A functional description of the advanced receiver

The breadboard Advanced Receiver 2 (ARX 2) that is currently being built for future use in NASA's Deep Space Network (DSN) is described. The hybrid analog/digital receiver performs multiple functions including carrier, subcarrier, and symbol synchronization. Tracking can be achieved for residual, suppressed, or hybrid carriers and for both sinusoidal and square-wave subcarriers. Other functions such as time-tagged Doppler extraction and monitor/control are also discussed, including acquisition algorithms and lock-detection schemes. System requirements are specified and a functional description of the ARX 2 is presented. The various digital signal-processing algorithms used are also discussed and illustrated with block diagrams

Direct-form adaptive equalization for underwater acoustic communication

Submitted in partial fulfillment of the requirements for the degree of Master of Science at the Massachusetts Institute of Technology and the Woods Hole Oceanographic Institution June 2012Adaptive equalization is an important aspect of communication systems in various environments. It is particularly important in underwater acoustic communication systems, as the channel has a long delay spread and is subject to the effects of time- varying multipath fading and Doppler spreading. The design of the adaptation algorithm has a profound influence on the performance of the system. In this thesis, we explore this aspect of the system. The emphasis of the work presented is on applying concepts from inference and decision theory and information theory to provide an approach to deriving and analyzing adaptation algorithms. Limited work has been done so far on rigorously devising adaptation algorithms to suit a particular situation, and the aim of this thesis is to concretize such efforts and possibly to provide a mathematical basis for expanding it to other applications. We derive an algorithm for the adaptation of the coefficients of an equalizer when the receiver has limited or no information about the transmitted symbols, which we term the Soft-Decision Directed Recursive Least Squares algorithm. We will demonstrate connections between the Expectation-Maximization (EM) algorithm and the Recursive Least Squares algorithm, and show how to derive a computationally efficient, purely recursive algorithm from the optimal EM algorithm. Then, we use our understanding of Markov processes to analyze the performance of the RLS algorithm in hard-decision directed mode, as well as of the Soft-Decision Directed RLS algorithm. We demonstrate scenarios in which the adaptation procedures fail catastrophically, and discuss why this happens. The lessons from the analysis guide us on the choice of models for the adaptation procedure. We then demonstrate how to use the algorithm derived in a practical system for underwater communication using turbo equalization. As the algorithm naturally incorporates soft information into the adaptation process, it becomes easy to fit it into a turbo equalization framework. We thus provide an instance of how to use the information of a turbo equalizer in an adaptation procedure, which has not been very well explored in the past. Experimental data is used to prove the value of the algorithm in a practical context.Support from the agencies that funded this research- the Academic Programs Office at WHOI and the Office of Naval Research (through ONR Grant #N00014-07-10738 and #N00014-10-10259)

Multidimensional Optimized Optical Modulation Formats

This chapter overviews the relatively large body of work (experimental and theoretical) on modulation formats for optical coherent links. It first gives basic definitions and performance metrics for modulation formats that are common in the literature. Then, the chapter discusses optimization of modulation formats in coded systems. It distinguishes between three cases, depending on the type of decoder employed, which pose quite different requirements on the choice of modulation format. The three cases are soft-decision decoding, hard-decision decoding, and iterative decoding, which loosely correspond to weak, medium, and strong coding, respectively. The chapter also discusses the realizations of the transmitter and transmission link properties and the receiver algorithms, including DSP and decoding. It further explains how to simply determine the transmitted symbol from the received 4D vector, without resorting to a full search of the Euclidean distances to all points in the whole constellation

