Mecanismos de HARQ usando códigos LDPC com retransmissão parcial e combinação por diversidade

Modern standards of communication systems are including in their standards retransmission mechanisms. This work presents mechanisms for retransmission of packets by simpler and more efficient than the classical mechanisms of packet retransmissions. The proposed schemes are presented that use packet retransmission with partial retransmission and diversity combining. In addition, theoretical analysis will be presented to validate the results of the simulations. These theoretical analysis are obtained through the analysis EXIT charts and mutual information. The results and analysis were performed on channels such as AWGN and Block-Fading. The computational complexity, ease implementation and low average energy consumption for transmission of the proposed methods are some of the reasons they become interesting both for academia and for industry.Os recentes padrões de sistemas de comunicação estão incluindo em suas normas mecanismos de retransmissão de pacotes. Este trabalho apresenta mecanismos de retransmissão de pacotes de formas simples e mais eficientes do que os mecanismos clássicos de retransmissão de pacotes. São apresentados esquemas de retransmissão de pacotes que utilizam retransmissões parciais e combinação por diversidade. Além disso, serão apresentadas análises teóricas para validar os resultados obtidos das simulações. Estas análises teóricas são obtidas através das análises de EXIT charts e da informação mútua. Os resultados e análises foram realizados em canais do tipo AWGN e com desvanecimento por blocos (Block-Fading). A complexidade computacional, facilidade de implementação e baixo consumo médio de energia por transmissão dos métodos propostos são alguns dos motivos pelos quais, tornam-se interessantes tanto para a área acadêmica quanto para a indústria

Variable Rate Transmission Over Noisy Channels

Hybrid automatic repeat request transmission (hybrid ARQ) schemes aim to provide system reliability for transmissions over noisy channels while still maintaining a reasonably high throughput efficiency by combining retransmissions of automatic repeat requests with forward error correction (FEC) coding methods. In type-II hybrid ARQ schemes, the additional parity information required by channel codes to achieve forward error correction is provided only when errors have been detected. Hence, the available bits are partitioned into segments, some of which are sent to the receiver immediately, others are held back and only transmitted upon the detection of errors. This scheme raises two questions. Firstly, how should the available bits be ordered for optimal partitioning into consecutive segments? Secondly, how large should the individual segments be? This thesis aims to provide an answer to both of these questions for the transmission of convolutional and Turbo Codes over additive white Gaussian noise (AWGN), inter-symbol interference (ISI) and Rayleigh channels. Firstly, the ordering of bits is investigated by simulating the transmission of packets split into segments with a size of 1 bit and finding the critical number of bits, i.e. the number of bits where the output of the decoder is error-free. This approach provides a maximum, practical performance limit over a range of signal-to-noise levels. With these practical performance limits, the attention is turned to the size of the individual segments, since packets of 1 bit cause an intolerable overhead and delay. An adaptive, hybrid ARQ system is investigated, in which the transmitter uses the number of bits sent to the receiver and the receiver decoding results to adjust the size of the first, initial, packet and subsequent segments to the conditions of a stationary channel

Transmission and Combining for Hybrid Automatic Repeat Request in Multiple-Input Multiple-Output Systems

