24 research outputs found

    A study of digital holographic filters generation. Phase 2: Digital data communication system, volume 1

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    An empirical study of the performance of the Viterbi decoders in bursty channels was carried out and an improved algebraic decoder for nonsystematic codes was developed. The hybrid algorithm was simulated for the (2,1), k = 7 code on a computer using 20 channels having various error statistics, ranging from pure random error to pure bursty channels. The hybrid system outperformed both the algebraic and the Viterbi decoders in every case, except the 1% random error channel where the Viterbi decoder had one bit less decoding error

    Bit flipping decoding for binary product codes

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    Error control coding has been used to mitigate the impact of noise on the wireless channel. Today, wireless communication systems have in their design Forward Error Correction (FEC) techniques to help reduce the amount of retransmitted data. When designing a coding scheme, three challenges need to be addressed, the error correcting capability of the code, the decoding complexity of the code and the delay introduced by the coding scheme. While it is easy to design coding schemes with a large error correcting capability, it is a challenge finding decoding algorithms for these coding schemes. Generally increasing the length of a block code increases its error correcting capability and its decoding complexity. Product codes have been identified as a means to increase the block length of simpler codes, yet keep their decoding complexity low. Bit flipping decoding has been identified as simple to implement decoding algorithm. Research has generally been focused on improving bit flipping decoding for Low Density Parity Check codes. In this study we develop a new decoding algorithm based on syndrome checking and bit flipping to use for binary product codes, to address the major challenge of coding systems, i.e., developing codes with a large error correcting capability yet have a low decoding complexity. Simulated results show that the proposed decoding algorithm outperforms the conventional decoding algorithm proposed by P. Elias in BER and more significantly in WER performance. The algorithm offers comparable complexity to the conventional algorithm in the Rayleigh fading channel

    Some new results on majority-logic codes for correction of random errors

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    The main advantages of random error-correcting majority-logic codes and majority-logic decoding in general are well known and two-fold. Firstly, they offer a partial solution to a classical coding theory problem, that of decoder complexity. Secondly, a majority-logic decoder inherently corrects many more random error patterns than the minimum distance of the code implies is possible. The solution to the decoder complexity is only a partial one because there are circumstances under which a majority-logic decoder is too complex and expensive to implement. [Continues.

    Viterbi algorithm in continuous-phase frequency shift keying

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    The Viterbi algorithm, an application of dynamic programming, is widely used for estimation and detection problems in digital communications and signal processing. It is used to detect signals in communication channels with memory, and to decode sequential error-control codes that are used to enhance the performance of digital communication systems. The Viterbi algorithm is also used in speech and character recognition tasks where the speech signals or characters are modeled by hidden Markov models. This project explains the basics of the Viterbi algorithm as applied to systems in digital communication systems, and speech and character recognition. It also focuses on the operations and the practical memory requirements to implement the Viterbi algorithm in real-time. A forward error correction technique known as convolutional coding with Viterbi decoding was explored. In this project, the basic Viterbi decoder behavior model was built and simulated. The convolutional encoder, BPSK and AWGN channel were implemented in MATLAB code. The BER was tested to evaluate the decoding performance. The theory of Viterbi Algorithm is introduced based on convolutional coding. The application of Viterbi Algorithm in the Continuous-Phase Frequency Shift Keying (CPFSK) is presented. Analysis for the performance is made and compared with the conventional coherent estimator. The main issue of this thesis is to implement the RTL level model of Viterbi decoder. The RTL Viterbi decoder model includes the Branch Metric block, the Add-Compare-Select block, the trace-back block, the decoding block and next state block. With all done, we further understand about the Viterbi decoding algorithm

    Anticodes and error-correcting for digital data transmission

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    The work reported in this thesis is an investigation in the field of error-control coding. This subject is concerned with increasing the reliability of digital data transmission through a noisy medium, by coding the transmitted data. In this respect, an extension and development of a method for finding optimum and near-optimum codes, using N.m digital arrays known as anticodes, is established and described. The anticodes, which have opposite properties to their complementary related error-control codes, are disjoined fron the original maximal-length code, known as the parent anticode, to leave good linear block codes. The mathematical analysis of the parent anticode and as a result the mathematical analysis of its related anticodes has given some useful insight into the construction of a large number of optimum and near-optimum anticodes resulting respectively in a large number of optimum and near-optimum codes. This work has been devoted to the construction of anticodes from unit basic (small dimension) anticodes by means of various systematic construction and refinement techniques, which simplifies the construction of the associated linear block codes over a wide range of parameters. An extensive list of these anticodes and codes is given in the thesis. The work also has been extended to the construction of anticodes in which the symbols have been chosen from the elements of the finite field GF(q), and, in particular, a large number of optimum and near-optimum codes over GF(3) have been found. This generalises the concept of anticodes into the subject of multilevel codes