High Capacity CDMA and Collaborative Techniques

The thesis investigates new approaches to increase the user capacity and improve the error performance of Code Division Multiple Access (CDMA) by employing adaptive interference cancellation and collaborative spreading and space diversity techniques. Collaborative Coding Multiple Access (CCMA) is also investigated as a separate technique and combined with CDMA. The advantages and shortcomings of CDMA and CCMA are analysed and new techniques for both the uplink and downlink are proposed and evaluated. Multiple access interference (MAI) problem in the uplink of CDMA is investigated first. The practical issues of multiuser detection (MUD) techniques are reviewed and a novel blind adaptive approach to interference cancellation (IC) is proposed. It exploits the constant modulus (CM) property of digital signals to blindly suppress interference during the despreading process and obtain amplitude estimation with minimum mean squared error for use in cancellation stages. Two new blind adaptive receiver designs employing successive and parallel interference cancellation architectures using the CM algorithm (CMA) referred to as ‘CMA-SIC’ and ‘BA-PIC’, respectively, are presented. These techniques have shown to offer near single user performance for large number of users. It is shown to increase the user capacity by approximately two fold compared with conventional IC receivers. The spectral efficiency analysis of the techniques based on output signal-to interference-and-noise ratio (SINR) also shows significant gain in data rate. Furthermore, an effective and low complexity blind adaptive subcarrier combining (BASC) technique using a simple gradient descent based algorithm is proposed for Multicarrier-CDMA. It suppresses MAI without any knowledge of channel amplitudes and allows large number of users compared with equal gain and maximum ratio combining techniques normally used in practice. New user collaborative schemes are proposed and analysed theoretically and by simulations in different channel conditions to achieve spatial diversity for uplink of CCMA and CDMA. First, a simple transmitter diversity and its equivalent user collaborative diversity techniques for CCMA are designed and analysed. Next, a new user collaborative scheme with successive interference cancellation for uplink of CDMA referred to as collaborative SIC (C-SIC) is investigated to reduce MAI and achieve improved diversity. To further improve the performance of C-SIC under high system loading conditions, Collaborative Blind Adaptive SIC (C-BASIC) scheme is proposed. It is shown to minimize the residual MAI, leading to improved user capacity and a more robust system. It is known that collaborative diversity schemes incur loss in throughput due to the need of orthogonal time/frequency slots for relaying source’s data. To address this problem, finally a novel near-unity-rate scheme also referred to as bandwidth efficient collaborative diversity (BECD) is proposed and evaluated for CDMA. Under this scheme, pairs of users share a single spreading sequence to exchange and forward their data employing a simple superposition or space-time encoding methods. At the receiver collaborative joint detection is performed to separate each paired users’ data. It is shown that the scheme can achieve full diversity gain at no extra bandwidth as inter-user channel SNR becomes high. A novel approach of ‘User Collaboration’ is introduced to increase the user capacity of CDMA for both the downlink and uplink. First, collaborative group spreading technique for the downlink of overloaded CDMA system is introduced. It allows the sharing of the same single spreading sequence for more than one user belonging to the same group. This technique is referred to as Collaborative Spreading CDMA downlink (CS-CDMA-DL). In this technique T-user collaborative coding is used for each group to form a composite codeword signal of the users and then a single orthogonal sequence is used for the group. At each user’s receiver, decoding of composite codeword is carried out to extract the user’s own information while maintaining a high SINR performance. To improve the bit error performance of CS-CDMA-DL in Rayleigh fading conditions, Collaborative Space-time Spreading (C-STS) technique is proposed by combining the collaborative coding multiple access and space-time coding principles. A new scheme for uplink of CDMA using the ‘User Collaboration’ approach, referred to as CS-CDMA-UL is presented next. When users’ channels are independent (uncorrelated), significantly higher user capacity can be achieved by grouping multiple users to share the same spreading sequence and performing MUD on per group basis followed by a low complexity ML decoding at the receiver. This approach has shown to support much higher number of users than the available sequences while also maintaining the low receiver complexity. For improved performance under highly correlated channel conditions, T-user collaborative coding is also investigated within the CS-CDMA-UL system

