• Korean Journal of Construction Engineering and Management
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• v.21 no.2
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• pp.15-21
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• 2020

The relationship between keywords, found in all productivity related papers published in the KSCE journal for last 15 years, were analyzed in order to reveal a research trend in the area using text mining and A-Priori algorithm. As the results, it is found that the word of 'productivity' is most closely related to the words of 'work' and 'labor'. Futhermore, the word is somewhat related to those of 'factor', 'model', simulation', and 'work time'. It is also revealed that, on the other hand, the words of 'machine' and 'equipment' have little relationships with the keyword. This research will be a great help for academia to understand a research trend in the area of construction productivity.

• The Journal of the Acoustical Society of Korea
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• v.30 no.7
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• pp.383-389
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• 2011

In this paper, we propose a improved speech absence probability estimation algorithm by applying environmental noise classification for speech enhancement. The previous speech absence probability required to seek a priori probability of speech absence was derived by applying microphone input signal and the noise signal based on the estimated value of a posteriori SNR threshold. In this paper, the proposed algorithm estimates the speech absence probability using noise classification algorithm which is based on Gaussian mixture model in order to apply the optimal parameter each noise types, unlike the conventional fixed threshold and smoothing parameter. Performance of the proposed enhancement algorithm is evaluated by ITU-T P.862 PESQ (perceptual evaluation of speech quality) and composite measure under various noise environments. It is verified that the proposed algorithm yields better results compared to the conventional speech absence probability estimation algorithm.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.5
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• pp.157-163
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• 2017

The binary and interpolation search algorithms are the most famous among search area algorithms, the former running in $O(log_2n)$ on average, and the latter in $O(log_2log_2n)$ on average and O(n) at worst. Also, the interpolation search use only the probability of key value location without priori information. This paper proposes another search algorithm, which I term a 'hybrid block and interpolation search'. This algorithm employs the block search, a method by which MSB index of a data is determined as a block, and the interpolation search to find the exact location of the key. The proposed algorithm reduces the search range with priori information and search the reduced range with uninformed situation. Experimental results show that the algorithm has a time complexity of $O(log_2log_2n_i)$, $n_i{\simeq}0.1n$ both on average and at worst through utilization of previously acquired information on the block search. The proposed algorithm has proved to be approximately 10 times faster than the interpolation search on average.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.2
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• pp.275-280
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• 2016

In this paper, we propose a novel relay selection technique which utilizes a priori decoding information at relays for buffer-aided successive relaying networks. In the conventional relaying schemes, a single relay pair is selected for receiving data from the source and transmitting data to the destination. In the proposed technique, however, all relays except the relay selected for transmitting data to the destination try to decode the received signal from the source, and they store the data if they succeed decoding. The proposed technique selects the relay such that it can succeed its own transmission and it maximizes the number of relays successfully decoding the data from the source at the same time. It is shown that the proposed relaying technique significantly outperforms the conventional buffer-aided relaying schemes in terms of outage probability through extensive computer simulations.

• The Journal of the Acoustical Society of Korea
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• v.32 no.3
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• pp.279-285
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• 2013

In this paper, we propose a novel speech enhancement method to improve the performance of the conventional global soft decision which is based on the spectral gradient method applied to the ratio of a priori speech absence and presence probability value (q). Conventional global soft decision scheme used a fixed value of q in accordance with the hypothesis assumed, but the proposed algorithm is a technique for improving the speech absence probability which is applied adaptively variable value of q according to the speech presence or absence in the previous two frames and the conditions of the spectral gradient value. Experimental results show that the proposed improved global soft decision method based on the spectral gradient method yields better results compared to the conventional global soft decision technique based on the performance criteria of the ITU-T P. 862 PESQ (Perceptual Evaluation of Speech Quality).

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.2
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• pp.385-391
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• 2007

In this paper, we propose and observe the Adaptive Modulation system with optimal Turbo Coded V-BLAST (Vertical-Bell-lab Layered Space-Time) technique that is applied the extrinsic information from MAP (Maximum A Posteriori) Decoder with Iterative Decoding to use as a priori probability in two decoding procedures of V-BLAST: ordering and slicing. Also, comparing with the Adaptive Modulation system using conventional Turbo Coded V-BLAST technique that is simply combined V-BLAST with Turbo Coding scheme, we observe how much throughput performance has been improved. As a result of simulation, in the Adaptive Modulation systems with several Turbo Coded V-BLAST techniques, the optimal Turbo Coded V-BLAST technique has higher throughput gain than the conventional Turbo Coded V-BLAST technique. Especially, the results show that the proposed scheme achieves the gain of 1.5 dB SNR compared to the conventional system at 2.5 Mbps throughput.

