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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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An Implementation of Inverse Filter Using SVD for Multi-channel Sound Reproduction
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 3~11
This paper describes an implementation of inverse filter using SVD in order to recover the input in multi-channel system. The matrix formulation in SISO system is extended to MIMO system. In time and frequency domain we investigates the inversion of minimum phase system and non-minimum phase system. To execute an effective inversion of non-minimum phase system, SVD is introduced. First of all we computes singular values of system matrix and then investigates the phase property of system. In case of overall system is non-minimum phase, system matrix has one (or more) very small singular value (s). The very small singular value (s) carries information about phase properties of system. Using this property, approximate inverse filter of overall system is founded. The numerical simulation shows potentials in use of the inverse filter.
A Study on the Recognition of Korean Numerals Using Recurrent Neural Predictive HMM
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 12~18
In this paper, we propose the Recurrent Neural Predictive HMM (RNPHMM). The RNPHMM is the hybrid network of the recurrent neural network and HMM. The predictive recurrent neural network trained to predict the future vector based on several last feature vectors, and defined every state of HMM. This method uses the prediction value from the predictive recurrent neural network, which is dynamically changing due to the effects of the previous feature vectors instead of the stable average vectors. The models of the RNPHMM are Elman network prediction HMM and Jordan network prediction HMM. In the experiment, we compared the recognition abilities of the RNPHMM as we increased the state number, prediction order, and number of hidden nodes for the isolated digits. As a result of the experiments, Elman network prediction HMM and Jordan network prediction HMM have good recognition ability as 98.5% for test data, respectively.
Real-time Implementation of Multi-channel AMR Speech Coder
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 19~23
DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.
Automatic Phonetic Segmentation of Korean Speech Signal Using Phonetic-acoustic Transition Information
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 24~30
This article is concerned with automatic segmentation for Korean speech signals. All kinds of transition cases of phonetic units are classified into 3 types and different strategies for each type are applied. The type 1 is the discrimination of silence, voiced-speech and unvoiced-speech. The histogram analysis of each indicators which consists of wavelet coefficients and SVF (Spectral Variation Function) in wavelet coefficients are used for type 1 segmentation. The type 2 is the discrimination of adjacent vowels. The vowel transition cases can be characterized by spectrogram. Given phonetic transcription and transition pattern spectrogram, the speech signal, having consecutive vowels, are automatically segmented by the template matching. The type 3 is the discrimination of vowel and voiced-consonants. The smoothed short-time RMS energy of Wavelet low pass component and SVF in cepstral coefficients are adopted for type 3 segmentation. The experiment is performed for 342 words utterance set. The speech data are gathered from 6 speakers. The result shows the validity of the method.
A Study on the Small Size Loudspeaker for Hi-Fi Low Frequency Sound Reproduction
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 31~37
Following the recent trends of reducing the size of multimedia devices, we tried for the development of a compact-sized speaker to produce low-frequency sounds efficiently. For this work, equivalent-circuit analysis was used to get fundamental resonant frequency and then the structure of speaker components has been changed appropriately. As a result, an 80mm small-sized speaker was developed. The performance test showed that the resonant frequency of our system is 79 Hz while that of numerical analysis was 81Hz. At a distance of 1m from our speaker, the frequency ranges 80 Hz to 15kHz and the average sound pressure was found to be 84±2 dB. The second (at 400 Hz) and the third (at 100 Hz) high-frequency distortions of our system were 0.5% and 1.8% respectively, which is to be compared with the distortions of 0.9% and 6% in conventional speakers.
A Study on Standing Wave Type Ultrasonic Linear Motors
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 38~43
We developed a new standing wave type ultrasonic linear motor that can be driven bi-directionally. The operation principle of the motor was derived in an analytical form and the detailed structure was designed by the finite element method. Based on the design, a motor sample and a driving circuit were fabricated, and validity of the structure was verified through experiments.
Performance Analysis of FFTSA Method in the Water Environment Using Conformal Towed Acoustic Array
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 44~57
This paper analyses the performance of FFTSA (Fast Fourier Transform Synthetic Aperture) in the effects of temporal coherence and oscillatory towed course, which is one of the techniques for passive synthetic aperture SONAR process using linearly distributed towed array. Also this paper proposes the FFTSA technique using towed array having conformal shape to alleviate the performance degradation for estimating the incident angle under inconsistent under water environments. And this paper analyses the performance of the proposed FFTSA technique making use of conformal structure throughout exhaustive computer simulations.
A Computationally Efficient Time Delay and Doppler Estimation for the LFM Signal
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 58~66
In this paper, a computationally efficient time delay and doppler estimation algorithm is proposed for active sonar with Linear Frequency Modulated (LFM) signal. To reduce the computational burden of the conventional estimation algorithm, an algebraic equation is used which represents the relationship between the time delay and doppler in cross-ambiguity function of the LFM signal. The algebraic equation is derived based on the Fast maximum Likelihood (FML) method. Using this algebraic relation, the time delay and doppler are estimated with two 1-D search instead of the conventional 2-D search. The estimation errors of the proposed algorithm are analyzed for various SNR's. The simulation result demonstrates the good performance of the proposed algorithm.
Underwater Moving Source Tracking Using a Coherent Broad-band Matched Field Processing Technology
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 67~73
The shallow-water environment presents additional challenges arising from the complex interaction patterns of the sound with the sea bed. In order to overcome the difficulties generated by shallow-water propagation, broad-band matched field processing has been employed in an effort to increase robustness by utilizing multiple frequency information. In this paper, a coherent broad-band matched field processor is introduced that incorporates the spatial coherence of the acoustic field not only over one frequency but across frequencies. The incoherent and coherent processors are applied to the experimental data where it is shown that both processors give a high probability of correct localization. Also it is found that a coherent processor has better performance in the sidelobe pattern of ambiguity surfaces.
Iterative Polynomial Fitting Technique for the Nonlinear Array Shape Estimation
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 74~80
Because of ocean waves, swell, steering corrections, etc, the hydrophones of a towed array will not live along a straight line. However the degradation of bearing estimation performance occurs when beamforming is carried out on the hydrophone outputs of an acoustic towed array which is not straight. So it is required to estimate the shape of the array for the improved beamformer output. In this paper, an iterative array shape estimation technique is presented, which is based on the use of the least squares polynomial fitting to the data from heading sensors. The estimation error and the influence of deformations on the performance of the conventional beamformer output are investigated. Finally, the suggested method is applied to the real system in order to investigate the applicability.
Modeling of the Time-frequency Auditory Perception Characteristics Using Continuous Wavelet Transform
The Journal of the Acoustical Society of Korea, volume 20, issue 8, 2001, Pages 81~87
The human auditory system is appropriate for the "constant Q"system. The STFT (Short Time Fourier Transform) is not suitable for the auditory perception model since it has constant bandwidth. In this paper, the CWT (continuous wavelet transform) is employed for the auditory filter model. In the CWT, the frequency resolution can be adjusted for auditory sensation models. The proposed CWT is applied to the modeling of the JNVF. In addition, other signal processing methods such as STFT, VER-FFT and VFR-STFT are discussed. Among these methods, the model of JNVF (Just Noticeable Variation in Frequency) by using the CWT fits in with the JNVF of auditory model although it requires quite a long time.