• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.283-286
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• 2000

Comparison for various empirical models of normal incident absorption was made, along with experiments. Comparative result indicates that Voronina model which is function of fiber diameter and porosity is more suitable than the other models. An investigation for correlation between normal and random incident absorption was carried out by experiment and analysis. It appears that at the low frequencies, the random incident absorption is higher than the normal one, whileas at the high frequencies, the random incident absorption is decreased due to the effect of grazing incident components.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1E
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• pp.42-48
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• 2002

In this paper, a computer simulation method is presented far the dynamic analysis of a hydraulic engine mount system. The hydraulic engine mount system controls the damping characteristics using the viscosity of fluid flow. The complex stiffnesses of the main rubber for the hydraulic engine mount system are computed using a finite element analysis. The equations of motion considering the parameters of the hydraulic engine mount system are derived. To investigate the effects of the hydraulic engine mount system, the computer simulation running over a typical rough road is carried out using a vehicle dynamic model. These results are compared with those of the conventional rubber mount system.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4E
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• pp.146-155
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• 2002

We present a new technique for achieving source separation when given only a single charmel recording. The main idea is based on exploiting the inherent time structure of sound sources by learning a priori sets of time-domain basis functions that encode the sources in a statistically efficient manner. We derive a learning algorithm using a maximum likelihood approach given the observed single charmel data and sets of basis functions. For each time point we infer the source parameters and their contribution factors. This inference is possible due to the prior knowledge of the basis functions and the associated coefficient densities. A flexible model for density estimation allows accurate modeling of the observation, and our experimental results exhibit a high level of separation performance for simulated mixtures as well as real environment recordings employing mixtures of two different sources. We show separation results of two music signals as well as the separation of two voice signals.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.42-46
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• 2002

In this paper, a new pre-processing algorithm is proposed to improve the performance of stereo acoustic echo canceller. The proposed algorithm has the improved performance by the estimation error reduction of filter coefficient using input signal which was reduced reverberation of room in the basis MINT (Mu1tip1e-input/output Inverse Theorem) filtering. For real stereo speech signal and real room impulse response the results of simulation, we showed that the proposed method could improved 3∼5 dB ERLE (Echo Return Loss Enhancement) regardless of NLMS (Normalized Least Mean Square) and Projection adaptive algorithm.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1
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• pp.45-52
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• 1999

In this thesis, a packer is proposed and designed for removing the bottle-neck effect and easy signal processing using a new algorithm with the operation frequency of 54MHz. To verify the performance of the proposed packer, DCT coefficient encoding block with ROM table using a combinational logic is designed and its output data are used the input data of the packer. Circuits in this thesis are designed by using VHDL code and its modeling and simulation are performed by SYNOPSYS tool using $0.65{\mu}m$ rule.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.299-302
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• 2004

본 논문에서는 내용 기반 음악 정보 검색에 음악의 temporal 특징을 이용한 검색 방법을 제안한다. 방송환경에 적용하기 위해 검색 범위를 드라마나 영화의 배경 음악으로 사용되는 OST 앨범으로 제한하였다. 오디오의 특징 벡터로써 UFCC(Mel Frequency Cepstral Coefficient)를 사용하였으며 이 특징 벡터를 이용하여 VQ(Vector Quantization)로 부호화한 codeword로 오디오 신호의 시변 특성을 표현한다. 본 논문에서는 제안한 음악의 temporal 특성을 반영한 codeword-sequence를 이용하는 방법을 pitch-histogram을 기반으로 하는 방법 및 MFCC codeword-histogram을 기반으로 하는 방법과 비교하고 성능 개선을 보여주었다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.369-372
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• 1998

본 논문은 어휘독립(Vocabulary-Independent) 환경에서 별도의 훈련과정 없이 인식대상 어휘를 추가 및 변경할 수 있는 가변어휘(Variable Vocabulary) 음성인식에 관한 연구를 다룬다. 가변어휘 인식은 처음에 대용량 음성 데이터베이스(DB)로 음소모델을 훈련하고 인식대상 어휘가 결정되면 발음사전에 의거하여 음소모델을 연결함으로써 별도의 훈련과정 없이 인식대상 어휘를 변경 및 추가할 수 있다. 문맥 종속형(Context-Dependent) 음소 모델인 triphone을 사용하여 인식실험을 하였고, 인식성능의 비교를 위해 어휘종속 모델을 별도로 구성하여 인식실험을 하였다. Unseen triphone 문제와 훈련 DB의 부족으로 인한 모델 파라메터의 신뢰성 저하를 방지하기 위해 state-tying 방법 중 음성학적 지식에 기반을 둔 tree-based clustering(TBC) 기법[1]을 도입하였다. Mel Frequency Cepstrum Coefficient(MFCC)와 대수에너지에 기반을 둔 3 가지 음성특징 벡터를 사용하여 인식 실험을 병행하였고, 연속 확률분포를 가지는 Hidden Markov Model(HMM) 기반의 고립단어 인식시스템을 구현하였다. 인식 실험에는 22 개 부서명 DB[3]를 사용하였다. 실험결과 어휘독립 환경에서 최고 98.4%의 인식률이 얻어졌으며, 어휘종속 환경에서의 인식률 99.7%에 근접한 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.8 no.4
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• pp.60-64
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• 1989

In this paper, an local path constraint Is proposed in order to increase the speech recognition rate. An input speech signal is analyzed by autocorrelation and LPC coefficient as parameters. The local path constraint of the proposed type was compared with the conventional five types. The speechs used in this search are the subway stops, and the 130 words pronounced 10 times for the different 13 words consisting of 11 characters of syllable by 2 male and 1 female are tested. As a result, we proved that this proposed type is the most optimal type and the recognition rate of $94.6\%$ is obtained .

• The Journal of the Acoustical Society of Korea
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• v.25 no.3E
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• pp.131-136
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• 2006

The PML developed for the radio wave propagation is a powerful numerical domain truncation technique. We perform an analytic study on the reflection from the PML inserted in the ocean bottom. In the ocean bottom, we show the PML to have the improved performance but simultaneously the degeneration below the critical angle of the fast ocean bottom. The degeneration of the PML can be simply relaxed by stretching the thickness of the PML or putting the attenuation coefficient to the ocean bottom. As a better solution, we propose the improved truncation technique based on the PML and the non-local boundary condition. Finally, we apply the PML to the acoustic wave propagation using split-step Pade PE solver. For the problems of the ocean waveguide, the numerical efficiency of the PML is examined and the usefulness of the PML is confirmed.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1E
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• pp.1-7
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• 2006

This study examines the reflection characteristic of a thin transition layer of the ocean bottom showing variability with respect to depth. In order to model the surficial sediment simply, we reduce the Biot model to the depth dependent wave equation for the pseudo fluid using the fluid approximation (weak frame approximation). From the reduced equation, the difference between the inherent frequency dependency of the reflection and the frequency dependency resulting from a thin transition layer is investigated. Using Tang's depth porosity profile model of the surficial sediment [D. Tang et al., IEEE J. Oceanic Eng., vol.27(3), 546-560(2002)], we numerically simulated the reflection loss and investigated the contribution from both frequency dependencies. In addition, the effects of different sediment type and varying depth structure of the sediment are discussed.