    802.11 Payload Iterative decoding between multiple transmission attempts

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    Abstract. The institute of electrical and electronics engineers (IEEE) 802.11 standard specifies widely used technology for wireless local area networks (WLAN). Standard specifies high-performance physical and media access control (MAC) layers for a distributed network but lacks an effective hybrid automatic repeat request (HARQ). Currently, the standard specifies forward error correction (FEC), error detection (ED), and automatic repeat request (ARQ), but in case of decoding errors, the previously transmitted information is not used when decoding the retransmitted packet. This is called Type 1 HARQ. Type 1 HARQ uses received energy inefficiently, but the simple implementation makes it an attractive solution. Unfortunately, research applying more sophisticated HARQ schemes on top of IEEE 802.11 is limited. In this Master’s Thesis, a novel HARQ technology based on packet retransmissions that can be decoded in a turbo-like manner, keeping as much as possible compatibility with vanilla 802.11, is proposed. The proposed technology is simulated with both the IEEE 802.11 code and with the robust, efficient and smart communication in unpredictable environments (RESCUE) code. An additional interleaver is added before the convolutional encoder in the proposed technology, interleaving either the whole frame or only the payload to enable effective iterative decoding. For received frames, turbo-like iterations are done between initially transmitted packet copy and retransmissions. Results are compared against the non-iterative combining method maximizing signal-to-noise ratio (SNR), maximum ratio combining (MRC). The main design goal for this technology is to maintain compatibility with the 802.11 standard while allowing efficient HARQ. Other design goals are range extension, higher throughput, and better performance in terms of bit error rate (BER) and frame error rate (FER). This technology can be used for range extension at low SNR range and may provide up to 4 dB gain at medium SNR range compared to MRC. At high SNR, technology can reduce the penalty from retransmission allowing higher average modulation and coding scheme (MCS). However, these gains come with the cost of computational complexity from the iterative decoding. The main limiting factors of the proposed technology are decoding errors in the header and the scrambler area, and resource-hungry-processing. In simulations, perfect synchronization and packet detection is assumed, but in reality, especially at low SNR, packet detection and synchronization would be challenging. 802.11 pakettien iteratiivinen dekoodaus lähetysten välillä. Tiivistelmä. IEEE 802.11-standardi määrittelee yleisesti käytetyn teknologian langattomille lähiverkoille. Standardissa määritellään tehokas fyysinen- ja verkkoliityntäkerros hajautetuille verkoille, mutta siitä puuttuu tehokas yhdistetty automaattinen uudelleenlähetys. Nykyisellään standardi määrittelee virheenkorjaavan koodin, virheellisen paketin tunnistuksen sekä automaattisen uudelleenlähetyksen, mutta aikaisemmin lähetetyn paketin informaatiota ei käytetä hyväksi uudelleenlähetystilanteessa. Tämä menetelmä tunnetaan tyypin yksi yhdistettynä automaattisena uudelleenlähetyksenä. Tyypin yksi yhdistetty automaattinen uudelleenlähetys käyttää vastaanotettua signaalia tehottomasti, mutta yksinkertaisuus tekee siitä houkuttelevan vaihtoehdon. Valitettavasti edistyneempien uudelleenlähetysvaihtoehtojen tutkimusta 802.11-standardiin on rajoitetusti. Tässä diplomityössä esitellään uusi yhdistetty uudelleenlähetysteknologia, joka pohjautuu pakettien uudelleenlähetykseen, sallien turbo-tyylisen dekoodaamisen säilyttäen mahdollisimman hyvän taaksepäin yhteensopivuutta alkuperäisen 802.11-standardin kanssa. Tämä teknologia on simuloitu käyttäen sekä 802.11- että nk. RESCUE-virheenkorjauskoodia. Teknologiassa uusi lomittaja on lisätty konvoluutio-enkoodaajan eteen, sallien tehokkaan iteratiivisen dekoodaamisen, lomittaen joko koko paketin tai ainoastaan hyötykuorman. Vastaanotetuille paketeille tehdään turbo-tyyppinen iteraatio alkuperäisen vastaanotetun kopion ja uudelleenlähetyksien välillä. Tuloksia vertaillaan eiiteratiiviseen yhdistämismenetelmään, maksimisuhdeyhdistelyyn, joka maksimoi yhdistetyn signaali-kohinasuhteen. Tärkeimpänä suunnittelutavoitteena tässä työssä on tehokas uudelleenlähetysmenetelmä, joka ylläpitää taaksepäin yhteensopivuutta IEEE 802.11-standardin kanssa. Muita tavoitteita ovat kantaman lisäys, nopeampi yhteys ja matalampi bitti- ja pakettivirhesuhde. Kehitettyä teknologiaa voidaan käyttää kantaman lisäykseen matalan signaalikohinasuhteen vallitessa ja se on jopa 4 dB parempi kohtuullisella signaalikohinasuhteella kuin maksimisuhdeyhdistely. Korkealla signaali-kohinasuhteella teknologiaa voidaan käyttää pienentämään häviötä epäonnistuneesta paketinlähetyksestä ja täten sallien korkeamman modulaatio-koodiasteen käyttämisen. Valitettavasti nämä parannukset tulevat kasvaneen laskennallisen monimutkaisuuden kustannuksella, johtuen iteratiivisesta dekoodaamisesta. Isoimmat rajoittavat tekijät teknologian käytössä ovat dekoodausvirheet otsikossa ja datamuokkaimen siemenessä. Tämän lisäksi käyttöä rajoittaa resurssisyöppö prosessointi. Simulaatioissa oletetaan täydellinen synkronisointi, mutta todellisuudessa, erityisesti matalalla signaali-kohinasuhteella, paketin tunnistus ja synkronointi voivat olla haasteellisia

