With the reality that diverse air interfaces and dissimilar access networks coexist, accompanied by the trend that dynamic spectrum access (DSA) is allowed and will be gradually employed, cognition and cooperation form a promising framework to achieve the ideality of seamless ubiquitous connectivity in future communication networks. In this dissertation, the cognitive gateway (CG), conceived as a special cognitive radio (CR) node, is proposed and designed to facilitate universal interoperability among incompatible waveforms. A proof-of-concept prototype is built and tested. Located in places where various communication nodes and diverse access networks coexist, the CG can be easily set up and works like a network server with differentiated service (Diffserv) architecture to provide automatic traffic relaying and link establishment. The author extracts a scalable '“source-CG-destination“ snapshot from the entire network and investigates the key enabling technologies for such a snapshot. The CG features provide universal interoperability, which is enabled by a generic waveform representation format and the reconfigurable software defined radio platform. According to the trend of an all IP-based solution for future communication systems, the term “waveform“ in this dissertation has been defined as a protocol stack specification suite. The author gives a generic waveform representation format based on the five-layer TCP/IP protocol stack architecture. This format can represent the waveforms used by Ethernet, WiFi, cellular system, P25, cognitive radios etc. A significant advantage of CG over other interoperability solutions lies in its autonomy, which is supported by appropriate signaling processes and automatic waveform identification. The service process in a CG is usually initiated by the users who send requests via their own waveforms. These requests are transmitted during the signaling procedures. The complete operating procedure of a CG is depicted as a waveform-oriented cognition loop, which is primarily executed by the waveform identifier, scenario analyzer, central controller, and waveform converter together. The author details the service process initialized by a primary user (e.g. legacy public safety radio) and that initialized by a secondary user (e.g. CR), and describes the signaling procedures between CG and clients for the accomplishment of CG discovery, user registration and un-registration, link establishment, communication resumption, service termination, route discovery, etc. From the waveforms conveyed during the signaling procedures, the waveform identifier extracts the parameters that can be used for a CG to identify the source waveform and the destination waveform. These parameters are called “waveform indicators.“ The author analyzes the four types of waveforms of interest and outlines the waveform indicators for different types of communication initiators. In particular, a multi-layer waveform identifier is designed for a CG to extract the waveform indicators from the signaling messages. For the physical layer signal recognition, a Universal Classification Synchronization (UCS) system has been invented. UCS is conceived as a self-contained system which can detect, classify, synchronize with a received signal and provide all parameters needed for physical layer demodulation without prior information from the transmitter. Currently, it can accommodate the modulations including AM, FM, FSK, MPSK, QAM and OFDM. The design and implementation details of a UCS have been presented. The designed system has been verified by over-the-air (OTA) experiments and its performance has been evaluated by theoretical analysis and software simulation. UCS can be ported to different platforms and can be applied for various scenarios. An underlying assumption for UCS is that the target signal is transmitted continually. However, it is not the case for a CG since the detection objects of a CG are signaling messages. In order to ensure higher recognition accuracy, signaling efficiency, and lower signaling overhead, the author addresses the key issues for signaling scheme design and their dependence on waveform identification strategy. In a CG, waveform transformation (WT) is the last step of the link establishment process. The resources required for transformation of waveform pairs, together with the application priority, constitute the major factors that determine the link control and scheduling scheme in a CG. The author sorts different WT into five categories and describes the details of implementing the four typical types of WT (including physical layer analog – analog gateway, up to link layer digital – digital gateway, up-to-network-layer digital gateway, and Voice over IP (VoIP) – an up to transport layer gateway) in a practical CG prototype. The issues that include resource management and link scheduling have also been addressed. This dissertation presents a CG prototype implemented on the basis of GNU Radio plus multiple USRPs. In particular, the service process of a CG is modeled as a two-stage tandem queue, where the waveform identifier queues at the first stage can be described as M/D/1/1 models and the waveform converter queue at the second stage can be described as G/M/K/K model. Based on these models, the author derives the theoretical block probability and throughput of a CG. Although the “source-CG-destination” snapshot considers only neighboring nodes which are one-hop away from the CG, it is scalable to form larger networks. CG can work in either ad-hoc or infrastructure mode. Utilizing its capabilities, CG nodes can be placed in different network architectures/topologies to provide auxiliary connectivity. Multi-hop cooperative relaying via CGs will be an interesting research topic deserving further investigation. / Ph. D.

This dissertation provides the theory, design, and implementation of a cognitive engine, the enabling technology of cognitive radio. A cognitive radio is a wireless communications device capable of sensing the environment and making decisions on how to use the available radio resources to enable communications with a certain quality of service. The cognitive engine, the intelligent system behind the cognitive radio, combines sensing, learning, and optimization algorithms to control and adapt the radio system from the physical layer and up the communication stack. The cognitive engine presented here provides a general framework to build and test cognitive engine algorithms and components such as sensing technology, optimization routines, and learning algorithms. The cognitive engine platform allows easy development of new components and algorithms to enhance the cognitive radio capabilities. It is shown in this dissertation that the platform can easily be used on a simulation system and then moved to a real radio system. The dissertation includes discussions of both theory and implementation of the cognitive engine. The need for and implementation of all of the cognitive components is strongly featured as well as the specific issues related to the development of algorithms for cognitive radio behavior. The discussion of the theory focuses largely on developing the optimization space to intelligently and successfully design waveforms for particular quality of service needs under given environmental conditions. The analysis develops the problem into a multi-objective optimization process to optimize and trade-of of services between objectives that measure performance, such as bit error rate, data rate, and power consumption. The discussion of the multi-objective optimization provides the foundation for the analysis of radio systems in this respect, and through this, methods and considerations for future developments. The theoretical work also investigates the use of learning to enhance the cognitive engine's capabilities through feed-back, learning, and knowledge representation. The results of this work include the analysis of cognitive radio design and implementation and the functional cognitive engine that is shown to work in both simulation and on-line experiments. Throughout, examples and explanations of building and interfacing cognitive components to the cognitive engine enable the use and extension of the cognitive engine for future work. / Ph. D.