Hybrid automatic repeat request (HARQ) schemes combine packet retransmission with forward error correction to ensure a reliable communications. In multiple-input multiple output (MIMO) systems, interference cancellation (IC) detection is widely used where the detection and cancellation steps of the simultaneously transmitted data streams occur. In principle, the signal stream estimated at one IC stage is utilized to cancel the interference of other signal streams at the next IC stage. Thus, the detection probabilities of the transmitted data streams are mutually dependent. With HARQ, the detection performance of a packet also depends on how many times the packet has been retransmitted. The dissertation consists of three main contributions. Firstly, we develop a HARQ transmission state control algorithm for MIMO systems with IC detection to improve throughput. The HARQ transmission state is defined as the distribution of the initial packets and retransmission packets transmitted during a packet transmission time interval (PTTI). The proposed algorithm generates the transmission state in which initial packets and retransmission packets are sent together. The outcome is that it achieves a lower error probability for initial packets by exploiting the IC process and a significantly higher throughput than the conventional HARQ system, which is verified by simulation results. However, the maximum allowable number of retransmission is limited to one in this algorithm. Secondly, in order to extend the analysis for a more general case, we define the concept of the effective interference level (EIL) as the performance parameter to choose the set of packets during one PTTI and establish a relationship between EIL and the effective signal-to-interference-plus-noise ratio (SINR). We then show that choosing the set of packets that minimize the EIL successively from the lowest to the highest HARQ round leads to a lower packet error and higher throughput than conventional HARQ, which is verified by simulation. Also, the proposed EIL based scheme uses only the acknowledgement feedback messages like a conventional HARQ, because the number of HARQ rounds of each packet is the only required information to calculate the EIL. Simulation results highlight the superiority of the proposed scheme over the conventional scheme in terms of throughput with the signal-to-noise ratio gain of about 4.2 dB at maximum for MIMO systems with four transmit and four receive antennas. Thirdly, a low-complexity symbol-level combining (SLC) scheme is developed for Chase combining based HARQ (CC-HARQ) in MIMO systems, when the linear detection is considered at the receiver. In the proposed scheme, instead of using the entire channel matrix as in the existing SLC schemes, a subset of row vectors in the channel matrix is selected in the proposed scheme, and the selected row vectors are sequentially used during the estimation procedures of the retransmitted symbols, where the sequential utilization is enabled by using the Sherman-Morrison-Woodbury (SMW) lemma. Therefore, according to the number of the selected row vectors, this approach enables the proposed SLC scheme to have an advantage in complexity compared to the existing SLC schemes. In addition, we develop a row vector selection criterion for the proposed scheme to compute the amount of the SINR improvement by using a squared norm of each row vector with a significantly lower computational complexity. Simulation results show that compared to the existing SLC schemes, the proposed SLC scheme achieves similar or better error performance, while its computational complexity is lower or in the worst case similar

Time diversity solutions to cope with lost packets

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

Communications in the observation limited regime

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2009.Cataloged from PDF version of thesis.Includes bibliographical references (p. 141-145).We consider the design of communications systems when the principal cost is observing the channel, as opposed to transmit energy per bit or spectral efficiency. This is motivated by energy constrained communications devices where sampling the signal, rather than transmitting or processing it, dominates energy consumption. We show that sequentially observing samples with the maximum a posteriori entropy can reduce observation costs by close to an order of magnitude using a (24,12) Golay code. This is the highest performance reported over the binary input AWGN channel, with or without feedback, for this blocklength. Sampling signal energy, rather than amplitude, lowers circuit complexity and power dissipation significantly, but makes synchronization harder. We show that while the distance function of this non-linear coding problem is intractable in general, it is Euclidean at vanishing SNRs, and root Euclidean at large SNRs. We present sequences that maximize the error exponent at low SNRs under the peak power constraint, and under all SNRs under an average power constraint. Some of our new sequences are an order of magnitude shorter than those used by the 802.15.4a standard.(cont.) In joint work with P. Mercier and D. Daly, we demonstrate the first energy sampling wireless modem capable of synchronizing to within a ns, while sampling energy at only 32 Msamples per second, and using no high speed clocks. We show that traditional, minimum distance classifiers may be highly sensitive to parameter estimation errors, and propose robust, computationally efficient alternatives. We challenge the prevailing notion that energy samplers must accurately shift phase to synchronize with high precision.by Manish Bhardwaj.Ph.D

