In digital communication systems symbol timing recovery is of fundamental importance. The accuracy in estimation of symbol timing has a direct effect on received data error rates. The primary objective of this thesis is to implement a practical Digital Phase Locked Loop capable of accurate synchronisation of symbols suffering channel corruption typical of modern mobile communications. This thesis describes an all-software implementation of a Digital Phase Locked in a real-time system. A timing error detection (TED) algorithms optimally implemented into a Digital Signal Processor. A real-time transmitter and receiver system is implemented in order to measure performance when the received signal is corrupted by both Additive White Gaussian Noise and Flat Fading. The Timing Error Detection algorithm implemented is a discrete time maximum likelihood one known as FFML1, developed at Canterbury University. FFML1 along with other components of the Digital Phase Locked loop are implemented entirely in software, using Motorola 56321 assembly language.

International Telemetering Conference Proceedings / October 27-30, 1997 / Riviera Hotel and Convention Center, Las Vegas, Nevada / Sandia National Laboratories (SNL) currently achieves a bandwidth efficiency (h ) of 0.5 to 1.0 bps/Hz by using traditional modulation schemes, such as, BPSK and QFSK. SNL has an interest in increasing the present bandwidth efficiency by a factor of 4 or higher with the same allocated bandwidth (about 10 MHz). Simulations have shown that 32- QAM trellis-coded modulation (TCM) gives a good bit error rate (BER) performance, and meets the requirements as far as the bandwidth efficiency is concerned. Critical to achieving this is that the receiver be able to achieve timing synchronization. This paper examines a particular timing recovery algorithm for all-digital receivers. Timing synchronization in a digital receiver can be achieved in different ways. One way of achieving this is by interpolating the original sampled sequence to produce another sampled sequence synchronized to the symbol rate or a multiple of the symbol rate. An adaptive sampling conversion algorithm which performs this function was developed by Floyd Gardner in 1993. In the present work, his algorithm was applied to two different modulation schemes, BPSK and 4-ary PAM. The two schemes were simulated in the presence of AWGN and ISI along with Gardner’s algorithm for timing recovery, and a fractionally spaced equalizer (T/2 FSE) for equalization. Simulations show that the algorithm gives good BER performance for BPSK in all the situations, and at different sampling frequencies, but unfortunately poor performance for the 4-ary PAM scheme. This indicates that Gardner’s algorithm for sampling conversion is not suitable for multi-level signaling schemes.

This thesis deals with the synchronization of digital communication systems. Synchronization (from the Greek syn (together) and chronos (time)) denotes the task of making two systems running at the same time. In communication systems, the synchronization of the transmitter and the receiver requires to accurately estimate a number of parameters such as the carrier frequency and phase offsets, the timing epoch... In the early days of digital communications, synchronizers used to operate in either data-aided (DA) or non-data-aided (NDA) modes. However, with the recent advent of powerful coding techniques, these conventional synchronization modes have been shown to be unable to properly synchronize state-of-the-art receivers. In this context, we investigate in this thesis a new family of synchronizers referred to as code-aided (CA) synchronizers. The idea behind CA synchronization is to take benefit from the structure of the code used to protect the data to improve the estimation quality achieved by the synchronizers. In a first part of the thesis, we address the issue of turbo synchronization, i.e., the iterative synchronization of continuous parameters. In particular, we derive several mathematical frameworks enabling a systematic derivation of turbo synchronizers and a deeper understanding of their behavior. In a second part, we focus on the so-called CA hypothesis testing problem. More particularly, we derive optimal solutions to deal with this problem and propose efficient implementations of the proposed algorithms. Finally, in a last part of this thesis, we derive theoretical lower bounds on the performance of turbo synchronizers.