Due to advances in digital computing much of the baseband signal processing of a communication system has moved into the digital domain from the analog domain. Within the domain of digital communication systems, Software Defined Radios (SDRs) allow for majority of the signal processing tasks to be implemented in reconfigurable digital hardware. However this comes at a cost of higher power and resource requirements. Therefore, highly efficient custom hardware implementations for SDRs are needed to make SDRs feasible for practical use. Efficient custom hardware motivates the use of fixed point arithmetic in the implementation of Digital Signal Processing (DSP) algorithms. This conversion to finite precision arithmetic introduces quantization noise in the system, which significantly affects the performance metrics of the system. As a result, characterizing quantization noise and its effects within a DSP system is an important challenge that needs to be addressed. Current models to do so significantly over-estimate the quantization effects, resulting in an over-allocation of hardware resources to implement a system. Polynomial Chaos Expansion (PCE) is a method that is currently gaining attention in modelling uncertainty in engineering systems. Although it has been used to analyze quantization effects in DSP systems, previous investigations have been limited to simple examples. The purpose of this thesis is to therefore introduce new techniques that allow the application of PCE to be scaled up to larger DSP blocks with many noise sources. Additionally, the thesis introduces design space exploration algorithms that leverage the accuracy of PCE simulations to estimate bitwidths for fixed point implementations of DSP systems. The advantages of using PCE over current modelling techniques will be presented though its application to case studies relevant to practice. These case studies include Sine Generators, Infinite Impulse Response (IIR) filters, Finite Impulse Response (FIR) filters, FM demodulators and Phase Locked Loops (PLLs). / Thesis / Master of Applied Science (MASc)

Digital Selective Calling (DSC) is a vital maritime communications and safety system, enabling ships in distress to alert nearby vessels and coast guard stations of their emergency. While DSC is suitable for calling, its technical format is substandard from a cybersecurity perspective. Specifically, this work aims to demonstrate that Very High Frequency (VHF) DSC distress calls can be spoofed using Software Defined Radio (SDR). A VHF DSC distress call encoder and VHF DSC SDR signal constructor were developed. The forged distress call was transmitted using various techniques to two different DSC decoder programs, as well as to the maritime VHF transceiver ICOM IC-M510. It was shown that all of the targeted DSC decoders were susceptible to spoofing. This thesis concludes that VHF DSC distress calls can be spoofed using SDR, and infers that the DSC system as a whole has inherent security vulnerabilities that need to be addressed to assure the safety of future seafaring.

Impulse Ultra Wideband (UWB) communications is an emerging technology that promises a number of benefits over traditional narrowband or broadband signals: extremely high data rates, extremely robust operation in dense multipath environments, low probability of intercept/detection, and the ability to operate concurrently with existing users. Unfortunately, most currently available UWB systems are based on dedicated hardware, preventing researchers from investigating algorithms or architectures that take advantage of some of the unique properties of UWB signals. This dissertation outlines the development of a general purpose software radio transceiver testbed for UWB signals. The testbed is an enabling technology that provides a development platform for investigating ultra wideband communication algorithms (e.g., acquisition, synchronization, modulation, multiple access), ranging or radar (e.g., precision position location, intrusion detection, heart and respiration rate monitoring), and could potentially be used in the area of ultra wideband based medical imaging or vital signs monitoring. As research into impulse ultra wideband expands, the need is greater now than ever for a platform that will allow researchers to collect real-world performance data to corroborate theoretical and simulation results. Additionally, this dissertation outlines the development of the Time-Interleaved Analog to Digital Converter array which served as the core of the testbed, along with a comprehensive theoretical and simulation-based analysis on the effects of Analog to Digital Converter mismatches in a Time-Interleaved Sampling array when the input signal is an ultra wideband Gaussian Monocycle. Included in the discussion is a thorough overview of the implementation of both a scaled-down prototype as well as the final version of the testbed. This dissertation concludes by evaluating the of the transceiver testbed in terms of the narrowband dynamic range, the accuracy with which it can sample and reconstruct a UWB pulse, and the bit error rate performance of the overall system. / Ph. D.