Multicast MAC extensions for high rate real-time traffic in wireless LANs

Nowadays we are rapidly moving from a mainly textual-based to a multimedia-based Internet, for which the widely deployed IEEE 802.11 wireless LANs can be one of the promising candidates to make them available to users anywhere, anytime, on any device. However, it is still a challenge to support group-oriented real-time multimedia services, such as video-on-demand, video conferencing, distance educations, mobile entertainment services, interactive games, etc., in wireless LANs, as the current protocols do not support multicast, in particular they just send multicast packets in open-loop as broadcast packets, i.e., without any possible acknowledgements or retransmissions. In this thesis, we focus on MAC layer reliable multicast approaches which outperform upper layer ones with both shorter delays and higher efficiencies. Different from polling based approaches, which suffer from long delays, low scalabilities and low efficiencies, we explore a feedback jamming mechanism where negative acknowledgement (NACK) frames are allowed from the non-leader receivers to destroy the acknowledgement (ACK) frame from the single leader receiver and prompts retransmissions from the sender. Based on the feedback jamming scheme, we propose two MAC layer multicast error correction protocols, SEQ driven Leader Based Protocol (SEQ-LBP) and Hybrid Leader Based Protocol (HLBP), the former is an Automatic Repeat reQuest (ARQ) scheme while the later combines both ARQ and the packet level Forward Error Correction (FEC). We evaluate the feedback jamming probabilities and the performances of SEQ-LBP and HLBP based on theoretical analyses, NS-2 simulations and experiments on a real test-bed built with consumer wireless LAN cards. Test results confirm the feasibility of the feedback jamming scheme and the outstanding performances of the proposed protocols SEQ-LBP and HLBP, in particular SEQ-LBP is good for small multicast groups due to its short delay, effectiveness and simplicity while HLBP is better for large multicast groups because of its high efficiency and high scalability with respect to the number of receivers per group.Zurzeit vollzieht sich ein schneller Wechsel vom vorwiegend textbasierten zum multimediabasierten Internet. Die weitverbreiteten IEEE 802.11 Drahtlosnetzwerke sind vielversprechende Kandidaten, um das Internet für Nutzer überall, jederzeit und auf jedem Gerät verfügbar zu machen. Die Unterstützung gruppenorientierter Echtzeit-Dienste in drahtlosen lokalen Netzen ist jedoch immer noch eine Herausforderung. Das liegt daran, dass aktuelle Protokolle keinen Multicast unterstützen. Sie senden Multicast-Pakete vielmehr in einer "Open Loop"-Strategie als Broadcast-Pakete, d. h. ohne jegliche Rückmeldung (feedback) oder Paketwiederholungen. In der vorliegenden Arbeit, anders als in den auf Teilnehmereinzelabfragen (polling) basierenden Ansätzen, die unter langen Verzögerungen, geringer Skalierbarkeit und geringer Effizienz leiden, versuchen wir, Multicast-Feedback bestehend aus positiven (ACK) und negativen Bestätigungen (NACK) auf MAC-Layer im selben Zeitfenster zu bündeln. Die übrigen Empfänger können NACK-Frames senden, um das ACK des Leaders zu zerstören und Paketwiederholungen zu veranlassen. Basierend auf einem Feedback-Jamming Schema schlagen wir zwei MAC-Layer-Protokolle für den Fehlerschutz im Multicast vor: Das SEQ-getriebene Leader Based Protocol (SEQ-LBP) und das Hybrid Leader Based Protocol (HLBP). SEQ-LBP ist eines Automatic Repeat reQuest (ARQ) Schema. HLBP kombiniert ARQ und paketbasierte Forward Error Correction (FEC). Wir evaluieren die Leistungsfähigkeit von ACK/NACK jamming, SEQ-LBP und HLBP durch Analysis, Simulationen in NS-2, sowie Experimenten in einer realen Testumgebung mit handelsüblichen WLAN-Karten. Die Testergebnisse bestätigen die Anwendbarkeit der Feedback-Jamming Schemata und die herausragende Leistungsfähigkeit der vorgestellten Protokolle SEQ-LBP und HLBP. SEQ-LBP ist durch seine kurze Verzögerung, seine Effektivität und seine Einfachheit für kleine Multicast-Gruppen nützlich, während HLBP auf Grund seiner hohen Effizienz und Skalierbarkeit im Bezug auf die Größe der Empfänger eher in großen Multicast-Gruppen anzuwenden ist