Ultra Wideband Communications: from Analog to Digital

Ultrabreitband-Signale (Ultra Wideband [UWB]) können einen signifikanten Nutzen im Bereich drahtloser Kommunikationssysteme haben. Es sind jedoch noch einige Probleme offen, die durch Systemdesigner und Wissenschaftler gelöst werden müssen. Ein Funknetzsystem mit einer derart großen Bandbreite ist normalerweise auch durch eine große Anzahl an Mehrwegekomponenten mit jeweils verschiedenen Pfadamplituden gekennzeichnet. Daher ist es schwierig, die zeitlich verteilte Energie effektiv zu erfassen. Außerdem ist in vielen Fällen der naheliegende Ansatz, ein kohärenter Empfänger im Sinne eines signalangepassten Filters oder eines Korrelators, nicht unbedingt die beste Wahl. In der vorliegenden Arbeit wird dabei auf die bestehende Problematik und weitere Lösungsmöglichkeiten eingegangen. Im ersten Abschnitt geht es um „Impulse Radio UWB”-Systeme mit niedriger Datenrate. Bei diesen Systemen kommt ein inkohärenter Empfänger zum Einsatz. Inkohärente Signaldetektion stellt insofern einen vielversprechenden Ansatz dar, als das damit aufwandsgünstige und robuste Implementierungen möglich sind. Dies trifft vor allem in Anwendungsfällen wie den von drahtlosen Sensornetzen zu, wo preiswerte Geräte mit langer Batterielaufzeit nötigsind. Dies verringert den für die Kanalschätzung und die Synchronisation nötigen Aufwand, was jedoch auf Kosten der Leistungseffizienz geht und eine erhöhte Störempfindlichkeit gegenüber Interferenz (z.B. Interferenz durch mehrere Nutzer oder schmalbandige Interferenz) zur Folge hat. Um die Bitfehlerrate der oben genannten Verfahren zu bestimmen, wurde zunächst ein inkohärenter Combining-Verlust spezifiziert, welcher auftritt im Gegensatz zu kohärenter Detektion mit Maximum Ratio Multipath Combining. Dieser Verlust hängt von dem Produkt aus der Länge des Integrationsfensters und der Signalbandbreite ab. Um den Verlust durch inkohärentes Combining zu reduzieren und somit die Leistungseffizienz des Empfängers zu steigern, werden verbesserte Combining-Methoden für Mehrwegeempfang vorgeschlagen. Ein analoger Empfänger, bei dem der Hauptteil des Mehrwege-Combinings durch einen „Integrate and Dump”-Filter implementiert ist, wird für UWB-Systeme mit Zeit-Hopping gezeigt. Dabei wurde die Einsatzmöglichkeit von dünn besetzten Codes in solchen System diskutiert und bewertet. Des Weiteren wird eine Regel für die Code-Auswahl vorgestellt, welche die Stabilität des Systems gegen Mehrnutzer-Störungen sicherstellt und gleichzeitig den Verlust durch inkohärentes Combining verringert. Danach liegt der Fokus auf digitalen Lösungen bei inkohärenter Demodulation. Im Vergleich zum Analogempfänger besitzt ein Digitalempfänger einen Analog-Digital-Wandler im Zeitbereich gefolgt von einem digitalen Optimalfilter. Der digitale Optimalfilter dekodiert den Mehrfachzugriffscode kohärent und beschränkt das inkohärente Combining auf die empfangenen Mehrwegekomponenten im Digitalbereich. Es kommt ein schneller Analog-Digital-Wandler mit geringer Auflösung zum Einsatz, um einen vertretbaren Energieverbrauch zu gewährleisten. Diese Digitaltechnik macht den Einsatz langer Analogverzögerungen bei differentieller Demodulation unnötig und ermöglicht viele Arten der digitalen Signalverarbeitung. Im Vergleich zur Analogtechnik reduziert sie nicht nur den inkohärenten Combining-Verlust, sonder zeigt auch eine stärkere Resistenz gegenüber Störungen. Dabei werden die Auswirkungen der Auflösung und der Abtastrate der Analog-Digital-Umsetzung analysiert. Die Resultate zeigen, dass die verminderte Effizienz solcher Analog-Digital-Wandler gering ausfällt. Weiterhin zeigt sich, dass im Falle starker Mehrnutzerinterferenz sogar eine Verbesserung der Ergebnisse zu beobachten ist. Die vorgeschlagenen Design-Regeln spezifizieren die Anwendung der Analog-Digital-Wandler und die Auswahl der Systemparameter in Abhängigkeit der verwendeten Mehrfachzugriffscodes und der Modulationsart. Wir zeigen, wie unter Anwendung erweiterter Modulationsverfahren die Leistungseffizienz verbessert werden kann und schlagen ein Verfahren zur Unterdrückung schmalbandiger Störer vor, welches auf Soft Limiting aufbaut. Durch die Untersuchungen und Ergebnissen zeigt sich, dass inkohärente Empfänger in UWB-Kommunikationssystemen mit niedriger Datenrate ein großes Potential aufweisen. Außerdem wird die Auswahl der benutzbaren Bandbreite untersucht, um einen Kompromiss zwischen inkohärentem Combining-Verlust und Stabilität gegenüber langsamen Schwund zu erreichen. Dadurch wurde ein neues Konzept für UWB-Systeme erarbeitet: wahlweise kohärente oder inkohärente Empfänger, welche als UWB-Systeme Frequenz-Hopping nutzen. Der wesentliche Vorteil hiervon liegt darin, dass die Bandbreite im Basisband sich deutlich verringert. Mithin ermöglicht dies einfach zu realisierende digitale Signalverarbeitungstechnik mit kostengünstigen Analog-Digital-Wandlern. Dies stellt eine neue Epoche in der Forschung im Bereich drahtloser Sensorfunknetze dar. Der Schwerpunkt des zweiten Abschnitts stellt adaptiven Signalverarbeitung für hohe Datenraten mit „Direct Sequence”-UWB-Systemen in den Vordergrund. In solchen Systemen entstehen, wegen der großen Anzahl der empfangenen Mehrwegekomponenten, starke Inter- bzw. Intrasymbolinterferenzen. Außerdem