• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.3 no.1 s.4
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• pp.117-125
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• 2004

Space-time coding and modulation exploit the presence of multiple transmit antennas to improve performance on multipath Rayleigh fading channels. In this paper, we propose the Trellis-Coded Differential Space Time Modulation-OFDM system with multiple symbol detection. The Trellis-code perform the set partition with unitary group codes. The Viterbi decoder containing new branch metrics is introduced in order to improve the bit error rate (BER) in the differential detection of the Unitary differential space time modulation. Also, we describe the Viterbi algorithm in order to use this branch metrics. Our study shows that such a Viterbi decoder improves BER performance without sacrificing bandwidth and power efficiency.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.66-71
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• 2001

This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.4
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• pp.569-574
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• 2017

In order to transmit information in a channel environment in which noise exists, a coding technique of information is required. One of the coding techniques used for error detection and correction close to the Shannon limit is Low Density Parity Code(LDPC). LDPC and decoding characteristic features by Sum-product algorithm are matched for the performance to Turbo Code, RA(Repeat Accumulate) code, in case of very long code length of LDPC surpass their performance. This paper explains LDPC coding scheme of image data and decoding scheme, implements LDPC decoder in FPGA.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.5
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• pp.887-892
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• 2003

In this paper, we propose a simple scaling algorithm for CDMA mobile communications where a voice traffic signals are transmitted by individual one of several data rates at every frames. The traditional method is based on using look-up table called SMT(symbol metric table), but the proposed algorithm is real-time direct scaling method through simple bit manipulations without lookup table. The bit error rate performance is calculated by computer simulation over AWGN and Rayleigh fading channels. From the results, it is shown that the proposed algorithm outperforms the traditional SMT method on Rayleigh channel by 0.3∼0.8dB, while achieving the less H/W complexity.

• Journal of IKEEE
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• v.2 no.2 s.3
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• pp.285-293
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• 1998

This paper presents unequal error protection and error concealment techniques far robust H.263 video transmission over mobile channels. The proposed error protection scheme has three major features. First, it has the capability of preventing the loss of synchronization information in H.263 video stream as much as possible that the H.263 decoder can resynchronize at the next decoding point, if errors are occurred. Secondly, it employs an unequal error protection scheme to support variable coding rates using rate compatible punctured convolutional (RCPC) codes, dividing the encoded stream into two classes. Finally, a macroblock-interleaving scheme is employed in order to minimize the corruption of consecutive macroblocks due to burst errors, which can make a proper condition for error concealment. In addition, to minimize the spatial error propagations due to the variable length codes, a fast resynchronization scheme at the group of block layer is developed for recovering subsequent error-free macroblocks following the damaged macroblock. futhermore, error concealment techniques based on both side match criterion and overlapped block motion compensation (OBMC) are employed at the source decoder so that it can not only recover the lost macroblock more accurately, but also reduce blocking artifacts. Experimental results show that the proposed scheme can be an effective error protection scheme since proper video quality can be maintained under various channel bit error rates.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.73-79
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• 2004

In this paper, we propose an efficient soft decision decoding algorithm for linear block codes. A conventional soft decision decoder have to invoke a hard decision decoder several times to estimate its soft decision values. However, in this method, we may not have candidate codewords, thus it is very difficult to produce soft decision values. We solve this problem by introducing an efficient algorithm to search candidate codewords. By using this, we can highly reduce the cases we cannot find candidate codewords. We estimate the performance of the proposed algorithm by using the computer simulations. The simulation is performed for binary (63, 36) BCH code in fading channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.3A
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• pp.324-330
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• 2004

In this paper, we propose the TCM(Trellis coded modulation) decoding scheme that reduces the number of operations in branch metric with STBC(space time block codes) channel information and present the implementation results. The proposed TCM decoding scheme needs only 1 signal point in each TCM subset. Using bias point scheme, It detects the minimum distance symbol. The proposed TCM decoding scheme can reduce the branch metric calculations. In case of 16QAM 8 subset, the reduction ratio is about 50％ and for 64QAM 8 subset, about 80％ reduction can be obtained. The results of logic synthesis for the TCM and STBC decoder with the proposed scheme are 57.6K gate count.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1567-1570
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• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

• Journal of electromagnetic engineering and science
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• v.11 no.4
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• pp.239-249
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• 2011

In this paper, we propose a reliable $8{\times}8$ up-down switching polar relay code based on 3GPP LTE standard, motivated by 3GPP LTE down link, which is 30 bps/Hz for $8{\times}8$ MIMO antennas, and by Arikan's channel polarization for the frequency selective fading (FSF) channels with the generator matrix $Q_8$. In this scheme, a polar encoder and OFDM modulator are implemented sequentially at both the source node and relay nodes, the time reversion and complex conjugation operations are separately implemented at each relay node, and the successive interference cancellation (SIC) decoder, together with the cyclic prefix (CP) removal, is performed at the destination node. Use of the scheme shows that decoding at the relay without any delay is not required, which results in a lower complexity. The numerical result shows that the system coded by polar codes has better performance than currently used designs.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.4C
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• pp.329-336
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• 2010

Distributed source coding (DSC) system moves computational burden from encoder to decoder, so it takes higher decoding complexity. This paper explores the problem of reducing the decoding complexity of practical Slepian-Wolf coding using low-density parity check accumulate (LDPCA) codes. It is shown that the convergence of mean magnitude (CMM) stopping criteria for LDPC codes help reduce the 85% of decoding complexity under the 2% of compression rate loss, and marginal initial rate request reduces complexity below complexity minimum bound. Moreover, inter-rate stopping criterion, modified for rate-adaptable characteristic, is proposed for LDPCA codes, and it makes decoder perform less iterative decoding than normal stopping criterion does when channel characteristic is unknown.