    Iterative decoding and detection for physical layer network coding

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    PhD ThesisWireless networks comprising multiple relays are very common and it is important that all users are able to exchange messages via relays in the shortest possible time. A promising technique to achieve this is physical layer network coding (PNC), where the time taken to exchange messages between users is achieved by exploiting the interference at the relay due to the multiple incoming signals from the users. At the relay, the interference is demapped to a binary sequence representing the exclusive-OR of both users’ messages. The time to exchange messages is reduced because the relay broadcasts the network coded message to both users, who can then acquire the desired message by applying the exclusive-OR of their original message with the network coded message. However, although PNC can increase throughput it is at the expense of performance degradation due to errors resulting from the demapping of the interference to bits. A number of papers in the literature have investigated PNC with an iterative channel coding scheme in order to improve performance. However, in this thesis the performance of PNC is investigated for end-to-end (E2E) the three most common iterative coding schemes: turbo codes, low-density parity-check (LDPC) codes and trellis bit-interleaved coded modulation with iterative decoding (BICM-ID). It is well known that in most scenarios turbo and LDPC codes perform similarly and can achieve near-Shannon limit performance, whereas BICM-ID does not perform quite as well but has a lower complexity. However, the results in this thesis show that on a two-way relay channel (TWRC) employing PNC, LDPC codes do not perform well and BICM-ID actually outperforms them while also performing comparably with turbo codes. Also presented in this thesis is an extrinsic information transfer (ExIT) chart analysis of the iterative decoders for each coding scheme, which is used to explain this surprising result. Another problem arising from the use of PNC is the transfer of reliable information from the received signal at the relay to the destination nodes. The demapping of the interference to binary bits means that reliability information about the received signal is lost and this results in a significant degradation in performance when applying soft-decision decoding at the destination nodes. This thesis proposes the use of traditional angle modulation (frequency modulation (FM) and phase modulation (PM)) when broadcasting from the relay, where the real and imaginary parts of the complex received symbols at the relay modulate the frequency or phase of a carrier signal, while maintaining a constant envelope. This is important since the complex received values at the relay are more likely to be centred around zero and it undesirable to transmit long sequences of low values due to potential synchronisation problems at the destination nodes. Furthermore, the complex received values, obtained after angle demodulation, are used to derive more reliable log-likelihood ratios (LLRs) of the received symbols at the destination nodes and consequently improve the performance of the iterative decoders for each coding scheme compared with conventionally coded PNC. This thesis makes several important contributions: investigating the performance of different iterative channel coding schemes combined with PNC, presenting an analysis of the behaviour of different iterative decoding algorithms when PNC is employed using ExIT charts, and proposing the use of angle modulation at the relay to transfer reliable information to the destination nodes to improve the performance of the iterative decoding algorithms. The results from this thesis will also be useful for future research projects in the areas of PNC that are currently being addressed, such as synchronisation techniques and receiver design.Iraqi Ministry of Higher Education and Scientific Research

    Codes on Graphs and More

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    Modern communication systems strive to achieve reliable and efficient information transmission and storage with affordable complexity. Hence, efficient low-complexity channel codes providing low probabilities for erroneous receptions are needed. Interpreting codes as graphs and graphs as codes opens new perspectives for constructing such channel codes. Low-density parity-check (LDPC) codes are one of the most recent examples of codes defined on graphs, providing a better bit error probability than other block codes, given the same decoding complexity. After an introduction to coding theory, different graphical representations for channel codes are reviewed. Based on ideas from graph theory, new algorithms are introduced to iteratively search for LDPC block codes with large girth and to determine their minimum distance. In particular, new LDPC block codes of different rates and with girth up to 24 are presented. Woven convolutional codes are introduced as a generalization of graph-based codes and an asymptotic bound on their free distance, namely, the Costello lower bound, is proven. Moreover, promising examples of woven convolutional codes are given, including a rate 5/20 code with overall constraint length 67 and free distance 120. The remaining part of this dissertation focuses on basic properties of convolutional codes. First, a recurrent equation to determine a closed form expression of the exact decoding bit error probability for convolutional codes is presented. The obtained closed form expression is evaluated for various realizations of encoders, including rate 1/2 and 2/3 encoders, of as many as 16 states. Moreover, MacWilliams-type identities are revisited and a recursion for sequences of spectra of truncated as well as tailbitten convolutional codes and their duals is derived. Finally, the dissertation is concluded with exhaustive searches for convolutional codes of various rates with either optimum free distance or optimum distance profile, extending previously published results

