Self-concatenated code design and its application in power-efficient cooperative communications

In this tutorial, we have focused on the design of binary self-concatenated coding schemes with the help of EXtrinsic Information Transfer (EXIT) charts and Union bound analysis. The design methodology of future iteratively decoded self-concatenated aided cooperative communication schemes is presented. In doing so, we will identify the most important milestones in the area of channel coding, concatenated coding schemes and cooperative communication systems till date and suggest future research directions

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

Near-capacity MIMOs using iterative detection

In this thesis, Multiple-Input Multiple-Output (MIMO) techniques designed for transmission over narrowband Rayleigh fading channels are investigated. Specifically, in order to providea diversity gain while eliminating the complexity of MIMO channel estimation, a Differential Space-Time Spreading (DSTS) scheme is designed that employs non-coherent detection. Additionally, in order to maximise the coding advantage of DSTS, it is combined with Sphere Packing (SP) modulation. The related capacity analysis shows that the DSTS-SP scheme exhibits a higher capacity than its counterpart dispensing with SP. Furthermore, in order to attain additional performance gains, the DSTS system invokes iterative detection, where the outer code is constituted by a Recursive Systematic Convolutional (RSC) code, while the inner code is a SP demapper in one of the prototype systems investigated, while the other scheme employs a Unity Rate Code (URC) as its inner code in order to eliminate the error floor exhibited by the system dispensing with URC. EXIT charts are used to analyse the convergence behaviour of the iteratively detected schemes and a novel technique is proposed for computing the maximum achievable rate of the system based on EXIT charts. Explicitly, the four-antenna-aided DSTSSP system employing no URC precoding attains a coding gain of 12 dB at a BER of 10-5 and performs within 1.82 dB from the maximum achievable rate limit. By contrast, the URC aidedprecoded system operates within 0.92 dB from the same limit.On the other hand, in order to maximise the DSTS system’s throughput, an adaptive DSTSSP scheme is proposed that exploits the advantages of differential encoding, iterative decoding as well as SP modulation. The achievable integrity and bit rate enhancements of the system are determined by the following factors: the specific MIMO configuration used for transmitting data from the four antennas, the spreading factor used and the RSC encoder’s code rate.Additionally, multi-functional MIMO techniques are designed to provide diversity gains, multiplexing gains and beamforming gains by combining the benefits of space-time codes, VBLASTand beamforming. First, a system employing Nt=4 transmit Antenna Arrays (AA) with LAA number of elements per AA and Nr=4 receive antennas is proposed, which is referred to as a Layered Steered Space-Time Code (LSSTC). Three iteratively detected near-capacity LSSTC-SP receiver structures are proposed, which differ in the number of inner iterations employed between the inner decoder and the SP demapper as well as in the choice of the outer code, which is either an RSC code or an Irregular Convolutional Code (IrCC). The three systems are capable of operating within 0.9, 0.4 and 0.6 dB from the maximum achievable rate limit of the system. A comparison between the three iteratively-detected schemes reveals that a carefully designed two-stage iterative detection scheme is capable of operating sufficiently close to capacity at a lower complexity, when compared to a three-stage system employing a RSC or a two-stage system using an IrCC as an outer code. On the other hand, in order to allow the LSSTC scheme to employ less receive antennas than transmit antennas, while still accommodating multiple users, a Layered Steered Space-Time Spreading (LSSTS) scheme is proposed that combines the benefits of space-time spreading, V-BLAST, beamforming and generalised MC DS-CDMA. Furthermore, iteratively detected LSSTS schemes are presented and an LLR post-processing technique is proposed in order to improve the attainable performance of the iteratively detected LSSTS system.Finally, a distributed turbo coding scheme is proposed that combines the benefits of turbo coding and cooperative communication, where iterative detection is employed by exchanging extrinsic information between the decoders of different single-antenna-aided users. Specifically, the effect of the errors induced in the first phase of cooperation, where the two users exchange their data, on the performance of the uplink in studied, while considering different fading channel characteristics

