• Title/Summary/Keyword: Reed Solomon Code

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Efficient Transmission of Scalable Video Streams Using Dual-Channel Structure (듀얼 채널 구조를 이용한 Scalable 비디오(SVC)의 전송 성능 향상)

  • Yoo, Homin;Lee, Jaemyoun;Park, Juyoung;Han, Sanghwa;Kang, Kyungtae
    • KIPS Transactions on Computer and Communication Systems
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    • v.2 no.9
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    • pp.381-392
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    • 2013
  • During the last decade, the multitude of advances attained in terminal computers, along with the introduction of mobile hand-held devices, and the deployment of high speed networks have led to a recent surge of interest in Quality of Service (QoS) for video applications. The main difficulty is that mobile devices experience disparate channel conditions, which results in different rates and patterns of packet loss. One way of making more efficient use of network resources in video services over wireless channels with heterogeneous characteristics to heterogeneous types of mobile device is to use a scalable video coding (SVC). An SVC divides a video stream into a base layer and a single or multiple enhancement layers. We have to ensure that the base layer of the video stream is successfully received and decoded by the subscribers, because it provides the basis for the subsequent decoding of the enhancement layer(s). At the same time, a system should be designed so that the enhancement layer(s) can be successfully decoded by as many users as possible, so that the average QoS is as high as possible. To accommodate these characteristics, we propose an efficient transmission scheme which incorporates SVC-aware dual-channel repetition to improve the perceived quality of services. We repeat the base-layer data over two channels, with different characteristics, to exploit transmission diversity. On the other hand, those channels are utilized to increase the data rate of enhancement layer data. This arrangement reduces service disruption under poor channel conditions by protecting the data that is more important to video decoding. Simulations show that our scheme safeguards the important packets and improves perceived video quality at a mobile device.

A Hybrid Decoding Algorithm for MPE-FEC based on DVB-SSP (DVB-SSP 기반 혼합형 MPE-FEC 복호 알고리즘)

  • Park, Tae-Doo;Kim, Min-Hyuk;Kim, Nam-Soo;Kim, Chul-Sung;Jung, Ji-Won;Lee, Seong-Ro
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.9C
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    • pp.848-854
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    • 2009
  • DVB-SSP is a new broadcasting system for hybrid satellite communications, which supports mobile handhold systems and fixed terrestrial systems. An upper layer, including erasure Reed-Solomon error correction combined with cyclic redundancy check. However, a critical factor must be considered in upper layer decoding. 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 upper layer decoding methods; hybrid decoding. By means of simulation we show that the performance of the proposed decoding algorithm is superior to that of the conventional one in AWGN channel and TI channel.

An Implementation on the Computing Algorithm for Inverse Finite Field using Composite Field (합성체를 이용한 유한체의 역원 계산 알고리즘 구현)

  • Noh Jin-Soo;Rhee Kang-Hyeon
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.43 no.3 s.309
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    • pp.76-81
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    • 2006
  • Recently, Finite field is applied the cryptography in the modern multimedia communication. Especially, block codes such as Elliptic Curve Cryptosystem and Reed-Solomon code among the error correcting codes are defined with finite field. Also, finite field algorithm is conducting the research actively because many kind of application parts need the real time operating ability therefore the exclusive hardware have been implementing. In this paper, we proposed the inverse finite field algorithm over GF($2^8$) using finite composite field and implemented in a hardware, and then compare this hardware with the currently used 'Itoh and Tsujii' hardware in respect to structure, area and computation time. Furthermore, this hardware was inserted into the AES SubBytes block and implemented on FPGA emulator board to confirm that the superiority of the proposed algorithm through the performance evaluation.