    Algebraic Codes For Error Correction In Digital Communication Systems

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    Access to the full-text thesis is no longer available at the author's request, due to 3rd party copyright restrictions. Access removed on 29.11.2016 by CS (TIS).Metadata merged with duplicate record (http://hdl.handle.net/10026.1/899) on 20.12.2016 by CS (TIS).C. Shannon presented theoretical conditions under which communication was possible error-free in the presence of noise. Subsequently the notion of using error correcting codes to mitigate the effects of noise in digital transmission was introduced by R. Hamming. Algebraic codes, codes described using powerful tools from algebra took to the fore early on in the search for good error correcting codes. Many classes of algebraic codes now exist and are known to have the best properties of any known classes of codes. An error correcting code can be described by three of its most important properties length, dimension and minimum distance. Given codes with the same length and dimension, one with the largest minimum distance will provide better error correction. As a result the research focuses on finding improved codes with better minimum distances than any known codes. Algebraic geometry codes are obtained from curves. They are a culmination of years of research into algebraic codes and generalise most known algebraic codes. Additionally they have exceptional distance properties as their lengths become arbitrarily large. Algebraic geometry codes are studied in great detail with special attention given to their construction and decoding. The practical performance of these codes is evaluated and compared with previously known codes in different communication channels. Furthermore many new codes that have better minimum distance to the best known codes with the same length and dimension are presented from a generalised construction of algebraic geometry codes. Goppa codes are also an important class of algebraic codes. A construction of binary extended Goppa codes is generalised to codes with nonbinary alphabets and as a result many new codes are found. This construction is shown as an efficient way to extend another well known class of algebraic codes, BCH codes. A generic method of shortening codes whilst increasing the minimum distance is generalised. An analysis of this method reveals a close relationship with methods of extending codes. Some new codes from Goppa codes are found by exploiting this relationship. Finally an extension method for BCH codes is presented and this method is shown be as good as a well known method of extension in certain cases

    Time diversity solutions to cope with lost packets

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    A dissertation submitted to Departamento de Engenharia Electrotécnica of Faculdade de Ciências e Tecnologia of Universidade Nova de Lisboa in partial fulfilment of the requirements for the degree of Doctor of Philosophy in Engenharia Electrotécnica e de ComputadoresModern broadband wireless systems require high throughputs and can also have very high Quality-of-Service (QoS) requirements, namely small error rates and short delays. A high spectral efficiency is needed to meet these requirements. Lost packets, either due to errors or collisions, are usually discarded and need to be retransmitted, leading to performance degradation. An alternative to simple retransmission that can improve both power and spectral efficiency is to combine the signals associated to different transmission attempts. This thesis analyses two time diversity approaches to cope with lost packets that are relatively similar at physical layer but handle different packet loss causes. The first is a lowcomplexity Diversity-Combining (DC) Automatic Repeat reQuest (ARQ) scheme employed in a Time Division Multiple Access (TDMA) architecture, adapted for channels dedicated to a single user. The second is a Network-assisted Diversity Multiple Access (NDMA) scheme, which is a multi-packet detection approach able to separate multiple mobile terminals transmitting simultaneously in one slot using temporal diversity. This thesis combines these techniques with Single Carrier with Frequency Division Equalizer (SC-FDE) systems, which are widely recognized as the best candidates for the uplink of future broadband wireless systems. It proposes a new NDMA scheme capable of handling more Mobile Terminals (MTs) than the user separation capacity of the receiver. This thesis also proposes a set of analytical tools that can be used to analyse and optimize the use of these two systems. These tools are then employed to compare both approaches in terms of error rate, throughput and delay performances, and taking the implementation complexity into consideration. Finally, it is shown that both approaches represent viable solutions for future broadband wireless communications complementing each other.Fundação para a Ciência e Tecnologia - PhD grant(SFRH/BD/41515/2007); CTS multi-annual funding project PEst-OE/EEI/UI0066/2011, IT pluri-annual funding project PEst-OE/EEI/LA0008/2011, U-BOAT project PTDC/EEATEL/ 67066/2006, MPSat project PTDC/EEA-TEL/099074/2008 and OPPORTUNISTICCR project PTDC/EEA-TEL/115981/200
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