Cross-layer hybrid automatic repeat request error control with turbo processing for wireless system

The increasing demand for wireless communication system requires an efficient design in wireless communication system. One of the main challenges is to design error control mechanism in noisy wireless channel. Forward Error Correction (FEC) and Automatic Repeat reQuest (ARQ) are two main error control mechanisms. Hybrid ARQ allows the use of either FEC or ARQ when required. The issues with existing Hybrid ARQ are reliability, complexity and inefficient design. Therefore, the design of Hybrid ARQ needs to be further improved in order to achieve performance close to the Shannon capacity. The objective of this research is to develop a Cross-Layer Design Hybrid ARQ defined as CLD_ARQ to further minimize error in wireless communication system. CLD_ARQ comprises of three main stages. First, a low complexity FEC defined as IRC_FEC for error detection and correction has been developed by using Irregular Repetition Code (IRC) with Turbo processing. The second stage is the enhancement of IRC_FEC defined as EM_IRC_FEC to improve the reliability of error detection by adopting extended mapping. The last stage is the development of efficient CLD_ARQ to include retransmission for error correction that exploits EM_IRC_FEC and ARQ. In the proposed design, serial iterative decoding and parallel iterative decoding are deployed in the error detection and correction. The performance of the CLD_ARQ is evaluated in the Additive White Gaussian Noise (AWGN) channel using EXtrinsic Information Transfer (EXIT) chart, bit error rate (BER) and throughput analysis. The results show significant Signal-to-Noise Ratio (SNR) gain from the theoretical limit at BER of 10-5. IRC_FEC outperforms Recursive Systematic Convolutional Code (RSCC) by SNR gain up to 7% due to the use of IRC as a simple channel coding code. The usage of CLD_ARQ enhances the SNR gain by 53% compared to without ARQ due to feedback for retransmission. The adoption of extended mapping in the CLD_ARQ improves the SNR gain up to 50% due to error detection enhancement. In general, the proposed CLD_ARQ can achieve low BER and close to the Shannon‘s capacity even in worse channel condition

On Bit-interleaved Coded Modulation with QAM Constellations

Bit-interleaved coded modulation (BICM) is a flexible modulation/coding scheme which allows the designer to choose a modulation constellation independently of the coding rate. This is because the output of the channel encoder and the input to the modulator are separated by a bit-level interleaver. In order to increase spectral efficiency, BICM can be combined with high-order modulation schemes such as quadrature amplitude modulation (QAM) or phase shift keying. BICM is particularly well suited for fading channels, and it only introduces a small penalty in terms of channel capacity when compared to the coded modulation capacity for both additive white Gaussian noise (AWGN) and fading channels. Additionally, if the so-called BICM with iterative decoding (BICM-ID) is used, the demapper and decoder iteratively exchange information, improving the system performance. <p>At the receiver's side of BICM, the reliability metrics are calculated for the coded bits under the form of logarithmic likelihood ratios, or simply L-values. These metrics are then deinterleaved and further used by the soft-input channel decoder. This thesis deals with the probabilistic characterization of the L-values calculated by the demapper when BICM is used in conjunction with high order QAM schemes. Three contributions are included in this thesis.</p> <p>In <b>Paper A</b> the issue of the probabilistic modelling of the extrinsic L-values for BICM-ID is addressed. Starting with a simple piece-wise linear model of the L-values obtained via the max-log approximation, expressions for the probability density functions (PDFs) for Gray-mapped 16-QAM are found. The developed analytical expressions are then used to efficiently compute the so-called extrinsic information transfer functions of the demapper, and they are also compared with the histograms of the L-values obtained through numerical simulations.</p> <p>In <b>Paper B</b> closed-form expressions for the PDFs of the L-values in BICM with Gray mapped QAM constellations are developed. Based on these expressions, two simple Gaussian mixture approximations that are analytically tractable are also proposed. The developments are used to efficiently calculate the BICM channel capacity and to develop bounds on the coded bit-error rate when a convolutional code is used. The coded performance of an hybrid automatic repeat request based on constellation rearrangement is also evaluated.</p> <p>In <b>Paper C</b> closed-form expressions for the PDFs of the L-values in BICM transmissions with Gray-mapped QAM constellations over fully-interleaved fading channels are derived. The results are particularized for a Rayleigh fading channel, however, developments for the general case of a Nakagami-$m$ case are also included. Using the developed expressions, the performance of BICM transmissions using convolutional and turbo codes is efficiently evaluated. The BICM channel capacity for different fading channels and constellation sizes is also calculated.</p