Denna studie genomför en detaljerad undersökning av två olika plattformar för programvarudefinierad radio (SDR), GNU Radio och Simulink, för att avgöra vilken som är mest lämpad för integration i en specifik kurs inom läroplanen vid mittuniversitet. Utvärderingen fokuserar på nyckelprestandamått såsom beräkningskapacitet, simuleringens hastighet och visualiseringsförmåga, vilket ger en omfattande jämförelse mellan dessa två plattformar. Undersökningen inleds med att simulera det fysiska lagret av WiFi, vilket är ett grundläggande krav för kursens laboratoriearbete. Studiens omfattning utvidgas sedan för att inkludera simuleringar av andra nätverkstekniker som 4G. Denna expansion syftar till att samla in omfattande data för en mer noggrann jämförelse och för att grundligt utvärdera varje plattforms förmåga att hantera olika nätverkssimuleringar. Dessutom fördjupar studien sig i olika simuleringstekniker genom att diskutera två distinkta angreppssätt till SDR-simuleringar, vilket belyser deras respektive styrkor och tillämpbarhet i ett utbildningssammanhang. Det slutgiltiga målet med denna forskning är att avgöra om GNU Radio erbjuder betydande fördelar jämfört med MATLABs Simulink och om det bör ersätta Simulink som det primära verktyget som används i denna kurs. I förväntan på potentiella förändringar har nya alternativa laborationsinstruktioner för GNU Radio också utvecklats och presenterats. Dessa instruktioner är utformade för att underlätta en smidig övergång om universitetet beslutar att anta GNU Radio, för att säkerställa att utbildningsmålen fortsätter att uppnås på ett effektivt och effektivt sätt. Även om studien visar att GNU Radio erbjuder bättre beräkningskapacitet var själva simuleringsprocessen något svårare. De två plattformarna uppnådde nästan samma resultat, men GNU Radio krävde extra arbete. Med tanke på att inlärningsresultaten var liknande men inlärningsprocessen med GNU Radio var mer komplicerad bedömdes GNU Radio som olämplig för denna kurs. / This study conducts a detailed examination of two distinct Software Defined Radio (SDR) platforms, GNU Radio and Simulink, to ascertain which is more suited for integration into a specific course within the curriculum at Mid University. The evaluation focuses on key performance metrics such as computing efficiency, simulation speed, and visualization capabilities, providing a comprehensive comparison between these two platforms. The investigation begins by simulating the physical layer of WiFi, which is a fundamental requirement of the course laboratory work. The scope of the study is then broadened to include simulations of other network technologies like 4G. This expansion aims to collect extensive data for a more accurate comparison and to thoroughly evaluate the capabilities of each platform in handling various network simulations. Moreover, the study delves into different simulation methodologies by discussing two distinct approaches to SDR simulations, highlighting their respective strengths and applicabilities in an educational context. The ultimate objective of this study is to determine whether GNU Radio offers a significant advantage over MATLAB’s Simulink and if it should replace Simulink as the primary tool used in this course. In anticipation of potential changes, new alternative laboratory instructions for GNU Radio are also developed and presented. These instructions are designed to facilitate a smooth transition should the university decide to adopt GNU Radio, ensuring that educational goals continue to be met efficiently and effectively. While the study shows that GNU Radio offers better computing efficiency, the process of simulating was somewhat more challenging. The two platforms accomplished almost the same tasks, but GNU Radio required extra effort. Considering that the learning outcomes were similar but the learning process with GNU Radio was more difficult, GNU Radio was deemed unsuitable for this course.

Αντικείμενο της διδακτορικής διατριβής αποτελεί η μελέτη και ανάλυση των μηχανισμών συγχρονισμού που εφαρμόζονται σε ψηφιακούς δορυφορικούς δέκτες διαγραμμάτων αστερισμού πολλαπλών δακτυλίων με σκοπό την ανάπτυξη νέων τεχνικών που παρουσιάζουν βελτιωμένη απόδοση καθώς και μεθόδων αξιολόγησης της απόδοσής τους. Οι σύγχρονες τάσεις στον τομέα των ψηφιακών επικοινωνιών και συγκεκριμένα στο πεδίο των τεχνικών διαμόρφωσης και διόρθωσης σφαλμάτων, καθώς και η εντεινόμενη ανάγκη για πιο αποδοτικές εφαρμογές και υπηρεσίες μέσω δορυφορικών ζεύξεων οδήγησαν στην ανάπτυξη νέων προτύπων δορυφορικών επικοινωνιών, όπως το DVB-S2, από τον Ευρωπαϊκό Οργανισμό Διαστήματος (ΕΟΔ-ESA). Βάσει των προτύπων αυτών, απαιτούνται νέες προσεγγίσεις και τεχνικές στο σχεδιασμό δορυφορικών δεκτών. Παράλληλα, η προσέγγιση Software Defined Radio (SDR) αποτελεί μια πολλά υποσχόμενη μεθοδολογία η οποία επιτρέπει την απαιτούμενη προσαρμοστικότητα και ευελιξία για την υποστήριξη πολλαπλών τύπων λειτουργίας και ρυθμών συμβόλων στους σύγχρονους δέκτες. Ο συγχρονισμός σε ένα δορυφορικό δέκτη (μονού φορέα) αποτελεί