<p>This master thesis has been done at Enea Epact AB. The purpose of this thesis is to develop and implement a Link Quality Control algorithm for GPRS/EPGRS in the current testing tool. A Link Quality Control (LQC) shall take quality values from mobile stations and base stations and decide a codingsscheme that opimizes the throughput of data. </p><p>The Advantage with LQC is that it adapts the used coding scheme to the channel quality. If the channel quality is too bad for the used coding scheme, a slower coding scheme with more redundancy should be selected. On the other hand, if the channel quality is too good for the used coding scheme, LQC should recommend a faster coding scheme with less redundancy. </p><p>The testing tool is now using a static coding schme that doesn’t change during a data session. An LQC is therefore necessary for better simulation of the traffic and to make the tests more real.</p>

This master thesis has been done at Enea Epact AB. The purpose of this thesis is to develop and implement a Link Quality Control algorithm for GPRS/EPGRS in the current testing tool. A Link Quality Control (LQC) shall take quality values from mobile stations and base stations and decide a codingsscheme that opimizes the throughput of data. The Advantage with LQC is that it adapts the used coding scheme to the channel quality. If the channel quality is too bad for the used coding scheme, a slower coding scheme with more redundancy should be selected. On the other hand, if the channel quality is too good for the used coding scheme, LQC should recommend a faster coding scheme with less redundancy. The testing tool is now using a static coding schme that doesn’t change during a data session. An LQC is therefore necessary for better simulation of the traffic and to make the tests more real.

The objective of this thesis is to improve the performance of real-time communication overa wireless channel, by means of specifically tailored channel coding. The deadlinedependent coding (DDC) communication protocol presented here lets the timeliness and thereliability of the delivered information constitute quality of service (QoS) parametersrequested by the application. The values of these QoS parameters are transformed intoactions taken by the link layer protocol in terms of adaptive coding strategies.Incremental redundancy hybrid automatic repeat request (IR-HARQ) schemes usingrate compatible punctured codes are appealing since no repetition of previously transmittedbits is made. Typically, IR-HARQ schemes treat the packet lengths as fixed and maximizethe throughput by optimizing the puncturing pattern, i.e. the order in which the coded bitsare transmitted. In contrast, we define an IR strategy as the maximum number of allowedtransmissions and the number of code bits to include in each transmission. An approach isthen suggested to find the optimal IR strategy that maximizes the average code rate, i.e., theoptimal partitioning of n-kparity bits over at most M transmissions, assuming a givenpuncturing pattern. Concatenated coding used in IR-HARQ schemes provides a new arrayof possibilities for adaptability in terms of decoding complexity and communication timeversus reliability. Hence, critical reliability and timing constraints can be readily evaluatedas a function of available system resources. This in turn enables quantifiable QoS and thusnegotiable QoS. Multiple concatenated single parity check codes are chosen as examplecodes due to their very low decoding complexity. Specific puncturing patterns for thesecomponent codes are obtained using union bounds based on uniform interleavers. Thepuncturing pattern that has the best performance in terms of frame error rate (FER) at a lowsignal-to-noise ratio (SNR) is chosen. Further, using extrinsic information transfer (EXIT)analysis, rate compatible puncturing ratios for the constituent component code are found.The puncturing ratios are chosen to minimize the SNR required for convergence.The applications targeted in this thesis are not necessarily replacement of cables inexisting wired systems. Instead the motivation lies in the new services that wireless real-time communication enables. Hence, communication within and between cooperatingembedded systems is typically the focus. The resulting IR-HARQ-DDC protocol presentedhere is an efficient and fault tolerant link layer protocol foundation using adaptiveconcatenated coding intended specifically for wireless real-time communications. / Doktorsavhandlingar vid Chalmers tekniska högskola. Ny serie, 2198, Technical report. D, 29,

No description available.

