We live in an information hungry age, we generate and process information at a rate never before recorded in the history of mankind. Today’s computing platforms are run on Gigahertz multi-core processors churning out Gigabits streams of data that need to be transmitted as quickly as possible. Often times the source and the destination are mobile which means wired connections are not a choice. This has led to an ever increasing need to develop wireless access technologies that support high throughput regardless of the transmission environment. Till date, many proprietary solutions exist that seek to bridge this gap with little or no support for interoperability. For the sheer scale of development that is required, a standard based solution is the key. The IEEE 802.1x committee oversees the development of standards for wireless systems, it formed the 802.16 working group to develop a standards-based Wireless Metropolitan Area Network (MAN) solution. One of the fruits of this effort is the 802.16e standard fondly referred to as mobile WiMAX and it is the subject of study in this thesis. This thesis seeks to analyze the transmission characteristics of two of the antenna systems defined in the standard i.e. Adaptive Beamforming Systems and Multiple-Input Multiple-Output Systems. Multiple-Input Multiple-Output (MIMO): utilizes multiple antennas at the transmitter and receiver to provide diversity gain, multiplexing gain or both. Adaptive Antenna Systems (AAS): Adaptive array system uses an antenna array to generate in real-time radiation patterns with the main lobes and/or nulls dynamically tuned to specific directions in order to increase or suppress signal power in that direction. / Worldwide Interoperability for Microwave Access (WiMAX) is the acronym for Institute of Electrical and Electronics Engineers (IEEE) 802.16 set of standards governing Air Interface for Fixed Broadband Wireless Access Systems. In the history of wireless systems, WiMAX is revolutionary technology as affords its users the Wi-Fi grade throughput and cellular system level of mobility. With WiMAX, broadband technology (traditionally ADSL and Fiber) goes wireless and WiMAX users can basically enjoy triple-play application, and split-second download and upload rates. WIMAX also offers full mobility much as traditional cellular systems do with features like seamless hand-over and roaming at vehicular speed; this is made possible because the system design covers the access network to core network. For the operator, WiMAX is a welcome development because it merges traditional cellular networks with broadband technology thus opening them to more business offerings and a larger client base and all this at a reduced cost of deployment. Base stations are comparatively cheaper and do not require extensive planning typical of other cellular systems thus WiMAX is aptly suited for emerging markets where infrastructure cost is a major issue; little wonder a lot of 3rd world countries have signified interest in the technology.

Acoustic Sound Source localization using signal processing is required in order to estimate the direction from where a particular acoustic source signal is coming and it is also important in order to find a soluation for hands free communication. Video conferencing, hand free communications are different applications requiring acoustic sound source localization. This applications need a robust algorithm which can reliably localize and position the acoustic sound sources. The Steered Response Power Phase Transform (SRP-PHAT) is an important and roubst algorithm to localilze acoustic sound sources. However, the algorithm has a high computational complexity thus making the algorithm unsuitable for real time applications. This thesis focuses on describe the implementation of the SRP-PHAT algorithm as a function of source type, reverberation levels and ambient noise. The main objective of this thesis is to present different approaches of the SRP-PHAT to verify the algorithm in terms of acoustic enviroment, microphone array configuration, acoustic source position and levels of reverberation and noise.

