• Korea Journal of Hospital Management
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• v.9 no.1
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• pp.1-21
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• 2004

The United States has a unique health care system, which is unlikely any other health care systems in the world. The major part of basic functional components of the system -financing, insurance, delivery, and payment- is in private hands. A market-oriented economy invites the participation of numerous private entities that are interested in carrying out the key functions of health systems. Due to this central feature, U.S.health care is not delivered through a network of interrelated components designed to work together coherently. For lack of standardization, the various components of the system fit together only loosely. The involvement of numerous players in the key functions leads to duplication, overlap, inadequacy, inconsistency, and waste, which add to the complexity and also make the system inefficient. Hence, cost containment remains an elusive goals. Moreover, the system falls short of delivering equitable services to all americans, though consumption of health care services is the largest in the world. On the other hand, United States leads the world in the latest and the best in medical technology, medical training, and research. It offers some of the most sophisticated institutions, products, and processes of health care delivery. This article discuss the characteristic features of the U.S. health care system. and its performance, trying to seek its implication on Korean health care system.

• 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.

• Journal of Advanced Navigation Technology
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• v.15 no.6
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• pp.1042-1046
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• 2011

In this paper, we analyze the STO (Symbol Timing Offset) problem in conventional ZP-OFDM (Zero Padding-Orthogonal Frequency Division Multiplexing) systems and propose a robust ZP-OFDM system with bi-directional overlap-add scheme to overcome the problem. The proposed ZP-OFDM system is able to preserve the orthogonality between subcarriers and reduce the interference from other ZP-OFDM symbols due to the BOA scheme, which exploits both ZP intervals of the previous and the current ZP-OFDM symbols, even though serious STOs result from inaccurate symbol timing synchronization. Simulation results verify that the proposed ZP-OFDM system is superior to the conventional ZP-OFDM system.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.43 no.7
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• pp.641-647
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• 2015

In this paper, Analyzed the multi-channel signal processing algorithms for digital-transponder. To analyze suitable multi-channel signal processing algorithms, compare algorithms about four criteria. Four criteria are as follows, perfect reconstruction, interference rejection, resource usage and power consumption. Analysis for each algorithm in accordance with these four criteria. then propose the multi-channel signal processing algorithms for digital satellite communication system.

• Language and Information
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• v.7 no.2
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• pp.31-58
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• 2003

The main goal of this paper is to describe and analyse intensifiers, especially non-head-bound-intensifiers (NHBIs), which can be included in the discussion and analysis of these elements as focus particles. In doing so, NHBIs such as Korean susulo, casin/cache, English x-self and German selbst are dealt with in a rather cross-linguistical perspective. The pure and strict comparison between Korean, English and German is not intended. This paper is mainly concerned with the semantic domain where the respective contributions of the expressions in question overlap, which offers the common base for the discussion regarding Korean, one of the non-European languages and English and German, two European languages. They share the semantic domain ‘intensification’ regarding relevant subject-NP. They introduce an ordering distinguishing center and periphery. In contrast to head-bound-intensifiers (HBIs), however, NHBIs add self-involvement (directness of involvement) of subject-NP to the meaning of the relevant sentence. I adopt the proposals of Konig (1991), Primus (1992) and Siemund (2000) in the treatment of intensifiers as focus particles. However, I reject Konig (1991) that just NHBIs talre scope over a whole clause, Primus (1992) that NHBIs focus VPs, not NPs, and Siemund (2000) that NHBIs can be further devided into two groups, viz. NHBIs with exclusive readings and NHBIs with inclusive readings. Evidence for my position is presented mainly in the course of describing and analysing some syntactic properties and the meaning and use of NHBIs. I come to the conclusion that both the common meaning of intensifiers as focus particles and the common meaning of NHBIs of three languages can be represented by a simple logical formalism.

• Journal of IKEEE
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• v.3 no.1 s.4
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• pp.141-149
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• 1999

If signal time duration of MC-CDMA method which has studied for next-generation high-speed data transmission is not sufficiently large compared to delay spread of channel, the performance is degraded by generation of intersymbol interference. In this paper, this problem will solve through serial to parallel convertor and make large sufficiently time duration of signal compared to delay spread of channel and rise variable spectral efficiency through the number of serial to parallel convertor subchannel we will add to parallel frequency diversity block for improve the performance in mobile Communication. Spectral efficiency of the proposed system is counted and compared to spectral efficiency of MC-CDMA and investigated through computer simulations by using multipath Rayleigh fading channel.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.6
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• pp.1250-1256
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• 2004

The movement of continuously changing positions should be stored and indexed for querying current and past positions. A simple extension of the original R-tree to add time as another dimension, called 3D R-tree, does not handle current position queries and does not address the problem of low space utilization due to high overlap of index nodes. In this paper, 1 propose the dynamic splitting policy for improving the 3D R-tree in order to improve space utilization of split nodes. 1 also extend the original 3D R-tree by introducing a new tagged index structure for being able to query the current and past positions of moving objects. 1 found out that my extension of the original R-tree, called the tagged dynamic 3DR-tree, outperforms both the 3D R-tree and 75-tree when querying current and past position.

• Journal of Satellite, Information and Communications
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• v.5 no.1
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• pp.32-37
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• 2010

The algorithm of multiple channel signal processing requires the flexibility of variable frequency band, efficient allocation of transmission power, and flexible frequency band reallocation to satisfy various service types which requires different transmission rates and frequency band. There are three methods including per-channel approach, multiple tree approach, and block approach performing frequency band reallocation method by channelization and dechannelization in the multiple-channel signal. This paper proposes an improved per-channel approach for converting the frequency band of multiple carrier signals efficiently. The proposed algorithm performs decimation and interpolation using CIC(cascaded integrator comb filter), half-band filter, and FIR filter. In addition, it performs filtering of each sub-channel, and reallocates channel band through FIR low-pass filter in the multiple-channel signal. The computer simulation result shows that the perfect reconstruction of output signal and the flexible frequency band reallocation is performed efficiently by the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1
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• pp.1-9
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• 2009

This paper deals with the realization of multi-channel live karaoke. In this study, 6-channel MP3 decoding and tempo/key scaling was operated in real time by using the TMS320C6713 DSP, which is 32 bit floating-point DSP made by TI Co. The 6 channel consists of front L/R instrument, rear L/R instrument, melody, and woofer. In case of the 4 channel, rear L/R instrument can be replaced with drum L/R channel. And the final output data is generated as adjusted to a 5.1 channel speaker. The SOLA algorithm was applied for tempo scaling, and key scaling was done with interpolation and decimation in the time domain. Drum channel was excluded in key scaling by separating instruments into drums and non-drums, and in processing SOLA, high-quality tempo scaling was made possible by differentiating SOLA frame size, which was optimized for real-time process. The use of 6 channels allows the composition of various channels, and the multi-channel audio system of this study can be effectively applied at any place where live music is needed.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.293-303
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• 2002

This paper implements optimized Time Scale Modification (TSM) system using ITU G.729 speech decoder and Dual SOLA algorithm. The proposed system assume 8 Kz sampling rate, 80 samples/frame input speech from the ITU G.729 speech Decoder and the TSM (Time Scale Modification) feature of Dual SOLA produces the high quality output speech that was slow-down or speed up as a user's choice. Especially, the proposed Optimized Dual SOLA base on various simulations and theoretical analysis, and the additional interpolation procedure of the speech makes it possible to setup high performance integrated TSM system at the maximum time scale modification rate. The system performance is analyzed and verified with various input speech and playback speed.