• The Korean Journal of Air & Space Law and Policy
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• v.29 no.1
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• pp.199-236
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• 2014

This paper analyses from the perspective of distributive justice the legal mechanisms for international allocation of orbital slots, which are of co-owned nature and thereby limited natural resources in outer space. The allocative function is delegated to the International Telecommunication Union. The Radio Regulation, amongst such other legal instruments as the Constitution and Convention, by which the ITU and contracting States thereof abides, dictates how the orbital positions are distributed. Thus, the RR is thoroughly reviewed in the essay. The mechanisms are in a broad sense categorized into two systems: 'a posteriori system' where the 'first come, first served' principle prevails; and 'a priori system' designed to foster the utilisation of the slots by those who lack space resources and are, in especial, likely to be marginalised under the former system. The argument proceeds on the premise that a posteriori system places the under-resourced States in unfavourable positions in the securement of the slots. In contrast with this notion, seven factors were instantiated for an assertion that the degradation of the distributive justice derived from the 'first come, first served' rule, which lays the foundation for the system, could be either mitigated or counterbalanced by the alleged exceptions to the rule. However, the author of this essay argues for counterevidences against the factors and thereby demonstrating that the principle still remains as an overwhelming doctrine, posing a threat to the pursuit of fair allocation. The elements he set forth are as in the following: 1) that the 'first come, first served' principle only applies to assignments capable of causing harmful interferences; 2) the interoperability of the principle with the 'rule of conformity' with the all the ITU instruments; 3) the viability of alternative registrations, as an exception of the application of the principle, on the condition of provisional and informational purposes; 4) another reference that matters in deciding the priority: the types of services in the TFA; 5) the Rule of Procedure H40 proclaiming a ban on taking advantage of coming first to the Register; 6) the technical factors and equity-oriented norms under international and municipal laws along with; 7) the changes of 'basic characteristics' of registered assignments. The second half of this essay illustrates by examining the relevant Annexes to the Regulation that the planned allocation, i.e., a priori system, bear the structured flaws that hinder the fulfillment of the original purpose of the system. The Broadcasting and Fixed Satellite Systems are the reviewed Plans in which the 'first come, first served' principle re-emerges in the end as a determining factor to grant the 'right to international recognition' to administrations including those who has not the allotted portions in the Plan.

• Journal of Intelligence and Information Systems
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• v.10 no.2
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• pp.65-77
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• 2004

This paper presents a priori and the local font classification method. The font classification uses ascenders, descenders, and serifs extracted from a word image. The gradient features of those sub-images are extracted, and used as an input to a neural network classifier to produce font classification results. The font classification determines 2 font styles (upright or slant), 3 font groups (serif sans-serif or typewriter), and 7-font names (Postscript fonts such as Avant Garde, Helvetica, Bookman, New Century Schoolbook, Palatine, Times, and Courier). The proposed a priori and local font classification method allows an OCR system consisting of various font-specific character segmentation tools and various mono-font character recognizers. Experiments have shown font classification accuracies reach high performance levels of about 95.4 percent even with severely touching characters. The technique developed for tile selected 7 fonts in this paper can be applied to any other fonts.

• Geophysics and Geophysical Exploration
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• v.10 no.4
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• pp.314-321
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• 2007

Recently, multi-dimensional joint inversion of geophysical data based on fundamentally different physical properties is being actively studied. Joint inversion can provide a way to obtaining much more accurate image of the subsurface structure. Through the joint inversion, furthermore, it is possible to directly estimate non-geophysical material properties from geophysical measurements. In this study, we developed a new algorithm for jointly inverting dc resistivity and seismic traveltime data based on the multiple constraints: (1) structural similarity based on cross-gradient, (2) correlation between two different material properties, and (3) a priori information on the material property distribution. Through the numerical experiments of surface dc resistivity and seismic refraction surveys, the performance of the proposed algorithm was demonstrated and the effects of different regularizations were analyzed. In particular, we showed that the hidden layer problem in the seismic refraction method due to an inter-bedded low velocity layer can be solved by the joint inversion when appropriate constraints are applied.

• The Journal of the Acoustical Society of Korea
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• v.23 no.7
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• pp.540-547
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• 2004

Since speech enhancement algorithms based on Kalman/Wiener filter require a priori knowledge of the noise and have focused on the minimization of the variance of the estimation error between clean and estimated speech signal, small estimation error on the noise statistics may lead to large estimation error. However, H/sub ∞/ filter does not require any assumptions and a priori knowledge of the noise statistics, but searches the best estimated signal among the entire estimated signal by applying least upper bound, consequently it is more robust to the variation of noise statistics than Kalman/Wiener filter. In this paper, we Propose a speech enhancement method using HMM and multi H/sub ∞/ filters. First, HMM parameters are estimated with the training data. Secondly, speech is filtered with multiple number of H/sub ∞/ filters. Finally, the estimation of clean speech is obtained from the sum of the weighted filtered outputs. Experimental results shows about 1dB∼2dB SNR improvement with a slight increment of computation compared with the Kalman filter method.