Self-concatenated coding for wireless communication systems

In this thesis, we have explored self-concatenated coding schemes that are designed for transmission over Additive White Gaussian Noise (AWGN) and uncorrelated Rayleigh fading channels. We designed both the symbol-based Self-ConcatenatedCodes considered using Trellis Coded Modulation (SECTCM) and bit-based Self- Concatenated Convolutional Codes (SECCC) using a Recursive Systematic Convolutional (RSC) encoder as constituent codes, respectively. The design of these codes was carried out with the aid of Extrinsic Information Transfer (EXIT) charts. The EXIT chart based design has been found an efficient tool in finding the decoding convergence threshold of the constituent codes. Additionally, in order to recover the information loss imposed by employing binary rather than non-binary schemes, a soft decision demapper was introduced in order to exchange extrinsic information withthe SECCC decoder. To analyse this information exchange 3D-EXIT chart analysis was invoked for visualizing the extrinsic information exchange between the proposed Iteratively Decoding aided SECCC and soft-decision demapper (SECCC-ID). Some of the proposed SECTCM, SECCC and SECCC-ID schemes perform within about 1 dB from the AWGN and Rayleigh fading channels’ capacity. A union bound analysis of SECCC codes was carried out to find the corresponding Bit Error Ratio (BER) floors. The union bound of SECCCs was derived for communications over both AWGN and uncorrelated Rayleigh fading channels, based on a novel interleaver concept.Application of SECCCs in both UltraWideBand (UWB) and state-of-the-art video-telephone schemes demonstrated its practical benefits.In order to further exploit the benefits of the low complexity design offered by SECCCs we explored their application in a distributed coding scheme designed for cooperative communications, where iterative detection is employed by exchanging extrinsic information between the decoders of SECCC and RSC at the destination. In the first transmission period of cooperation, the relay receives the potentially erroneous data and attempts to recover the information. The recovered information is then re-encoded at the relay using an RSC encoder. In the second transmission period this information is then retransmitted to the destination. The resultant symbols transmitted from the source and relay nodes can be viewed as the coded symbols of a three-component parallel-concatenated encoder. At the destination a Distributed Binary Self-Concatenated Coding scheme using Iterative Decoding (DSECCC-ID) was employed, where the two decoders (SECCC and RSC) exchange their extrinsic information. It was shown that the DSECCC-ID is a low-complexity scheme, yet capable of approaching the Discrete-input Continuous-output Memoryless Channels’s (DCMC) capacity.Finally, we considered coding schemes designed for two nodes communicating with each other with the aid of a relay node, where the relay receives information from the two nodes in the first transmission period. At the relay node we combine a powerful Superposition Coding (SPC) scheme with SECCC. It is assumed that decoding errors may be encountered at the relay node. The relay node then broadcasts this information in the second transmission period after re-encoding it, again, using a SECCC encoder. At the destination, the amalgamated block of Successive Interference Cancellation (SIC) scheme combined with SECCC then detects and decodes the signal either with or without the aid of a priori information. Our simulation results demonstrate that the proposed scheme is capable of reliably operating at a low BER for transmission over both AWGN and uncorrelated Rayleigh fading channels. We compare the proposed scheme’s performance to a direct transmission link between the two sources having the same throughput

