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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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Calculation Model of Roughness for Searching Roughness-contributed Components
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 3~12
It is known that the roughness is one of the most important metrics in assessing the sound quality. In this study, a new roughness model is suggested by combing the previous auditory filter model and several signal processing methods for the enhancement of calculation efficiency and accuracy. For testing the usefulness of the present model, the predicted responses are compared with the experimental data and it is observed that they are in good agreements. Also, it is found that the previous models have limitations to search frequency components mainly contributed to overall roughness. By modifying the correlation criteria of the present model, the revised model for the proper estimation of roughness-contributed components is embedded.
Noise Reduction of Anti-phase Shifting to Maximum Amplitude Response in a Helmet
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 13~20
The active noise cancellation system offers a better low frequency performance with a smaller and lighter system compared to a passive one. This paper presents an active noise control system capable of reducing the noise in a helmet after attenuating the external noise using the helmet as the passive noise reduction system, which consists of a controller for inverting and compensating the phase delay, a microphone for picking up the external noise, and a loudspeaker for radiating the acoustic anti-phase signal to reduce the external noise. In this paper, external noise can be reduced by noise controller by compensating the phase difference to be 180°in the frequency of maximun value in the amplitude response. The noise of the phase delay covered from 50°to 310°was reduced in this system and it is possible to obtain a noise reduction of up to approximately 20 dB at the ears in the enclosure.
Development of Sound Radiation Analysis System Using the Results of Power Flow Finite Element Method
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 21~30
The analysis system implementing a serial process from structural vibration to sound radiation has been developed using both the power flow finite element method (PFFEM) known as a new vibrational analysis technique in medium to high frequency ranges and the acoustic boundary element method (BEM) which is effective in analyzing the sound radiation problems. The vibration analysis for arbitrary shape structures composed of plates is performed, and using the vibration energy density obtained from this analysis as the velocity boundary conditions for an acoustic analysis, vibro-acoustic analysis has been processed. To verify the developed system, we select a simple structure model and compare the results of developed system with those of SYSNOISE, and also the developed system is applied for the vibro-acoustic analysis of various structures in shapes.
A Study on the Linear Array Beamforming by Cross Correlation Matrix
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 31~36
Passive sonar system forms the various beams in any desired directions to obtain the improvement in Signal-to-Noise (S/N) ratio, bearing detection and localization of targets, and the attenuation of interferences from other directions. The improvement of beamforming is very important to detect modern underwater targets as noise reduction technology leads to considerably low-level acoustic emissions in the long range in complex environmental sea. In this paper, we proposed the spatial cross correlation beamforming (SCCBF) algorithm using cross correlation matrix of individual hydrophone pairs of linear array sensors. By the theoretical analysis and simulation, the proposed SCCBF is demonstrated that its performances compared to conventional beamforming (CBF) output can be obtain above 3dB of array gain and about half of beam width represented the bearing accuracy in target detection. Also, this paper presents sea test result of linear passive sonar system that the proposed algorithm implemented.
Design of the Noise Suppressor Using Wavelet Transform
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 37~46
This paper proposes a new noise suppression method using the Wavelet transform analysis. The noise suppressor using the Wavelet transform shows the more effective advantages in a babble noise than one using the short-time Fourier transform. We designed a new channel structure based on spectral subtraction of Wavelet transform coefficients and used the Wavelet mask pattern with more higher time resolution in high frequency. It showed a good adaptation capability for babble noise with a non-stationary property. To evaluate the performance of proposed noise canceller, the informal subjective listening tests (Mos tests) were performed in background noise environments (car noise, street noise, babble noise) of mobile communication. The proposed noise suppression algorithm showed about MOS 0.2 performance improvements than the suppression algorithm of EVRC in informal listening tests. The noise reduction by the proposed method was shown in spectrogram of speech signal.
Efficient Variable Dimension Quantization of Harmonic Magnitude
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 47~54
In this paper, we present a variable dimension vector quantization for spectral magnitudes. Espectially, spectral magnitudes of the Harmonic coder, need variable dimension quantizer because those are not fixed dimension. So, this paper present efficient quantization methods. These methods use variable Discrete Cosine Transform(DCT) for spectral magnitude parameters and NSTVQ which is combined odd/even, split and multi-stage structure, proposed quantization methods use Spectral Distortion(SD) for performance measure. Consequently, Multi-Stage Nonsquare Transform Vector Quantization(MSNSTVQ) is the best in performance measure.
An Adaptive Pruning Threshold Algorithm for the Korean Address Speech Recognition
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 55~62
In this paper, we propose a new adaptative pruning algorithm which effectively reduces the search space during the recognition process. As maximum probabilities between neighbor frames are highly interrelated, an efficient pruning threshold value can be obtained from the maximum probabilities of previous frames. The main idea is to update threshold at the present frame by a combination of previous maximum probability and hypotheses probabilities. As present threshold is obtained in on-going recognition process, the algorithm does not need any pre-experiments to find threshold values even when recognition tasks are changed. In addition, the adaptively selected threshold allows an improvement of recognition speed under different environments. The proposed algorithm has been applied to a Korean Address recognition system. Experimental results show that the proposed algorithm reduces the search space of average 14.4% and 9.14% respectively while preserving the recognition accuracy, compared to the previous method of using fixed pruning threshold values and variable pruning threshold values.
