• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.1920-1923
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• 2003

This paper presents compression rate improvement for SPIHT algorithm though redundancy bit removing. Proposed SPIHT algorithm uses a method to select of optimized threshold from feature of wavelet transform coefficients and removes sign bit if coefficient of LL area. Experimental results show that the proposed algorithm achieves more improvement bit rate and more fast progressive transmission with low bit rate.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1989년도 하계종합학술대회 논문집
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• pp.457-460
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• 1989

In this paper, an improvement of the EBP(error back propagation) learning speed using an acceleration algorithm is described. Using an acceleration algorithm known as the Partan method in the gradient search algorithm, learning speed is 25% faster than the original EBP algorithm in the simulaion results.

• 한국통신학회논문지
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• 제25권6B호
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• pp.1050-1059
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• 2000

In this paper, we propose a method to improve JBIG2 compression ratio which can be applied to Hangul text. Hangul character is composed of a few symbols which is called JASO, which needs inevitable increase of position information to be transmitted. To reduce this disadvantage, we have proposed an algorithm that generate aggregated symbol in combination of JASO symbols. Proposed algorithm shows better performance in Huffman coding than in arithmetic coding. In lossless coding, proposed algorithm showed 4.5∼16.7(%) improvement for Huffman coding and 2.9∼10.4(%) improvement for arithmetic coding. In lossy coding, proposed algorithm showed 3.7∼17.0(%) improvement for Huffman coding and 2.1∼10.5(%) improvement for arithmetic coding.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2010년도 하계학술대회
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• pp.74-76
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• 2010

Digital images or videos are used in modern digital devices. The resolution of HDTV in digital broadcasting system is higher than that of previous analog systems. Also, mobile phone with 3G can provide images as well as video streaming services in realtime. In these circumstances, the visual quality of images has become an important factor. We can make image clear by transient improvement process that reduces transient in edges. In this paper, we present an transient improvement algorithm. The proposed algorithm improves edges by making smooth edge to steep edge. Before performing transient improvement algorithm, edge detection algorithm should be operated. Laplacian operator is used in edge detection, and the absolute value of it is used to calculate gain value. Then, local maximum and minimum values are computed to discriminate current pixel value to raise up or pull down. Compensating value that gain value multiplies with the difference between maximum (or minimum) value and current pixel value adds (or subtracts) to current pixel value. That is, improved signal is generated by making the narrow transient of edge. The advantage of proposed algorithm is that it doesn't produce shooting problem like overshoot or undershoot.

• Journal of information and communication convergence engineering
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• 제2권3호
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• pp.202-204
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• 2004

An improvement of tracking capacity to find a position of the Iris images is presented in this paper. The propose algorithm is used the Four Points Diagonal Positioning algorithm that the image is positioned with arbitrary 4 points on the edge of iris and the selective 4 points are drawn by a diagonal line on the cross. The experiment result shows that the algorithm is efficient to track on the eyelid.

• 정보처리학회논문지A
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• 제8A권2호
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• pp.133-146
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• 2001

In a dynamic load distribution policies, each node gathers the current system sates information before making a decision on load balancing. Load balancing policies based on this strategy can suffer from processor thrashing. In this paper, we propose a new algorithm which attempts to decrease the frequency of the processor thrashing, the algorithm is based on the integration of three components. The first, the algorithm of which determine the size of jobs be transferred. The second, negotiation protocol with obtains a mutual agreement between a sender and a receiver on the transferring job size. And the third, a symmetrically-initiated location policy. The algorithm proposed in this paper used Siman IV as simulation tool to prove the improvement of performance. I analyzed the result of simulation, and compared with related works. The mean response time shows that there are no difference with existing policy, but appear a outstanding improvement in high load. The thrashing coefficient that shows the average response time, CPU overhead and the thrashing ratio at both the receiving and sending node has been used in the analysis. A significant improvement in the average response time and the CPU overhead ratio was detected using our algorithm when an overhead occurred in the system over other algorithm. The thrashing coefficient differed in the sending node and the receiving node of the system. Using our algorithm, the thrashing coefficient at the sending node showed more improvement when there was an overhead in the system, proving to be more useful. Therefore, it can be concluded that the thrashing ratio can be reduce by properly setting the maximum and minimum value of the system’s threshold queue.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 하계종합학술대회 논문집(2)
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• pp.553-556
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• 2004

In this paper, we present a new efficient multi-level hardware/software partitioning algorithm for system-on-a-chip design. Originally the multi-level partitioning algorithm are proposed to enhance the performance of previous iterative improvement partitioning algorithm for large scale circuits. But when designing very complex and heterogeneous SoCs, the HW/SW partitioning decision needs to be made prior to refining the system description. In this paper, we present a new method, based on multi-level algorithm, which can cover SoC design. The different variants of algorithm are evaluated by a randomly generated test graph. The experimental results on test graphs show improvement average $9.85\%$ and $8.51\%$ in total communication costs over FM and CLIP respectively.

• 전자공학회논문지A
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• 제33A권12호
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• pp.1-8
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• 1996

In this paper, we propose a hybrid blind equalizer with TEA and SG (stop & Go) algorithm with switching point a 0 dB of MSE value for improvement of convergence performance, where TEA is used initially to open the eye and then SG algorithm as rapid convergence is employed. The switching point is selected at the point of 0 dB MSE level because of settling the coefficients of blind equalier. As a result of computer simulatons for 8-PAM in the non-minimum phase channel, the proposed algorithm has better convergence speed as 3,500 ~ 4,500 iterations and has better MsE about 3 ~ 6 dB than those of original TEA. Also, computational cost of proposed algorithm is reduced as 5 ~ 16% than that of original TEA. and, the proposed algorithm has better convergence than SG algorithm as 8,500 ~ 17,500 iteratins but, the MSE is similar to original SG.

• 대한전기학회논문지:전력기술부문A
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• 제48권11호
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• pp.1394-1400
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• 1999

This paper presents and efficient improvement of the iterative eigenvalue calculation method and the selection of initial values in AESOPS algorithm. To determine the initial eigenvalues of the system, system state matrix is constructed with the two-axis generator model. From the submatrices including synchronous and damping coefficients, the initial eigenvalues are calculated by the QR method. Participation factors are also calculated from the above submatrices in order to determine the generators which have a important effect to the specific oscillation mode. Also, the heuristically approximated eigenvalue calculation method in the AESOPS algorithm is transformed to the Newton Raphson Method which is largely used in the nonlinear numerical analysis. The new methods are developed from the AESOPS algorithm and thus only a few calculation steps are added to practice the proposed algorithm.

• 대한의용생체공학회:의공학회지
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• 제34권1호
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.