kann die Funktionalität des Systems durch Mehrnutzerinterferenz und Schmalbandstörungen deutlich beeinflusst werden. Um sie zu eliminieren, wird die „Widely Linear”-Rangreduzierung benutzt. Dabei verbessert die Rangreduzierungsmethode das Konvergenzverhalten, besonders wenn der gegebene Vektor eine sehr große Anzahl an Abtastwerten beinhaltet (in Folge hoher einer Abtastrate). Zusätzlich kann das System durch die Anwendung der R-linearen Verarbeitung die Statistik zweiter Ordnung des nicht-zirkularen Signals vollständig ausnutzen, was sich in verbesserten Schätzergebnissen widerspiegelt. Allgemeine kann die Methode der „Widely Linear”-Rangreduzierung auch in andern Bereichen angewendet werden, z.B. in „Direct Sequence”-Codemultiplexverfahren (DS-CDMA), im MIMO-Bereich, im Global System for Mobile Communications (GSM) und beim Beamforming.The aim of this thesis is to investigate key issues encountered in the design of transmission schemes and receiving techniques for Ultra Wideband (UWB) communication systems. Based on different data rate applications, this work is divided into two parts, where energy efficient and robust physical layer solutions are proposed, respectively. Due to a huge bandwidth of UWB signals, a considerable amount of multipath arrivals with various path gains is resolvable at the receiver. For low data rate impulse radio UWB systems, suboptimal non-coherent detection is a simple way to effectively capture the multipath energy. Feasible techniques that increase the power efficiency and the interference robustness of non-coherent detection need to be investigated. For high data rate direct sequence UWB systems, a large number of multipath arrivals results in severe inter-/intra-symbol interference. Additionally, the system performance may also be deteriorated by multi-user interference and narrowband interference. It is necessary to develop advanced signal processing techniques at the receiver to suppress these interferences. Part I of this thesis deals with the co-design of signaling schemes and receiver architectures in low data rate impulse radio UWB systems based on non-coherent detection.● We analyze the bit error rate performance of non-coherent detection and characterize a non-coherent combining loss, i.e., a performance penalty with respect to coherent detection with maximum ratio multipath combining. The thorough analysis of this loss is very helpful for the design of transmission schemes and receive techniques innon-coherent UWB communication systems.● We propose to use optical orthogonal codes in a time hopping impulse radio UWB system based on an analog non-coherent receiver. The “analog” means that the major part of the multipath combining is implemented by an integrate and dump filter. The introduced semi-analytical method can help us to easily select the time hopping codes to ensure the robustness against the multi-user interference and meanwhile to alleviate the non-coherent combining loss.● The main contribution of Part I is the proposal of applying fully digital solutions in non-coherent detection. The proposed digital non-coherent receiver is based on a time domain analog-to-digital converter, which has a high speed but a very low resolution to maintain a reasonable power consumption. Compared to its analog counterpart, itnot only significantly reduces the non-coherent combining loss but also offers a higher interference robustness. In particular, the one-bit receiver can effectively suppress strong multi-user interference and is thus advantageous in separating simultaneously operating piconets.The fully digital solutions overcome the difficulty of implementing long analog delay lines and make differential UWB detection possible. They also facilitate the development of various digital signal processing techniques such as multi-user detection and non-coherent multipath combining methods as well as the use of advanced modulationschemes (e.g., M-ary Walsh modulation).● Furthermore, we present a novel impulse radio UWB system based on frequency hopping, where both coherent and non-coherent receivers can be adopted. The key advantage is that the baseband bandwidth can be considerably reduced (e.g., lower than 500 MHz), which enables low-complexity implementation of the fully digital solutions. It opens up various research activities in the application field of wireless sensor networks. Part II of this thesis proposes adaptive widely linear reduced-rank techniques to suppress interferences for high data rate direct sequence UWB systems, where second-order non-circular signals are used. The reduced-rank techniques are designed to improve the convergence performance and the interference robustness especially when the received vector contains a large number of samples (due to a high sampling rate in UWB systems). The widely linear processing takes full advantage of the second-order statistics of the non-circular signals and enhances the estimation performance. The generic widely linear reduced-rank concept also has a great potential in the applications of other systems such as Direct Sequence Code Division Multiple Access (DS-CDMA), Multiple Input Multiple Output (MIMO) system, and Global System for Mobile Communications (GSM), or in other areas such as beamforming

