• Journal of Broadcast Engineering
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• v.16 no.1
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• pp.14-22
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• 2011

The conventional satellite image fusion methods usually add the same amount of higher frequency components extracted from the panchromatic image to all the multispectral images. However, it is noted that each of multispectral images has different amount of overlap with the panchromatic image in terms of its spectrum, and also has different intensities. Thus giving the same amount of high frequency contents to all the spectral bands does not match with this observation, which causes color distortion in the fused image. In this paper, we propose a new wavelet-domain satellite image fusion algorithm that can compensate for these differences in intensity and spectrum overlap. For the compensation of intensity differences, we first estimate the high resolution multispectral images from P, considering the relative intensity ratios. For the compensation of the amount of spectral overlap, their wavelet coefficients are appended to the conventional wavelet-domain method where the coefficients for the addition is determined by the amount of spectrum overlap. Experiments are conducted for the IKONOS satellite images whose spectrums are well known, and the results show that the proposed algorithm gives higher PSNR and correlation coefficients compared to the conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.717-726
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• 2003

This paper presents the methods that design the Wavelet-based pitch detector for 2,4 kbps Harmonic-CELP Coder, and that achieve the effective waveform interpolation by decision window shape of the transition region, Waveform interpolation coder operates by encoding one pitch-period-sized segment, a prototype segment, of speech for each frame, generate the smooth waveform interpolation between the prototype segments for voiced frame, But, harmonic synthesis of the prototype waveforms between previous frame and current frame occur not only waveform errors but also discontinuity at frame boundary on that case of pitch halving or doubling, In addtion, in transition region since waveform interpolation coder synthesizes the excitation waveform by using overlap-add with triangularity window, therefore, Harmonic-CELP fail to model the instantaneous increasing speech and synthesis waveform linearly increases, First of all, in order to detect the precise pitch period, we use the hybrid 1st pitch detector, and increse the precision by using 2nd ACF-pitch detector, Next, in order to modify excitation window, we detect the onset, offset of frame by GCI, As the result, pitch doubling is removed and pitch error rate is decreased 5.4% in comparison with ACF, and is decreased 2,66% in comparison with wavelet detector, MOS test improve 0.13 at transition region.

• 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.

• 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.

• 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.

• 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.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.31-34
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• 2000

This paper proposes a reconstruction scheme for overlapping characters in Thai printed document. Overlapping characters are characters that overlap with surrounding characters. The problem of overlapping characters is still an unsolved problem In commercially available software of Thai character recognition systems. The algorithm of reconstruction scheme is based on structural analysis of overlapping Thai printed characters. It consists of 2 steps: overlapping point determination and reconstruction of segmented characters. The overlapping point is defined as the intersection point between characters and can be determined by using templates. Then, an overlapping character is separated into segments at the intersection point. The structure of each segment may be an incomplete character and is not identical to the original one. Therefore, the reconstruction process is employed to add the incomplete part of these segments. The proposed scheme has been implemented and tested with 70 patterns of conventionally found in overlapping printed Thai characters with different typefaces and type sizes. The experimental results show that the proposed scheme can segment and reconstruct overlapping characters correctly. The proposed scheme can improve the recognition rate of commercially available software, ThaiOCR1.5 and ArnThai1.0, more than 60 percents

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.427-434
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• 2012

The overlap-add technique based on waveform similarity (WSOLA) method is known to be an efficient high-quality algorithm for time scaling of speech signal. The computational load of WSOLA is concentrated on the repeated normalized cross-correlation (NCC) calculation to evaluate the similarity between two signal waveforms. To reduce the computational complexity of WSOLA, this paper proposes a fast NCC computation method, in which NCC is obtained through pre-calculated sum tables to eliminate redundancy of repeated NCC calculations in the adjacent regions. While the denominator part of NCC has much redundancy irrespective of the time-scale factor, the numerator part of NCC has less redundancy and the amount of redundancy is dependent on both the time-scale factor and optimal shift value, thereby requiring more sophisticated algorithm for fast computation. The simulation results show that the proposed method reduces about 40%, 47% and 52% of the WSOLA execution time for the time-scale compression, 2 and 3 times time-scale expansions, respectively, while maintaining exactly the same speech quality of the conventional WSOLA.

• 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.

• 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.