Abstract In this doctoral thesis, initial code synchronization (i.e., code acquisition) of a direct sequence spread spectrum (DS/SS) system is studied when a smart antenna is used in a receiver. Code synchronization means time synchronization of the used spreading code in the receiver. After an introduction to the topic, a literature review of code acquisition is presented. In addition, a review of the results in the literature under fading, data modulation, Doppler, intentional interference, multiple-access interference, other system interference, and multiple antennas is given. After that, an overview of the smart antennas, especially focusing on digital beamforming and direction-of-arrival (DOA) estimation algorithms is presented. The end part of the thesis concentrates on the author’s own research of the topic. Original articles of this article dissertation have been classified according to their contents into two groups. The first group covers DS/SS code acquisition performance in intentional interference by exploiting how well different beamforming algorithms can eliminate narrowband and wideband interfering signals in the case, where the DOA of the desired signal is known. The obtained results show that most spatial beamforming algorithms are capable of cancelling multiple different types of interfering signals if they are not arriving from the same direction as the desired signal. If angle separation between desired and interfering signals is not sufficient, then more complex methods have to be used. The second group of articles focuses on a theoretical analysis of synchronization probabilities and mean acquisition times. If the DOA of the desired signal is unknown, the whole angular uncertainty region can be divided into small angular cells using beamforming techniques, as is proposed in the literature. Then there is a two-dimensional (delay-angle) acquisition problem. In this thesis, the research work of that area is expanded to cover also advanced beamforming techniques, since they offer increased interference suppression capability. It is shown that the code acquisition performance of the delay-angle method can be improved in some cases by adding a DOA estimator into the system, because it may reduce the number of required angular cells. In addition, such a minimum mean square error (MMSE) beamforming structure is proposed, where only one period of the known pseudo noise spreading code is used as a reference signal. The method was shown to have better acquisition performance than the delay-angle method has, since MMSE beamforming does not need DOA information. However, in this thesis, such a method was not found which outperforms the rest of the methods in all scenarios. / Tiivistelmä Tässä väitöstyössä tutkitaan suorahajotushajaspektrijärjestelmän (DS/SS, direct sequence spread spectrum) koodisynkronoinnin etsintävaihetta, kun vastaanottimessa käytetään älyantennia. Koodisynkronoinnilla tarkoitetaan järjestelmän käyttämän hajotuskoodin ajastuksen synkronointia vastaanottimessa. Johdannon jälkeen esitetään kirjallisuuskatsaus koodisynkronointiin sekä tuodaan esille kirjallisuudesta löytyviä tutkimustuloksia aihepiiristä seuraavissa tilanteissa: häipyvä kanava, Doppler-ilmiö, tahallinen häirintä, monikäyttöhäiriö, muiden järjestelmien aiheuttama häiriö sekä moniantennijärjestelmät. Tämän jälkeen esitetään yleiskatsaus älyantenneihin kohdistuen erityisesti digitaalisiin keilanmuodostus- sekä suuntaestimointialgoritmeihin. Työn loppuosa keskittyy kirjoittajan omaan tutkimukseen aiheesta. Tämän nippuväitöskirjan alkuperäiset artikkelit on luokiteltu kahteen ryhmään niiden sisältöön perustuen. Ensimmäinen ryhmä käsittelee DS/SS-järjestelmän koodisynkronoinnin etsintävaiheen suorituskykyä tahallisessa häirinnässä tutkimalla, miten hyvin erilaiset keilanmuodostusalgoritmit kykenevät poistamaan kapea- ja leveäkaistaisia häirintäsignaaleja tilanteessa, jossa hyötysignaalin tulosuunta tiedetään. Tutkimustulokset osoittavat, että monet tilatason keilanmuodostusalgoritmit kykenevät poistamaan useita erityyppisiä häirintäsignaaleita, jos ne eivät saavu hyötysignaalin kanssa samasta suunnasta. Mikäli kulmaero hyöty- ja häirintäsignaalien välillä ei ole riittävä, joudutaan käyttämään rakenteeltaan monimutkaisempia menetelmiä. Toinen ryhmä artikkeleita keskittyy synkronointiin liittyvien todennäköisyyksien ja keskimääräisen etsintäajan teoreettiseen analyysiin. Jos hyötysignaalin tulosuunta on tuntematon, voidaan kulmaepävarmuusalue jakaa pieniin kulmasoluihin käyttäen keilanmuodostustekniikoita, kuten kirjallisuudessa esitetään. Tällöin kyseessä on kaksiulotteinen (viive-kulma) etsintäongelma. Tässä työssä kyseistä tutkimusaihetta laajennetaan koskemaan myös edistyneet keilanmuodostusmenetelmät, koska ne tarjoavat parantuneen häiriönvaimennuskyvyn. Työssä osoitetaan, että viive-kulma menetelmän suorituskykyä voidaan parantaa joissakin tilanteissa lisäämällä järjestelmään suuntaestimaattori, koska se saattaa vähentää tarvittavien kulmasolujen lukumäärää. Lisäksi tutkitaan sellaista pienimmän keskineliövirheen (MMSE, minimum mean square error) keilanmuodostusmenetelmää, jossa ainoastaan yhtä hajotuskoodin koodijaksoa käytetään opetukseen. Kyseisellä menetelmällä todettiin olevan parempi suorituskyky kuin viive-kulma etsinnällä, koska MMSE-menetelmä ei tarvitse suuntainformaatiota. Tässä työssä ei kuitenkaan löydetty yhtä sellaista menetelmää, jonka suorituskyky on muita parempi kaikissa tilanteissa.

Tato diplomová práce se zabývá problematikou vzdáleného rozpoznávání mluvčích. V případě dat zachycených odlehlým mikrofonem se přesnost standardního rozpoznávání značně snižuje, proto jsem navrhl dva přístupy pro zlepšení výsledků. Prvním z nich je použití mikrofonního pole (záměrně rozestavené sady mikrofonů), které je schopné nasměrovat virtuální "paprsek" na pozici řečníka. Dále jsem prováděl adaptaci komponent systému (PLDA skórování a extraktoru i-vektorů). S využitím simulace pokojových podmínek jsem syntetizoval trénovací a testovací data ze standardní datové sady NIST 2010. Ukázal jsem, že obě techniky a jejich kombinace vedou k výraznému zlepšení výsledků. Dále jsem se zabýval společným určením identity a pozice mluvčího. Zatímco výsledky ve venkovním simulovaném prostředí (bez ozvěn) jsou slibné, výsledky z interiéru (s ozvěnami) jsou smíšené a vyžadují další prozkoumání. Na závěr jsem mohl systémem vyhodnotit omezené množství reálných dat získaných přehráním a záznamem nahrávek ve skutečné místnosti. Zatímco výsledky pro mužské nahrávky odpovídají simulaci, výsledky pro ženské nahrávky nejsou přesvědčivé a vyžadují další analýzu.