Digital Audio Broadcast: Modulation, Transmission & Performance Analysis

Radio broadcasting technology has evolved rapidly over the last few years due to ever increasing demands for as high quality sound services with ancillary data transmission in mobile environment. In order to accomplish this, Members of European Broadcasting Union (EBU), the European Telecommunications Standards Institute (ETSI) and International Telecommunications Union (ITU-R) developed a completely new digital radio broadcasting technology called the Eureka- 147 Digital Audio Broadcasting (DAB) system which improves the overall broadcasting performance by delivering near CD quality audio and data services in mobile receivers along with efficient use of the available radio frequency spectrum. Digital Audio Broadcasting (DAB) system developed within the Eureka 147 Project is a new digital radio technology for broadcasting radio stations that provides high-quality audio and data services to both fixed and mobile receivers. The system uses COFDM technology that combats the effect of multipath fading & ISI and makes it spectrally more efficient compared with existing AM/FM systems. This project presents the performance analysis of Eureka-147 DAB system. DAB transmission mode-II is implemented first and then extended successfully to other modes. A frame-based processing is used in this study. Performance studies for AWGN, Rayleigh and Rician channels have been conducted. For all studies BER has been used as performance criteria. This project also discusses issues related to system performance using concatenated coding technique, including the outer Block code, the inner convolutional code, outer BCH code and the inner convolutional code

Constellation Shaping for WDM systems using 256QAM/1024QAM with Probabilistic Optimization

In this paper, probabilistic shaping is numerically and experimentally investigated for increasing the transmission reach of wavelength division multiplexed (WDM) optical communication system employing quadrature amplitude modulation (QAM). An optimized probability mass function (PMF) of the QAM symbols is first found from a modified Blahut-Arimoto algorithm for the optical channel. A turbo coded bit interleaved coded modulation system is then applied, which relies on many-to-one labeling to achieve the desired PMF, thereby achieving shaping gain. Pilot symbols at rate at most 2% are used for synchronization and equalization, making it possible to receive input constellations as large as 1024QAM. The system is evaluated experimentally on a 10 GBaud, 5 channels WDM setup. The maximum system reach is increased w.r.t. standard 1024QAM by 20% at input data rate of 4.65 bits/symbol and up to 75% at 5.46 bits/symbol. It is shown that rate adaptation does not require changing of the modulation format. The performance of the proposed 1024QAM shaped system is validated on all 5 channels of the WDM signal for selected distances and rates. Finally, it was shown via EXIT charts and BER analysis that iterative demapping, while generally beneficial to the system, is not a requirement for achieving the shaping gain.Comment: 10 pages, 12 figures, Journal of Lightwave Technology, 201

