• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.61-64
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• 1996

To improve the performance of a recognition system, namely the recognition rate, we propose a hew implementation of perceptual distance using LPC cepstrum(perceptual cepstrum, PLC). The PLC is caculated by convolution of a usual LPC cepstrum and a perceptual lifter(PL). To caculate PL, we define a new weighting function in the linear frequency domain considering the frequency scale(Bark-scale) characteristics. The PL is the inverse Fourier transform of the exponents of the weighting function. We verified our method through the speech recognition experiments. The performance of PLC was compared with that of the rasied sine liftering method.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.80-84
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• 1999

In this paper, the split model analysis algorithm, which can generate the wideband speech signal from the spectral information of narrowband signal, is developed. The split model analysis algorithm deals with the separation of the 10/sup th/ order LPC model into five cascade-connected 2/sup nd/ order model. The use of the less complex 2/sup nd/ order models allows for the exclusion of the complicated nonlinear relationships between model parameters and all the poles of the LPC model. The relationships between the model parameters and its corresponding analog poles is proved and applied to each 2/sup nd/ order model. The wideband speech signal is obtained by changing only the sampling rate.

• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.90-98
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• 1996

An auditory model is proposed for robust speech recognition in noisy environments. The model consists of cochlear bandpass filters and nonlinear stages, and represents frequency and intensity information efficiently even in noisy environments. Frequency information of the signal is obtained by zero-crossing intervals, and intensity information is also incorporated by peak detectors and saturating nonlinearities. Also, the robustness of the zero-crossings in estimating frequency is verified by the developed analytic relationship of the variance of the level-crossing interval perturbations as a function of the crossing level values. The proposed auditory model is computationally efficient and free from many unknown parameters compared with other auditory models. Speaker-independent speech recognition experiments demonstrate the robustness of the proposed method.

• Journal of the Korean Society for Nondestructive Testing
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• v.30 no.5
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• pp.458-463
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• 2010

Since the acoustic nonlinearity is sensitive to the minute variation of material properties, the nonlinear ultrasonic technique(NUT) has been considered as a promising method to evaluate the material degradation or fatigue. However, there are certain limitations to apply the conventional NUT using the bulk wave to thin plates. In case of plates, the use of Lamb wave can be considered, however, the propagation characteristics of Lamb wave are completely different with the bulk wave, and thus the separate study for the nonlinearity of Lamb wave is required. For this work, this paper analyzed first the conditions of mode pair suitable for the practical application as well as for the cumulative propagation of quadratic harmonic frequency and summarized the result in for conditions; (1) phase matching, (2) non-zero power flux, (3) group velocity matching, and (4) non-zero out-of-plane displacement. Experimental results in aluminum plates showed that the amplitude of the secondary Lamb wave and nonlinear parameter growed up with increasing propagation distance at the mode pair satisfying the above all conditions and that the ration of nonlinear parameters measured in Al6061-T6 and Al1100-H15 was closed to the ratio of the absolute nonlinear parameters.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.386-393
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• 2008

This paper proposes a new approach for speech enhancement in highly nonstationary noisy environments. The spectral subtraction (SS) is a well known technique for speech enhancement in stationary noisy environments. However, in real world, noise is mostly nonstationary. The proposed method uses an auto control parameter for an adaptive threshold to work well in highly nonstationary noisy environments. Especially, the auto control parameter is affected by a linear function associated with an a posteriori signal to noise ratio (SNR) according to the increase or the decrease of the noise level. The proposed algorithm is combined with spectral subtraction (SS) using a hangover scheme (HO) for speech enhancement. The performances of the proposed method are evaluated ITU-T P.835 signal distortion (SIG) and the segment signal to-noise ratio (SNR) in various and highly nonstationary noisy environments and is superior to that of conventional spectral subtraction (SS) using a hangover (HO) and SS using a minimum statistics (MS) methods.

• Journal of the Korean Society for Nondestructive Testing
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• v.34 no.4
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• pp.277-282
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• 2014

Recently, much attention has been paid to nonlinear ultrasonic technology as a potential tool to assess hidden damages that cannot be detected by conventional ultrasonic testing. One nonlinear ultrasonic technique is measurement of the resonance frequency shift, which is based on the hysteresis of the material elasticity. Sophisticated measurement of resonance frequency is required, because the change in resonance frequency is usually quite small. In this investigation, the nonlinear electromagnetic acoustic resonance (NEMAR) method was employed. The NEMAR method uses noncontact electromagnetic acoustic transducers (EMATs) in order to minimize the effect of the transducer on the frequency response of the object. Aluminum plate specimens that underwent three point bending fatigue were tested with a shear wave EMAT. The hysteretic nonlinear parameter ${\alpha}$, a key indicator of damage, was calculated from the resonance frequency shift at several levels of input voltage. The hysteretic nonlinear parameter of a damaged sample was compared to that of an intact one, showing a difference in the values.