μια πολύπλοκη και απαιτητική διαδικασία που αφορά την εκτίμηση των παραμέτρων της μετάδοσης, οι οποίες και ανταποκρίνονται στον πραγματικό ρυθμό συμβόλων, στη συχνότητα και φάση του φορέα μετάδοσης καθώς και στη γνώση των ορίων των πλαισίων φυσικού επιπέδου. Οι μηχανισμοί συγχρονισμού αποτελούν σημαντικό, από άποψη κρισιμότητας και απαιτήσεων σε επεξεργαστική ισχύ, τμήμα των αποδιαμορφωτών, οι οποίοι σε περίπτωση λειτουργικής αποτυχίας οδηγούν στην απώλεια της αξιοπιστίας του δέκτη. Εξαιτίας της σπουδαιότητας των μηχανισμών αυτών, η αναζήτηση αποδοτικών και υλοποιήσιμων αλγορίθμων συγχρονισμού αποτελεί σημαντική παράμετρο στον σχεδιασμό συστημάτων δεκτών. Ένα σημαντικό πρόβλημα που αρχικά αντιμετώπισε η παρούσα διδακτορική διατριβή αφορά την ανάπτυξη βέλτιστης αρχιτεκτονικής διαχείρισης του σήματος εισόδου IF σε ένα δέκτη SDR μέσω κατάλληλης ψηφιακής επεξεργασίας των δειγμάτων εισόδου. Σκοπός της βαθμίδας είναι να υποβιβάσει το φάσμα του ψηφιακού σήματος εισόδου IF στη βασική ζώνη, υπολογίζοντας τις αντίστοιχες συνιστώσες του σήματος βασικής ζώνης. Περιορισμό στο πρόβλημα, αποτελεί η μέγιστη συχνότητα δειγματοληψίας του κυκλώματος ψηφιοποίησης. Η λύση που προτείνεται αντιμετωπίζει τις παραπάνω συνθήκες με μια νέα αρχιτεκτονική που βασίζεται σε δύο βαθμίδες μετατόπισης συχνότητας, μια σταθερής και μια προγραμματιζόμενης συχνότητας. Η προγραμματιζόμενη οδηγείται από την εκτίμηση του σφάλματος μετατόπισης συχνότητας που πραγματοποιείται σε επόμενο στάδιο επεξεργασίας του σήματος βασικής ζώνης. Το πλεονέκτημα της αρχιτεκτονικής αυτής, είναι η διπλάσια ακρίβεια στη ρύθμιση της συχνότητας σε σχέση με την κλασική προσέγγιση για δεδομένη συχνότητα δειγματοληψίας και αριθμό bits στον καταχωρητή συσσώρευσης φάσης του ταλαντωτή. Τέλος, ο παραπάνω υποβιβαστής προορίζεται για χρήση σε δέκτες SDR με χρήση μετατροπέων σήματος (ADC) περιορισμένης συχνότητας δειγματοληψίας. Στο πλαίσιο της παρούσας εργασίας, μελετήθηκε ο μηχανισμός ανάκτησης χρονισμού συμβόλου (Symbol Timing Recovery - STR) που υλοποιείται με τη χρήση κλειστού βρόχου δεύτερης τάξης και βασίζεται στο σήμα ενός ανιχνευτή σφάλματος χρονισμού (Timing Error Detector - TED). Τα θεμελιώδη χαρακτηριστικά του βρόχου, όπως ο χρόνος και η ποιότητα σύγκλισης, καθορίζονται από τις τιμές του κέρδους των δύο κλάδων του φίλτρου του βρόχου πρώτης τάξης τύπου P-I (Proportional-Integral) που αποτελεί μια ευρέως διαδεδομένη λύση για τηλεπικοινωνιακές εφαρμογές συγχρονισμού. Αφού περιγράφηκε και αναλύθηκε η γενικευμένη μεθοδολογία παραμετροποίησης του βρόχου, στη συνέχεια δόθηκε έμφαση σε βρόχους που αξιοποιούν τον ανιχνευτή Gardner. Τα χαρακτηριστικά ανεξαρτησίας του από τις τιμές των συμβόλων που χρησιμοποιεί καθώς και του παραμένοντος σφάλματος συχνότητας, τον καθιστούν μια αξιόπιστη λύση για τον συγχρονισμό συμβόλων πριν από το συγχρονισμό συχνότητας σε ψηφιακούς δέκτες. Κάνοντας χρήση της ανάλυσης αυτής και λόγω της υστέρησης των διαγραμμάτων πολλαπλών δακτυλίων τύπου M-APSK, ως προς την απόδοση του κλειστού βρόχου ανάκτησης χρονισμού συμβόλου, σε σχέση με τα διαγράμματα μονού δακτυλίου ίδιας μέσης ενέργειας, η διατριβή προτείνει μια παραλλαγή του τυπικού βρόχου για τη βελτίωση της συμπεριφοράς τους. Η αυξημένη διακύμανση στο σήμα εισόδου του ανιχνευτή λόγω της εναλλαγής των συμβόλων διαφορετικού πλάτους στην είσοδο του ανιχνευτή σφάλματος χρονισμού αποτελεί την κύρια αιτία για την αυξημένη διακύμανση κατά την παρακολούθηση του σφάλματος χρονισμού από τις δομές τέτοιων βρόχων. Η προσέγγιση που προτείνεται, βασίζεται στην εισαγωγή μιας υπομονάδας στον τυπικό βρόχο που προσαρμόζει τα πλάτη των συμβόλων όλων των δακτυλίων σε ένα δακτύλιο αναφοράς πριν την εισαγωγή τους στον ανιχνευτή σφάλματος χρονισμού. Επίσης κάνει χρήση του τοπικού ρολογιού του βρόχου με στόχο τη ρύθμιση του πλάτους συγκεκριμένων δειγμάτων του σήματος εισόδου και χωρίς να επηρεάζει τα πλάτη των συμβόλων που εισάγονται στο προσαρμοσμένο φίλτρο εξόδου. Η εφαρμογή της υπομονάδας έχει ως αποτέλεσμα τη μείωση του θορύβου κατά την παρακολούθηση του σφάλματος μετά την αρχική σύγκλιση του βρόχου, γεγονός που μεταφράζεται στη μείωση της τυπικής απόκλισης του σφάλματος εκτίμησης του χρονισμού σε σύγκριση με τον τυπικό βρόχο. Η απόδοση των βαθμίδων συγχρονισμού καθορίζεται συνήθως με βάση εσωτερικές παραμέτρους οι οποίες και επηρεάζονται σημαντικά από την αρχιτεκτονική του εκάστοτε μηχανισμού, την παράμετρο εκτίμησης καθώς και την κατάσταση λειτουργίας του μηχανισμού. Η διατριβή αξιοποιώντας την ύπαρξη πολλαπλών δακτυλίων στα διαγράμματα αστερισμού της μεθόδου διαμόρφωσης προτείνει ένα νέο ενιαίο μέγεθος εκτίμησης της απόδοσης των βαθμίδων συγχρονισμού σε δέκτες διαγραμμάτων πολλαπλών δακτυλίων M-APSK. Σημαντικό πλεονέκτημα του μέγεθος αποτελεί η αποκλειστική χρήση του σήματος εξόδου των βαθμίδων συγχρονισμού μέσω κατάλληλης επεξεργασίας (των παραγόμενων τιμών συμβόλων), παρέχοντας τη δυνατότητα στο μέγεθος να χρησιμοποιηθεί σε συνθήκες μετατόπισης συχνότητας/φάσης φορέα και/ή σφάλματος στο χρονισμό συμβόλου. Ένα άλλο πλεονέκτημα του παραπάνω μεγέθους σχετίζεται με το γεγονός