Scalable and rate adaptive wireless multimedia multicast

The methods that are described in this work enable highly efficient audio-visual streaming over wireless digital communication systems to an arbitrary number of receivers. In the focus of this thesis is thus point-to-multipoint transmission at constrained end-to-end delay. A fundamental difference as compared to point-to-point connections between exactly two communicating sending and receiving stations is in conveying information about successful or unsuccessful packet reception at the receiver side. The information to be transmitted is available at the sender, whereas the information about successful reception is only available to the receiver. Therefore, feedback about reception from the receiver to the sender is necessary. This information may be used for simple packet repetition in case of error, or adaptation of the bit rate of transmission to the momentary bit rate capacity of the channel, or both. This work focuses on the single transmission (including retransmissions) of data from one source to multiple destinations at the same time. A comparison with multi-receiver sequentially redundant transmission systems (simulcast MIMO) is made. With respect to feedback, this work considers time division multiple access systems, in which a single channel is used for data transmission and feedback. Therefore, the amount of time that can be spent for transmitting feedback is limited. An increase in time used for feedback transmissions from potentially many receivers results in a decrease in residual time which is usable for data transmission. This has direct impact on data throughput and hence, the quality of service. In the literature, an approach to reduce feedback overhead which is based on simultaneous feedback exists. In the scope of this work, simultaneous feedback implies equal carrier frequency, bandwidth and signal shape, in this case orthogonal frequency-division multiplex signals, during the event of the herein termed feedback aggregation in time. For this scheme, a constant amount of time is spent for feedback, independent of the number of receivers giving feedback about reception. Therefore, also data throughput remains independent of the number of receivers. This property of audio-visual digital transmission is taken for granted for statically configured, single purpose systems, such as terrestrial television. In the scope of this work are, however, multi-user and multi-purpose digital communication networks. Wireless LANs are a well-known example and are covered in detail herein. In suchlike systems, it is of great importance to remain independent of the number of receivers, as otherwise the service of ubiquitous digital connectivity is at the risk of being degraded. In this regard, the thesis at hand elaborates at what bit rates audio-visual transmission to multiple receivers may take place in conjunction with feedback aggregation. It is shown that the scheme achieves a multi-user throughput gain when used in conjunction with adaptivity of the bit rate to the channel. An assumption is the use of an ideal overlay packet erasure correcting code in this case. Furthermore, for delay constrained transmission, such as in so-called live television, throughput bit rates are examined. Applications have to be tolerant to a certain level of residual error in case of delay constrained transmission. Improvement of the rate adaptation algorithm is shown to increase throughput while residual error rates are decreased. Finally, with a consumer hardware prototype for digital live-TV re-distribution in the local wireless network, most of the mechanisms as described herein can be demonstrated.Die in vorliegender Arbeit aufgezeigten Methoden der paketbasierten drahtlosen digitalen Kommunikation ermöglichen es, Fernsehinhalte, aber auch audio-visuelle Datenströme im Allgemeinen, bei hoher Effizienz an beliebig große Gruppen von Empfängern zu verteilen. Im Fokus dieser Arbeit steht damit die Punkt- zu Mehrpunktübertragung bei begrenzter Ende-zu-Ende Verzögerung. Ein grundlegender Unterschied zur Punkt-zu-Punkt Verbindung zwischen genau zwei miteinander kommunizierenden Sender- und Empfängerstationen liegt in der Übermittlung der Information über erfolgreichen oder nicht erfolgreichen Paketempfang auf Seite der Empfänger. Da die zu übertragende Information am Sender vorliegt, die Information über den Erfolg der Übertragung jedoch ausschließlich beim jeweiligen Empfänger, muss eine Erfolgsmeldung auf dem Rückweg von Empfänger zu Sender erfolgen. Diese Information wird dann zum Beispiel zur einfachen Paketwiederholung im nicht erfolgreichen Fall genutzt, oder aber um die Übertragungsrate an die Kapazität des Kanals anzupassen, oder beides. Grundsätzlich beschäftigt sich diese Arbeit mit der einmaligen, gleichzeitigen Übertragung von Information (einschließlich Wiederholungen) an mehrere Empfänger, wobei ein Vergleich zu an mehrere Empfänger sequentiell redundant übertragenden Systemen (Simulcast MIMO) angestellt wird. In dieser Arbeit ist die Betrachtung bezüglich eines Rückkanals auf Zeitduplexsysteme beschränkt. In diesen Systemen wird der Kanal für Hin- und Rückweg zeitlich orthogonalisiert. Damit steht für die Übermittlung der Erfolgsmeldung eine beschränkte Zeitdauer zur Verfügung. Je mehr an Kanalzugriffszeit für die Erfolgsmeldungen der potentiell vielen Empfänger verbraucht wird, desto geringer wird die Restzeit, in der dann entsprechend weniger audio-visuelle Nutzdaten übertragbar sind, was sich direkt auf die Dienstqualität auswirkt. Ein in der Literatur weniger ausführlich betrachteter Ansatz ist die gleichzeitige Übertragung von Rückmeldungen mehrerer Teilnehmer auf gleicher Frequenz und bei identischer Bandbreite, sowie unter Nutzung gleichartiger Signale (hier: orthogonale Frequenzmultiplexsignalformung). Das Schema wird in dieser Arbeit daher als zeitliche Aggregation von Rückmeldungen, engl. feedback aggregation, bezeichnet. Dabei wird, unabhängig von der Anzahl der Empfänger, eine konstante Zeitdauer für Rückmeldungen genutzt, womit auch der Datendurchsatz durch zusätzliche Empfänger nicht notwendigerweise sinkt. Diese Eigenschaft ist aus statisch konfigurierten und für einen einzigen Zweck konzipierten Systemen, wie z. B. der terrestrischen Fernsehübertragung, bekannt. In dieser Arbeit werden im Gegensatz dazu jedoch am Beispiel von WLAN Mehrzweck- und Mehrbenutzersysteme betrachtet. Es handelt sich in derartigen Systemen zur digitalen Datenübertragung dabei um einen entscheidenden Vorteil, unabhängig von der Empfängeranzahl zu bleiben, da es sonst unweigerlich zu Einschränkungen in der Güte der angebotenen Dienstleistung der allgegenwärtigen digitalen Vernetzung kommen muss. Vorliegende Arbeit zeigt in diesem Zusammenhang auf, welche Datenraten unter Benutzung von feedback aggregation in der Verteilung an mehrere Empfänger und in verschiedenen Szenarien zu erreichen sind. Hierbei zeigt sich, dass das Schema im Zusammenspiel mit einer Adaption der Datenrate an den Übertragungskanal inhärent einen Datenratengewinn durch Mehrbenutzerempfang zu erzielen vermag, wenn ein überlagerter idealer Paketauslöschungsschutz-Code angenommen wird. Des weiteren wird bei der Übertragung mit zeitlich begrenzter Ausführungsdauer, z. B. dem sogenannten Live-Fernsehen, aufgezeigt, wie sich die erreichbare Datenrate reduziert und welche Restfehlertoleranz an die Übertragung gestellt werden muss. Hierbei wird ebenso aufgezeigt, wie sich durch Verbesserung der Ratenadaption erstere erhöhen und zweitere verringern lässt. An einem auf handelsüblichen Computer-Systemen realisiertem Prototypen zur Live-Fernsehübertragung können die hierin beschriebenen Mechanismen zu großen Teilen gezeigt werden