One of the most important technologies used in modern communication systems is channel coding. Channel coding dates back to a paper published by Shannon in 1948 [1] entitled “A Mathematical Theory of Communication”. The basic idea behind channel coding is to send redundant information (parity) together with a message to make the transmission more error resistant. There are different types of codes that can be used to generate the parity required, including block, convolutional and concatenated codes. A special subclass of codes consisting of the codes mentioned in the previous paragraph, is sparse graph codes. The structure of sparse graph codes can be depicted via a graphical representation: the factor graph which has sparse connections between its elements. Codes belonging to this subclass include Low-Density-Parity-Check (LDPC) codes, Repeat Accumulate (RA), Turbo and fountain codes. These codes can be decoded by using the belief propagation algorithm, an iterative algorithm where probabilistic information is passed to the nodes of the graph. This dissertation focuses on noisy decoding of fountain codes using belief propagation decoding. Fountain codes were originally developed for erasure channels, but since any factor graph can be decoded using belief propagation, noisy decoding of fountain codes can easily be accomplished. Three fountain codes namely Tornado, Luby Transform (LT) and Raptor codes were investigated during this dissertation. The following results were obtained: <ol> <li>The Tornado graph structure is unsuitable for noisy decoding since the code structure protects the first layer of parity instead of the original message bits (a Tornado graph consists of more than one layer).</li> <li> The successful decoding of systematic LT codes were verified.</li> <li>A systematic Raptor code was introduced and successfully decoded. The simulation results show that the Raptor graph structure can improve on its constituent codes (a Raptor code consists of more than one code).</li></ol> Lastly an LT code was used to replace the convolutional incremental redundancy scheme used by the 2G mobile standard Enhanced Data Rates for GSM Evolution (EDGE). The results show that a fountain incremental redundancy scheme outperforms a convolutional approach if the frame lengths are long enough. For the EDGE platform the results also showed that the fountain incremental redundancy scheme outperforms the convolutional approach after the second transmission is received. Although EDGE is an older technology, it still remains a good platform for testing different incremental redundancy schemes, since it was one of the first platforms to use incremental redundancy. / Dissertation (MEng)--University of Pretoria, 2008. / Electrical, Electronic and Computer Engineering / MEng / unrestricted

Dans cette thèse, on s'est intéressé à l'étude des canaux de diffusion avec des contraintes de transmission pratiques. Tout d'abord, on a étudié l'impact de la contrainte pratique de l'utilisation d'un alphabet fini à l'entrée du canal de diffusion Gaussien avec deux utilisateurs. Deux modèles de canaux de diffusion sont considérés lorsqu'il y a, en plus d'un message commun pour les deux utilisateurs, (i) un message privé pour l'un des deux utilisateurs sans contrainte de sécurité (ii) un message confidentiel pour l'un des deux utilisateurs qui doit être totalement caché de l'autre utilisateur. On a présenté plusieurs stratégies de diffusion distinguées par leur complexité d'implémentation. Plus précisément, on a étudié les régions des débits atteignables en utilisant le partage de temps, la superposition de modulation et le codage par superposition. Pour la superposition de modulation et le cas général du codage par superposition, les régions des débits atteignables maximales sont obtenues en maximisant par rapport aux positions des symboles dans la constellation et la distribution de probabilité jointe. On a étudié le compromis entre la complexité d'implémentation des stratégies de transmission et leurs efficacités en termes de gains en débits atteignables. On a étudié aussi l'impact de la contrainte de sécurité sur la communication en comparant les débits atteignables avec et sans cette contrainte. Enfin, on a étudié les performances du système avec des schémas d'accusés de réception hybrides (HARQ) pour un canal à écoute à évanouissement par blocs lorsque l'émetteur n'a pas une information parfaite sur l'état instantané du canal mais connait seulement les statistiques. On a considéré un schéma adaptatif pour la communication sécurisée en utilisant des canaux de retour à niveaux multiples vers l'émetteur pour changer la longueur des sous mots de code à chaque retransmission afin que le débit utile secret soit maximisé sous des contraintes d'"outages". / In this thesis, we aim to study broadcast channels with practical transmission constraints. First, we study the impact of finite input alphabet constraint on the achievable rates for the Gaussian broadcast channel with two users. We consider two models of broadcast channels, when there is in addition of a common message for two users, (i) a private message for one of them without secrecy constraint (ii) a confidential message for one of them which should be totally hidden from the other user. We present several broadcast strategies distinguished by their complexity of implementation. More precisely, we study achievable rate regions using time sharing, superposition modulation and superposition coding. For superposition modulation and superposition coding strategies, maximal achievable rate regions are obtained by maximizing over both symbol positions in the constellation and the joint probability distribution. We study the tradeoff between the complexity of implementation of the transmission strategies and their efficiency in terms of gains in achievable rates. We study also the impact of the secrecy constraint on communication by comparing the achievable rates with and without this constraint. Finally, we study the system performance using HARQ schemes for the block-fading wiretap channel when the transmitter has no instantaneous channel state information but knows channel statistics. We consider an adaptive-rate scheme for the secure communication by using multilevel feedback channels to change sub-codeword lengths at each retransmission, in order to maximize the secrecy throughput under outage probabilities constraints.