ότι δεν είναι αναγκαία η γνώση των μεταδιδόμενων συμβόλων, σε αντίθεση με αντίστοιχα μεγέθη απόδοσης που χρησιμοποιούνται στην έξοδο των αποδιαμορφωτών, όπως το Error Vector Magnitude (EVM). Η μαθηματική ανάλυση της μέσης τιμής του μεγέθους σε συνθήκες προσθετικού λευκού προσθετικού θορύβου (AWGN) που παρουσιάζεται στη διατριβή αυτή αφορά τόσο την περίπτωση όπου ο δέκτης γνωρίζει τον δακτύλιο προέλευσης των λαμβανομένων συμβόλων, όσο και την περίπτωση όπου ο δέκτης αγνοεί τον δακτύλιο προέλευσης των λαμβανομένων συμβόλων και υπολογίζει το μέγεθος σύμφωνα με τον πλησιέστερο σε αυτά δακτύλιο. Το δεύτερο από τα παραπάνω σενάρια αφορά ρεαλιστικά συστήματα δεκτών όπου η πληροφορία του δακτυλίου προέλευσης των συμβόλων λήψης δεν είναι διαθέσιμη. Και στις δύο παραπάνω περιπτώσεις, αποδεικνύεται μια σταθερή σχέση του προτεινόμενου μεγέθους με το λόγο των ισχύων συμβόλου και θορύβου AWGN (Es/No). Βάσει των παραπάνω χαρακτηριστικών, το προτεινόμενο μέγεθος είναι σε θέση να αξιοποιηθεί για την εκτίμηση των συνθηκών στο κανάλι υπό συνθήκες λευκού Gaussian θορύβου μέσω επεξεργασίας του σήματος εξόδου από τον βρόχο STR ο οποίος αποτελεί συνήθως και τον πρώτο μηχανισμό συγχρονισμού σε ψηφιακούς δέκτες δορυφορικών επικοινωνιών. Αξίζει να σημειωθεί ότι η εκτίμηση των συνθηκών αυτών είναι εφικτή ακόμα και υπό συνθήκες σημαντικού παραμένοντος σφάλματος στη συχνότητα του φορέα. Η σπουδαιότητα της εκτίμησης αυτής έγκειται στο γεγονός ότι μπορεί να αξιοποιηθεί από τις ακόλουθες βαθμίδες συγχρονισμού (συχνότητας φορέα και φάσης) για την κατάλληλη προσαρμογή και επιτάχυνση των λειτουργιών τους. Μία δεύτερη μορφή αξιοποίησης του μεγέθους αποτελεί και η εκτίμηση-διόρθωση μεγάλων αποκλίσεων στη συχνότητα του φορέα κάνοντας χρήση προς επεξεργασία παραγόμενων, από το συγκεκριμένο βρόχο, σημάτων. Τα σήματα αυτά σχετίζονται με την είσοδο και την έξοδο του προσαρμοσμένου φίλτρου του βρόχου STR. Ο έλεγχος της απόκλισης στη συχνότητα του φορέα στο συγκεκριμένο σημείο επεξεργασίας των ψηφιακών δεκτών κάτω από συγκεκριμένα όρια, είναι ιδιαίτερα κρίσιμος καθώς επηρεάζει σημαντικά την απόδοση και αποτελεσματικότητα των ακόλουθων βαθμίδων συγχρονισμού. Στο τελικό στάδιό της, η διατριβή αναλύει και παρουσιάζει την υλοποίηση ενός πλήρους αποδιαμορφωτή SDR τεχνολογίας DVB-S2 σε πλατφόρμα επαναπρογραμματιζόμενης λογικής που συνδυάζει κυκλώματα υλικού και λογισμικού (FPGA, DSP). O αποδιαμορφωτής υποστηρίζει τα διαγράμματα μονού (QPSK/8PSK), διπλού (16APSK) και τριπλού (32APSK) δακτυλίου, ενώ αποτελεί τμήμα ενός συνολικού δέκτη DVB-S2 που υλοποιεί όλες τις λειτουργίες, από τη διαχείριση του σήματος εισόδου ΙF μέχρι την προώθηση της ανακτώμενης ψηφιακής πληροφορίας σε τοπικό δίκτυο GbE-LAN. Στην υλοποίηση του αποδιαμορφωτή περιλαμβάνεται η υλοποίηση σε κύκλωμα FPGA του προτεινόμενου υποβιβαστή συχνότητας IF, η υλοποίηση σε DSP του βρόχου STR (βάσει του ανιχνευτή Gardner) και όλων των υπόλοιπων μηχανισμών συγχρονισμού που είναι απαραίτητοι για τη σωστή αποδιαμόρφωση του σήματος εισόδου. Οι μηχανισμοί αυτοί είναι: συγχρονισμός πλαισίου, συγχρονισμός συχνότητας και φάσης φορέα καθώς και κανονικοποίηση πλάτους πριν την αντιστοίχιση των bits. Επίσης δίνονται πληροφορίες για την υλοποίηση των μηχανισμών αντιστοίχισης (Demapping), διόρθωσης σφαλμάτων (FEC - LDPC/BCH) καθώς και του μηχανισμού διαχείρισης και προώθησης (BBFRAME Processing) της ανακτώμενης πληροφορίας προς τη διεπαφή τοπικού δικτύου του δέκτη DVB-S2. / The objective of this thesis is the analysis and study of the synchronization mechanisms performed by digital satellite terminal receivers when multiple ring constellation diagrams are used. The aim of this thesis is to develop new synchronization techniques that exhibit improved performance and to also propose new methods and ways for evaluating the effectiveness of such receiver submodules. The new trends in the field of digital communications systems and, especially, in modulation and error coding techniques, along with the increasing demand for more effective and interactive applications and services through limited satellite links, have initiated the development of new satellite communications standards. The newest standard is DVB-S2, by the European Space Agency (ESA), in which modern and up-to-date techniques for the design of satellite terminal receiver are required. Meanwhile, the Software Defined Radio (SDR) technology comprises a promising implementation approach as it incorporates the necessary flexibility and versatility for supporting various functionalities and rates into modern receiver structures. Synchronization functions of satellite receivers are complicated and demanding procedures that are related to the estimation of transmission parameters, which correspond to the nominal symbol rate, carrier frequency, phase and to the boundaries of the physical layer frames. These functions determine the complexity and performance of receiver realizations. Thus developing more efficient and simple, in terms of implementation complexity, algorithms and mechanisms is a key objective in such processing