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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
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 21, Issue 8 - Nov 2002
Volume 21, Issue 7 - Oct 2002
Volume 21, Issue 6 - Aug 2002
Volume 21, Issue 5 - Jul 2002
Volume 21, Issue 4 - May 2002
Volume 21, Issue 3 - Apr 2002
Volume 21, Issue 2 - Feb 2002
Volume 21, Issue 1 - Jan 2002
Volume 21, Issue 1E - 00 2002
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16kbps Windeband Sideband Speech Codec
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 5~10
This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.
Recognition for Noisy Speech by a Nonstationary AR HMM with Gain Adaptation Under Unknown Noise
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 11~18
In this paper, a gain-adapted speech recognition method in noise is developed in the time domain. Noise is assumed to be colored. To cope with the notable nonstationary nature of speech signals such as fricative, glides, liquids, and transition region between phones, the nonstationary autoregressive (NAR) hidden Markov model (HMM) is used. The nonstationary AR process is represented by using polynomial functions with a linear combination of M known basis functions. When only noisy signals are available, the estimation problem of noise inevitably arises. By using multiple Kalman filters, the estimation of noise model and gain contour of speech is performed. Noise estimation of the proposed method can eliminate noise from noisy speech to get an enhanced speech signal. Compared to the conventional ARHMM with noise estimation, our proposed NAR-HMM with noise estimation improves the recognition performance about 2-3%.
A Study on Spatio-temporal Features for Korean Vowel Lipreading
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 19~26
This paper defines the visual basic speech units, visemes and investigates various visual features of a lip for the effective Korean lipreading. First, we analyzed the visual characteristics of the Korean vowels from the database of the lip image sequences obtained from the multi-speakers, thereby giving a definition of seven Korean vowel visemes. Various spatio-temporal features of a lip are extracted from the feature points located on both inner and outer lip contours of image sequences and their classification performances are evaluated by using a hidden Markov model based classifier for effective lipreading. The experimental results for recognizing the Korean visemes have demonstrated that the feature victor containing the information of inner and outer lip contours can be effectively applied to lipreading and also the direction and magnitude of the movement of a lip feature point over time is quite useful for Korean lipreading.
An Extension of the VoiceXML Platform for Push-based Voice Applications
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 27~36
VoiceXML is a standard dialog mark-up language for the neat generation voice applications. The current VoiceXML 1.0 specification is silent on who place outbound calls for push-based voice applications. The push-barred voice applications become very important in modern information systems such as CRM. In this paper, we design and implement an extended VoiceXML platform that supports both inbound and outbound voice information services. We also extend the VoiceXML DTD so as to be able to inbound/outbound fax based on Call Control Requirements of W3C.
A Study on the to Shorten of Early Decay Time in the Reverberation Curve Using MINT
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 37~41
In this paper, we made shorter EDT(early decay time) of room reverberation curve using multiple-channel. The speech signal was processed inverse filtering with full-band and sub-band in the basis MINT, and then the multiple-channel adaptive filters were used LMS (Least Mean Square) and NLMS (Normalized Least Mean Square) algorithm. Experimental results, we could get 1/3 of time reduction at 20dB level in the reverberation curve using full-band NLMS when two microphones were used. Also, it is shown that the speech articulation was improved 80% from the test listeners with the speech, which was to shorten EDT by MINT in the subjective assessments using real room impulse response.
Performance Improvement of Stereo Acoustic Echo Canceller Using MINT Filtering
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 42~46
In this paper, a new pre-processing algorithm is proposed to improve the performance of stereo acoustic echo canceller. The proposed algorithm has the improved performance by the estimation error reduction of filter coefficient using input signal which was reduced reverberation of room in the basis MINT (Mu1tip1e-input/output Inverse Theorem) filtering. For real stereo speech signal and real room impulse response the results of simulation, we showed that the proposed method could improved 3∼5 dB ERLE (Echo Return Loss Enhancement) regardless of NLMS (Normalized Least Mean Square) and Projection adaptive algorithm.