Synchronization is a very important part of every digital communication receiver. While in bandpass coherent transmission, frequency and phase synchronization play a very important role in reliable transmission, symbol timing recovery is a necessary part of every baseband and bandpass coherent receiver. This dissertation deals with the problem of synchronization in the presence of fading and interference. First, the performance of an automatic frequency control loop is investigated using two parameters of average switching rate and mean time to loss of lock. These parameters are derived in closed-form or as integral-form formulas for different scenarios of modulated and unmodulated signals in different fading channels when there is one interference signal present at the input of the AFC. Then, the results are generalized to the noisy fading scenario and it is shown that in Rayleigh fading case, the performance of AFC becomes better when the desired signal is noisier. In the second part, the problem of symbol timing recovery is investigated in systems that are subject to intersymbol interference and non-data-aided maximum likelihood synchronizer is derived in these channels. Then, a new simple bound on the performance of synchronizers is derived and compared to the previously known lower bounds. It is shown that while this lower bound solves the shortcomings of the well known modified Cramer-Rao bound at small values of signal-to-noise-ratio, it is much easier to compute compared to another well known bound, the detection theory bound. / Communications

The purpose of this master thesis was to investigate how effiecient the extractionof radiofrequency quality metrics from digital video broadcast (DVB) streamscan become using software defined radio. Software defined radio (SDR) is a fairlynew technology that offers you the possibility of very flexible receivers and transmitters where it is possible to upgrade the modulation and demodulation overtime. Agama is interested in SDR for use in the Agama Analyzer, a widely deployedmonitoring probe running on top of standard services. Using SDR, Agama coulduse that in all deployments, such as DVB by cable/terrestrial/satellite (DVBC/T/S), which would simplify logistics. This thesis is an implementation of a SDR to be able to receive DVB-C. TheSDR must perform a number of adaptive algorithms in order to prevent the received symbols from being significantly different from the transmitted ones. Themain parts of the SDR include timing recovery, carrier recovery and equalization.Timing recovery performs synchronization between the transmitted and receivedsymbols and the carrier recovery performs synchronization between the carrierwave of the transmitter and the local oscillator in the receiver. The thesis discusses various methods to perform the different types of synchronizations andequalizations in order to find the most suitable methods.

Digital communication systems invariably employ an underlying analog communication channel. At the transmitter, data is modulated to obtain an analog waveform which is input to the channel. At the receiver, the output of the channel needs to be mapped back into the discrete domain. To this effect, the continuous-time received waveform is sampled at instants chosen by the timing recovery block. Therefore, timing recovery is an essential component of digital communication systems. A widely used timing recovery method is based on a phase-locked loop (PLL), which updates its timing estimates based on a decision-directed device. Timing recovery performance is a strong function of the reliability of decisions, and hence, of the channel signal-to-noise ratio (SNR). Iteratively decodable error-control codes (ECCs) like turbo codes and LDPC codes allow operation at SNRs lower than ever before, thus exacerbating timing recovery. We propose iterative timing recovery, where the timing recovery block, the equalizer and the ECC decoder exchange information, giving the timing recovery block access to decisions that are much more reliable than the instantaneous ones. This provides significant SNR gains at a marginal complexity penalty over a conventional turbo equalizer where the equalizer and the ECC decoder exchange information. We also derive the Cramer-Rao bound, which is a lower bound on the estimation error variance of any timing estimator, and propose timing recovery methods that outperform the conventional PLL and achieve the Cramer-Rao bound in some cases. At low SNR, timing recovery suffers from cycle slips, where the receiver drops or adds one or more symbols, and consequently, almost always the ECC decoder fails to decode. Iterative timing recovery has the ability to corrects cycle slips. To reduce the number of iterations, we propose cycle slip detection and correction methods. With iterative timing recovery, the PLL with cycle slip detection and correction recovers most of the SNR loss of the conventional receiver that separates timing recovery and turbo equalization.

No description available.