Link level performance evaluation and link abstraction for LTE/LTE-advanced downlink

Els objectius principals d'aquesta tesis són l'avaluació del rendiment a nivell d'enllaç i l'estudi de l'abstracció de l'enllaç pel LTE/LTE-Advanced DL. S’ha desenvolupat un simulador del nivell d'enllaç E-UTRA DL basat en la tecnologia MIMO-OFDM. Es simulen els errors d'estimació de canal amb un model d'error de soroll additiu Gaussià anomenat CEEM. El resultat d'aquest simulador serveix per avaluar el rendiment a nivell d'enllaç del LTE/LTE-Advanced DL en diferents entorns . La idea bàsica dels mètodes d'abstracció de l'enllaç és mapejar el vector de SNRs de les subportadores a un valor escalar, l'anomenada ESNR, la qual és usada per a predir la BLER. Proposem un innovador mètode d'abstracció de l'enllaç que pot predir la BLER amb bona precisió en esvaïments multicamí i que inclouen els efectes de les retransmissions HARQ. El mètode proposat es basa amb l'estimació de la informació mútua entre els bits transmesos i els LLRs rebuts.The main objectives of this dissertation are the evaluation of the link level performance and the study of link abstraction for LTE/LTE-Advanced DL. An E-UTRA DL link level simulator has been developed based on MIMO-OFDM technology. We simulate channel estimation errors by a Gaussian additive noise error model called CEEM. The result of this simulator serves to evaluate the MIMO-OFDM LTE/LTE-Advanced DL link level performance in different environments. The basic idea of link abstraction methods is to map the vector of the subcarrier SNRs to a single scalar, the ESNR, which is then used to predict the BLER. We propose a novel link abstraction method that can predict the BLER with good accuracy in multipath fading and including the effects of HARQ retransmissions. The proposed method is based on estimating the mutual information between the transmitted bits and the received LLRs.Postprint (published version