Architectures multi-Asip pour turbo récepteur flexible

Rapidly evolving wireless standards use modern techniques such as turbo codes, Bit Interleaved coded Modulation (BICM), high order QAM constellation, Signal Space Diversity (SSD), Multi-Input Multi-Output (MIMO) Spatial Multiplexing (SM) and Space Time Codes (STC) with different parameters for reliable high rate data transmissions. Adoption of such techniques in the transmitter can impact the receiver architecture in three ways: (1) the complex processing related to advanced techniques such as turbo codes, encourage to perform iterative processing in the receiver to improve error rate performance (2) to satisfy high throughput requirement for an iterative receiver, parallel processing is mandatory and finally (3) to allow the support of different techniques and parameters imposed, programmable yet high throughput hardware processing elements are required. In this thesis, to address the high throughput requirement with turbo processing, first of all a study of parallelism on turbo decoding is extended for turbo demodulation and turbo equalization. Based on the results acquired from the parallelism study a flexible high throughput heterogeneous multi-ASIP NoC based unified turbo receiver is proposed. The proposed architecture fulfils the target requirements in a way that: (a) Application Specific Instruction-set Processor (ASIP) exploits metric generation level parallelism and implements the required flexibility, (b) throughputs beyond the capacity of single ASIP in a turbo process are achieved through multiple ASIP elements implementing sub-block parallelism and shuffled processing and finally (c) Network on Chip is used to handle communication conflicts during parallel processing of multiple ASIPs. In pursuit to achieve a hardware model of the proposed architecture two ASIPs are conceived where the first one, namely EquASIP, is dedicated for MMSE-IC equalization and provides a flexible solution for multiple MIMO techniques adopted in multiple wireless standards with a capability to work in turbo equalization context. The second ASIP, named as DemASIP, is a flexible demapper which can be used in MIMO or single antenna environment for any modulation till 256-QAM with or without iterative demodulation. Using available TurbASIP and NoC components, the thesis concludes on an FPGA prototype of heterogeneous multi-ASIP NoC based unified turbo receiver which integrates 9 instances of 3 different ASIPs with 2 NoCs.Les normes de communication sans fil, sans cesse en évolution, imposent l'utilisation de techniques modernes telles que les turbocodes, modulation codée à entrelacement bit (BICM), constellation MAQ d'ordre élevé, diversité de constellation (SSD), multiplexage spatial et codage espace-temps multi-antennes (MIMO) avec des paramètres différents pour des transmissions fiables et de haut débit. L'adoption de ces techniques dans l'émetteur peut influencer l'architecture du récepteur de trois façons: (1) les traitement complexes relatifs aux techniques avancées comme les turbocodes, encourage à effectuer un traitement itératif dans le récepteur pour améliorer la performance en termes de taux d'erreur (2) pour satisfaire l'exigence de haut débit avec un récepteur itératif, le recours au parallélisme est obligatoire et enfin (3) pour assurer le support des différentes techniques et paramètres imposées, des processeurs de traitement matériel flexibles, mais aussi de haute performance, sont nécessaires. Dans cette thèse, pour répondre aux besoins de haut débit dans un contexte de traitement itératif, tout d'abord une étude de parallélisme sur le turbo décodage a été étendue aux applications de turbo démodulation et turbo égalisation. Partant des résultats obtenus à partir de l'étude du parallélisme, un récepteur itératif unifié basé sur un modèle d'architecture multi-ASIP hétérogène intégrant un réseau sur puce (NoC) a été proposé. L'architecture proposée répond aux exigences visées d'une manière où: (a) le concept de processeur à jeu d'instruction dédié à l'application (ASIP) exploite le parallélisme du niveau de génération de métriques et met en oeuvre la flexibilité nécessaire, (b) les débits au-delà de la capacité d'un seul ASIP dans un processus itératif sont obtenus au moyen de multiples ASIP implémentant le parallélisme de sous-blocs et le traitement combiné et enfin (c) le concept de réseau sur puce (NoC) est utilisé pour gérer les conflits de communication au cours du traitement parallèle itératif multi-ASIP. Dans le but de parvenir à un modèle matériel de l'architecture proposée, deux ASIP ont été conçus où le premier, nommé EquASIP, est dédié à l'égalisation MMSE-IC et fournit une solution flexible pour de multiples techniques multi-antennes adoptés dans plusieurs normes sans fil avec la capacité de travailler dans un contexte de turbo égalisation. Le deuxième ASIP, nommé DemASIP, est un démappeur flexible qui peut être utilisé dans un environnement multi-antennes et pour tout type de modulation jusqu'à MAQ-256 avec ou sans démodulation itérative. En intégrant ces ASIP, en plus des NoC et TurbASIP disponibles à Télécom Bretagne, la thèse conclut sur un prototype FPGA d'un récepteur itératif unifié multi-ASIP qui intègre 9 coeurs de 3 différents types d'ASIP avec 2 NoC

