• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.599-609
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• 2009

An overview of the recently finalized multi-channel audio coding standard MPEG Surround is provided. This audio coding standard downmixes multi-channel signals to mono or stereo signals and, simultaneously, extracts spatial parameters for its encoding process. In its decoding process, it reconstructs multi-channel signals based on the downmix signals and spatial parameters. Since the downmix signals are coded in conventional audio coding format such as AAC and MP3 and the spatial parameters require a small amount of information MPEG Surround guarantees high sound quality multi-channel audio at low bit rates. Besides, it is backward-compatible to conventional audio coding techniques because the downmix signals can be played on portable audio devices ignoring the spatial parameter information. In this paper, Part 1 presents an overview of the basic structure of MPEG Surround and Part 2 describes various modes and tools including the binaural mode which supports the virtual 5.1-channel playback via headphones or earphones. The listening test results by various companies and organizations are also presented.

• Journal of the Korean Society for Nondestructive Testing
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• v.30 no.2
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• pp.139-145
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• 2010

Bending fatigue of aluminium alloy was characterized by acoustic nonlinearity of narrow band laser-generated surface wave. The higher harmonic components generated intrinsically by arrayed line laser beam were analyzed theoretically and acoustic nonlinearity was measured successfully on the surface of fatigue damaged aluminium 6061 alloy. The acoustic nonlinearity increased as a function of fatigue cycles and has close relation with damage level. Consequently, the nonlinear acoustic technique of laser-generated surface wave could be potential to characterize surface damages subjected to fatigue.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.5
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• pp.162-166
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• 2009

In this paper, we propose a novel approach of an acoustic echo cancellation (AEC) algorithm which is differently adopted in the relevant period condition by the suppression parameter of a parametric wiener filter (PWF). The PWF uses the suppression parameter to compensate uncertainty of acoustic echo signal estimation. The existing PWF method using the fixed suppression parameter derives the distortion of the near-end signal at the double-talk. To solve this problem, the boundary condition is devised using decision of the double-talk detection (DTD) algorithm and voice activity detector (VAD). The boundary condition makes it possible to treat differently depending on the case of the single-talk and double-talk. According to the experimental results, the proposed approach is found to be effective for acoustic echo cancellation using the boundary condition.

• The Journal of the Acoustical Society of Korea
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• v.42 no.4
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• pp.364-369
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• 2023

Time-delay estimation between two receivers is a technique that has been applied in a variety of fields, from underwater acoustics to room acoustics and robotics. There are two types of time delay estimation techniques: one that estimates the amount of time delay from the correlation between receivers, and the other that parametrically models the time delay between receivers and estimates the parameters by system recognition. The latter has the characteristic that only a small fraction of the system's parameters are directly related to the delay. This characteristic can be exploited to improve the accuracy of the estimation by methods such as Lasso regularization. However, in the case of Lasso regularization, the necessary information is lost. In this paper, we propose a method using Elastic Net that adds Ridge regularization to Lasso regularization to compensate for this. Comparing the proposed method with the conventional Generalized Cross Correlation (GCC) method and the method using Lasso regularization, we show that the estimation variance is very small even for white Gaussian signal sources and colored signal sources.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.231-234
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• 1998

시간영역에서 유성음의 피치를 조절하기 위해 한피치구간의 신호 중에서 성문이 닫힌 구간의 특성을 추정한 파라미터로 성문 닫힌 구간의 신호에 연속하여 선형적으로 연장 또는 축소하므로써 고 음질을 유지하면서도 자유롭게 피치를 조절할 수 있는 방법을 제안하였다. 제안된 방법은 PSOLA 기법에서와 같은 window의 적용이나 신호의 겹침에 의한 영향이 최소화되므로 보다 명료한 합성음을 얻을 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.1 no.1
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• pp.70-75
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• 1982

본 논문에서는 이차원스펙트럼 분석기를 구성하여 진동 파라미터들을 추정하고 기존 방법인 Power Spectrum 방법과 비교 검토하여 이차원스펙트럼이 gaussian noise를 제거하기 때문에 유효함을 알 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.15-23
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• 2001

This paper presents a voice conversion technique that modifies the utterance of a source speaker as if it were spoken by a target speaker. Feature parameter conversion methods to perform the transformation of vocal tract and prosodic characteristics between the source and target speakers are described. The transformation of vocal tract characteristics is achieved by modifying the LPC cepstral coefficients using Linear Multivariate Regression (LMR). Prosodic transformation is done by changing the average pitch period between speakers, and it is applied to the residual signal using the LP-PSOLA scheme. Experimental results show that transformed speech by LMR and LP-PSOLA synthesis method contains much characteristics of the target speaker.

• The Journal of the Acoustical Society of Korea
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• v.24 no.6
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• pp.303-308
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• 2005

In this paper, the acoustic properties of the light bulb are presented based on a new light bulb source system of continuously transmitting implosive signal . We describe the results of analysis of bulb signals and comparison with Previous works. The results show that Peak-source-level and Primary resonant frequency are increasing with increasing source depth. This bulb source can be used for the purpose of geoacoustic parameter inversion and source tracking in sha]low water via matched field processing.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.325-332
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• 2008

The prediction method of TS parameters, frequency response, and electrical input impedance is proposed with physical properties of parts and results of electromagnetic FEA(Finite Element Analysis) in a loudspeaker driver design. In design for weight reduction and improvement of flux density asymmetry, the prediction results are well coincided with measurement ones. As the applications, it can be applied in design for improvement of the $2^{nd}$ harmonic distortion with flux density distribution analysis. The proposed method is expected to be utilized for reducing trial-and-error process in electromagnetic parts design. It can also be used for providing guidelines for parts selection in the early stages.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4
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• pp.31-37
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• 1999

A bimodal speech recognition using visual and audio information has been proposed and researched to improve the performance of ASR(Automatic Speech Recognition) system in noisy environments. The integration method of two modalities can be usually classified into an early integration and a late integration. The early integration method includes a method using a fixed weight of lip parameters and a method using a variable weight according to speech SNR information. The 4 late integration methods are a method using audio and visual information independently, a method using speech optimal path, a method using lip optimal path and a way using speech SNR information. Among these 6 methods, the method using the fixed weight of lip parameter showed a better recognition rate.