• The Journal of the Acoustical Society of Korea
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• v.15 no.1E
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• pp.89-94
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• 1996

This paper deals with the application of the LMS algorithm as a decision-directed adaptive equalizer in a communication receiver which also employs a sophisticated decoding scheme such as the Viterbi algorithm, in which the desired signal, hence the error, is not available until several symbol intervals later because of decoding delay. In such applications the implemented weight updating algorithm becomes DLMS and major penalty is reduced convergence speed. Therefore, every effort should by made to keep the delay as small as possible if it is not avoidable. In this paper we present a modified implementation in which the effects of the decoding delay can be avioded and perform some computer simulations to check the validity and the performance of the new implementation.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.14 no.8
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• pp.816-822
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• 2003

In this paper, we propose an efficient iteration control method with low complexity for Turbo TCM(Turbo Trellis Coded Modulation) decoding which will be used fur power-limited environment. As the decoding approaches the performance limit of a given turbo code, any further iteration results in very little improvement. Therefore, it is important to devise an efficient criterion to stop the iteration process and prevent unnecessary computations and decoding delay. This paper presents an efficient algorithm for turbo TCM decoding that can greatly reduce the delay and iteration number. The proposed method use adaptive iteration number according to the criterion using the extrinsic information variance parameter in turbo TCM decoding process. The simulation results show that the proposed technique effectively can reduce the decoding delay and computation with very little performance degradation.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.39-42
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• 2004

Turbo codes, proposed by Berrou, that increase the interleaver size and the number of iteration have better than conventional convolutional codes, in case of BER performance. However, because turbo codes has been required much decoding delay to increase iteration number, it demands unnecessary iterative decoding. Therefore, in this paper, we propose iterative decoding stop criterion that uses the variance of absolute value of LLR. This algorithm can be reduced average iterative decoding number and had lossless performance of BER, because of decreasing unnecessary iterative decoding.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.12
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• pp.2376-2385
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• 1994

In receiving the digital TV signal, both decoding delay and the channel hopping delay are very critical factors in applications. The intra-slice coding in the MPEG-2 SIMPLE PROFLE of No B-picture is one of the primary methods for short delay time in video decoding. It has the advantage of short decoding delay, but has the drawback of long channel hopping delay time. In this paper, we propose a method to reduce the channel delay with negligible loss in SNR performance. It is shown that dividing pictures into several regions of slices and adding some restriction in motion vector search for inter-frame coding. hence the random acess points are effectively increased. and the channel hopping delay is reduced.

• IE interfaces
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• v.4 no.1
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• pp.63-69
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• 1991

In this paper, the Go-Back-N ARQ protocol with decoding in communication network is considered. The time delay and throughput are respectively analyzed as a function of window size and decoding time out. Packets arrive continuously at the decoder, and are stored in a buffer if the decoder is busy upon its arrival. The decoder devotes no more than a time-out period of predetermined length to the decoding of any single packet. If packet decoding is completed within that period, the packet leaves the system. Otherwise, it is retransimitted and its decoding starts anew. The time delay and throughput are obtained using recursive formula and difference equation. An appropriate time out and window size that satisfies the grade of service can be determined.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.30A no.3
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• pp.16-33
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• 1993

New decoding algorithm of double-error-correction Reed-Solmon codes over GF(2$^{8}$) for optical compact disks is proposed and decoding algorithm of RS codes with triple-error-correcting capability is presented in this paper. First of all. efficient algorithms for estimating the number of errors in the received code words are presented. The most complex circuits in the RS decoder are parts for soving the error-location numbers from error-location polynomial, so the complexity of those circuits has a great influence on overall decoder complexity. One of the most known algorithm for searching the error-location number is Chien's method, in which all the elements of GF(2$^{m}$) are substituted into the error-location polynomial and the error-location number can be found as the elements satisfying the error-location polynomial. But Chien's scheme needs another 1 frame delay in the decoder, which reduces decoding speed as well as require more stroage circuits for the received ocode symbols. The ther is Polkinghorn method, in which the roots can be resolved directly by solving the error-location polynomial. Bur this method needs additional ROM (readonly memory) for storing tthe roots of the all possible coefficients of error-location polynomial or much more complex cicuit. Simple, efficient, and high speed method for solving the error-location number and decoding algorithm of double-error correction RS codes which reudce considerably the complexity of decoder are proposed by using Hilbert theorems in this paper. And the performance of the proposed decoding algorithm is compared with that of conventional decoding algorithms. As a result of comparison, the proposed decoding algorithm is superior to the conventional decoding algorithm with respect to decoding delay and decoder complexity. And decoding algorithm of RS codes with triple-error-correcting capability is presented, which is suitable for error-correction in digital audio tape, also.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1427-1436
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• 1997

Turbo codes, decoded through an iterative decoding algorithm, habe recently been shown to yidel remarkable coding gains close to theoretical limits in the Gaussian channel environment. This thesis presents the performance of Turbo code through the computer simulation. The performance of modified Turbo code is compared to that of the conventional Turbo codes. The modified Turbo code reduces the time delay in decoding with minimal effect to the performance for voice transmission sytems. To achieve the same performance, random interleaver the size of which is no less than the square root of the original one should be used. Also, the modified Turbo code is applied to MC-CDMA system, and its performance is analyzed under the Rayleigh Fading channel environment. In Rayleigh fading channel environment, due to the amplitude distortion caused by fading, the interleaver of the size twice no less than that in the Gaussian channel enironment was required. In overall, the modified Turbo code maintained the performance of the conventional Turbo code while the time delay in transmission and decoding was reduced at the rate of multiples of two times the squared root of the interleaver size.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.3
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• pp.1144-1158
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• 2016

In this paper, delay-constrained data transmission is considered over error-prone networks. Network coding is deployed for efficient information exchange, and an approximate decoding approach is deployed to overcome potential all-or-nothing problems. Our focus is on determining the cluster size and its impact on approximate decoding performance. Decoding performance is quantified, and we show that performance is determined only by the number of packets. Moreover, the fundamental tradeoff between approximate decoding performance and data transfer rate improvement is analyzed; as the cluster size increases, the data transfer rate improves and decoding performance is degraded. This tradeoff can lead to an optimal cluster size of network coding-based networks that achieves the target decoding performance of applications. A set of experiment results confirms the analysis.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.2
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• pp.62-67
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• 2001

In the recent mobile communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that need a large time delay over the reception process. Moreover, Turbo Code has been known as the robust ending method with the confidence over the fading channel. The International Telecommunication Union(ITU) has recently adopted as the standardization of the channel coding over the third generation mobile communications such as IMT-2000. Therefore, in this paper, we proposed of the method to improve the conventional performance with the parallel concatenated 4-New Turbo Decoder using MAP a1gorithm in spite of complexity increasement. In the real-time video and video service over the third generation mobile communications, the performance of the proposed method was analyzed by the reduced decoding delay using the variable decoding method by computer simulation over AWGN and fading channels.

• 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).