• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.12C
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• pp.997-1005
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• 2008

In this paper, we proposed advanced T-DMB system which is based on Eureka-147 using UEP coding methods and hiererchical modulation for providing additional services while maintaining BER performance. We simulated the proposed advanced T-DMB system using unequal l6QAM modulation scheme combined with various bit split methods and coding methods such as double binary turbo code of DVB-RCS standard and LDPC code of DVB-S2 standard. In the simulation results, double binary turbo code and LDPC code of proposed advanced T-DMB system have coding gain of 2dB and 3.5dB compared to conventional T-DMB system respectively.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.10
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• pp.40-47
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• 1996

A method for computing effectiv ebandwidth of aggregated varable bit rate (VBR) moving picture experts group (MPEG) video traffic is proposed. To compute statistical characteristics of aggregated MPEG traffic, first we split input MPEG traffic into I, B, and P frame traffics and aggregate respective I, B, and P frame traffics according to the frame type. Second statisticsal characteristics of the aggregated MPEG traffic are obtained using those of aggregated I, B, and P frame traffics. The effective bandwidth of the aggregated I frame traffic is computed by using the gaussian bound. Using the modified equivalent capacity, we obtain the effective bandwidths of aggregated B and P frame traffics and then compute the effective bandwidth of the combined B and P frame traffic. Finally the effective bandwidth of the aggregated MPEG traffic is computed by adding the gaussian bound of the aggregated I frame traffic and modifed equivalent capacity of combined B and P frame traffic. Computer simulation shows that the proposed method estimates effective bandwidth of the aggregated MPEG traffic well.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.12
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• pp.3033-3045
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• 1996

In this paper, we propse an object-oriented coding method in low bit-rate channels using pyramid structure and residual image compensation. In the motion estimation step, global motion is estimated using a set of multiresolution images constructed in a pyramid structure. We split an input image into two regions based on the gradient value. Regions with larte motions obtain observation points at low resolution level to guarantee robustness to noise and to satisfy a motion constraint equation whereas regions with local motions such as eye, and lips get observation points at the original resolution level. Local motion variations and intesity variations of an image reconstructed by the golbal motion are compensated additionally by using the previous residual image component. Finally, the model failure (MF) region is compensated by the pyramid mapping of the previous displaced frame difference (DFD). Computer simulation results show that the proposed method gives better performance that the convnetional one in terms of the peak signal to noise ratio (PSNR), compression ratio (CR), and computational complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.9
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• pp.927-938
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• 1992

In 1-D subband coding system, a quadrature mirror filter (QMF) pair can be used to split a signal into two subbands and to reconstruct the original signal. In this paper, a pair of 1-D reconstruction filter for the subband coding system is introduced and a coding technique with classified vector quantization, based on energy distributions, for 16 subband images is presented. As computer simulation results show that the method can give similar perceptual quality with bit rate reduced by $20{\sim}30%$ of that of ordinary vector quantization.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.248-254
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• 2005

In this Paper, the structure of a split bandwidth wideband speech coder and its highband coder for tone qualify elevation are Proposed. The lowband and highband by the split bandwidth method are encoded independently applying the G.729E and MLT (Modulated Lapped Transform) residual model. In the highband structure which is encoded by low bit rate of 4kbps, the MLT residual signals are distinguished to voice and unvoice signal . The voice signals are applied to MLT peak picking method by lowband pitch period. Because transformed MLT residual signals are represented by periodic signal that have periodic peak. The unvoice signals are applied to MLT which linear prediction spectral response is added and do vector quantization. Performance for proposed 15.8kbps wideband speech coder was verified through subjective listening test.

• Journal of Broadcast Engineering
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• v.9 no.1
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• pp.17-31
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• 2004

This paper proposes an algorithm for the robust transmission of video in error Prone environment using multiple description codingby optimal split of DCT coefficients and rate-distortionoptimization framework. In MDC, a source signal is split Into several coded streams, which is called descriptions, and each description is transmitted to the decoder through different channel. Between descriptions, structured correlations are introduced at the encoder, and the decoder exploits this correlation to reconstruct the original signal even if some descriptions are missing. It has been shown that the MDC is more resilient than the singe description coding(SDC) against severe packet loss ratecondition. But the excessive redundancy in MDC, i.e., the correlation between the descriptions, degrades the RD performance under low PLR condition. To overcome this Problem of MDC, we propose a hybrid MDC method that controls the SDC/MDC switching according to channel condition. For example, the SDC is used for coding efficiency at low PLR condition and the MDC is used for the error resilience at high PLR condition. To control the SDC/MDC switching in the optimal way, RD optimization framework are used. Lagrange optimization technique minimizes the RD-based cost function, D+M, where R is the actually coded bit rate and D is the estimated distortion. The recursive optimal pet-pixel estimatetechnique is adopted to estimate accurate the decoder distortion. Experimental results show that the proposed optimal split of DCT coefficients and SD/MD switching algorithm is more effective than the conventional MU algorithms in low PLR conditions as well as In high PLR condition.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.3
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• pp.235-240
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• 2018

FTN method which transmits signals faster the Nyquist rate is representative method for improving throughput efficiency sacrificed performance due to inter-symbol interference. To compensate performance loss, in this paper, we propose a multiband FTN method which split the coded bits into several bands and transmits signals applying FTN method. As coded bits are being assigned different bands, number of samples per bit of each band is increased, it induced performance improvement by noise averaging effect. In the simulations, compared the performance of single band FTN method and multiband FTN method when the interference rate is 25%. The results of simulation show the performance of proposed method is better than that of single band FTN one by 0.3dB~0.5dB.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.15 no.7
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• pp.2610-2630
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• 2021

The aim of this paper is to propose the alternative algorithm to finish the process in public key cryptography. In general, the proposed method can be selected to finish both of modular exponentiation and point multiplication. Although this method is not the best method in all cases, it may be the most efficient method when the condition responds well to this approach. Assuming that the binary system of the exponent or the multiplier is considered and it is divided into groups, the binary system is in excellent condition when the number of groups is small. Each group is generated from a number of 0 that is adjacent to each other. The main idea behind the proposed method is to convert the exponent or the multiplier as the subtraction between two integers. For these integers, it is impossible that the bit which is equal to 1 will be assigned in the same position. The experiment is split into two sections. The first section is an experiment to examine the modular exponentiation. The results demonstrate that the cost of completing the modular multiplication is decreased if the number of groups is very small. In tables 7 - 9, four modular multiplications are required when there is one group, although number of bits which are equal to 0 in each table is different. The second component is the experiment to examine the point multiplication process in Elliptic Curves Cryptography. The findings demonstrate that if the number of groups is small, the costs to compute point additions are low. In tables 10 - 12, assigning one group is appeared, number of point addition is one when the multiplier of a point is an even number. However, three-point additions are required when the multiplier is an odd number. As a result, the proposed method is an alternative way that should be used when the number of groups is minimal in order to save the costs.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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• pp.52-60
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• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.

• The Korean Journal of Applied Statistics
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• v.16 no.1
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• pp.45-53
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• 2003

One of decision tree method is regression trees which are used to predict a continuous response. The general splitting criteria in tree growing are based on a compromise in the impurity between the left and the right child node. By picking or the more interesting subsets and ignoring the other, the proposed new splitting criteria in this paper do not split based on a compromise of child nodes anymore. The tree structure by the new criteria might be unbalanced but plausible. It can find a interesting subset as early as possible and express it by a simple clause. As a result, it is very interpretable by sacrificing a little bit of accuracy.