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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
Editor in Chief :
Volume & Issues
Volume 11, Issue 6 - Dec 1992
Volume 11, Issue 4 - Aug 1992
Volume 11, Issue 3 - Jun 1992
Volume 11, Issue 2 - Apr 1992
Volume 11, Issue 1 - Feb 1992
Volume 11, Issue 1E - 00 1992
Selecting the target year
Identification of Acoustic Signals of Vehicles Using Bispectrum
An, Chong-Koo ; Lee, Dong-Min ; Lee, Tai-Ho ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 5~13
Since power spectrum has no information about the phase of a signal, the power spectral analysis technique can not be used to interpret the phase coherency of the signal produced by some nonlinear process. In this case, third-order spectrum, the so called bispectrum, is very useful in analyzing such signals. some typical computer simulation results are shown in order to represent the usefulness of the bispectrum, and the bispectra of the measured acoustic signals of three vehicles are shown in order to use to identify the sources of those signals.
Assessment of Telephone Speech Transmission Quality by Opinion Test
Kwon, Yoon-Ju ; Jang, Dae-Young ; Kang, Kyeong-Ok ; Kang, Seong-Hoon ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 14~21
In order to establish the speech transmission quality of networks, a series of subjective tests for loudness rating(LR) and sidetone masking rating(STMR) among transmission impairments were carried out. As a result of subjective tests, relationships of mean opinion score(MOS) with LR and STMR, respectively, were obtained. Also, we obtained the cumulative MOS characteristics which represent the percentage of scores that subjects voted. Thus it is easy to achieve a strategic objective of customer satisfaction for present networks and new services.
A Training Algorithm for the Transform Trellis Code with Applications to Stationary Gaussian Sources and Speech
Kim, Dong-Youn ; Park, Yong-Seo ; Whang, Keum-Chan ; Pearlman, William A. ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 22~34
There exists a transform trellis code that is optimal for stationary Gaussian sources and the squared-error distortion measure at all rates. In this paper, we train an asymptotically optimal version of such a code to obtain one which is matched better to the statistics of real world data. The training algorithm uses the M algorithm to search the trellis codebook and the LBG algorithm to update the trellis codebook. We investigate the trained transform trellis coding scheme for the first-order AR(autoregressive) Gaussian source whose correlation coefficient is 0.9 and actual speech sentences. For the first-order AR source, the achieved SNR for the test sequence is from 0.6 to 1.4 dB less than the maximum achievable SNR as given by Shannon's rate-distortion function for this source, depending on the rate and surpasses all previous known results for this source. For actual speech data, to achieve improved performance, we use window functions and gain adaptation at rate 1.0 bits/sample.
Automatic Speaker Identification by Sustained Vowel Phonation
Bae, Geon-Seong ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 35~41
A speaker identification scheme using the speaker-based VQ codecook of a sustained vowel is proposed and tested. With the pitch synchronous LPC vector of the sustained vowel /i/ as a feature vector, a VQ codebook size of 4 was found to be suitable to characterize each speaker's feature space. For 40 normal speakers (20 males, 20 females), we achieved the correct identification rate of 99.4% with a training data set, and 89.4% with a test data set with speech samples of only 50 pitch periods.
Range Data Acquistion and Shape Feature Extraction
Cho, Dong-Uk ; Kim, Ji-Yeong ; Lee, Boo-Ho ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 42~51
This paper proposes an acquisition and the representation method of the 3-dimensional information. The proposed range finder system can reduce the computation time by only calculating the
of each pixel compared to the existing methods. We also propose a shape feature extraction method by considering the sign change of the acquired range data. Finally, the effectiveness of this system is demonstrated by several experiments.
Effects of Talker Sidetone and Room Noise on the Speech Level of a Talker
Kang, Kyeong-Ok ; Kang, Seong-Hoon ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 52~59
In order to see the effects of talker sidetone on a talker's speech level quantitatively when he converses with others on a telephone, we reviewed the measuring algorithm of speech level and assessed variation of speech level due to that of sidetone masking rating(STMR). We measured room noise effects on speech level, when STMR values were changed, as well. If we consider the effects of talker sidetone and room noise on speech level, the results of experiments suggest that a talker continuously tries to keep the psychological loudness of his own speech, heard by himeself via a telephone handset, at the constant and comfortable level by controlling his speaking level according as STMR value and room noise are change. That is, because the amount of his speech masked by his talker sidetone and room noise is different when STMR value and room noise are changed, we can see the tendency that he controls his speaking level in order to keep the perceived psychological loudness of his own speech to be constant.
On Realizing the Voice Response and Recoding System for a Home Visitor - A Predictor for the waveform Coding of Speech Signals by using the Dual First-Order Difference Values-
Bae, Myung-Jin ; Lee, Mi-Suk ; Lim, Un-Chun ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 60~66
We can see the fact in the autocorrelation of the speech samples that the autocorrelation of adjacent past and next sample is larger than the autocorrelation of several order time delayed samples. It is more effective to use the adjacent past and next sample for prediction of present sample than only use the several order time delayed past. Thus, in this paper, we proposed a new predictor for the wave form coding that predict the present sample by using the one past and next samples. The proposed predictor has higher prediction gain up to 9dB than that of the CCITT-ADPCM.
A Covariance Type ARMA Fast Transversal Filter
Lee, Chul-Heui ; Jang, Young-Soo ;
The Journal of the Acoustical Society of Korea, volume 11, issue 1, 1992, Pages 67~79
For effective on-line ARMA parameter estimation, a covariance type ARMA fast transversal filter (FTF) algorithm is presented. The proposed algorithm is a covariance type implementation of ELS(Extended Least Squares) estimator and it is a fast time update recursion which is based on the fact that the correlation matrix of ARMA model satisfies the shift invariance property in each sub-block. The geometric approach is used in the derivation of the proposed algorithm. It takes small computational burden of 13N+37 MADPR(Multiplication And Division Per Recursion). Also, AR and MA orders can be independetly and arbitrarily specified.