Near-capacity fixed-rate and rateless channel code constructions

Fixed-rate and rateless channel code constructions are designed for satisfying conflicting design tradeoffs, leading to codes that benefit from practical implementations, whilst offering a good bit error ratio (BER) and block error ratio (BLER) performance. More explicitly, two novel low-density parity-check code (LDPC) constructions are proposed; the first construction constitutes a family of quasi-cyclic protograph LDPC codes, which has a Vandermonde-like parity-check matrix (PCM). The second construction constitutes a specific class of protograph LDPC codes, which are termed as multilevel structured (MLS) LDPC codes. These codes possess a PCM construction that allows the coexistence of both pseudo-randomness as well as a structure requiring a reduced memory. More importantly, it is also demonstrated that these benefits accrue without any compromise in the attainable BER/BLER performance. We also present the novel concept of separating multiple users by means of user-specific channel codes, which is referred to as channel code division multiple access (CCDMA), and provide an example based on MLS LDPC codes. In particular, we circumvent the difficulty of having potentially high memory requirements, while ensuring that each user’s bits in the CCDMA system are equally protected. With regards to rateless channel coding, we propose a novel family of codes, which we refer to as reconfigurable rateless codes, that are capable of not only varying their code-rate but also to adaptively modify their encoding/decoding strategy according to the near-instantaneous channel conditions. We demonstrate that the proposed reconfigurable rateless codes are capable of shaping their own degree distribution according to the nearinstantaneous requirements imposed by the channel, but without any explicit channel knowledge at the transmitter. Additionally, a generalised transmit preprocessing aided closed-loop downlink multiple-input multiple-output (MIMO) system is presented, in which both the channel coding components as well as the linear transmit precoder exploit the knowledge of the channel state information (CSI). More explicitly, we embed a rateless code in a MIMO transmit preprocessing scheme, in order to attain near-capacity performance across a wide range of channel signal-to-ratios (SNRs), rather than only at a specific SNR. The performance of our scheme is further enhanced with the aid of a technique, referred to as pilot symbol assisted rateless (PSAR) coding, whereby a predetermined fraction of pilot bits is appropriately interspersed with the original information bits at the channel coding stage, instead of multiplexing pilots at the modulation stage, as in classic pilot symbol assisted modulation (PSAM). We subsequently demonstrate that the PSAR code-aided transmit preprocessing scheme succeeds in gleaning more information from the inserted pilots than the classic PSAM technique, because the pilot bits are not only useful for sounding the channel at the receiver but also beneficial for significantly reducing the computational complexity of the rateless channel decoder

A Study on Efficient Receiver Design for UWA Communication System

Underwater Acoustic Channels are fast varying channel according to environmental conditions and exhibit strong random fluctuations in amplitude as well as phase due to reflection, refraction, and diffraction. Due to these highly space, time and frequency dependent channel characteristics, it is very difficult to establish reliable and long-range underwater acoustic communication. In this project, channel modeling has been done showing the different channel characteristics of underwater and their dependencies on frequency, temperature, pressure, salinity etc. Also, it has been shown through some theoretical and practical results that the nakagami fading is the best suitable generalized fading to be used in underwater. In this research work various techniques such as equalization, pilot based OFDM and LDPC Coding has also been done to mitigate the channel fading effect and to improve the performance. An adaptive equalizer has been implemented through three different algorithms LMS, NLMS and RLS for linear as well as non-linear channels to mitigate ISI and, their convergence characteristics along with bit error rate performance has been compared. Two types of pilot insertion, block and Comb type has also been done while implementing OFDM. Block type pilot based OFDM is suitable for slow fading and comb type pilot based OFDM is suitable for a fast fading channel. As in underwater, both types of fading exist, hence, lattice type pilot based OFDM is the best suitable for underwater acoustic communication. LDPC channel coding through which almost Shannon capacity performance can be achieved; has also been implemented taking nakagami channel fading. Bit error rate performance has been compared for different LDPC decoding techniques and for different code rate

Study of information transfer optimization for communication satellites

The results are presented of a study of source coding, modulation/channel coding, and systems techniques for application to teleconferencing over high data rate digital communication satellite links. Simultaneous transmission of video, voice, data, and/or graphics is possible in various teleconferencing modes and one-way, two-way, and broadcast modes are considered. A satellite channel model including filters, limiter, a TWT, detectors, and an optimized equalizer is treated in detail. A complete analysis is presented for one set of system assumptions which exclude nonlinear gain and phase distortion in the TWT. Modulation, demodulation, and channel coding are considered, based on an additive white Gaussian noise channel model which is an idealization of an equalized channel. Source coding with emphasis on video data compression is reviewed, and the experimental facility utilized to test promising techniques is fully described