• Journal of Korean Institute of Industrial Engineers
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• v.25 no.2
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• pp.162-172
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• 1999

For VBR video traffic, Motion-Picture Experts Group(MPEG) coding algorithm was adopted as the standard coding algorithm by International Telecommunication Union(ITU). In this paper, we propose a traffic model of an MPEG coded video traffic in frame level and cell level, and develop an approximate model for evaluating performance of a ATM multiplexer with homogeneous MPEG video sources by considering burst-level variation of aggregated traffics. For homogeneous MPEG video traffics which are frame-synchronized, the performance of the ATM multiplexer is influenced by source correlation at the multiplexing time. When sources are highly correlated, we decompose the aggregated cell streams by the frame-type and model multiplexing process during a frame time as n*D/D/1/K queueing model and suggest an approximate method for obtaining CLP of the ATM multiplexer. In the case that sources are highly correlated, the solution has the meaning of the upper bounds of performance of the ATM multiplexer. For the verification of our model, we compare the solution of our model with simulation resets. As the number of sources increases. The CLP obtained from our model approaches to simulation results, and gives upper bounds of simulation results.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.6
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• pp.1175-1179
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• 2006

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. In this paper, I propose a new method of TSIUVC(Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. As a result, a method by using Least Mean Square was obtained a high quality approximation-synthesis waveform . The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and synthesis.

• Genomics & Informatics
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• v.20 no.4
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• pp.43.1-43.18
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• 2022

Atropa belladonna is a valuable medicinal plant and a commercial source of tropane alkaloids, which are frequently utilized in therapeutic practice. In this study, bioinformatic methodologies were used to examine the pattern of coding sequences and the factors that might influence codon usage bias in the chloroplast genome of Atropa belladonna and other nightshade genomes. The chloroplast engineering being a promising field in modern biotechnology, the characterization of chloroplast genome is very important. The results revealed that the chloroplast genomes of Nicotiana tabacum, Solanum lycopersicum, Capsicum frutescens, Datura stramonium, Lyciumbarbarum, Solanum melongena, and Solanum tuberosum exhibited comparable codon usage patterns. In these chloroplast genomes, we observed a weak codon usage bias. According to the correspondence analysis, the genesis of the codon use bias in these chloroplast genes might be explained by natural selection, directed mutational pressure, and other factors. GC12 and GC3S were shown to have no meaningful relationship. Further research revealed that natural selection primarily shaped the codon usage in A. belladonna and other nightshade genomes for translational efficiency. The sequencing properties of these chloroplast genomes were also investigated by investing the occurrences of palindromes and inverted repeats, which would be useful for future research on medicinal plants.

• The Journal of the Acoustical Society of Korea
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• v.29 no.2
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• pp.102-110
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• 2010

Music transcription is extracting pitch (the height of a musical note) and rhythm (the length of a musical note) information from audio file and making a music score. In this paper, we decomposed a waveform into frequency and rhythm components using Non-Negative Matrix Factorization (NMF) and Non-Negative Sparse coding (NNSC) which are often used for source separation and data clustering. And using the subharmonic summation method, fundamental frequency is calculated from the decomposed frequency components. Therefore, the accurate pitch of each score can be estimated. The proposed method successfully performed music transcription with its results superior to those of the conventional methods which used either NMF or NNSC.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.81-84
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• 2008

최근 센서네트워크와 같은 에너지 제한 환경을 위한 경량화 부호화 기술의 필요성이 대두됨에 따라 분산 소스 부호화 기술(Distributed Source Coding)의 응용기술로 비디오 부호화 복잡도의 대부분을 차지하는 움직임 예측/보상과정을 부호화기가 아닌 복호화기에서 수행하는 분산 비디오 부호화 기술(Distributed Video Coding)에 대한 연구가 활발히 이루어져 왔다. 이에 가장 대표적인 기술인 Wyner-Ziv 코딩 기술은 채널 코드를 이용하여 원본 프레임과 이에 대한 복호화기의 예측영상인 보조정보 사이의 잡음을 제거하여 영상을 복원한다. 일반적으로 보조정보는 원본영상에 유사한 키 프레임간의 프레임 보간을 통하여 생성되며 채널 코드는 Shannon limit에 근접한 성능을 보이는 Turbo 코드나 LDPC 코드가 사용된다. 이와 같은 채널 코드의 복호화는 채널 잡음 모델에 기반하여 수행되어지며 Wyner-Ziv 코딩 기술에서는 이 채널 잡음 모델을 '상관 잡음 모델' (Correlation Noise Modeling)이라 하고 일반적으로 Laplacian이나 Gaussian으로 모델화 한다. 하지만 복호화기에는 원본 영상에 대한 정보가 없기 때문에 정확한 상관 잡음 모델을 알 수 없으며 잡음 모델에 대한 예측의 부정확성은 잡음 제거를 위한 패리티 비트의 증가를 야기해 부호화 기술의 압축 성능 저하를 가져온다. 이에 본 논문은 원본 프레임과 보조정보 사이의 잡음을 정확하게 예측하여 잡음을 정정할 수 있는 향상된 상관 잡음 모델을 제안한다. 제안 방법은 잘못된 잡음 예측에 의해 Laplacian 계수가 너무 커지는 것을 방지하면서 영상내의 잡음의 유무에 별다른 영향을 받지 않는 새로운 문턱값을 사용한다. 다양한 영상에 대한 제안 방법의 실험 결과는 평균적으로 약 0.35dB에 해당하는 율-왜곡 성능 향상을 보여주었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1C
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• pp.72-82
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• 2004

