Synchronization for capacity -approaching coded communication systems

The dissertation concentrates on synchronization of capacity approaching error-correction codes that are deployed in noisy channels with very low signal-to-noise ratio (SNR). The major topics are symbol timing synchronization and frame synchronization.;Capacity-approaching error-correction codes, like turbo codes and low-density parity-check (LDPC) codes, are capable of reaching very low bit error rates and frame error rates in noisy channels by iterative decoding. To fully achieve the potential decoding capability of Turbo codes and LDPC codes, proper symbol timing synchronization, frame synchronization and channel state estimation are required. The dissertation proposes a joint estimator of symbol time delay and channel SNR for symbol timing recovery, and a maximum a posteriori (MAP) frame synchronizer for frame synchronization.;Symbol timing recovery is implemented by sampling and interpolation. The received signal is sampled multiple times per symbol period with unknown delay and unknown SNR. A joint estimator estimates the time delay and the SNR. The signal is rebuilt by interpolating available samples using estimated time delay. The intermediate decoding results enable decision-feedback estimation. The estimates of time delay and SNR are refined by iterative processing. This refinement improves the system performance significantly.;Usually the sampling rate is assumed to be a strict integer multiple of the symbol rate. However, in a practical system the local oscillators in the transmitter and the receiver may have random drifts. Therefore the sampling rate is no longer an exact multiple of the symbol rate, and the sampling time follows a random walk. This random walk may harm the system performance severely. The dissertation analyzes the effect of random time walks and proposes to mitigate the effect by overlapped sliding windows and iterative processing.;Frame synchronization is required to find the correct boundaries of codewords. MAP frame synchronization in the sense of minimizing the frame sync failure rate is investigated. The MAP frame synchronizer explores low-density parity-check attributes of the capacity-approaching codes. The accuracy of frame synchronization is adequate for considered coded systems to work reliably under very low SNR

Synchronous and asynchronous sequential symbol synchronizers

In this work, we present two synchronizer groups: the synchronous and the asynchronous. The synchronous group is based in forward logic with flip flops and the asynchronous group is based in forward logic with delay line feedback. In each group we consider two versions: the manual and the automatic. The main objective is to study the two groups, each one with two versions and to observe its jitter performance as function of the noise

Synchronization in all-digital QAM receivers

The recent advance in Field Programmable Gate Array (FPGA) technology has been largely embraced by the communication industry, which views this technology as an effective and economical alternative to the design of Application Specific Integrated Circuits (ASICs). The primary reasons for switching to FPGAs are lower development and non-recurring engineering costs, the flexibility to design to a preliminary standard and adapt the design as the standard evolves, as well as the option of performing software updates in the field. A sector with strong interest in FPGAs is the coaxial cable TV/Internet distribution industry. The creation of soft preliminary standards by the standards organization governing the industry has been the main catalyst for the massive adoption of FPGAs by small to medium size companies, which see this technology as an opportunity to compete in this open market. Both the circuit speed and the economy of FPGA technology depend upon using algorithms that map efficiently into its fabric. Often it is prudent to sacrifice performance to improve either clock speed or economy when developing with FPGAs. The purpose of this research is to both revise and devise synchronization algorithms / structures for cable digital receivers that are to be implemented in FPGA. The main communication scheme used by the coaxial cable distribution industry is digital Quadrature Amplitude Modulation (QAM). The problem of synchronizing to the QAM signal in the receiver is not a new topic and several synchronization-related circuits, which were devised with ASICs implementation in mind, can be found in the open literature. Of interest in this thesis is the non-data-aided digital timing synchronizer that was proposed by D'Andrea to recover timing with no knowledge of the transmitted data. Accurate timing estimation was achieved by reshaping the received signal with a prefilter prior to estimating the timing. A problem with D'Andrea's synchronizer is that the prefilter for reshaping the signal is a relatively long Finite Impulse Response (FIR) filter, whose implementation requires a large number of multipliers. This may not have been an issue with ASICs in as much as the number of hardwired multipliers on a chip is not limited as it is in an FPGA chip. One contribution in this research is to propose an alternative to D'Andrea's synchronizer by replacing the long FIR filter with two single-pole Infinite Impulse Response (IIR) filters that are directly placed inside the timing recovery loop. This novel architecture, which drastically reduces the number of multipliers, is well suited for FPGA implementation. Non-data-aided feedforward synchronizers, which use the same prefilter as D'Andrea's synchronizer, have been receiving significant attention in recent years. Detailed performance analysis for these synchronizers can be found in the open literature. These synchronizers have the advantage of using a feedfordward structure rather than a feedback structure, as it is the case in D'Andrea's synchronizer, to estimate the timing. While D'Andrea's synchronizer has an advantage in performance over a non-data-aided feedforward synchronizer, this has not been reported in the literature. In this thesis a second contribution consists of thoroughly analyzing the steady state timing jitter in D'Andrea synchronizer by deriving a closed-form expression for the noise power spectrum and a simple equation to estimate the timing jitter variance. A third contribution is a novel low-complexity and fast acquisition coherent detector for the detection of Quadrature Phase Shift Keying (QPSK) (i.e., 4-QAM) symbols. This detector performs carrier phase synchronization much faster than a conventional coherent detector. The acquisition time is comparable to that of a differential detector. The fast acquisition comes at the expense of phase jitter, and the end result is a 1 dB performance loss over theoretical coherent detection. This detector can be used in place of the differential detector with no economic penalty. Doing so yields a performance advantage of about 2 dB over differential detection