During the race towards the4th generation (4G) cellular-based digital communication systems, a growth in the demand for high capacity, multi-media capable, improved Quality-of-Service (QoS) mobile communication systems have caused the developing mobile communications world to turn towards betterMultiple Access (MA) techniques, like Code Division Multiple Access (CDMA) [5]. The demand for higher throughput and better QoS in future 4G systems have also given rise to a scheme that is becoming ever more popular for use in these so-called ‘bandwidth-on-demand’ systems. This scheme is known as adaptive channel coding, and gives a system the ability to firstly sense changes in conditions, and secondly, to adapt to these changes, exploiting the fact that under good channel conditions, a very simple or even no channel coding scheme can be used for Forward Error Correction(FEC). This will ultimately result in better system throughput utilization. One such scheme, known as incremental redundancy, is already implemented in the Enhanced Data Rates for GSM Evolution (EDGE) standard. This study presents an extensive simulation study of a Multi-User (MU), adaptive channel coded Direct Sequence Spread Spectrum Multiple Access (DS/SSMA) communication system. This study firstly presents and utilizes a complex Base Band(BB) DS/SSMA transmitter model, aimed at user data diversity [6] in order to realize the MU input data to the system. This transmitter employs sophisticated double-sideband (DSB)Constant-Envelope Linearly Interpolated Root-of-Unity (CE-LI-RU) filtered General Chirp-Like (GCL) sequences [34, 37, 38] to band limit and spread user data. It then utilizes a fully user-definable, complex Multipath Fading Channel Simulator(MFCS), first presented by Staphorst [3], which is capable of reproducing all of the physical attributes of realistic mobile fading channels. Next, this study presents a matching DS/SSMA receiver structure that aims to optimally recover user data from the channel, ensuring the achievement of data diversity. In order to provide the basic channel coding functionality needed by the system of this study, three simple, but well-known channel coding schemes are investigated and employed. These are: binary Hamming (7,4,3) block code, (15,7,5) binary Bose-Chadhuri-Hocquenghem (BCH) block code and a rate 1/3 <i.Non-Systematic (NS) binary convolutional code [6]. The first step towards the realization of any adaptive channel coded system is the ability to measure channel conditions as fast as possible, without the loss of accuracy or inclusion of known data. In 1965, Gooding presented a paper in which he described a technique that measures communication conditions at the receiving end of a system through a device called a Performance Monitoring Unit (PMU) [12, 13]. This device accelerates the system’sBit Error Rate (BER) to a so-called Pseudo Error Rate(PER) through a process known as threshold modification. It then uses a simple PER extrapolation algorithm to estimate the system’s true BER with moderate accuracy and without the need for known data. This study extends the work of Gooding by applying his technique to the DS/SSMA system that utilizes a generic Soft-Output Viterbi Algorithm(SOVA) decoder [39] structure for the trellis decoding of the binary linear block codes [3, 41-50], as well as binary convolutional codes mentioned, over realistic MU frequency selective channel conditions. This application will grant the system the ability to sense changes in communication conditions through real-time BER measurement and, ultimately, to adapt to these changes by switching to different channel codes. Because no previous literature exists on this application, this work is considered novel. Extensive simulation results also investigate the linearity of the PER vs. modified threshold relationship for uncoded, as well as all coded cases. These simulations are all done for single, as well as multiple user systems. This study also provides extensive simulation results that investigate the calculation accuracy and speed advantages that Gooding’s technique possesses over that of the classic Monte-Carlo technique for BER estimation. These simulations also consider uncoded and coded cases, as well as single and multiple users. Finally, this study investigates the experimental real-time performance of the fully functional MU, adaptive coded, DS/SSMA communication system over varying channel conditions. During this part of the study, the channel conditions are varied over time, and the system’s adaptation (channel code switching) performance is observed through a real-time observation of the system’s estimated BER. This study also extends into cases with multiple system users. Since the adaptive coded system of this study does not require known data sequences (training sequences), inclusion of Gooding’s technique for real-time BER estimation through threshold modification and PER extrapolation in future 4G adaptive systems will enable better Quality-of-Service (QoS) management without sacrificing throughput. Furthermore, this study proves that when Gooding’s technique is applied to a coded system with a soft-output, it can be an effective technique for QoS monitoring, and should be considered in 4G systems of the future. / Dissertation (MEng (Computer Engineering))--University of Pretoria, 2007. / Electrical, Electronic and Computer Engineering / MEng / unrestricted

