• Journal of applied mathematics & informatics
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• v.8 no.2
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• pp.327-343
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• 2001

This is a survey article on finite element a posteriori error estimates with an emphasize on gradient recovery type error estimators. As an example, the error estimator based on the ZZ patch recovery technique will be discussed in some detail.

• Journal of the Korean Mathematical Society
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• v.48 no.3
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• pp.529-550
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• 2011

In this work, a numerical solution of the incompressible Navier-Stokes equations is proposed. The method suggested is based on an algorithm of discretization by mixed finite elements with a posteriori error estimation of the computed solutions. In order to evaluate the performance of the method, the numerical results are compared with some previously published works or with others coming from commercial code like Adina system.

• Journal of applied mathematics & informatics
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• v.28 no.5_6
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• pp.1331-1345
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• 2010

This paper describes a numerical solution of Navier-Stokes equations. It includes algorithms for discretization by finite element methods and a posteriori error estimation of the computed solutions. In order to evaluate the performance of the method, the numerical results are compared with some previously published works or with others coming from commercial code like ADINA system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.2
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• pp.277-286
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• 2010

In this paper amplitude and phase distortion of the received signal through a fading channel results in a severe performance degradation of the communication system, Therefore we consider the variance of the maximum a posteriori phase estimate of sinusoidal signal by the Cramer-Rao bound in wireless fading channel. To find the Cramer-Rao lower bound for the variance of the phase, We use the derived probability density function(pdf) of the phase in Nakagami fading channel. We analyze the error performance of modulation signals using order statistics on generalized combining reception and find adequate diversity branch number.

• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.590-596
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• 2004

This paper Presents a new self-adaptation algorithm based on maximum a posteriori (MAP) eigenvoice for Korean connected digit recognition. The proposed MAP eigenvoice is developed by introducing a probability density model for the eigenvoice coefficients. The Proposed approach provides a unified framework that incorporates the Prior model into the conventional eigenvoice estimation. In self-adaptation system we use only one adaptation utterance that will be recognized, we use MAP eigenvoice that is most robust adaptation. In series of self-adaptation experiments on the Korean connected digit recognition task. we demonstrate that the performance of the proposed approach is better than that of the conventional eigenvoice algorithm for a small amount of adaptation data.

• The Journal of the Acoustical Society of Korea
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• v.27 no.5
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• pp.256-261
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• 2008

In this paper, we propose a novel approach to improve the performance of minima controlled recursive averaging (MCRA) which is based on the conditional maximum a posteriori criterion. A crucial component of a practical speech enhancement system is the estimation of the noise power spectrum. One state-of-the-art approach is the minima controlled recursive averaging (MCRA) technique. The noise estimate in the MCRA technique is obtained by averaging past spectral power values based on a smoothing parameter that is adjusted by the signal presence probability in frequency subbands. We improve the MCRA using the speech presence probability which is the a posteriori probability conditioned on both the current observation the speech presence or absence of the previous frame. With the performance criteria of the ITU-T P.862 perceptual evaluation of speech quality (PESQ) and subjective evaluation of speech quality, we show that the proposed algorithm yields better results compared to the conventional MCRA-based scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3C
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• pp.304-310
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• 2009

This paper proposes a successive MAP (maximum a posteriori probability) detection scheme with SoIC(soft interference cancellation) to reduce the receiver complexity of hierarchical M-ary QAM system. For the successive MAP detection, modulation symbols generated from the other data streams are treated as Gaussian noise or eliminated as the soft interference according to their priorities. The log-likelihood ratio of the a posteriori probability (LAPRP) of each bit is calculated by the MAP detector with an adjusted noise variance in order to take the elimination and Gaussian assumption effect into account. By separating the detection process into the successive steps, the detection complexity is reduced to increase linearly with the number of bits per hierarchical M-ary QAM symbol. Simulation results show that the proposed detection provides a small performance degradation as compared to the optimal MAP detection.

• Phonetics and Speech Sciences
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• v.4 no.3
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• pp.111-117
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• 2012

In this paper, we compare the performance of several speaker adaptation methods for a HMM-based Korean speech synthesis system with small amounts of adaptation data. According to objective and subjective evaluations, a hybrid method of constrained structural maximum a posteriori linear regression (CSMAPLR) and maximum a posteriori (MAP) adaptation shows better performance than other methods, when only five minutes of adaptation data are available for the target speaker. During the objective evaluation, we find that the duration models are insufficiently adapted to the target speaker as the spectral envelope and pitch models. To alleviate the problem, we propose the duration rectification method and the duration interpolation method. Both the objective and subjective evaluations reveal that the incorporation of the proposed two methods into the conventional speaker adaptation method is effective in improving the performance of the duration model adaptation.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1E
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• pp.3-6
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• 1999

In digital mobile communication system, the transmission errors affect the quality of output speech seriously. There are many error concealment techniques using a posteriori probability which provides information about any transmitted parameter. They need knowledge about channel transition probability as well as the 1st order Markov transition probability of codec parameters for estimation of transmitted parameters. However, in applications of mobile communication systems, the channel transition probability varies depending on nonstationary channel characteristics. The mismatch of designed channel transition probability of the estimator to actual channel transition probability degrades the performance of the estimator. In this paper, we proposed a new parameter estimator which adapts to the channel characteristics using short time average of maximum a posteriori probabilities(MAPs). The proposed scheme, when applied to the LSP parameter estimation, performed better than the conventional estimator which do not adapt to the channel characteristics.

• Proceedings of the Computational Structural Engineering Institute Conference
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• 2002.10a
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• pp.86-93
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• 2002

A new error estimation method is proposed. This utilizes the variation of energy functional about the mapping function between the global and the master elements. The resultant system of equations is the weak form of the generalized conservation checks. However, This formulation has an important information about the relations between the connected elements. In other words, some relations between the connected elements are obtained and these can be used very usefully to measure it posteriori error. In this paper, the explicit formulations are presented for the 1-dimensional model and the 2-dimensional model problems. Numerical results are provided for first order shear deformation theody of beam model and the plane stress problem.