This paper proposes more robust error concealment techniques (ECTs) for MPEG-2 intra coded frame. MPEG-2 source coding algorithm is very sensitive to transmission errors due to the use of variable-length coding. The transmission errors are corrected by error correction scheme, however, they cannot be revised properly. Error concealment (EC) is used to conceal the errors which are not corrected by error correction and to provide minimum visual distortion at the decoder. If errors are generated in intra coded frame, that is the starting frame of GOP, they are propagated to other inter coded frames due to the nature of motion compensated prediction coding. Such propagation of error may cause severe visual distortion. The proposed algorithm in this paper utilizes the spatio-temporal information of neighboring inter coded frames to conceal the successive slices errors occurred in I-frame. The proposed method also overcomes the problems that previous ECTs reside. The proposed algorithm generates consistent performance even in network where the violent transmission errors frequently occur. Algorithm is performed in MPEG-2 video codec and we can confirm that the proposed algorithm provides less visible distortion and higher PSNR than other approaches through simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.12A
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• pp.1156-1164
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• 2008

In this paper, we propose a bi-directional coding method (BTC) using time reversal mirror for effective data exchange between mobile stations and basestations. In conventional relay systems, while one node transmits data, other nodes should keep silent to avoid interference. Thus, to interchange data between basestations and mobile stations, it needs four stages. However, using the proposed BTC method, a relay station transmits a probe source before data transmission, and using this information, mobile stations and basestations transmit data to the relay station simultaneously. Therefore, since it exchanges data in only two steps, the system achieves improved system capacity. In addition, owing to no need for complex calculation, the relay system can be implemented with low complexity. The simulation results demonstrate that the proposed method achieves enhanced system capacity.

• Journal of Information Science Theory and Practice
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• v.10 no.spc
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• pp.56-65
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• 2022

Korea Institute of Science and Technology Information (KISTI) is a Worldwide LHC Computing Grid (WLCG) Tier-1 center mandated to preserve raw data produced from A Large Ion Collider Experiment (ALICE) experiment using the world's largest particle accelerator, the Large Hadron Collider (LHC) at European Organization for Nuclear Research (CERN). Physical medium used widely for long-term data preservation is tape, thanks to its reliability and least price per capacity compared to other media such as optical disk, hard disk, and solid-state disk. However, decreasing numbers of manufacturers for both tape drives and cartridges, and patent disputes among them escalated risk of market. As alternative to tape-based data preservation strategy, we proposed disk-only erasure-coded archival storage system, Custodial Disk Storage (CDS), powered by Exascale Open Storage (EOS), an open-source storage management software developed by CERN. CDS system consists of 18 high density Just-Bunch-Of-Disks (JBOD) enclosures attached to 9 servers through 12 Gbps Serial Attached SCSI (SAS) Host Bus Adapter (HBA) interfaces via multiple paths for redundancy and multiplexing. For data protection, we introduced Reed-Solomon (RS) (16, 4) Erasure Coding (EC) layout, where the number of data and parity blocks are 12 and 4 respectively, which gives the annual data loss probability equivalent to 5×10^-14. In this paper, we discuss CDS system design based on JBOD products, performance limitations, and data protection strategy accommodating EOS EC implementation. We present CDS operations for ALICE experiment and long-term power consumption measurement.

• The Journal of the Acoustical Society of Korea
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• v.38 no.5
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• pp.494-510
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• 2019

Underwater acoustic communication channel parameters consist of impulse response, delay spreading, scattering function, coherence bandwidth, frequency selective fading, coherence time and time variant magnitude fading statistics on which communication system modem and channel coding are designed. These parameters are influenced by sound velocity profile, platform motion and sea surface roughness in given acoustical oceanography condition. In this paper, channel model based on phasor, channel simulator, measurement and analysis method of channel parameters are given in a fixed source-to-receiver system and the parameters are analyzed using shallow water experimental data. For two different source-to-receiver ranges of 300 m and 600 m, the parameters are characterized by three multipaths such as a direct, a surface reflection path with time variant scattering and a bottom reflection path. The results present a channel modelling method of a fixed source source-to-receiver system, channel parameters measurement and analysis methods and a system design and performance assessment method in shallow water.

• 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.