platforms. A significant problem that was encountered during the research for the present thesis, was the design and implementation of an efficient digital IF down-converter architecture that is able to manipulate the input IF signal of an SDR receiver through proper processing of its digital input sample stream. The objective of this unit is the shifting of the IF input signal to baseband and the generation of the corresponding baseband I, Q signals. A usual limitation in such realizations is the maximum sampling frequency of front-end ADC circuits. The presented solution addresses this constraint with an architecture that is based on two cascaded units of frequency down-conversion, one with fixed and one with programmable frequency. The programmable unit is driven by the frequency offset estimations of a following baseband processing stage. The advantage of this architecture is the double precision that is achieved compared to the typical approach and for a given sampling frequency. It is worth mentioning that the frequency converter is intended for use in SDR receivers utilizing ADC circuits of moderate sampling frequency. Additionally, in the framework of this thesis, the Symbol Timing Recovery (STR) mechanism based on a second order feedback loop driven by the signal of a timing error detector (TED), was studied and analyzed. The fundamental characteristics of such a control loop, mainly the duration and quality of the initial acquisition are defined through the gain value of the two paths included into the first order loop filter (Proportional-Integral, P-I). This structure comprises a usual approach for communications applications. Conforming to this general analysis for the configuration and the design of the feedback loop, the thesis focuses on the feedback loop incorporating the Non-Data-Aided (NDA) Gardner TED. Using the above analysis and due to the fact that multiple ring constellation diagrams exhibit insufficient performance in such closed loops in comparison to the single ring counterparts of the same mean energy, this thesis proposes a modification of the typical loop deploying the Gardner TED that improves its performance. The increased variance of the input signal of the TED that stems from the changes of symbols with variable magnitude comprises the main reason for the increased variance during the tracking of the timing error in such loop structures. The proposed approach is based on the insertion of a subunit inside the loop structure that adjusts the symbol magnitudes of all rings to a reference magnitude before they are fed into the Gardner TED logic. The above subunit makes use of the internally generated clock of the loop in order to control the magnitude of specific signal samples and does not affect the sample stream at the matched filter input. The application of the specific subunit has the advantage of minimizing the noise during the tracking operation of the loop, which leads to the decrease of the standard deviation of the estimation error when compared to the typical loop structure. The performance of synchronization mechanisms is usually evaluated based on internal parameters that are strongly related to the utilized architecture, the estimated parameter and the operational status of the specific mechanism. The present thesis exploits the use of multiple ring constellation diagrams in modulation process and proposes a generic and new `figure of merit' that is able to determine the performance of various synchronization mechanisms that are incorporated into multiple ring constellation (M-APSK) receivers. A significant advantage of this metric is that it solely based on the processing of the signal at the mechanism's output (extracted symbol values) which enables the utilization of this metric in the presence of frequency, phase and symbol rate offset errors. Another advantage of the proposed metric is that it does not require any knowledge on the transmitted symbols, in contrast to other widely used performance metrics that are applied at the demodulator output, such as the Error Vector Magnitude (EVM) e.t.c. The mathematical analysis of the mean value of the metric under additive white Gaussian noise (AWGN) that is exhibited in this document, includes the theoretical and practical cases. In the first, the receiver is aware of the ring derivation of received symbols, whereas in the second case this information is absent and the receiver determines the metric according to the nearest ring for each symbol. The second case corresponds to realistic receiver realizations. As is shown, in both cases there is a fixed relation between the proposed metric and the commonly used performance metric ratio Es/No for AWGN channels. According to the characteristics described above, the proposed metric can be utilized for the estimation of channel condition under additive white Gaussian noise. This is accomplished through the processing of the STR output signal (symbol values) which usually comprises the first synchronization mechanism in digital