Real-time Implementation of the G.729 Annex A Using ARM9
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 63~68
This paper describes the details of ITU-T SGIS G.729A speech coder implementation using ARM9 Thumb/sup R/ processor core and various techniques used in the optimization process. ITU-T G.729 speech coder is the standard of the toll quality 8 kbit/s speech coding. The input to the speech encoder is assumed to be a 16 bits PCM signal at a sampling rate of 8000 samples per second. G.729A is reduced complexity version of the G.729 coder. This version is bit stream interoperable with the full version. The implemented coder requires 34.8 MIPS for the encoder and 8.1 MIPS for the decoder, 36.5 kBytes of program ROM and 6.3 kBytes of data RAM, respectively. The implemented coder is tested against the set of 9 test vectors provided by ITU-T for bit exact implementation.
Speaker Verification System Based on HMM Robust to Noise Environments
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 69~75
Intra-speaker variation, noise environments, and mismatch between training and test conditions are the major reasons for the speaker verification system unable to use it practically. In this study, we propose robust end-point detection algorithm, noise cancelling with the microphone property compensation technique, and inter-speaker discriminate technique by weighting cepstrum for robust speaker verification system. Simulation results show that the average speaker verification rate is improved in the rate of 17.65% with proposed end-point detection algorithm using LPC residue and is improved in the rate of 36.93% with proposed noise cancelling and microphone property compensation algorithm. The proposed weighting function for discriminating inter-speaker variations also improves the average speaker verification rate in the rate of 6.515%.
Selective Quantization Based on Band Property for Wideband Signal Codec
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 76~82
In this paper, a novel quantization method for wideband signal codec with 7 kHz bandwidth is proposed. In the transform-based wideband signal codecs, the signal is transformed to frequency domain and the spectral coefficients in each frequency band are quantized based on human perceptual model, followed by Huffman coding. However, the property of each band varies with frequency, and the codec has poor performance when all bands are quantized with the same method. Therefore, a selective quantization method is proposed, which analyzes the band property and selects the quantization domain between frequency domain and time domain based on the quantization efficiency. It is confirmed that the proposed method has better performance than the quantizer of G722.1 codec.
A Study on Robust Matched Field Processing Based on Feature Extraction
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 83~88
In this paper, matched field processing algorithm robust to environmental mismatches in an ocean waveguide based on feature extraction is summarized. However, in applying this processor to localize a source there are two preliminary issues to be resolved. One is the number of eigenvectors to be extracted and the other is the number of environmental samples to be used. To determine these issues, the relation between the number of dominant modes propagating in a given ocean waveguide and that of eigenvectors to be extracted is analyzed. Then, the analysis results are confirmed by the subspace analysis. This analysis quantifies the similarity between the subspace spanned by the signal vectors and that spanned by the eigenvectors to be extracted. The error index is defined as a relative difference between the location estimated by the current processor and the real source location. It is identified that in the case of extracting the largest eigenvectors equal to the number of dominant modes in a given environment, the processor localizes the source successfully. From the numerical simulations, it is shown that use of at least 30 environmental samples guarantee stable performance of the proposed processor.
A Study on a Multi-channel Fiber Optic Hydrophone System
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 89~93
In recent years Fiber optic hydrophone systems have been the focus of much attention in the sonar world. For sonar arrays, a fiber optic approach offers the major benefit of passive multiplexing of large numbers of hydrophones without underwater electronics. This paper describes recent development work covering array construction, opto-electronics development, hydrohpone design and sea trials. And the development of an interferometric mult-channel fiber optic hydrophone system which uses time division multiplexing capable of driving in excess of 32 channel is described. For this, a 12 channel time division multiplexing array has been constructed, and the performance of this system is demonstrated by sea trial.
Effect of Bias for Snapshots Using Minimum Variance Processor in MFP
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 94~100
When using a sample covariance matrix data in paucity of snapshots, adaptive matched field processing will have problem in inverting covariance matrix due to the rank deficiency. The general solutions are diagonal loading and eigenanalysis methods, but there is a significant bias in the power output. This paper presents a quantitative study of bias of power output and the performance of source localization through the simulation and the measured data analysis in fixed source case using the diagonal loading method for the minimum variance processor. Results show that the bias in power output is reduced and the performance of source localization is improved when the number of snapshots is greater than the number of array sensors.
Reserved Slot Allocation Scheme for Voice Service in WATM MAC
The Journal of the Acoustical Society of Korea, volume 20, issue 7, 2001, Pages 101~108
In this paper we focus on dynamic reservation slot allocation scheme for supporting QoS of a voice traffic in WATM MAC. Especially, voice traffic is the most important real-time object, and so we propose a new MAC protocol for voice traffic over WATM networks in the multimedia environment. According to the characteristics of voice traffic which is repeatedly in silent state and active state, new protocol allocates reservation slots dynamically with respect to the number of silent voice source of which starting time is stored to the state table in base station (BS). The simulation results show that the proposed protocol has better performance than slotted ALOHA in average access delay, collision rate, better than NC-PRMA(Non Collision Packet Reservation Multiple Access) in band width efficiency, and can provide a certain level of QoS requirement by the given slot assignment even though the number of voice terminals is increased.