• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.58-62
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• 1999

This paper proposes use of acoustic parameters to improve the discriminability among digit models in Korean connected digit recognition. The proposed method used the logarithmic values of energy ratio between the predetermined frequency bands as additional feature parameters, based on the acoustic-phonetic knowledge. The results of our experiment show that the proposed method reduced the error rate by 46% in comparison with the baseline system. And incorporation of channel compensation technique in the proposed method yielded error reduction of about 69%.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1E
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• pp.59-64
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• 1996

When the input-output record is available, the identification of a bilinear system is considered. It is assumed that the input is noise free and the output is contaminated by an additive noise. It is further assumed that the covariance matrix of the noise is known up to a factor of proportionality. The extended generalized total least squares (e-GTLS) method is proposed as one of the consistent estimators of the bilinear system parameters. Considering that the input is noise-free and that bilinear system equation is linear with respect to the system parameters, we extend the GTLS problem. The extended GTLS problem is reduced to an unconstrained minimization problem, and is solved by the Newton-Raphson method. We compare the GTLS method and the e-GTLS method in the point of the accuracy of the estimated system parameters.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4E
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• pp.35-40
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• 1998

In this paper, an efficient quantization method of line spectrum pairs(LSP) with cascaded structure of vector quantizer and scalar quantizer is proposed. First, input LSP parameters is vector-quantized using a codebook a with a moderate number of entries. In the second stage of quantization, the components of residual vector are individually quantized by the scalar quantizer. The utilization of ordering property of LSP parameters and the inclusion of interframe prediction improve the quantizer performance and remove the stability check routine after quantization procedure. The new vector-scalar hybrid quantizer using 26 bits/frame shows a transparent quality of speech that an average spectral distortion is 1 dB and the frame proportion with above 2 dB spectral distortion is less than 2%. The performances of proposed quantization method is evaluated in the transmission errors.

• The Journal of the Acoustical Society of Korea
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• v.19 no.1
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• pp.48-53
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• 2000

To analyze a room reverberation, many objective parameters have been suggested as measures of subjective perception in concert hall acoustics. However, it is difficult to understand the relationship among the individual parameters and to have a whole image of them. In this paper five primary objective parameters were selected and a reverberation model based on them was established by the reverberation decay curve. The proposed model was interpreted geometrically in terms of subjective characteristics. And finally we discuss the usefulness and the limitation of the model when it is applied to the analysis and.

• The Journal of the Acoustical Society of Korea
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• v.27 no.3E
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• pp.77-83
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• 2008

In the process of the sonar image textures produced, the orientation and scale factors are very significant. However, most of the related methods ignore the directional information and scale invariance or just pay attention to one of them. To overcome this problem, we apply Gabor wavelet to extract the features of sonar images, which combine the advantages of both the Gabor filter and traditional wavelet function. The mother wavelet is designed with constrained parameters and the optimal parameters will be selected at each orientation, with the help of bandwidth parameters based on the Fisher criterion. The Gabor wavelet can have the properties of both multi-scale and multi-orientation. Based on our experiment, this method is more appropriate than traditional wavelet or single Gabor filter as it provides the better discrimination of the textures and improves the recognition rate effectively. Meanwhile, comparing with other fusion methods, it can reduce the complexity and improve the calculation efficiency.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.45-49
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• 1991

In area of the speech synthesis, the waveform coding with high quality are mainly used to the synthesis by analysis. However, it is difficult to applying the waveform coding to the synthesis by rule, because the parameters of this coding are not classified as either excitation parameters and vocal tract parameters. In this paper, we proposed a new pitch change method that can alter the pitch periods in the waveform coding. The proposed method expands the pitch period by the LPC synthesis method, and then the period is compressed by the waveform halving technique. Thus, it is possible that the waveform coding is carried out the synthesis by rule in speech processing.

• Proceedings of the KSPS conference
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• 2003.05a
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• pp.101-104
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• 2003

The purpose of this study was to compare acoustic differences of fricative /s/ between the dysarthric subjects and normal subjects. In additional, the subjects' speeches were evaluated in terms of word intelligibility containing /s/ and perceptual severity. Acoustic parameters were duration, peak frequency and intensity of /s/. The results showed that the first peak frequency and intensity of /s/ were significantly different between dysarthric subjetcts and normal subjects. Second, peceptual parameters were significantly different between dysarthric subjetcts and normal subjects. The Pearson correlation coefficient was used to determine the relationship between the acoustical and perceptual data. The results showed that there was a strong correlation between perceptual parameters and peak frequency of /s/.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.893-898
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• 1994

In this paper, the influence of voice source estimation and modeling on speech synthesis and coding is examined and then their new estimation and modeling techniques are proposed and verified by computer simulation. It is known that the existing speech synthesizer produced the speech which is dull and inanimated. These problems are arised from the fact that existing estimation and modeling techniques can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can represent a variety of source characteristics. First, we divide speech samples in one pitch region into four parts having different characteristics. Second, the vocal-tract parameters and voice source waveforms are estimated in each regions differently using sequential SVD. Third, we propose composite source model as a new voice source model which is represented by weighted sum of pre-defined basis functions. And finally, the weights and time-shift parameters of the proposed composite source model are estimeted uning EM(estimate maximize) algorithm. Experimental results indicate that the proposed estimation and modeling methods can estimate more accurate voice source waveforms and represent various source characteristics.

• The Journal of the Acoustical Society of Korea
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• v.26 no.6
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• pp.276-285
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• 2007

In this paper it was discussed and compared measured loudspeaker impedance curve with ones reconstructed by TS parameters estimated using four kinds of parameter estimation methods developed in frequency domain. Frequency dependent parameters were introduced and derived using least square error minimization technique. For known dynamic mass TS parameter estimation methods were reviewed and also proved non-uniqueness of these parameters by simulation method. Minimum phase transformation was adopted to derive phase information from magnitude of loudspeaker electrical impedance curve measured by one channel analyzer.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.443-448
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• 2005

This study aimed to evaluate the room acoustical performances of multi-purpose halls, analyzing the architectural features by survey and the acoustical characteristics by measurements. Moreover, vocabularies appropriate for the evaluation of the room acoustical performance were determined through questionnaire and subjective responses. From the correlation analysis and factor analysis to the physical values through measurements and the subjective evaluation using vocabularies, useful vocabularies for the evaluation of multi-purpose halls' acoustical performances were determined and physical parameters corresponding to the vocabulary were identified.