On distributed coding, quantization of channel measurements and faster-than-Nyquist signaling

This dissertation considers three different aspects of modern digital communication systems and is therefore divided in three parts. The first part is distributed coding. This part deals with source and source- channel code design issues for digital communication systems with many transmitters and one receiver or with one transmitter and one receiver but with side information at the receiver, which is not available at the transmitter. Such problems are attracting attention lately, as they constitute a way of extending the classical point-to-point communication theory to networks. In this first part of this dissertation, novel source and source-channel codes are designed by converting each of the considered distributed coding problems into an equivalent classical channel coding or classical source-channel coding problem. The proposed schemes come very close to the theoretical limits and thus, are able to exhibit some of the gains predicted by network information theory. In the other two parts of this dissertation classical point-to-point digital com- munication systems are considered. The second part is quantization of coded chan- nel measurements at the receiver. Quantization is a way to limit the accuracy of continuous-valued measurements so that they can be processed in the digital domain. Depending on the desired type of processing of the quantized data, different quantizer design criteria should be used. In this second part of this dissertation, the quantized received values from the channel are processed by the receiver, which tries to recover the transmitted information. An exhaustive comparison of several quantization cri- teria for this case are studied providing illuminating insight for this quantizer design problem. The third part of this dissertation is faster-than-Nyquist signaling. The Nyquist rate in classical point-to-point bandwidth-limited digital communication systems is considered as the maximum transmission rate or signaling rate and is equal to twice the bandwidth of the channel. In this last part of the dissertation, we question this Nyquist rate limitation by transmitting at higher signaling rates through the same bandwidth. By mitigating the incurred interference due to the faster-than-Nyquist rates, gains over Nyquist rate systems are obtained

Data reliability and error correction for NAND Flash Memory System

NAND flash memory has been widely used for data storage due to its high density, high throughput, and low power. However, as the flash memory scales to smaller process technologies and stores more bits per cell, its reliability is decreasing. The error correction coding can be used to significantly improve the data reliability; nevertheless, the advanced ECCs such as low-density parity-check (LDPC) codes generally demand soft decisions while NAND flash memory channel provides hard-decisions only. Extracting the soft information requires the accurate characterization of flash memory channel and the effective design of coding schemes. To this end, we have presented a novel LDPC-TCM coding scheme for the Multilevel Cell (MLC) flash memories. The a posteriori TCM decoding algorithm is used in the scheme to generate soft information, which is fed to the LDPC decoder for further correction of data bits. It has been demonstrated that the proposed scheme can achieve higher error correction performance than the traditional hard-decisions based flash coding algorithms, and is feasible in the design practice. Further with the LDPC-TCM, we believe it is important to characterize the flash memory channel and investigate a method to calculate the soft decision for each bit, with the available channel outputs. We studied the various noises and interferences occurring in the memory channel and mathematically formulated the probability density function of the overall noise distribution. Based on the results we derived the final distribution for the cell threshold voltages, which can be used to instruct the calculation of soft decisions. The discoveries on the theoretical level have been demonstrated to be consistent with the real channel behaviours. The channel characterization and model provided in this dissertation can enable more design of soft-decisions based ECCs for future NAND flash memories. The data pattern processing algorithm deals with the write patterns and targets to lower the proportion of patterns that would introduce data errors. On the other hand, the voltages applied to the memory cells charges the MOSFET capacitances frequently on programming these data patterns, leading to the power problem. The high energy consumption and current spikes also cause reliability issue to the data stored in the flash memory. This dissertation proposes a write pattern formatting algorithm (WPFA) attempting to solve the two problems together. We have designed and implemented the algorithm and evaluated its performance through both the software simulations and hardware synthesis