• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.3
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• pp.29-38
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• 2004

In this paper, we present an efficient architecture of a versatile Reed-Solomon (RS) decoder which can be programmed to decode RS codes continuously with my message length k as well as any block length n. This unique feature eliminates the need of inserting zeros for decoding shortened RS codes. Also, the values of the parameters n and k, hence the error-correcting capability t can be altered at every codeword block. The decoder permits 3-step pipelined processing based on the modified Euclid's algorithm (MEA). Since each step can be driven by a separate clock, the decoder can operate just as 2-step pipeline processing by employing the faster clock in step 2 and/or step 3. Also, the decoder can be used even in the case that the input clock is different from the output clock. Each step is designed to have a structure suitable for decoding RS codes with varying block length. A new architecture for the MEA is designed for variable values of the t. The operating length of the shift registers in the MEA block is shortened by one, and it can be varied according to the different values of the t. To maintain the throughput rate with less circuitry, the MEA block uses both the recursive technique and the over-clocking technique. The decoder can decodes codeword received not only in a burst mode, but also in a continuous mode. It can be used in a wide range of applications because of its versatility. The adaptive RS decoder over GF(2$^{8}$ ) having the error-correcting capability of upto 10 has been designed in VHDL, and successfully synthesized in an FPGA chip.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.494-501
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• 2013

The convolution code(CC) of code rate 1/2 as a forward error correction (FEC) in Quadrature Phase Shift Keying (QPSK) is applied to decrease bit error rate (BER) by background noise and multipath in shallow water acoustic channel. Ratio of transmitting signal bandwidth to channel coherence bandwidth is defined as frequency selectivity index. BER and bit energy-to-noise ratio gain of transmitted signal according to frequency selectivity index are evaluated. In the results of indoor water tank experiment, BER is well matched theoretical results at frequency selectivity index less than about 1.0. And bit energy-to-noise ratio gain is also matched theoretical value of 5 dB. BER is effectively decreased at frequency selective multipath channel with frequency selectivity index higher than 1.0. But bit energy-to-noise ratio greater than a certain size in terms of CC weaving is effective in reducing bit errors. In the results, the defined frequency selectivity index in this study could be applied to evaluate a performance of CC in multipath channel. Also it could effectively reduced BER in a low speed underwater acoustic communication system without an equalizer.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8A
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• pp.859-867
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• 2004

To provide high-quality media streaming service over the best-effort Internet, a streaming methodology is required to response to the dynamic fluctuation of underlying networks. In this paper, we implement the MPEG-4 streaming system with adaptive transport based on priorities of media packets. The implemented system is composed of the common MPEG-4 streaming components such as elementary stream provider, sync and DMIF layer, and adaptive transport module including data prioritization and FEC control. More specifically, the prioritized sync layer packets (based on object level) are delivered to the transport module and then are encoded by an adaptive FEC encoder to help reliable transport. The FEC combination is dynamically adjusted by the feedback information from the receiver. In addition, low priority packets are selectively dropped to meet the limitation of available bandwidth. The experimental results over the emulation-based testbed show that the Proposed system can mitigate the impact of network fluctuation and thus improve the quality of streaming.

• Journal of Satellite, Information and Communications
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• v.4 no.2
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• pp.28-33
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• 2009

In this paper, we evaluate the performance of various diversity techniques which can contribute to provide efficient multimedia broadcasting services via hybrid/integrated satellite and terrestrial network. Space-time coding can achieve the diversity gain in a multi-path environment without additional bandwidth requirement. Recent study results reported that satellite systems can achieve high diversity gains by appropriate utilization of STC and/or forward error correction. Based on these previous study results, we present various cooperative diversity techniques by combing STC and rate compatible turbo codes in order to realize the transmit diversity for the mobile satellite system. The satellite and several terrestrial repeaters operate in unison to send the encoded signals, so that receiver may realize diversity gain. The results demonstrated in this paper can be utilized in future system implementation.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.1
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• pp.69-77
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• 2005

