catenoidal curvature to be as short as possible to achieve larger amplification of the transducer deformation. Based on the analysis results, a magneto-strictive transducer sample was fabricated and its performance was evaluated experimentally. The transducer has the resonance frequency of 19.3 ㎑ as well as the maximum SPL of 199 dB, and shows the omni-directional radiation pattern.
Development of Ultrasonic Magnetostrictive Sensors System to Measure in Very High Temperatures
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 66~74
The temperature measurement of yen high temperature core melt is of importance in LAVA (Lower-plenum Arrested Vessel Attack) experiment in which gap formation between core melt and the reactor lower head, and the effect of the gap on thermal behavior are to be measured. The delay time of ultrasonic wavelets due to high temperature is suggested. As a first stage, a molten material temperature was measured up to 2300℃. Also, the optimization design of the ultrasonic temperature sensor with persistence at the high temperature was suggested in this paper. And the utilization of the theory suggested in the reference〔1〕and the efficiency of the developed system are certified by performing experiments. This sensor welded magnetostrictive element and tungsten element will be able to measure a temperature range of 3000℃ hereafter.
Error Analysis of the Passive Localization Using Near-field Effect in the Sea
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 75~81
In this paper we analyzed the localization error of near-field detection algorithm in the sea. The near-field detection algorithms using triangulation and wavefront curvature basically assume a signal in two dimension of bearing and range. But the assumption causes localization error because there is three dimension of bearing, range, and depth in the sea. Even through three dimensional effect is considered, the localization error is occurred if multipath propagation in the sea is ignored. To analyze the localization error in the sea, we simulate the near-field localization using acoustic propagation model and focused beamforming considering wavefront curvature. The simulation results indicate that localization error always occurs in the sea and the error varied with sound velocity profile, water depth, bottom slope, source range, etc.
Synthetic Aperture Processing in Beamspace Using Twin-line Array
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 82~86
In this Paper, we Propose synthetic aperture technique for twin-line may. Sin91e-line way is required long aperture size in order to achieve high SNR and angular resolution in shallow water Ultra low frequency signal from far-field has left-right ambiguity at sing1e-line array. To resolve these Problems, we'd like to adopt the synthetic aperture technique to twin-line array. The synthetic aperture method adopts coherent processing of sub-aperture signals at successive tine intervals in the beam domain. The proposed method shows low nile error and improved angular resolution. In simulation result, average sidelobe level is reduced about 7〔dB〕when the array Peformed 5-synthesis.
Performance Analysis of a Receiver for WCDMA Systems
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 87~93
As a new type of a linear decorrelating receiver, the Pseudo-Decorrelator was presented for asynchronous code division multiple access systems by the author. In this paper, the concept of the Pseudo-Decorrelator is extended to derive a receiver for WCDMA uplink systems over an additive white Gaussian noise channel. Starting with the analysis of the multiple access components of the decision statistics, a non-square cross-correlation matrix for each bit is obtained. This cross-correlation matrix is then inverted, and the inverted matrix is applied to the decision statistics obtained from a conventional receiver. In this receiver, the detection process can be started after the first three consecutive bits are received. Simulation results are presented for K-user systems over an additive white Gaussian noise channel under the circumstances in which synchronization errors, including time delay errors and carrier phase errors exist. It is shown that the proposed receiver performs better than a conventional receiver and parallel interference canceller.
Numerical Simulation of Head Related Transfer Functions and Sound Fields
;V. Kahana;P. A. Nelson;M. Petyt;
The Journal of the Acoustical Society of Korea, volume 20, issue 6, 2001, Pages 94~103
The goal of using numerical methods in this study is two-fold: to replicate a set of measured, individualized HRTFs by a computer simulation, and also to visualise the resultant sound field around the head. Two methods can be wed: the Boundary Element Method (BEM) and the Infinite-Finite Element Method (IFEM). This paper presents the results of a preliminary study carried out on a KEMAR dummy-head, the geometry of which was captured with a high accuracy 3-D laser scanner and digitiser. The scanned computer model was converted to a few valid BEM and IFEM meshes with different polygon resolutions, enabling us to optimise the simulation for different frequency ranges. The results show a good agreement between simulations and measurements of the sound pressure at the blocked ear-canal of the dummy-head. The principle of reciprocity provides an effect method to simulate HRTF database. The BEM was also used to investigate the total sound field around the head, providing a tool to visualise the sound field for different arrangements of virtual acoustic imaging systems.