Η παρούσα διπλωματική εργασία αφορά στην πειραματική αξιολόγηση του μηχανισμού ανάκτησης ρυθμού συμβόλου για ψηφιακούς δέκτες τεχνολογίας SDR που λαμβάνουν δεδομένα μέσω δορυφόρου. Η ορολογία SDR/SR (Software Defined Radio/Software Radio) χρησιμοποιείται για να χαρακτηρίσει τους πομποδέκτες που μπορούν να καθορίζουν σημαντικές παραμέτρους τους και βασικές αρχές της λειτουργίας τους μέσω αναβάθμισης ή ενημέρωσης του λογισμικού τους. Ο μηχανισμός ανάκτησης του ρυθμού συμβόλου (Symbol Timing Recovery, STR) αναπτύχθηκε στα πλαίσια της διδακτορικής διατριβής του διδάκτορος Παναγιώτη Σαββόπουλου. Η παρούσα εργασία μελετά τη σύγκλιση του βρόχου υπό συνθήκες παραμένοντος σφάλματος συχνότητας καθώς επίσης και τον προσδιορισμό του λόγου σήματος προς θόρυβο στην έξοδο του βρόχου κάνοντας χρήση ενός νέου μεγέθους, metric, το οποίο έχει εισαχθεί στα πλαίσια της προαναφερθείσας διδακτορικής διατριβής, υπό συνθήκες λευκού Gaussian θορύβου. Το μέγεθος αυτό είναι σε θέση να δώσει αξιόπιστα αποτελέσματα στις ενδιάμεσες υπομονάδες του δέκτη υπό συνθήκες παραμένοντος σφάλματος συχνότητας. Στην παρούσα εργασία μελετώνται οι QPSK, 8PSK, 16-APSK και 32-APSK διαμορφώσεις διότι αυτές οι διαμορφώσεις χρησιμοποιούνται από το πρότυπο DVB-S2. / The purpose of this project is the experimental evaluation of a mechanism for the symbol timing recovery which is used in digital Software Defined Radio receivers. SDR/SR (Software Defined Radio/Software Radio) technology is used to characterise the transmitters and the receivers which are able to determine important parameters and basic primciples for their function through upgrade or briefing of their software. The symbol timing recovery mechanism (STR) was developped in terms of the doctora of dr Panagiotis Savopoylos. The precent project examines the loop’s convergence when there is frequency error as well as the signal to noise ratio in the output of STR with the use of a new size, metric, which was also developped in terms of the doctora which was mentioned before, when there is white Gaussian noise. The metric is able to give reliable results in the intermediate stages of the receiver when there is frequency error. In the precent project are examined the QPSK, 8PSK,16-APSK, 32-APSK modulations because these modulations are used in DVB-S2 standard.

In this thesis, symbol timing recovery based on matched filtering in Gaussian Minimum Shift Keying (GMSK) with bandwidth-bit period product (BT) of 0.3 is investigated. GMSK is the standard modulation type for GSM. Although GMSK modulation is non-linear, it is approximated to Offset Quadrature Amplitude Modulation (OQAM), which is a linear modulation, so that Maximum Likelihood Sequence Estimation (MLSE) method is possible in the receiver part. In this study Typical Urban (TU) channel model developed in COST 207 is used. Two methods are developed on the construction of the matched filter. In order to obtain timing recovery for GMSK signals, these methods are investigated. The fractional time delays are acquired by using interpolation and an iterative maximum search process. The performance of the proposed symbol timing recovery (STR) scheme is assessed by using computer simulations. It is observed that the STR tracks the variations of the frequency selective multipath fading channels almost the same as the Mazo criterion.

In this thesis, two key issues regarding M-ary orthogonal signaling systems, namely channel estimation and symbol timing recovery are investigated. Kasami codes, which are also called quasi orthogonal codes, are used for transmission of the information in place of orthogonal waveforms. In order to achieve symbol synchronization, a timing recovery scheme based on the Maximum Likelihood (ML) estimation of timing offset is proposed and the effects of proposed structure over the receiver performance are examined by using computer simulations. Moreover, the receiver performance of M-ary orthogonal signals transmitted over multipath fading channel is investigated. Least Square (LS) approach, based on the transmission of known training sequence, is used to estimate the channel impulse response. In addition to this, frame synchronization is employed at the receiver to extract the timing information by determining the start time of the received symbols. Computer simulations related to the proposed receiver structure are carried out in order to observe how the system performance is affected under multipath fading channel. Parameter selection guides regarding a good performance are also provided.