• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.5
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• pp.2427-2445
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• 2016

In contrast to traditional "store-and-forward" routing mechanisms, network coding offers an elegant solution for achieving maximum network throughput. The core idea is that intermediate network nodes linearly combine received data packets so that the destination nodes can decode original files from some authenticated packets. Although network coding has many advantages, especially in wireless sensor network and peer-to-peer network, the encoding mechanism of intermediate nodes also results in some additional security issues. For a powerful adversary who can control arbitrary number of malicious network nodes and can eavesdrop on the entire network, cryptographic signature schemes provide undeniable authentication mechanisms for network nodes. However, with the development of quantum technologies, some existing network coding signature schemes based on some traditional number-theoretic primitives vulnerable to quantum cryptanalysis. In this paper we first present an efficient network coding signature scheme in the standard model using lattice theory, which can be viewed as the most promising tool for designing post-quantum cryptographic protocols. In the security proof, we propose a new method for generating a random lattice and the corresponding trapdoor, which may be used in other cryptographic protocols. Our scheme has many advantages, such as supporting multi-source networks, low computational complexity and low communication overhead.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.393-398
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• 2002

The conventional bit allocation scheme for MPEG-1 audio encoding searches the subband with minimum MNR(mask-to-noise ratio) repetitively until its operation is completed, which occupies most of its total computational complexity. In this paper, as a computationally efficient approximation of it, we propose a new bit allocation scheme with a simple subband search and compare it with the existing schemes[1][2] in terms of the computational complexity and sound quality. For the performance comparison, we used the pop music signal contained in SQAM(sound quality assess material) CD from EBU. Simulation results show that the computational complexity of the proposed method is about 42% of that of the existing one in [1] and the sound quality difference in terms of MNR between the two schemes is within the 0.2 ㏈, for the case of using the layer II at the bit rate of 128 kbps.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.150-159
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• 2014

The latest video coding standard, HEVC can improve the coding efficiency significantly compared with the H.264/AVC. However the HEVC encoder requires much larger computational complexities. The longer 8-tap interpolation filter of the HEVC which is used in a non-integer motion estimation is one of the reasons and this paper aims to reduce the computational complexities. First of all, three shorter-tap interpolation filters for a motion estimation process are tested rather than the use of a standard interpolation filter. In addition, the fast searching strategies to reduce the number of comparisons for choosing the best non-integer motion vector are proposed. Finally, the interpolation process is selectively applied according to the searching strategy. By combining all of the techniques, the experimental results show that the encoding times can be reduced by 13.6%, 18.5% and 21.1% with the coding efficiency penalties of 0.7%, 1.5% and 2.5%, respectively. For the full-HD video sequences, the coding efficiency penalties are reduced to 0.4%, 1.1% and 1.6% at the same level of the encoding time savings, which shows the effectiveness of the proposed schemes for the high resolution video sequences.

• Journal of KIISE:Information Networking
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• v.36 no.5
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• pp.425-436
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• 2009

In recent years, the user demands have increased for multimedia service of high quality over the broadband convergence network. These rising demands for high quality multimedia service led the popularization of various user terminals and large scale display equipments, which needs a variety type of QoS (Quality of Service). In order to support demands for QoS, numerous research projects are in progress both from the perspective of network as well as end system; For example, at the network perspective, QoS guaranteeing by improving of internet performance such as Active Queue Management, while at the end system perspective, SVC (Scalable Video Coding) encoding scheme to guarantee media quality. However, existing AQM algorithms have problems which do not guarantee QoS, because they did not consider the essential characteristics of video encoding schemes. In this paper, it is proposed to solve this problem by deploying the TS- AQM (Temporal Scalability Active Queue Management) which employs the differentiated packet dropping for dependency of the temporal level among the frames, based on SVC encoding characteristics by exploiting the TID (Temporal ID) field of the SVC NAL unit header. The proposed TS-AQM guarantees multimedia service quality through video decoding reliability for SVC streaming service, by differentiated packet dropping when congestion exists.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.81-90
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• 2010

This paper presents H.264 coding schemes for multi-view video using the concept of layered depth image(LDI) representation and efficient compression technique for LDI. After converting those data to the proposed representation, we encode color, depth, and auxiliary data representing the hierarchical structure, respectively, Two kinds of preprocessing approaches are proposed for multiple color and depth components. In order to compress auxiliary data, we have employed a near lossless coding method. Finally, we have reconstructed the original viewpoints successfully from the decoded approach that is useful for dealing with multiple color and depth data simultaneously.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.5 no.1
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• pp.33-44
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• 2012

A class of recurrent neural networks is proposed and proven to be capable of identifying any discrete-time dynamical system. The applications of the proposed network are addressed in the encoding, identification, and extraction of finite state automata. Simulation results show that the identification of finite state automata using the proposed network, trained by the hybrid greedy simulated annealing with a modified error function in the learning stage, exhibits generally better performance than other conventional identification schemes.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.45 no.9
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• pp.57-64
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• 2008

This paper proposes a new test data compression scheme which has good performance. The proposed scheme is composed of adding transition stage and shifting transition stage using don't care remapping technique. The experimental results show that the new scheme provides higher compression ratio than RL-huffman encoding which is the one of the highest performance schemes, and requires smaller hardware overhead. Therefore it can be widely used as a practical solution for test data compression.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2092-2099
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• 2001

In the three-dimensional image system using stereoscopic images, efficient coding schemes which can get rid of redundancy between the left and right images are usually used. In this paper, we propose an efficient coding method by using relationship between a reference image and residual image. In the proposed algorithm, zero-tree method which guaranty a good quality in low bit rate is used for encoding the residual image. Zero-tree algorithm gives good coding performance, but it has computational complexity so that we used ADLS method to reduce time for the disparity estimation. Using the wavelet based zero-tree method, it is shown that high quality of image in the limited band-width can be preserved through computer simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5A
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• pp.355-362
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• 2009

In real-time communication services, delay constraints are among the most important QoS (Quality of Service) factors. In particular, it is difficult to guarantee the delay requirement over wireless channels, since they exhibit dynamic time-varying behavior and even severe burst-errors during periods of deep fading. Channel throughput may be increased, but at the cost of the additional delays when ARQ (Automatic Repeat Request) schemes are used. For real-time communication services, it is very essential to predict data deliverability. This paper derives the delay distribution and the successful delivery probability within a given delay budget using a priori channel model and a posteriori information from the perspective of queueing theory. The Gilbert-Elliot burst-noise channel is employed as an a Priori channel model, where a two-state Markov-modulated Bernoulli process $(MMBP_2)$ is used. for a posteriori information, the channel parameters, the queue-length and the initial channel state are assumed to be given. The numerical derivation is verified and analyzed via Monte Carlo simulations. This numerical derivation is then applied to a rate control scheme for real-time video transmission, where an optimal encoding rate is determined based on the future channel capacity and the distortion of the reconstructed pictures.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1694-1701
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• 2013

Distributed video coding technique(DVC) is needed for several applications with low bit rate or low encoding resources. However, since conventional DVC schemes have performance limitations and so as one of the several methods to overcome this problem and to improve the coding efficiency, DVC scheme with the moving information of side information has been developed. Based on this system, in this paper, an efficient reliability estimation is introduced to evaluate the blocks composing of side information and send these blocks location to the encoder side. Then, these enable the encoder to selectively encode and to improve the coding performance. Experimental results show that the proposed scheme performs better than the conventional coding scheme.