• The Journal of the Acoustical Society of Korea
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• v.23 no.2
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• pp.125-133
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• 2004

Most of the way shape estimation method using reference sources assume that the reference sources are in the farfield. That is, the reference sources are assumed to be far from the array. However, in applications of the array with reference sources, the reference sources are not far from the way, so that in practical ocean environments, the conventional method using farfield source model fail to estimate the positions of the hydrophones. In this paper, based on the nearfield source model, a subspace-based array shape estimation method was proposed. In the proposed method, nearfield reference source is modeled using the differential time delay at each hydrophone, and nearfield parameters are derived. Using these parameters, a subspace-based array shape estimation method that generalizes the existing farfield subspace fitting method which can work regardless of the range of the source is proposed. The Cramer-Rao lower bound for the proposed method is investigated. The results of the numerical experiments indicate that the proposed method performs well in estimating the shape of a perturbed way regardless of the ranges of the reference sources.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.32-37
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• 1991

고해상도 신호처리의 기본적인 문제는, 관찰 데이터의 개수가 작고 신호 대 잡음비(SNR)가 낮아서, 푸리에 분석기법에 의해 주기신호가 분해되지 않는 경우에, 신호의 파라미터를 추정하는 것이라 할 수 있다. 주기신호의 주파수 추정 문제에서는 일반적으로 주기신호의 개수를 알고 있다고 가정하는데, 주기신호의 개수가 사전에 알려져 있지 않은 경우, 주파수 추정은 결국 주기신호의 개수결정문제가 되어, EVD나 SVD를 이용한 개수 결정방법이 활발히 연구되어 왔다. 고해상도 신호처리에서는 EVD나 SVD의 비선형 특성 상임게치 신호 대 잡음비가 존재하며 이 SNR보다 낮은 경우 심각한 왜곡현상을 보이게 되어, 주파수 추정 또는 주기신호의 개수결정에 큰 오차를 보이게 된다. 주기신호의 개수를 사전에 알고 있는 경우, 임게치 SNR를 낮추려는 노력으로는 overdetermined over-ranked structured correlation matrix의 rank reduction과 averaging을 이용한 신호 향상방법(signal enhancement)이 연구되어 왔다. 그러나 사전에 주기신호의 개수를 알아야만 하는 결점이 있고, 잡음이 백색이여야 하는 제약이 있었다. 일반적으로 환경 잡음은 유색이고, 주기신호의 개수를 사전에 모르는 경우이므로, 낮은 SNR에서의 주파수 추정문제는 유색잡음을 고려한 신호향상으로 임게치 SNR을 낮추고 주기신호의 개수를 결정한 후 주파수 추정이 이루어져야 한다. 본 논문에서는 이를 위해 광대협 유색잡음에서의 신호향상과 그 과정 중 중 주기신호의 개수를 결정하는 알고리즘ㅇ르 제시하고자 한다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.148-154
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• 2009

In audio coding, SBR technology synthesizes the high-bands using patched time-frequency information from low-bands and the correction parameters, Since SBR transmits only correction parameters for high-bands, it provides a low-rate coding of high-bands, and is used as a core module of MPEG-4 HE-AAC, SBR was originally designed for audio signal and its performance for speech signal tends to decrease, and the major reason is an excessive noise floor in high-bands which is caused by incorrect tonality computation, In this paper, a new method to determine noise floor level in an adaptive fashion according to the speech characteristics is proposed in order to solve the problem of SBR for speech signal, The proposed method maintains the compatibility with the standard SBR, and the subjective performance evaluation shows that the proposed method improves the SBR performance especially for male speech signal compared with the standard SBR.

• Journal of the Korean Society of Propulsion Engineers
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• v.21 no.5
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• pp.46-53
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• 2017

This paper deals with an acoustic model for a lean premixed gas turbine combustor composed of three stages: premixing chamber, nozzle and flame tube. Our model is given as an acoustic transfer function whose input is a heat release rate perturbation and output is a velocity perturbation at a flame location. We have shown that the resonance frequencies are functions of three round-trip frequencies of acoustic wave in each stage, and area ratios between stages. By analyzing poles of the acoustic transfer function, we could characterize resonant frequencies and their dependency on various system parameters of a combustor. It was found that our analytic findings match with existing numerical and experimental results in literature.