• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1631-1634
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• 2002

멀티미디어 서버는 클라이언트가 요청한 멀티미디어 데이터 스트림을 효율적으로 제공하기 위해 사용된다. 현재, 정보통신 기술의 발달로 인해 멀티미디어 서버는 멀티미디어 정보를 온라인으로 서비스 할 수 있게 되었다. 하지만 네트워크 상에서 발생하는 Packet Delay 에 때문에 서버에서 사용자에게 연속적이고 엄격한 실시간 제약이 있는 비디오 전송과 같은 멀티미디어 서비스를 제공하는 것은 매우 힘들다. 따라서, 본 논문에서는 서버에서 필요한 Packet 를 먼저 전송하여 Packet Delay 를 줄이는 방법을 제안하였다. MPEG-4 에서 오브젝트의 AU(Access Unit)들의 디코딩 시간을 표시하는 DTS(Decoding Time Stamp)를 참조하여 생성된 Deadline threshold 를 기준으로 Deadline 이 가장 빠른 AU부터 전송하는 스케줄링 알고리즘을 이용하여 MPEC-4 미디어 오브젝트를 스트리밍한다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.9
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• pp.62-70
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• 1995

In some practical applications of the LMS Algorithm the coefficient adaptation can be performed only after some fixed delay. The resulting algorithm is known as the Delayed Least Mean Square (DLMS) algorithm in the literature. There exist analyses for this algorithm, but most of them are based on the unrealistic independence assumption between successive input vectors. Inthis paper we consider the DLMS algorithm with decreasing step size .mu.(n)=n/a, a>0 and prove the almost-sure convergence ofthe weight vector W(n) to the Wiener solution W$_{opt}$ as n .rarw. .inf. under the mixing unput condition and the satisfaction of the law of large numbers. Computer simulations for decision-directed adaptive equalizer with decoding delay are performed to demonstrate the functioning of the proposed algorithm.m.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.696-708
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• 2002

In this paper, a novel transcoding algorithm for the Adaptive Multi Rate(AMR) and the Enhanced Variable Rate Codec(EVRC) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. The proposed algorithm consists of the parameter decoding, frame classification, mode decision, and transcoders for two frame types. The transcoders convert the parameters such as LSP, frame energy, pitch delay for the adaptive codebook, fixed codebook vector, and codebook gains. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent speech quality to that produced by the tandem transcoding algorithm.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8
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• pp.2473-2483
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• 2000

In this paper, The turbo code is used to effectively remove noise which is generated on the image communication channel. Turbo code had excellent decoding performance. However, it had limitations for real time communication because of the system complexity and time delay in decoding procedure. To overcome this problem, this paper proposed a new SRI(Semi Random Interleaved algorithm, which decrease the time delay, when the image data, which reduced the interleaver size of turbo code encoder and decoder, transmitted. The SRI algorithm was composed of 0.5 interleaver size from input frame sequence. When the data inputs in interleaver, the data recorded by row such as block interleaver. But, When the data read in interleaver, the data was read by randomly and the next data located by the just address simultaneously. Therefore, the SRI reduced half-complexity when it was compared with pre-existing method such as block, helical, random interleaver. The image data could be the real time processing when the SRI applied to turbo code.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.3
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• pp.125-131
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• 2002

In this paper, a new concatenation scheme of serial concatenated convolutional codes is proposed and the performance analyzed. In the proposed scheme, each of information and parity bits of outer code is entered into inner code through interleaver and deinterleaver. Therefore, the interleaver size is same as the length of input frame. Since the interleaver size of proposed type is reduced to half of the conventional Benedetto type, the interleaver delay time required for iterative decoding is reduced. In addition the multiplexer and demultiplexer are not used in the decoder of the proposed type, the complexity of decoder can be also reduced. As results of simulation, the performance of proposed type shows the better error performance as compared to that of the conventional Benedetto type in case of the same interleaver size. And it can be observed that the difference of BER performance is increased with the increase of Eb/No. In case of the same length of input frame, the proposed type shows almost same performance with Benedetto type despite that the interleaver size is reduced by half.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.3 s.345
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• pp.103-110
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• 2006

Turbo code, a kind of error correction coding technique, has been used in the field of digital mobile communication system. As the number of iterations increases, it can achieves remarkable BER performance over AWGN channel environment. However, if the number of iterations Is increases in the several channel environments, any further iteration results in very little improvement, and requires much delay and computation in proportion to the number of iterations. To solve this problems, it is necessary to device an efficient criterion to stop the iteration process and prevent unnecessary delay and computation. In this paper, it proposes an efficient and simple criterion for stopping the iteration process in turbo decoding. By using variance values of noise derived from mean values of LLR in turbo decoder, the proposed algorithm can largely reduce the computation and average number of iterations without BER performance degradation. As a result of simulations, the computation of the proposed algorithm is reduced by about $66{\sim}80%$ compared to conventional algorithm. The average number of iterations is reduced by about $13.99%{\sim}15.74%$ compared to CE algorithm and about $17.88%{\sim}18.59%$ compared to SCR algorithm.

• The Journal of the Acoustical Society of Korea
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• v.27 no.8
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• pp.427-434
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• 2008

In this paper, a transcoding algorithm for SMV and AMR speech coder is proposed. In the application requiring the interoperability of different networks, two speech coders must work together with the structure of cascaded connection, tandem. The tandem which is one of the simplest methods has several problems such as long delay, high complexity and the quality degradation due to twice complete encoding/decoding process. These problems can be solved by using transcoding algorithm. The proposed algorithm consists of LSP (Line Spectral Pair) conversion, pitch delay conversion, and fast fixed codebook search. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem with reduced computational complexity and delay.

• Journal of IKEEE
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• v.20 no.4
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• pp.435-438
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• 2016

HEVC CABAC binary arithmetic decoder operates in three decoding modes i.e. regular, bypass, and termination modes, where their decoding operations and time differ a lot. Furthermore, when renormalization occurs, its internal feedback loop induces large delay. In this paper, a binary arithmetic decoder was designed to solve this problem. In advance, it checks all range values with possible renormalization. When renormalization occurs, it immediately updates range value and finishes all calculation in a cycle. When implemented in 0.18 um process technology, its maximum operating frequency and gate counts are 215 MHz and 5,423 gates, respectively.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.61-70
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• 2003

In this paper, a transcoding algorithm for the Selectable Mode Vocoder (SMV) and the G.723.1 speech coder via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding process. The proposed algorithm is composed of four parts: the parameter decoding, line spectral pair (LSP) conversion, pitch period conversion, excitation conversion and rate selection. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem transcoding with reduced computational complexity and delay.

• Journal of Communications and Networks
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• v.16 no.5
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• pp.479-484
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• 2014

In this paper, we consider the construction of cyclic and quasi-cyclic structured q-ary low-density parity-check (LDPC) codes over a designated small field. The construction is performed with a pre-defined sliding-window, which actually executes the regular mapping from original field to the targeted field under certain parameters. Compared to the original codes, the new constructed codes can provide better flexibility in choice of code rate, code length and size of field. The constructed codes over small fields with code length from tenths to hundreds perform well with q-ary sum-product decoding algorithm (QSPA) over the additive white Gaussian noise channel and are comparable to the improved spherepacking bound. These codes may found applications in wireless sensor networks (WSN), where the delay and energy are extremely constrained.