• Journal of the Korea Academia-Industrial cooperation Society
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• v.8 no.3
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• pp.522-527
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• 2007

In this paper, we propose an error control coding scheme for constant-amplitude multicode-biorthogonal modulation. The product code is appropriate to the constant-amplitude multicode-biorthogonal modulation for the bit error rate performance improvement. In the constant-amplitude multicode-biorthogonal modulation, the vertical redundant bits are used for the constant amplitude coding. The proposed product code can be constructed by using the additional horizontal redundant bits. The hardware complexity of the encoder/decoder pair is very low. The simulation results show that the bit error rate performance of the proposed coding scheme is improve

• Proceedings of the KIEE Conference
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• 1999.11c
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• pp.524-526
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• 1999

In the construction of successful fuzzy models and/or controllers for nonlinear systems, the identification of a good fuzzy inference system is an important yet difficult problem, which is traditionally accomplished by a time-consuming trial-and-error process. In this paper, we propose a systematic identification procedure for complex multi-input single-output nonlinear systems with DNA coding method. A DNA coding method is optimization algorithm based on biological DNA as conventional genetic algorithms(GAs) are. The strings in the DNA coding method are variable-length strings, while standard GAs work with a fixed-length coding scheme. the DNA coding method is well suited to learning because it allows a flexible representation of a fuzzy inference system. We also propose a new coding method fur applying the DNA coding method to the identification of fuzzy models. This coding scheme can effectively represent the zero-order Takagi-Sugeno(TS) fuzzy model. To acquire optimal TS fuzzy model with higher accuracy and economical size, we use the DNA coding method to optimize the parameters and the number of fuzzy inference system. In order to demonstrate the superiority and efficiency of the proposed scheme, we finally show its application to a Duffing-forced oscillation system.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.1
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• pp.209-213
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• 2010

In this paper, an efficient predictor for entropy coding is proposed. It adaptively selects one of two prediction errors obtained by MED(median edge detector) or GAP(gradient adaptive prediction). The reduced error is encoded by existing entropy coding method. Experimental results show that the proposed algorithm can compress higher than existing predictive methods.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• Journal of KIISE:Information Networking
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• v.33 no.4
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• pp.333-342
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• 2006

The cdma2000 lxEV - DO mobile communication system provides broadcast and multicast services (BCMCS) to meet an increasing demand from multimedia data services. The servicing of video streams over a BCMCS network must, however, face a challenge from the unreliable and error-prone nature of the radio channel. The BCMCS network uses Reed-Solomon coding integrated with the MAC protocol for error recovery. We analyze this coding technique and show that it is not effective in the case of slowly moving mobiles. To improve the playback quality of an MPEG-4 FGS video stream, we propose the Hybrid error recovery scheme, which combines Reed-Solomon with ARQ, using slots which are saved by reducing the Reed-Solomon coding overhead. The target packets to be retransmitted are prioritized by a utility function to reduce the packet error rate in the application layer within a fixed retransmission budget. This is achieved by considering of the map of the error control block at each mobile node. The proposed Hybrid error recovery scheme also uses the characteristics of MPEG-4 FGS (fine granularity scalability) to improve the video quality even when conditions are adverse: slow-moving nodes and a high error rate in the physical channel.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.7
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• pp.12-21
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• 2002

In this paper, we propose a new hybrid ARQ scheme to consider real-time and non real-time services in a wireless ATM network. Real-time and non-real-time services require different error control schemes according to each service characteristics. Therefore, in the next generation mobile communication environments where these service scenarios should be deployed, hybrid ARQ scheme using RCPC code with variable coding rate becomes one of the most suitable solutions. Because the variable coding rate is applied according to traits of transmitted frame and channel status, hybrid ARQ scheme using RCPC code can expect UEP effect. The UEP scheme does not apply equal error protection level to all information, but does high error protection level to more important information. In Our scheme, UEP of high error protection level is applied to real-time service, and UEP of low error protection and retransmission techniques are applied to non real-time service. We show that the proposed hybrid ARQ scheme improves channel utilization efficiency and yields high error correction behaviors.

• Proceedings of the Korean Reliability Society Conference
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• 2000.04a
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• pp.247-253
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• 2000

본 논문은 정보 여분(Information Redundancy)에서의 에러 검출 코딩(Error Detect Coding) 기법을 이용하여 3-out-of-6 자체 검사기를 설계하고, 주기적인 코드(Frequency Coding) 주입을 통해 고장-안전 제어 논리를 모델링 했다. 고장-안전 제어 논리 모듈과 TMR(Triple Modular Redundancy)의 단일 모듈간에 대해서 신뢰성 병렬 수치 해석을 수행하였고, 이때 고장-안전 제어 논리가 기존의 하드웨어 여분 기법보다 시스템 소모비용과 기능적 오버헤드가 감소되어 기능신뢰성이 증가되는 결과를 얻었다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.8
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• pp.109-118
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• 1994

We proposed new field/frame mode selection quantization and buffer control strategy for interlaced digital HDTV image coding system. The field/frame mode selection is performed based on the mean square error and number of generated bits for each superblock. The quantization factor for each superblock is determined by the characteristics of human visual system and buffer status. The statistical characteristics of the number of generated bits for basis block and the prediction of buffer status are used for buffer control. Simulation results show that the proposed field/frame mode selection and rate buffer control. Simulation results show that the proposed field/frame mode selection and rate buffer control strategy have good subjective image quality and have stable buffer status.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.3
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• pp.23-31
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• 2010

The increasing demands on low-power, and low-complexity video encoder have been motivating extensive research activities on distributed video coding (DVC) in which the encoder compresses frames without utilizing inter-frame statistical correlation. In DVC encoder, contrary to the conventional video encoder, an error control code compresses the video frames by representing the frames in the form of syndrome bits. In the meantime, the DVC decoder generates side information which is modeled as a noisy version of the original video frames, and a decoder of the error-control code corrects the errors in the side information with the syndrome bits. The noisy observation, i.e., the side information can be understood as the output of a virtual channel corresponding to the orignal video frames, and the conditional probability of the virtual channel model is assumed to follow a Laplacian distribution. Thus, performance improvement of DVC systems depends on performances of the error-control code and the optimal reconstruction step in the DVC decoder. In turn, the performances of two constituent blocks are directly related to a better estimation of the parameter of the correlation channel. In this paper, we propose an algorithm to estimate the parameter of the correlation channel and also a low-complexity version of the proposed algorithm. In particular, the proposed algorithm minimizes squared-error of the Laplacian probability distribution and the empirical observations. Finally, we show that the conventional algorithm can be improved by adopting a confidential window. The proposed algorithm results in PSNR gain up to 1.8 dB and 1.1 dB on Mother and Foreman video sequences, respectively.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.