Dans cette thèse, les techniques de coopération ont été étudiées pour un canal multi-accès multi-relais composé d'au moins deux sources qui communiquent avec une seule destination à l'aide d'au moins deux nœuds de relayage en mode semi-duplex. Le multiplexage par répartition dans le temps est supposé. Tout d'abord, l’algorithme d’adaptation de lien est exécuté par l'ordonnanceur centralisé. Durant la première phase de transmission, les sources transmettent chacune à leur tour leur message respectif pendant des intervalles de temps consécutifs. Dans chaque intervalle de temps dans la deuxième phase, la destination planifie un nœud pour transmettre les redondances, mettant en œuvre un protocole coopératif d'Hybrid Automatic Repeat reQuest (HARQ), où les canaux de contrôle limités bidirectionnels sont disponibles depuis les sources et les relais vers la destination. Dans la première partie de la thèse, les stratégies de sélection des nœuds centralisé sont proposées pour la deuxième phase de transmission. Les décisions d’ordonnancement sont prises en fonction de la connaissance des ensembles de sources correctement décodées par chaque noeud et ayant comme objectif de maximiser l’efficacité spectrale moyenne. L'analyse de la probabilité de coupure de l'information ainsi que les simulations Monte-Carlo (MC) sont effectués afin de valider ces stratégies. Dans la seconde partie, un algorithme d’adaptation de lien lent est proposé afin de maximiser l’efficacité spectrale moyenne sous contrainte de vérification d'une qualité de service individuelle cible pour une famille donnée de schémas de modulation et de codage, réposant sur l'information sur la distribution des canaux signalée. Les débits des sources discrets sont déterminés en utilisant l’approche "Genie-Aided" suivie d’un algorithme itératif de correction de débit. Les simulations MC montrent que l’algorithme d’adaptation de lien proposé offre des performances proches de celles de la recherche exhaustive. Dans la troisième partie, les performances de protocole HARQ à redondance incrémentale (IR) avec codage mono et multi-utilisateur, ainsi que l'HARQ de type Chase Combining avec codage mono-utilisateur sont comparées. Les simulations MC montrent que l'IR-HARQ avec codage mono-utilisateur offre le meilleur compromis entre performance et complexité pour le scénario de petit nombre de sources. Un schéma de codage pratique est proposé et validé à l'aide de simulations MC. / In this thesis, cooperation techniques have been studied for Multiple Access Multiple Relay Channel, consisted of at least two sources which communicate with a single destination with the help of at least two half-duplex relaying nodes. Time Division Multiplexing is assumed. First, the link adaptation algorithm is performed at the centralised scheduler. Sources transmit in turns in consecutive time slots during the first transmission phase. In each time slot of the second phase, the destination schedules a node to transmit redundancies, implementing a cooperative Hybrid Automatic Repeat reQuest (HARQ) protocol, where bidirectional limited control channels are available from sources and relays towards the destination. In the first part of the thesis, centralized node selection strategies are proposed for the second phase. The scheduling decisions are made based on the knowledge of the correctly decoded source sets of each node, with the goal to maximize the average spectral efficiency. An information outage analysis is conducted and Monte-Carlo (MC) simulations are performed to evaluate their performance. In the second part, a slow-link adaptation algorithm is proposed which aims at maximizing the average spectral efficiency under individual QoS targets for a given modulation and coding scheme family relying on the reported Channel Distribution Information of all channels. Discrete source rates are first determined using the "Genie-Aided" assumption, which is followed by an iterative rate correction algorithm. The resulting link adaptation algorithm yields performance close to the exhaustive search approach as demonstrated by MC simulations. In the third part, performances of Incremental Redundancy (IR) HARQ with Single and Multi User encoding, as well as the Chase Combining HARQ with Single User encoding are compared. MC simulations demonstrate that IR-HARQ with Single User encoding offers the best trade-off between performance and complexity for a small number of sources in our setting. Practical coding scheme is proposed and validated using MC simulations.