• 한국지반공학회:학술대회논문집
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• 한국지반공학회 2006년도 춘계 학술발표회 논문집
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• pp.552-561
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• 2006

The purpose of this study is to propose a pavement analysis system which considers dynamic effects resulted from the various vehicle speeds. Vehicle loading effects were estimated by loading frequency and dynamic loads under various vehicle speeds. In addition, a proposed analysis model takes the non-linear temperature using a predictive model for dynamic modulus in asphalt layer and the non-linear stress in the unbound material. To examine adequacy of existing multi-layer elastic analysis of non-linear temperature in asphalt layer and non-linear stress conditions in unbound material, this study divided layers of asphalt pavement structures with 10 layers in asphalt, 2 layers in subbase and 1 layer in subgrade. In order to verify the pavement analysis system that considers various speeds, deflections of pavement calculated using ABAQUS, a three dimensional finite element program, were compared with the results of field tests under various speeds.

• 대한음성학회지:말소리
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• 제48호
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• pp.117-126
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• 2003

source of speech signal is actually composed of combination of periodic and aperiodic components, although it is often modeled to either one of those. In the paper an experiment which can separate periodic and aperiodic components from speech source. Linear predictive residual signal was used as a approximated vocal source the original speech to obtain the estimated aperiodic part. Iterative extrapolation method was used to compute the aperiodic part.

• Communications for Statistical Applications and Methods
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• 제8권2호
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• pp.407-416
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• 2001

Sample surveys are usually designed and analyzed to produce estimates for a large area or populations. Therefore, for the small area estimations, sample sizes are often not large enough to give adequate precision. Several small area estimation methods were proposed in recent years concerning with sample sizes. Here, we will compare simple Bayesian approach with Bayesian prediction for small area estimation based on linear regression model. The performance of the proposed method was evaluated through unemployment population data form Economic Active Population(EAP) Survey.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2212-2215
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• 2003

In this paper, we proposed speech feature vector extractor for embedded system using TMS 320C31 DSP chip. For this extractor, we used algorithm using cepstrum coefficient based on LPC(Linear Predictive Coding) that is reliable algorithm to be is widely used for speech recognition. This system extract the speech feature vector in real time, so is used the mobile system, such as cellular phones, PDA, electronic note, and so on, implemented speech recognition.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1995년도 제12회 음성통신 및 신호처리 워크샵 논문집 (SCAS 12권 1호)
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• pp.184-187
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• 1995

This paper proposes an algorithm for machine recognition of phonemes in continuous speech. The proposed algorithm is static strategy neural network. The algorithm uses, at the stage of training neuron, features such as PARCOR coefficient and auditory-like perceptual liner prediction . These features are extracted from speech samples selected by a sliding 25.6msec windows with s sliding gap being 3 msec long, then interleaved and summed up to 7 sets of parmeters covering 171 msec worth of speech for use of neural inputs. Perfomances are compared when either PARCOR or auditory-like PLP is included in the feture set.

• 대한의용생체공학회:의공학회지
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• 제11권1호
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• pp.75-82
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• 1990

The purpose of this study is to help the hearing Impaired's speech training with a visible speech processing system. In brief, this system converts the features of speech signals into graphics on monitor, and adjusts the features of hearing impaired to normal ones. There are formant and pitch in the features used for this system. They are extracted using the digital signal processing such as linear predictive method or AMDF(Average Magnitude Difference Function). In order to effectively train for the hearing impaired's abnormal speech, easilly visible feature has been being studied.

• 한국음향학회지
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• 제4권1호
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• pp.43-54
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• 1985

이 논문에서는 Frequency weighting을 이용하여 선형예측 부호화기의 명료성을 개선하는 방법 을 연구한다. 잡음이 섞이지 않은 음성에 대해서는 음성을 분석하기전에 frequency weighting을 행한다. 또한 잡음이 섞인 음성인 경우에는 잡음성분을 spectral subtraction 방법에 의해서 제거한 다음에 frequency weighting을 준다. 이 때 frequency weighting을 주기 위해서 귀의 특성과 연관되어 잘 알려 진 C- message weighting 함수, flanagan weighting 함수 및 articulation index를 약간 수정한 weighting 함수를 사용했다. 여러 객관적인 distance measure를 사용하여 frequency weighting 방법의 성능을 측정하고 귀로 들어 본 결과, frequency weighting 방법을 사용하여 선형예측 방법에 의한 합성 음의 명료도를 효율적으로 개선할 수 있었다.

• 대한전기학회논문지
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• 제37권4호
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• pp.251-256
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• 1988

This work describes a speech analysis of Korean number ('1'-'10') which are spoken by several speakers using Fast Walsh Transform(FWHT) method. FWHT includes only addition and subtraction operations, therefore faster and needs less memory than FFT(Fast Fourier Transfifrm) or LPC(Linear Predictive Coding) analysis method. We have investigated that FWHT method can find speaker independent feature(which represents same cue about some word independent of different speakers) The results of this experiment, the 70% of same words(korean number '2')which spoken by several speakers have had slmilar patterns.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 추계종합학술대회 논문집(4)
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• pp.163-166
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• 2000

In this paper, we will present a lossy data compression method for coding multispectral images. The proposed method uses both spatial and spectra] correlation inherent in multispectral images. First, band 2 and band 6 are vector quantized. Secondly, band 4 is estimated with the quantized band 2 using the predictive coding. Errors of band 4 are encoded at a second stage based on the magnitude of the errors. Thirdly, remaining bands are calculated with the quantized band 2 and band 4. Errors of residual bands are wavelet transformed and then we apply the SPIHT coding on the transformed coefficients. We classify classes without extra information transmitting and then use linear predictor. And errors can be encoded by SPIHT coding at any target rate we are want. It is shown that this method has better performance than FPVQ. Average PSNR rises 0.645 dB at the same bit rate.

• Korea-Australia Rheology Journal
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• 제14권1호
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• pp.25-32
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• 2002

We report the first steps of a collaborative project between the University of Queensland, Polyflow, Michelin, SK Chemicals, and RMIT University, on simulation, validation and application of a recently introduced constitutive model designed to describe branched polymers. Whereas much progress has been made on predicting the complex flow behaviour of many - in particular linear - polymers, it sometimes appears difficult to predict simultaneously shear thinning and extensional strain hardening behaviour using traditional constitutive models. Recently a new viscoelastic model based on molecular topology, was proposed by McLeish and carson (1998). We explore the predictive power of a differential multi-mode version of the porn-pom model for the flow behaviour of two commercial polymer melts: a (long-chain branched) low-density polyethylene (LDPE) and a (linear) high-density polyethylene (HDPE). The model responses are compared to elongational recovery experiments published by Langouche and Debbaut (19c99), and start-up of simple shear flow, stress relaxation after simple and reverse step strain experiments carried out in our laboratory.