Theoretical Study on the Effects of the Withdrawal Weighting on the Performance of Resonator Type SAW Filters
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 47~55
This paper proposes a new improved lumped element equivalent circuit analysis method to analyze withdrawal weighted SAW resonators of irregular electrode configurations, which enables to calculate the frequency response of withdrawal weighted SAW resonators. This method has led to the derivation of Smith equivalent circuit's y-parameters for a single ground electrode and formulated the resonator's admittance by calculating the total current into an IDT assembly. To illustrate the effectiveness of the technique, this method was applied to the design of a simple ladder filter and the change of the filter performance was investigated in relation to the weighting of the series and parallel resonators, respectively. The results shows that the withdrawal weighted resonator ladder filters provide better performance in their bandwidth and transition characteristics than normal ones. This new equivalent circuit analysis method can also serve as a better tool to design and analyze general SAW resonator filters.
A Study on the Cross Talk Level in a Piezoelectric Ultrasonic Array Transducer
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 56~61
In piezoelectric ultrasonic linear array transducers widely used for diagnosis, the cross talk caused by the structural cross-coupling between adjacent elements inside the transducer affects the probe performance in a significant manner. In this study, we constructed a finite element model of a piezoelectric ultrasonic transducer, and analyzed its cross talk level with respect to the shape of and materials inside the kerf, The results of this work can be utilized in optimal design of the transducers for medical diagonosis and treatment as well as W applications.
MVDR Beamformer for High Frequency Resolution Using Subband Decomposition
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 62~68
It is well known that the MDVR beamforming outperforms the conventional delay-sum beamformer in the sense of noise rejection and bearing resolution. However, the MDVR method requires long observation time to achieve high frequency resolution. The STMV method uses the steered covariance matrix of sensor data, so it has an ability to form an adaptive weight vector from a single time-series snapshot. But it uses the same weight vector across all frequencies. In this paper, we propose an SSMV method. The basic idea of the SSMV method is to decompose a full frequency band into several subbands to acquire a weight vector for each subband, individually. Also the wrap may be divided into several subarrays in order to reduce a computational load and the bandwidth of each subband. Simulations using real sea trial data show that the proposed SSMV method has good performance with short observation time.
Initial Codebook Design by Modified splitting Method
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 69~72
We propose a modified splitting method to obtain an initial codebook, which is used to design a codebook. The principle of the proposed method is that the more representative vectors are assigned to the class, which has the mere member training vectors or a lower squared error. The conventional K-means algorithm and the method provided from reference (5) are used to estimate the performance of the designed codebook. In thin work, the proposed method shows better results than the conventional splitting method in all experiments.
Vibration Intensity Analysis of Penetration Beam-plate Coupled Structures
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 73~81
The transmission of vibration energy through beam-plate junctions in vibration intensity analysis called power new analysis (PFA) has been studied. PFA is an analytic tool for the prediction of frequency averaged vibration response of built-up structures at medium to high frequency ranges. The power transmission and reflection coefficients between the semi-infinite beam and plate are estimated using the wave transmission approach. For the application of the power coefficients to practical complex structures, the numerical methods, such as finite element method are needed to be adapted to the power flow governing equation. To solve the discontinuity of energy density at the joint, joint matrix is developed using energy flow coupling relationships at the beam-plate joint. Using the joint matrix developed in this paper, an idealized ship stem part is modeled with finite element program, and vibration energy density and intensity are calculated.
Implementation of an Efficient Wavelet Based Audio Data Retrieval System
The Journal of the Acoustical Society of Korea, volume 21, issue 1, 2002, Pages 82~88
In this paper, we proposed a audio indexing method that is used wavelet transform for audio data retrieval. It is difficult for audio data to make a efficient audio data index because of its own particular properties, such as requirement of large storage, real time to transfer and wide bandwidth. An audio data in del using wavelet transform make it possible to index and retrieval by using the particular wavelet transform properties. Our proposed indexing method doesn't separate data to several blocks. Therefore we use both high-pass and low-pass parts of last level coefficient of wavelet transform. Audio data indexing is made by applying the string matching algorithm to high-pass part and zero-crossing histogram to low-pass part. These are transformed to the continued strings, Through this method, we described a retrieval efficiency. The retrieval method is done by comparing the database index string to the query string and then data of minimum values is chosen to the result. Our simulation decided proper comparative coefficient and made known changing of retrieval efficiency versus audio data length. The results show that the proposed method improves retrieval efficiency compared to conventional method.