satellite terminal receivers. It is worth mentioning that the channel estimation is feasible even under significant carrier frequency offset errors. The significance of the above process is related to the fact that this estimation can be exploited by the following synchronization subunits (of carrier frequency and phase) of the receiver in order to properly adjust and make their operations faster. A second application of the proposed metric is the recovery of large frequency offset errors by processing the signal at the input and the output of the matched filter of the previously mentioned STR structure. The control of frequency offset errors at such point of the receiver processing chain under specific limits, is critical as it strongly affects the performance and efficiency of the following synchronization mechanisms. Finally, this thesis analyzes and presents the implementation of a complete SDR IF demodulator that is compliant to DVB-S2 technology and is based on a reconfigurable hardware platform. This platform incorporates hardware (FPGA) and software (DSP) circuits in a unified environment. The IF demodulator supports single (QPSK/8PSK), two (16APSK) and three (32APSK) ring constellations and comprises a significant part of a full receiver implementation that includes all the necessary functions ranging from the manipulation of the input IF signal to the forwarding of the recovered user information to a Gigabit Ethernet (GbE) LAN. In addition, the IF demodulator implementation includes the hardware realization of the IF digital down-converter into an FPGA device and the software realization of the remaining synchronization procedures starting from the STR into the available DSP processors of the reconfigurable platform. The other necessary procedures for the proper demodulation of the input signal, are: frame synchronization, carrier frequency/phase recovery and amplitude normalization. Furthermore, information is also given on the implementation of the corresponding demapping, error correction and LAN interfacing procedures that are performed in the following processing stages of the DVB-S2 receiver.

Given the ever increasing reliance of today’s society on ubiquitous wireless access, the paradigm of dynamic spectrum access (DSA) as been proposed and implemented for utilizing the limited wireless spectrum more efficiently. Orthogonal frequency division multiplexing (OFDM) is growing in popularity for adoption into wireless services employing DSA frame- work, due to its high bandwidth efficiency and resiliency to multipath fading. While these advantages have been proven for many wireless applications, including LTE-Advanced and numerous IEEE wireless standards, one potential drawback of OFDM or its non-contiguous variant, NC-OFDM, is that it exhibits high peak-to-average power ratios (PAPR), which can induce in-band and out-of-band (OOB) distortions when the peaks of the waveform enter the compression region of the transmitter power amplifier (PA). Such OOB emissions can interfere with existing neighboring transmissions, and thereby severely deteriorate the reliability of the DSA network. A performance-enhancing digital pre-distortion (DPD) technique compensating for PA and in-phase/quadrature (I/Q) modulator distortions is proposed in this dissertation. Al- though substantial research efforts into designing DPD schemes have already been presented in the open literature, there still exists numerous opportunities to further improve upon the performance of OOB suppression for NC-OFDM transmission in the presence of RF front-end impairments. A set of orthogonal polynomial basis functions is proposed in this dissertation together with a simplified joint DPD structure. A performance analysis is presented to show that the OOB emissions is reduced to approximately 50 dBc with proposed algorithms employed during NC-OFDM transmission. Furthermore, a novel and intuitive DPD solution that can minimize the power regrowth at any pre-specified frequency in the spurious domain is proposed in this dissertation. Conventional DPD methods have been proven to be able to effectively reduce the OOB emissions that fall on top of adjacent channels. However more spectral emissions in more distant frequency ranges are generated by employing such DPD solutions, which are potentially in violation of the spurious emission limit. At the same time, the emissions in adjacent channel must be kept under the OOB limit. To the best of the author’s knowledge, there has not been extensive research conducted on this topic. Mathematical derivation procedures of the proposed algorithm are provided for both memoryless nonlinear model and memory-based nonlinear model. Simulation results show that the proposed method is able to provide a good balance of OOB emissions and emissions in the far out spurious domain, by reducing the spurious emissions by 4-5 dB while maintaining the adjacent channel leakage ratio (ACLR) improvement by at least 10 dB, comparing to the PA output spectrum without any DPD.

