• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• 제13권2호
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• pp.176-187
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• 2002

The digital Audio Broadcasting (DAB) system adopts Coded OFDM(COFDM) for channel coding. The COFDM is a combined technique of multicarrier transmission(OFDM) and punctured convolutional coding with viterbi error correction. Because the channel coding is an important topic for OFDM systems, this paper proposes a new turbo coded OFDM system that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the frequency selective Rician fading channel and the frequency selective Rayleigh fading channel in conjunction with DAB transmission mode I suitable for the terrestrial single frequency network(SFN) broadcasting. The standard system's performance was improved with the aid of turbo codec.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제37권6A호
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• pp.420-425
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• 2012

A forward error correction (FEC) coding techniques is one of time diversity techniques with which the effect of channel impairments due to noise and fading are spreaded over independently, and thus the performance could be improved. Therefore, the performance of the FEC scheme can be maximized if we minimize the correlation of channel information across over a codeword. In this paper, we propose a block turbo code with the maximized time diversity effect which may be reduced due to utilization of high order modulation schemes and due to transmission over a comparatively fast fading environment. Especially, we propose a very simple formula to calculate the address of coded bit allocation, and thus we do not need any additional outer interleavers, i.e., inter-codeword interleavers. The simulation resuts investigated in this paper reveal that the proposed scheme can provide the performance gain of more than a few decibels compared to the conventional schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제35권1A호
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• pp.62-70
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• 2010

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 (STC) 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 schemes. Based on these previous study results, we present various cooperative diversity techniques by combining 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제30권2A호
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• pp.20-25
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• 2005

A high performance Viterbi decoder is designed using modified register exchange scheme and block decoding method. The elimination of the trace-back operation reduces the operation cycles to determine the merging state and the amount of memory. The Viterbi decoder has low latency, efficient memory organization, and low hardware complexity compared with other Viterbi decoding methods in block decoding architectures. The elimination of trace-back also reduces the power consumption for finding the merging state and the access to the memory. The proposed decoder can be designed with emphasis on either efficient memory or low latency. Also, it has a scalable structure so that the complexity of the hardware and the throughput are adjusted by changing a few design parameters before synthesis.

• Journal of Advanced Navigation Technology
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• 제8권2호
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• pp.190-196
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• 2004

The development of a new hardware structure which can implement the viterbi algorithm efficiently is required for applications such as a software radio because the viterbi algorithm, which is an error correction code function for the second and the third generation of mobile communication, needs a lot of arithmetic operations. The length of K in the viterbi algorithm different from each standard, for examples, K=7 in case of IS-95 standard and GSM standard, and K=9 in case of WCDMA and CDMA2000. In this paper, we propose a new hardware structure of an adaptive viterbi decoder which can decode the constraint length in K=3~9 and the data rate in 1/2 ~ 1/3. Prototyping results targeted to Altera Cyclon EPIC20F400C8, shows that the proposed hardware structure needs maximum 19,276 logic elements and power dissipation of 222.6 mW.

• The Korea Journal of Herbology
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• 제28권3호
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• pp.61-68
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• 2013

Objectives : A precise and simple system of nomenclature was required to avoid error, ambiguity or confusion. Although medicinal plants must be produced or distributed based on a pharmacopoeia described origin including scientific name, the Korean Pharmacopoeia tenth edition (KP 10) had many names against the nomenclature. Therefore, this study aimed at searching correct scientific names for 241 plants in KP 10. Methods : Authoritative databases - The Plant List, International Plant Name Index, YList, Tropicos, eFloras, World Checklist of Selected Plant Families, The Global Compositae Checklist, The International Legume Database and Information Service, et al. - and previously performed researches, floras were cross-checked. Results : The arrangement of this list was designed for four cases, errors including illegitimate, nomenclatural synonyms, recommended names and decision reserved names. Consideration about the scientific names produced nine correct names for ten misspellings and illegitimate, and thirty-six correct names for forty-one nomenclatural synonyms. These results should be reflected in the next of KP 10. Separately, ten recommended names were also suggested for taxonomic synonyms which had been used indiscriminately due to diverse taxonomic opinions. In addition to those, decision reserved names were suggested for thirteen species which had been corridor of uncertainty. Then again, there was need to study about authorship, because KP 10 did not keep recommendations for author citations. Conclusions : Correction of scientific names for some medicinal plants which violated the International Code of Nomenclature would be useful to improve the accuracy of a Pharmacopoeia as the criterional materials.

• Journal of the Institute of Electronics Engineers of Korea TC
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• 제42권2호
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• pp.61-74
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• 2005

Digital Multimedia Broadcasting(DMB) is a reliable multi-service system for reception by mobile and portable receivers. DMB system allows interference-free reception under the conditions of multipath propagation and transmission errors using COFDM modulation scheme, simultaneously, needs powerful channel error's correction ability. Viterbi Decoder for DMB receiver uses punctured convolutional code and needs lots of computations for real-time operation. So, it is desired to design a high speed and low-power hardware scheme for Viterbi decoder. This paper proposes a combined add-compare-select(ACS) and path metric normalization(PMN) unit for computation power. The proposed PMN architecture reduces the problem of the critical path by applying fixed value for selection algorithm due to the comparison tree which has a weak point from structure with the high-speed operation. The proposed ACS uses the decomposition and the pre-computation technique for reducing the complicated degree of the adder, the comparator and multiplexer. According to a simulation result, reduction of area $3.78\%$, power consumption $12.22\%$, maximum gate delay $23.80\%$ occurred from punctured viterbi decoder for DMB system.

• Journal of the Institute of Electronics Engineers of Korea TC
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• 제43권7호
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• pp.143-149
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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 increased 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 sign changed ratio of extrinsic information values in turbo decoder, the proposed algorithm can largely reduce the average number of iterations without BER performance degradation. As a result of simulations, the average number of iterations is reduced by about $12.48%{\sim}22.22%$ compared to CE algorithm and about $20.43%{\sim}54.02%$ compared to SDR algorithm.

• Journal of Intelligence and Information Systems
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• 제24권2호
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• pp.21-35
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• 2018

Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.

• Journal of Broadcast Engineering
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• 제14권2호
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• pp.228-237
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• 2009

This paper examines the potential of Ultra-High Definition TV (UD-TV) broadcasting transmission systems beyond HD-TV over cable channel. Firstly, we analyze the trend of TOV(Threshold of Visibility) by extending the OpenCable (J.83 Annex B) system 256QAM which is the standard of Korean and American cable television transmission to 1024QAM, and realize that the OpenCable 1024QAM has nearly 30% higher data rate than 256QAM at the expense of impractically higher TOV (Threshold of Visibility). To achieve practical TOV, we control code rates of inner convolutional coder and replace turbo coder in forward error correction (FEC) part, thereby analyzing the best performance of the OpenCable systems having conventional FEC. In that result, it is necessary to modify conventional FEC of the OpenCable system to achieve under 31.5dB TOV. Moreover we study the potential of UD-TV transmission via two or more TV channels, so called channel bonding, through the Shannon capacity in 6MHz channel and the relationship with next generation A/V codec technologies.