Design and implementation of a downlink MC-CDMA receiver

Cette thèse présente une étude d'un système complet de transmission en liaison descendante utilisant la technologie multi-porteuse avec l'accès multiple par division de code (Multi-Carrier Code Division Multiple Access, MC-CDMA). L'étude inclut la synchronisation et l'estimation du canal pour un système MC-CDMA en liaison descendante ainsi que l'implémentation sur puce FPGA d'un récepteur MC-CDMA en liaison descendante en bande de base. Le MC-CDMA est une combinaison de la technique de multiplexage par fréquence orthogonale (Orthogonal Frequency Division Multiplexing, OFDM) et de l'accès multiple par répartition de code (CDMA), et ce dans le but d'intégrer les deux technologies. Le système MC-CDMA est conçu pour fonctionner à l'intérieur de la contrainte d'une bande de fréquence de 5 MHz pour les modèles de canaux intérieur/extérieur pédestre et véhiculaire tel que décrit par le "Third Genaration Partnership Project" (3GPP). La composante OFDM du système MC-CDMA a été simulée en utilisant le logiciel MATLAB dans le but d'obtenir des paramètres de base. Des codes orthogonaux à facteur d'étalement variable (OVSF) de longueur 8 ont été choisis comme codes d'étalement pour notre système MC-CDMA. Ceci permet de supporter des taux de transmission maximum jusquà 20.6 Mbps et 22.875 Mbps (données non codées, pleine charge de 8 utilisateurs) pour les canaux intérieur/extérieur pédestre et véhiculaire, respectivement. Une étude analytique des expressions de taux d'erreur binaire pour le MC-CDMA dans un canal multivoies de Rayleigh a été réalisée dans le but d'évaluer rapidement et de façon précise les performances. Des techniques d'estimation de canal basées sur les décisions antérieures ont été étudiées afin d'améliorer encore plus les performances de taux d'erreur binaire du système MC-CDMA en liaison descendante. L'estimateur de canal basé sur les décisions antérieures et utilisant le critère de l'erreur quadratique minimale linéaire avec une matrice' de corrélation du canal de taille 64 x 64 a été choisi comme étant un bon compromis entre la performance et la complexité pour une implementation sur puce FPGA. Une nouvelle séquence d'apprentissage a été conçue pour le récepteur dans la configuration intérieur/extérieur pédestre dans le but d'estimer de façon grossière le temps de synchronisation et le décalage fréquentiel fractionnaire de la porteuse dans le domaine du temps. Les estimations fines du temps de synchronisation et du décalage fréquentiel de la porteuse ont été effectués dans le domaine des fréquences à l'aide de sous-porteuses pilotes. Un récepteur en liaison descendante MC-CDMA complet pour le canal intérieur /extérieur pédestre avec les synchronisations en temps et en fréquence en boucle fermée a été simulé avant de procéder à l'implémentation matérielle. Le récepteur en liaison descendante en bande de base pour le canal intérieur/extérieur pédestre a été implémenté sur un système de développement fabriqué par la compagnie Nallatech et utilisant le circuit XtremeDSP de Xilinx. Un transmetteur compatible avec le système de réception a également été réalisé. Des tests fonctionnels du récepteur ont été effectués dans un environnement sans fil statique de laboratoire. Un environnement de test plus dynamique, incluant la mobilité du transmetteur, du récepteur ou des éléments dispersifs, aurait été souhaitable, mais n'a pu être réalisé étant donné les difficultés logistiques inhérentes. Les taux d'erreur binaire mesurés avec différents nombres d'usagers actifs et différentes modulations sont proches des simulations sur ordinateurs pour un canal avec bruit blanc gaussien additif