The Viterbi decoder which is used for the forward error correction(FEC) is a crucial component for successful modern communication systems. As modern communication speed rapidly high, the development of high speed communication module is important. However, since the feedback loop in ACS operation, high speed of Viterbi decoder is very difficult. In this paper, we propose an area reduced, high speed ACS Architecture of Viterbi decoder based on the radix-4 architecture. The area is reduced by rearranging the ACS operations, and the speed is improved by retiming of path metric memory. The proposed ACS architecture of Viterbi decoder is implemented in VHDL and synthesized in Xilinx ISE 6.2i. The area-time product of the proposed architecture is improved by 11% compared to that of the previous high speed radix-4 ACS architecture.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.6 s.312
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• pp.110-116
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• 2006

In this paper, we implement a speech recognition system to support ubiquitous sensor network application services such as switch control, authentication, etc. using wireless audio sensors. The proposed system is consist of the wireless audio sensor, the speech recognition algorithm using psychoacoustic model and LDPC(low density parity check) for correcting errors. The proposed speech recognition system is inserted in a HOST PC to use the sensor energy effectively mil to improve the accuracy of speech recognition, a FEC(Forward Error Correction) system is used. Also, we optimized the simulation coefficient and test environment to effectively remove the wireless channel noises and correcting wireless channel errors. As a result, when the distance between sensor and the source of voice is less then 1.0m FAR and FRR are 0.126% and 7.5% respectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.6A
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• pp.577-586
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• 2011

Modem digital communication systems are required to use forward error correction (FEC) codes to combat inevitable channel impairment. Turbo codes or low density parity check (LDPC) codes, using iterative decoding with soft decision detection (SDD) information, are the most common examples. The excellent performance of these codes should be conditioned on accurate estimation of soft decision detection information. In order to use FEC codes with iterative decoding for Multi-Input Multi-Output (MIMO) system, reliable soft decision channel gain should be provided. In this paper, we investigate efficient SDD methods for turbo-coded MIMO system, and derive the corresponding formulas of SDD for various MIMO detection schemes. We present simulation results of the derived SDD schemes for turbo-coded MIMO systems, and show that the presented results almost approximate to maximum likelihood detection performance with much less computational load.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.10C
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• pp.597-604
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• 2011

In this paper, we propose group Manchester code (GM) modulation scheme for medical in-body wireless body area network (WBAN) systems. In IEEE, the WBAN system is assigned as 802.15. Task Group 6 (TG 6), and the related standardization is being progressed, Recently, in this Group, group pulse position modulation (GPPM), which can obtain data rate increase by grouping pulse position modulation (PPM) symbols, is proposed as a new modulation scheme for low-power operation of WBAN system. However, the conventional method suffers from BER performance degradation due to the absence of gray coding and its demodulation characteristics. Therefore, in this paper, we propose a modified GM scheme which groups Manchester code instead of PPM. In the proposed GM scheme, a low-complexity maximum likelihood (ML) demodulation method is employed in order to maximize the BER performances, Also, log likelihood ratio (LLR) decision method is proposed to employ the Turbo code as forward error correction (FEC), Finally, we verified that the proposed method has a good performance and is an appropriate scheme for in-body WBAN system through extensive performance evaluation.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.2
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• pp.104-113
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• 2013

A high-throughput Quasi-Cyclic Low-Density Parity-Check (QC-LDPC) decoder architecture is proposed for 60GHz multi-gigabit wireless personal area network (WPAN) applications. Two novel techniques which can apply to our selected QC-LDPC code are proposed, including a four block-parallel layered decoding technique and fixed wire network. Two-stage pipelining and four block-parallel layered decoding techniques are used to improve the clock speed and decoding throughput. Also, the fixed wire network is proposed to simplify the switch network. A 672-bit, rate-1/2 QC-LDPC decoder architecture has been designed and implemented using 90-nm CMOS standard cell technology. Synthesis results show that the proposed QC-LDPC decoder requires a 794K gate and can operate at 290 MHz to achieve a data throughput of 3.9 Gbps with a maximum of 12 iterations, which meet the requirement of 60 GHz WPAN applications.