A Study on the Variable Transmission of xHE-AAC Audio Frame (xHE-AAC 오디오 프레임의 가변 전송에 관한 연구)

  • Lee, Bongho;Yang, Kyutae;Lim, Hyoungsoo;Hur, Namho
    • Journal of Broadcast Engineering
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    • v.21 no.3
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    • pp.357-368
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    • 2016
  • In DAB+, HE-AAC v2 codec is applied for the fixed rate transmission of audio stream. In case that xHE-AAC codec including USAC, a more efficiency is expected when the variable frame is used in a given same bandwidth compared to the fixed frame transmission. For this to be realized, audio streams need to be multiplexed in a sub-channel before transmission, then a method is required to identify the border of each audio frames. In this paper, the toggled sync byte and additional identification field being sequentially placed between AU borders are proposed in order to deal with the AU border identification. In addition, the Reed-Solomon based error correction code which is compliant to DAB+ is proposed.

Performance of 3D HDTV Transmission with Block LDPC Codes (블록 LDPC 부호를 사용한 3D HDTV 전송 성능개선 방안 연구)

  • Kim, Min-Ki;Kim, Dong Ho
    • Journal of Satellite, Information and Communications
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    • v.8 no.4
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    • pp.21-25
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    • 2013
  • The dual-stream based stereoscopic 3D HDTV broadcasting service was launched recently. Although the dual-stream based HDTV service has been successfully provided, the 3D HDTV broadcasting system requires more bandwidth efficient transmission schemes because it should convey both left and right HD resolution images simultaneously in the finite 6MHz bandwidth. In this paper, we consider more advanced ATSC transmission schemes that use higher modulation such as 16-QAM and concatenated RS code and block LDPC codes. Compared with conventional ATSC system and the modified ATSC system in [2], the proposed system has about 2.97dB and 1.12dB SNR gain at the payload data rate of 19.44Mbps compared with the existing ATSC system and the modified ATSC system [2]. Also, the proposed scheme requires only 1.05dB power increase for the 3D HDTV service, which is reasonable SNR increase value and applicable to the advanced 3D high definition broadcasting realization in limited 6MHz bandwidth.

A Study on the Design and Implementation of a DSSS-based MODEM for a Right Termination System(FTS) (대역확산방식 비행종단시스템의 모뎀설계와 구현에 관한 연구)

  • Lim Keumsang;Kim Jaehwan;Cho Hyangduck;Kim Wooshik
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.2C
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    • pp.175-183
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    • 2006
  • This letter proposes a Direct Sequence Spread Spectrum (DS-SS)-based Flight Termination System(FTS) and show the simulation results and implements the system using FRGAs. The DS-SS FTS has immunity interference signals and the influence of jamming signal. Moreover, a DS-SS FTS can provides effects on an authentication and encryption with spread codes. And the system uses more less power than an analog FM system. We used Reed-Solomon (32, 28) code and triple Data Encryption Standard(3DES) for error correction and data encryption. Also we used counter algorithm for unauthenticated device's attack The spread codes of In-phase channel and Quadrature channel were generated by Gold sequence generators. The system was implemented in Altera APEX20K100E FPGA for the ground system and EPF10K100ARC240-3 for the airborne system.

On The Size of The Subgroup Generated by Linear Factors (선형 요소에 의해 생성된 부분그룹의 크기에 관한 연구)

  • Cheng, Qi;Hwang, Sun-Tae
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.45 no.6
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    • pp.27-33
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    • 2008
  • Given a polynomial ${\hbar}(x){\in}F_q[x]$ of degree h, it is an important problem to determine the size of multiplicative subgroup of $\(F_q[x]/({\hbar(x))\)*$ generated by $x-s_1,\;x-s_2,\;{\cdots},\;x-s_n$, where $\{s_1,\;s_2,\;{\cdots},\;s_n\}{\sebseteq}F_q$, and for all ${\hbar}(x){\neq}0$. So far the best known asymptotic lower bound is $(rh)^{O(1)}\(2er+O(\frac{1}{r})\)^h$, where $r=\frac{n}{h}$ and e(=2.718...) is the base of natural logarithm. In this paper, we exploit the coding theory connection of this problem and prove a better lower bound $(rh)^{O(1)}\(2er+{\frac{e}{2}}{\log}r-{\frac{e}{2}}{\log}{\frac{e}{2}}+O{(\frac{{\log}^2r}{r})}\)^h$, where log stands for natural logarithm We also discuss about the limitation of this approach.

