• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.3
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• pp.197-203
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• 2009

Finite field arithmetic is very important in the area of cryptographic applications and coding theory, and it is efficient to use normal bases in hardware implementation. Using the fact that $GF(2^{mk})$ having a type-I optimal normal basis becomes the extension field of $GF(2^m)$, we, in this paper, propose a new serial multiplier which reduce the critical XOR path delay of the best known Reyhani-Masoleh and Hasan's serial multiplier by 25% and the number of XOR gates of Kwon et al.'s multiplier by 2 based on the Reyhani-Masoleh and Hasan's serial multiplier for type-I optimal normal basis.

• 한국HCI학회:학술대회논문집
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• 2006.02a
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• pp.931-936
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• 2006

This paper presents different issues of the real-time compression algorithms without compromising the video quality in the distributed environment. The theme of this research is to manage the critical processing stages (speed, information lost, redundancy, distortion) having better encoded ratio, without the fluctuation of quantization scale by using IP configuration. In this paper, different techniques such as distortion measure with searching method cover the block phenomenon with motion estimation process while passing technique and floating measurement is configured by discrete cosine transform (DCT) to reduce computational complexity which is implemented in this video codec. While delay of bits in encoded buffer side especially in real-time state is being controlled to produce the video with high quality and maintenance a low buffering delay. Our results show the performance accuracy gain with better achievement in all the above processes in an encouraging mode.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.531-542
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• 2002

Novel tandemless transcoding method for AMR and EVRC speech coders is proposed in this paper. In contrast to conventional tandem method, the parameters which is used commonly in speech coder where CELP algorithm is adapted are directly transcoded. The proposed algorithm is composed of LSP transcoding, pitch delay transcoding, gains transcoding and fixed codebook vector transcoding Evaluation results show that the novel algorithm achieves better speech quality than tandem method and reduce computational complexity and delay.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1645-1650
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• 2005

FOPDT(First-Order Plus Dead-Time) and SOPDT(Second-Order Plus Dead-Time) process, which are used as the most useful process in industry, are difficult about process identification because of the long dead-time problem and the model mismatch problem. Thus, the accuracy of process identification is the most important problem in FOPDT and SOPDT process control. In this paper, we proposed the real-coded genetic algorithm for identification of FOPDT and SOPDT processes. The proposed method using real-coding genetic algorithm shows better performance characteristic comparing with the existing an area-based identification method and a directed identification method that use step-test responses. The proposed strategy obtained useful result through a number of simulation examples.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2017.11a
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• pp.107-109
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• 2017

HEVC 표준은 정수 화소로 표현된 신호에 DCT-II 를 기반으로 하는 보간 필터를 사용하여 부화소 신호를 생성한다. 움직임 보상 및 예측의 성능 향상을 위해서 부화소 신호를 생성하는 방법을 이용한다. HEVC 표준은 부화소를 각각 1/4-화소 단위로 생성을 하며, 부화소를 생성하기 위해서 길이가 다른 각각의 DCT 보간 필터를 사용하고 있다. 1/2-화소를 생성하는 경우에는 필터의 길이가 8 인 DCT 기반 보간 필터를 사용하며, 1/4-화소와 3/4-화소의 경우에는 필터의 길이가 7 인 DCT 기반 보간 필터를 사용한다. 본 논문에서는 DST-VII을 기반으로 하는 보간 필터를 제안하여, CTU-Level 단위로 RDO 과정을 통해 최적화된 보간 필터를 선택하여 사용한다. 본 논문에서 제안하는 방법은 HEVC 표준보다 BD-rate 가 Low Delay B 와 Random Access configurations 에서 각각 0.6%와 0.6%의 성능 향상을 가져오며 Low Delay P configuration 에서 0.5%의 성능감소를 보인다.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.8 no.4
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• pp.207-215
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• 2009

In recent digital communication systems, the performance of Turbo Code used as the error correction coding method depends on the interleaver size influencing the free distance determination and iterative decoding algorithms of the turbo decoder. However, some iterations are needed to get a better performance, but these processes require large time delay. Recently, methods of reducing the number of iteration have been studied without degrading original performance. In this paper, the new method combining ME (Mean Estimate) stopping criterion with SDR (sign difference ratio) stopping criterion of previous stopping criteria is proposed, and the fact of compensating each method's missed detection is verified Faster decoding realizes that reducing the number of iterative decoding about 1~2 times by adopting our proposed method into serially concatenation of both decoder. System Environments were assumed DS-CDMA forward link system with intense MAI (multiple access interference).

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.257-258
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• 2007

In real-time video communications, the large fluctuation of bits generated from contiguous frames may lead to the buffer oveflow and the delay of transmitting video data. This paper proposes a method that the bits generated from each frame may be in the range of specified fluctuation. The method uses a scene change detector and ignores the time information of detected frames. The experimental results indicate that our scheme achieves average PSNR gain of 2.0dB over TMN8.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.510-517
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• 2012

This paper analyzes the performance of voice-over-Internet protocol (VoIP) services in terms of the system throughput, the packet delay, and the signaling overhead in a mobile WiMAX system with a hybrid automatic repeat request (HARQ) mechanism. Furthermore, a queueing analytical model is developed with due consideration of adaptive modulation and coding, the signaling overhead, and the retransmissions of erroneous packets. The arrival process is modeled as the sum of the arrival rate at the initial transmission queue and the retransmission queue, respectively. The service rate is calculated by taking the HARQ retransmissions into consideration. This paper also evaluates the performance of VoIP services in a mobile WiMAX system with and without persistent allocation; persistent allocation is a technique used to reduce the signaling overhead for connections with a periodic traffic pattern and a relatively fixed payload. As shown in the simulation results, the HARQ mechanism increases the system throughput as well as the signaling overhead and the packet delay.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.213-216
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• 2000

In this paper, performance of a CDMA-based packet transmission system is analyzed and simulated in a wireless infrared channel. The indoor wireless infrared channel is modeled as a non-directed diffuse link. The pulse position modulation (PPM) is used as a modulation scheme. From the simulation results, it is shown that the RS coding is very effective in improving the packet-delay characteristics of the CDMA-based packet transmission system. The performance improvement is more significant especially when the infrared channel is in a worse condition.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.979-982
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• 2000

In this paper we suggest the adaptive QoS-based VOD system design, which can diagnose and reflect the variable network situation. This system is based on the changeable Forward Error Correction(FEC) coding and re-transmission to enhance system quality by using the measured real-time network value such as loss, delay and jitter.