• Journal of the Institute of Convergence Signal Processing
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• v.3 no.4
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• pp.67-73
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• 2002

There have been increased concerns on mobile stratospheric communication system(SCS) for the purpose of advanced service of personal and high speed communication systems. In fact, this SCS is considered and studied for IMT-2000 service by ITU. Although, it is important to make accurate channel model for prediction of the SCS performance, there is no measured channel data in this system. Thus, in this paper, we estimate the performance of SCS bye use of channel model provided by Corazza(2) and modified by You(3). And also, the effects of channel codings on system performance are analyzed by deriving bit error performance based on realistic Rician log-normal fading channel models. The performance results are divided into three kinds of areas with three kinds of elevation angles 20$^\cire$, 45$^\cire$, and 80$^\cire$. And also the effects of forward error correction channel codings on system performance with Hamming(7,4), HCH( IS,7) and convolutional code of constraint length 3 and code rate R=1/2.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.405-409
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• 2009

In this paper, the effects of LLR (Log-Likelihood Ratio) approximation on LDPC (Low-Density Parity-Check) decoder performance are analyzed, and optimal design conditions of LDPC decoder are derived. The min-sum LDPC decoding algorithm which is based on an approximation of LLR sum-product algorithm is modeled and simulated by MATLAB, and it is analyzed that the effects of LLR approximation bit-width and maximum iteration cycles on the bit error rate (BER) performance of LDCP decoder. The parity check matrix for IEEE 802.11n standard which has block length of 1,944 bits and code rate of 1/2 is used, and AWGN channel with QPSK modulation is assumed. The simulation results show that optimal BER performance is achieved for 7 iteration cycles and LLR bit-width of (7,5).

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.9
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• pp.835-842
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• 2009

DVB-SSP is a new broadcasting system for hybrid satellite communications, which supports mobile handheld systems and fixed terrestrial systems. However, a critical factor must be considered in upper layer decoding which including erasure Reed-Solomon error correction combined with cyclic redundancy check. If there is only one bit error in an IP packet, the entire IP packet is considered as unreliable bytes, even if it contains correct bytes. IF, for example, there is one real byte error, in an If packet of 512 bytes, 511 correct bytes are erased from the frame. Therefore, this paper proposed two kinds of upper layer decoding methods; LLR-based decoding and hybrid decoding. By means of simulation we show that the performance of the proposed decoding algorithm is superior to that of the conventional one.

• 전자공학회논문지 IE
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• v.43 no.4
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• pp.115-121
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• 2006

In this paper, we proposed about data error check and correction on channel transmission in the communication system. LDPC codes are used for minimizing channel errors by modeling AWGN Channel as a VDSL system. Because LDPC Codes use low density parity bit, mathematical complexity is low and relating processing time becomes shorten. Also the performance of LDPC code is better than that of turbo code in long code word on iterative decoding algorithm. This algorithm is better than conventional algorithms to correct errors, the proposed algorithm assigns weights for errors concerning parity bits. The proposed weighted Bit-flipping algorithm is better than the conventional Bit-flipping algorithm and we are recognized improve gain rate of 1 dB.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.6
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• pp.1355-1362
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• 2011

This paper describes a multi-mode LDPC decoder which supports three block lengths(648, 1296, 1944) and four code rates(1/2, 2/3, 3/4, 5/6) of IEEE 802.11n WLAN standard. Our LDPC decoder adopts a block-serial architecture based on min-sum algorithm and layered decoding scheme. A novel way to store check-node values and parity check matrix reduces the sizes of check-node memory and H-ROM. An efficient scheme for check-node memory addressing is used to achieve stall-free read/write operations. The designed LDPC decoder is verified by FPGA implementation, and synthesized with a $0.18-{\mu}m$ CMOS cell library. It has 219,100 gates and 45,036 bits RAM, and the estimated throughput is about 164~212 Mbps at 50 MHz@2.5v.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.2
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• pp.363-371
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• 2013

This paper describes a multi-standard LDPC decoder which supports 19 block lengths(576~2304) and 6 code rates(1/2, 2/3A, 2/3B, 3/4A, 3/4B, 5/6) of IEEE 802.16e mobile WiMAX standard and 3 block lengths(648, 1296, 1944) and 4 code rates(1/2, 2/3, 3/4, 5/6) of IEEE 802.11n WLAN standard. To minimize hardware complexity, it adopts a block-serial (partially parallel) architecture based on the layered decoding scheme. A DFU(decoding function unit) based on sign-magnitude arithmetic is used for hardware reduction. The designed LDPC decoder is verified by FPGA implementation, and synthesized with a 0.13-${\mu}m$ CMOS cell library. It has 312,000 gates and 70,000 bits RAM. The estimated throughput is about 79~210 Mbps at 100 MHz@1.8v.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.132-135
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• 2011

This paper describes a multi-mode LDPC decoder which supports three block lengths(648, 1296, 1944) and four code rates(1/2, 2/3, 3/4, 5/6) of IEEE 802.11n WLAN standard. To minimize hardware complexity, it adopts a block-serial (partially parallel) architecture based on the layered decoding scheme. A novel memory reduction technique devised using the min-sum decoding algorithm reduces the size of check-node memory by 47% as compared to conventional method. The designed LDPC decoder is verified by FPGA implementation, and synthesized with a $0.18-{\mu}m$ CMOS cell library. It has 219,100 gates and 45,036 bits RAM, and the estimated throughput is about 164~212 Mbps at 50 MHz@2.5v.

• 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.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.215-223
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• 2003

Turbo codes are selected as FEC(Forward error correction) codes with convolution code in 3GFP(3rd generation partnership project) and 3GPP2 standard of IMT2000. Especially, l/3 turbo code with K=4 is employed for 3GPP standard. In this paper, we proposed a hardware structure of a turbo decoder and denveloped the decoder for 3GPP standard turbo code. For its efficient operation, we design a SOVA decoder by employing a register exchange decoding block and new path metric normalization block as a SISO constituent decoder. In addition, we estimate its performance under MATLAB 6.0 and designed the turbo decoder including control block, input control buffer, SOVA constituent decoder with VHDL. Finally, we synthesized the developed turbo decoder under Synopsys FPGA Express and verified it with ALTERA EPF200SRC240-3 FPGA device.