Design of an Adaptive Reed-Solomon Decoder with Varying Block Length (가변 블록길이를 갖는 적응형 리드솔로몬 복호기의 설계)

  • Song, Moon-Kyou;Kong, Min-Han
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.4C
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    • pp.365-373
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    • 2003
  • In this paper, we design a versatle RS decoder which can decode RS codes of any block length n as well as any message length k, based on a modified Euclid's algorithm (MEA). This unique feature is favorable for a shortened RS code of any block length it eliminates the need to insert zeros before decoding a shortened RS code. Furthermore, the value of error correcting capability t can be changed in real time at every codeword block. Thus, when a return channel is available, the error correcting capability can be adaptiverly altered according to channel state. The decoder permits 4-step pipelined processing : (1) syndrome calculation (2) MEA block (3) error magnitude calculation (4) decoder failure check. Each step is designed to form a structure suitable for decoding a RS code with varying block length. A new architecture is proposed for a MEA block in step (2) and an architecture of outputting in reversed order is employed for a polynomial evaluation in step (3). To maintain to throughput rate with less circuitry, the MEA block uses not only a multiplexing and recursive technique but also an overclocking technique. The adaptive RS decoder over GF($2^8$) with the maximal error correcting capability of 10 has been designed in VHDL, and successfully synthesized in a FPGA.

Effect of frequency dependent multipath fading on non-coherent underwater communication system (주파수 종속 다중경로 페이딩이 비코히어런트 수중통신시스템에 미치는 영향)

  • Kim, Jongjoo;Park, Jihyun;Bae, Minja;Park, Kyu-Chil;Yoon, Jong Rak
    • The Journal of the Acoustical Society of Korea
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    • v.35 no.4
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    • pp.295-302
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    • 2016
  • Underwater acoustic communication channel is often defined as a multipath fading channel since the multipath arrivals from various paths interfere with each other and cause frequency dependent constructive or destructive interference in received signals. Therefore signal-to-noise ratio (SNR) of received signal fluctuates as a function of frequency. In addition, sea surface fluctuation induces frequency dependent time variant signal fading due to coherent component variation of surface bounce path. The frequency shift keying (FSK) system is known to be less sensitive and more robust under these interference and fading, and M-ary frequency shift keying (MFSK) system is adopted to increase a data rate. In this study, a bit error rate (BER) of 4 channels 4FSK system are examined in shallow sea multipath channel. Experimental results show that RS code reduces efficiently the BER of 4FSK system since frequency dependent time-varying fading is characterized to give burst errors. The BER of a different data rate or different source-to-receiver range depends on not only the channel coherent bandwidth but also frequency dependent multipath fading.

Characteristics of Underwater Acoustic Channel and Performance of Multi-Carrier System in Littoral Ocean near Busan City (부산인근 해역의 수중음향통신 채널특성과 다중반송파 시스템의 성능)

  • Kim, Jongjoo;Park, Jihyun;Bae, Minja;Yoon, Jong Rak
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.21 no.12
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    • pp.2394-2402
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    • 2017
  • The frequency selective fading by multipaths determines a performance of underwater acoustic communication system in shallow littoral ocean. In this study, a characteristics of underwater acoustic channel and performance of multi-carrier system is evaluated in littoral ocean with a 50m deep water, an effective wave height of 0.5m and sandy mud bottom near Busan city. A multipath delay spread and time and frequency domain are presented as a function of a transmitter-to-receiver range. A bit-error-rate of a 5 channel 4FSK(Frequency Shift Keying) with a transmission rate of 1kbps, is examined and RS(Reed-Solomon) code is also adopted to remove a burst error due to time domain fading. A number of multipath are less than four and a bit-error-rate is decreased as an increase of a transmitter-to-receiver range which gives a congestion of multi-paths resulting in a decrease of time and frequency domain fading. The measured bit-error-rate is about 10-4 at greater than 600m of transmitter-to-receiver range.