Based on different sampling theorem, for example classic Shannon’s sampling theorem and Papoulis’ generalized sampling theorem, signals are processed by the sampling devices without loss of information. As an interface between radio receiver front-ends and digital signal processing blocks, sampling devices play a dominant role in digital radio communications. Under the concept of Software Defined Radio (SDR), radio systems are going through the second evolution that mixes analog, digital and software technologies in modern radio designs. One design goal of SDR is to put the A/D converter as close as possible to the antenna. BandPass Sampling (BPS) enables one to have an interface between the RF or the higher IF signal and the A/D converter, and it might be a solution to SDR. However, three sources of performance degradation present in BPS systems, harmful signal spectral overlapping, noise aliasing and sampling timing jitter, hinder the conventional BPS theory from practical circuit implementations. In this thesis work, Generalized Quadrature BandPass Sampling (GQBPS) is first invented and comprehensively studied with focus on the noise aliasing problem. GQBPS consists of both BPS and FIR filtering that can use either real or complex coefficients. By well-designed FIR filtering, GQBPS can also perform frequency down-conversion in addition to noise aliasing reduction. GQBPS is a nonuniform sampling method in most cases. With respect to real circuit implementations, uniform sampling is easier to be realized compared to nonuniform sampling. GQBPS has been also extended to Generalized Uniform BandPass Sampling (GUBPS). GUBPS shares the same property of noise aliasing suppression as GQBPS besides that the samples are uniformly spaced. Due to the moving average operation of FIR filtering, the effect of sampling jitter is also reduced to a certain degree in GQBPS and GUBPS. By choosing a suitable sampling rate, harmful signal spectral overlapping can be avoided. Due to the property of quadrature sampling, the “self image” problem caused by I/Q mismatches is eliminated. Comprehensive theoretical analyses and program simulations on GQBPS and GUBPS have been done based on a general mathematic model. Circuit architecture to implementing GUBPS in Switched-Capacitor circuit technique has been proposed and analyzed. To improve the selectivity at the sampling output, FIR filtering is extended by adding a 1st order complex IIR filter in the implementation. GQBPS and GUBPS operate in voltage-mode. Besides voltage sampling, BPS can also be realized by charge sampling in current-mode. Most other research groups in this area are focusing on bandpass charge sampling. However, the theoretical analysis shows that our GQBPS and GUBPS in voltage mode are more efficient to suppress noise aliasing as compared to bandpass charge sampling with embedded filtering. The aliasing bands of sampled-data spectrum are always weighted by continuous-frequency factors for bandpass charge sampling with embedded filtering while discrete-frequency factors for GQBPS and GUBPS. The transmission zeros of intrinsic filtering will eliminate the corresponding whole aliasing bands of both signal and noise in GQBPS and GUBPS, while it will only cause notches at a limited set of frequencies in bandpass charge sampling. In addition, charge sampling performs an intrinsic continuous-time sinc function that always includes lowpass filtering. This is a drawback for a bandpass input signal. / QC 20100921

Software radios (SWR) have been proposed for wireless communication systems to enable them to operate according to incompatible wireless communication standards by implementing most analog functions in the digital section on software-reprogrammable hardware. However, this significantly increases the required computations for SWR functionality, mainly because of the digital front-end computationally intensive filtering functions, such as sample rate conversion (SRC), channelization, and equalization. For increasing the computational efficiency of SWR systems, two new SRC methods with better performance than conventional SRC methods are presented. In the first SRC method, we modify the conventional CIC filters to enable them to perform SRC on slightly oversampled signals efficiently. We also describe a SRC method with high efficiency for SRC by factors greater than unity at which SRC in SWR systems may be computationally demanding. This SRC method efficiently increases the sample rate of wideband signals, especially in SWR base station transmitters, by applying Lagrange interpolation for evaluating output samples hierarchically using a low-rate signal that is computed with low cost from the input signal. A new channelizer/synthesizer is also developed for extracting/combining frequency multiplexed channels in SWR transceivers. The efficiency of this channelizer/synthesizer, which uses modulated perfect reconstruction (PR) filter banks, is higher than polyphase filter banks (when applicable) for processing few channels, and significantly higher than discrete filter banks for processing any number of variable-bandwidth channels where polyphase filter banks are inapplicable. Because the available methods for designing modulated PR filter banks are inapplicable due to the required number of subchannels and stopband attenuation of the prototype filters, a new design method for these filter banks is introduced. This method is reliable and significantly faster than the existing methods. Modulated PR filter banks are also considered for implementing a frequency-domain block blind equalizer capable of equalizing SWR signals transmitted though channels with long impulse responses and severe intersymbol interference (ISI). This blind equalizer adapts by using separate sets of weights to correct for the magnitude and phase distortion of the channel. The adaptation of this blind equalizer is significantly more reliable and its computational requirements increase at a lower rate compared to conventional time-domain equalizers making it efficient for equalizing long channels that exhibit severe ISI.