Digital resampling and timing recovery in QAM systems

Digital resampling is a process that converts a digital signal from one sampling rate to another. This process is performed by means of interpolating between the input samples to produce output samples at an output sampling rate. The digital interpolation process is accomplished with an interpolation filter. The problem of resampling digital signals at an output sampling rate that is incommensurate with the input sampling rate is the first topic of this thesis. This problem is often encountered in practice, for example in multiplexing video signals from different sources for the purpose of distribution. There are basically two approaches to resample the signals. Both approaches are thoroughly described and practical circuits for hardware implementation are provided. A comparison of the two circuits shows that one circuit requires a division to compute the new sampling times. This time scaling operation adds complexity to the implementation with no performance advantage over the other circuit, and makes the 'division free' circuit the preferred one for resampling. The second topic of this thesis is performance analysis of interpolation filters for Quadrature Amplitude Modulation (QAM) signals in the context of timing recovery. The performance criterion of interest is Modulation Error Ratio (MER), which is considered to be a very useful indicator of the quality of modulated signals in QAM systems. The methodology of digital resampling in hardware is employed to describe timing recovery circuits and propose an approach to evaluate the performance of interpolation filters. A MER performance analysis circuit is then devised. The circuit is simulated with MATLAB/Simulink as well as implemented in Field Programmable Gate Array (FPGA). Excellent agreement between results obtained from simulation and hardware implementation proves the validity of the methodology and practical application of the research works

Statistical characterization of correlation-based time/frequency synchronizers for OFDM

Orthogonal Frequency Division Multiplexing (OFDM) has been widely adopted as a modulation format for reliable digital communication over multipath fading channels, e.g. IEEE 802.11g WiFi networks, as well as broadband wireline channels, e.g. DSL modems. However, its robustness to channel impairments comes at the cost of increased sensitivity to symbol timing and carrier frequency offset errors, and thus requires more complex synchronization methods than conventional single-carrier modulation formats. In this thesis, a class of synchronization methods based upon the intrinsic autocorrelation structure of the OFDM signal is studied from a statistical perspective. In particular, the reasons for the existence of irreducible time and frequency offset estimation errors in the limit of increasing signal-to-noise ratio (SNR) are investigated for correlator-based synchronizers for the non-fading channel case and several fading channel models. It is demonstrated that the primary source of irreducible synchronization errors at high SNR is the natural random distribution of signal energy in the cyclic prefix of the OFDM symbol. Comparisons of the distribution of correlator output magnitude between the non-fading and fading channel cases demonstrates that fading skews the distribution with respect to the non-fading case. A potential mechanism for reducing the effect of innate signal energy variability, correlator output windowed averaging, is studied from the perspective of its influence on the distribution of interpeak intervals in the temporal correlator output signal. While improved performance is realized through averaging for the non-fading channel case, this technique is not as effective for fading channels. In either instance, the windowed averaging method increases the latency of the synchronization process and thus introduces delay in the overall demodulation process