• MALSORI
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• v.55
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• pp.119-130
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• 2005

The aim of this paper is to investigate some acoustical characteristics of the voice quality features from the emotional speech database. Six different parameters are measured and compared for 6 different emotions (normal, happiness, sadness, fear, anger, boredom) and from 6 different speakers. Inter-speaker variability and intra-speaker variability are measured. Some intra-speaker consistency of the parameter change across the emotions are observed, but inter-speaker consistency are not observed.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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• pp.155-159
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• 2004

In this paper, it was analyzed the characteristics of electrical and acoustical variations for loudspeaker due to fabrication processes. First, mass of each components of loudspeaker was measured by electric precision scale and performed statistical analysis. Second. Thiele-Small parameters of sample loudspeakers produced by unskilled students were identified by known mass parameter identification method using electrical impedance method and investigated on the variations of each parameter. Electrical impedance tests and acoustic frequency responses were measured on sample loudspeakers and variations were examined to grasp relationship between components variation and fabrication processes. Main factors to effect the changes of electrical impedance were concluded by fabrication processes errors not by components of loudspeaker.

• Proceedings of the Acoustical Society of Korea Conference
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• 1987.11a
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• pp.82-85
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• 1987

Preliminary results of using the LPC parameter for text-independent speaker identification problem are presented. The idetification process includes log likelihood ratio for distance measure and dynamic programming for time normalization. To generate the data base for experiments, ten times. Experimental results show 99.4% of identification accuracy, incorrect identification were made when the speaker uses a dialect.

• Speech Sciences
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• v.9 no.1
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• pp.77-88
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• 2002

Acoustical correlates of stress can be classified as duration, intensity and fundamental frequency. This study examined the acoustical differences in the first two syllables of stressed English words produced by ten American and Korean speakers. The Korean subjects scored very high on the TOEFL. They read at a normal speed a fable from which the acoustical parameters of eight words were analyzed. In order to make the data comparison meaningful, each parameter was collected at 100 dynamic time points proportional to the total duration of the two syllables. Then the ratio of the parameter sum of the first rime to that of the second rime was calculated to determine the relative prominence of the syllables. Results showed that the durations of the first two syllables were almost comparable between the Americans and Koreans. However, statistically significant differences showed up in the diphthong pronunciations and in the words with the second syllable stressed. Also, remarkably high r-squared values were found between pairs of the three acoustical parameters, which suggests that either one or a combination of two or more parameters may account for the prominence of a syllable within a word.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1E
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• pp.65-69
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• 1996

This paper presents an algorithm for multipath channel parameter estimation by an improved Prony method. This algorithm applies a modified regularized spectral estimation to the conventional SVD Prony method. This method requires no a priori information on the number of multipath. The performance of the proposed algorithm is almost the same as that of the SVD based multipath channel parameter estimation algorithm.

• The Journal of the Acoustical Society of Korea
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• v.9 no.2
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• pp.73-81
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• 1990

• The Journal of the Acoustical Society of Korea
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• v.17 no.3E
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• pp.47-52
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• 1998

This paper prolposes a VFM (Variable Fading Memory)Kalman filtering and applies it to the parameter estimation for time-varying signals. By adaptively calculating the fading memory, the proposed algorithm does not require any predetermined fading memory when estimating the time-varying signal parameter. Moreover, the proposed algorithm has faster convergence speed than fixed fading memory one in case the signal contains an impulsive outlier. The performance of parameter estimation for time-varying signal is evaluated by computer simulation for two cases, one of which is the chirp signal whose frequency varies linearly with time and the other is the chip signal with an impulsive outlier. The experimental results show that the VFM Kalman filtering estimates the parameter of the chirp signal more rapidly than the fixed fading memory one in the region of an outlier.

• The Journal of the Acoustical Society of Korea
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• v.26 no.7
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• pp.323-327
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• 2007

In this paper, objectively quantified consonance of complex sound is proposed as a new psychoacoustical parameter. Proposing algorithm quantifies consonance of complex sound after applying psycho acoustical models which are parts of human perception such as masking effect, equal loudness contour, and critical band. To verify proposing algorithm, experiments with 10 car horn signals which have different complex sound were performed. The experiments show cross correlation of 0.95 between objectively quantified consonance by proposing algorithm and subjectively assessed consonance by listening tests. Considering the fact that there are few psychoacoustical parameter except Zwicker parameter, proposing algorithm will help to quantify psychoacoustical effect of complex sounds objectively.

• Proceedings of the Acoustical Society of Korea Conference
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• 1984.12a
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• pp.66-69
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• 1984

This paper is a study on the Endpoint Detection for Korean Speech Recognition. In speech signal process, analysis parameter was classification from Zero Crossing Rate(Z.C.R), Log Energy(L.E), Energy in the predictive error(Ep) and fundamental Korean Speech digits, /영/-/구/ are selected as date for the Recognition of Speech. The main goal of this paper is to develop techniques and system for Speech input ot machine. In order to detect the Endpoint, this paper makes choice of Log Energy(L.E) from various parameters analysis, and the Log Energy is very effective parameter in classifying speech and nonspeech segments. The error rate of 1.43% result from the analysis.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1E
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• pp.31-37
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• 2001

Matched Field Processing (MFP) is a successive process of correcting mismatches between true and assumed parameters by matching the measured acoustic field data with numerically simulated data which we call replica. The MFP is widely used both in geo-acoustic parameter inversions and in source localizations. Whether a certain parameter can be inverted effectively or whether a source can be localized correctly depends on the amount of the influence that a parameter has on the acoustic field during the matching process. Sensitive parameters can be better estimated than the less sensitive ones in MFP. On the contrary, the sensitive parameters affect adversely on the source localization results when they have uncertainties. In this paper, a sensitivity index is defined based upon the field variation resulting from the perturbed parameters. Numerical test results show that the index behaves in accordance with the results of source localization under a mismatched environment and also with the inversion solutions.