This thesis deals with design and realization of acoustic camera for sound source location. The essence of the device is microphone field listening for acoustic impulses which are processed in real-time in order to locate the source. Found location is displayed on LCD together with images from regular video camera. FPGA is used as a computational unit.

The aim of this master´s thesis is to work out a comparison and evaluation of selected methods used for sound field mapping. Thesis will include software tools and results of sample measurement on test object for each mapping method.

This thesis exploits index modulation techniques to design energy- and spectrum-efficient system models to operate in future wireless networks. In this respect, index modulation techniques are studied considering two different media: mapping the information onto the frequency indices of multicarrier systems, and onto the antenna array indices of a platform that comprises multiple antennas. The index modulation techniques in wideband communication scenarios considering orthogonal and generalized frequency division multiplexing systems are studied first. Single cell multiuser networks are considered while developing the system models that exploit the index modulation on the subcarriers of the multicarrier systems. Instead of actively modulating all the subcarriers, a subset is selected according to the index modulation bits. As a result, there are subcarriers that remain idle during the data transmission phase and the activation pattern of the subcarriers convey additional information. The transceivers for the orthogonal and generalized frequency division multiplexing systems with index modulation are both designed considering the uplink and downlink transmission phases with a linear combiner and precoder in order to reduce the system complexity. In the developed system models, channel state information is required only at the base station. The linear combiner is designed adopting minimum mean square error method to mitigate the inter-user-interference. The proposed system models offer a flexible design as the parameters are independent of each other. The parameters can be adjusted to design the system in favor of the energy efficiency, spectrum efficiency, peak-to-average power ratio, or error performance. Then, the index modulation techniques are studied for large-scale multiple-input multiple-output systems that operate in millimeter wave bands. In order to overcome the drawbacks of transmission in millimeter wave frequencies, channel properties should be taken in to account while envisaging the wireless communication network. The large-scale multiple-input multiple-output systems increase the degrees of freedom in the spatial domain. This feature can be exploited to focus the transmit power directly onto the intended receiver terminal to cope with the severe path-loss. However, scaling up the number of hardware elements results in excessive power consumption. Hybrid architectures provide a remedy by shifting a part of the signal processing to the analog domain. In this way, the number of bulky and high power consuming hardware elements can be reduced. However, there will be a performance degradation as a consequence of renouncing the fully digital signal processing. Index modulation techniques can be combined with the hybrid system architecture to compensate the loss in spectrum efficiency to further increase the data rates. A user terminal architecture is designed that employs analog beamforming together with spatial modulation where a part of the information bits is mapped onto the indices of the antenna arrays. The system is comprised a switching stage that allocates the user terminal antennas on the phase shifter groups to minimize the spatial correlation, and a phase shifting stage that maximizes the beamforming gain to combat the path-loss. A computationally efficient optimization algorithm is developed to configure the system. The flexibility of the architecture enables optimization of the hybrid transceiver at any signal-to-noise ratio values. A base station is designed in which hybrid beamforming together with spatial modulation is employed. The analog beamformer is designed to point the transmit beam only in the direction of the intended user terminal to mitigate leakage of the transmit power to other directions. The analog beamformer to transmit the signal is chosen based on the spatial modulation bits. The digital precoder is designed to eliminate the inter-user-interference by exploiting the zero-forcing method. The base station computes the hybrid beamformers and the digital combiners, and only feeds back the digital combiners of each antenna array-user pair to the related user terminals. Thus, a low complexity user architecture is sufficient to achieve a higher performance. The developed optimization framework for the energy efficiency jointly optimizes the number of served users and the total transmit power by utilizing the derived upper bound of the achievable rate. The proposed transceiver architectures provide a more energy-efficient system model compared to the hybrid systems in which the spatial modulation technique is not exploited. This thesis develops low-complexity system models that operate in narrowband and wideband channel environments to meet the energy and spectrum efficiency demands of future wireless networks. It is corroborated in the thesis that adopting index modulation techniques both in the systems improves the system performance in various aspects.:1 Introduction 1 1.1 Motivation 1 1.2 Overview and Contribution 2 1.3 Outline 9 2 Preliminaries and Fundamentals 13 2.1 Multicarrier Systems 13 2.2 Large-scale Multiple Input Multiple Output Systems 17 2.3 Index Modulation Techniques 19 2.4 Single Cell Multiuser Networks 22 3 Multicarrier Systems with Index Modulation 27 3.1 Orthogonal Frequency Division Multiplexing 28 3.2 Generalized Frequency Division Multiplexing 40 3.3 Summary 52 4 Hybrid Beamforming with Spatial Modulation 55 4.1 Uplink Transmission 56 4.2 Downlink Transmission 74 4.3 Summary 106 5 Conclusion and Outlook 109 5.1 Conclusion 109 5.2 Outlook 111 A Quantization Error Derivations 113 B On the Achievable Rate of Gaussian Mixtures 115 B.1 The Conditional Density Function 115 B.2 Tight Bounds on the Differential Entropy 116 B.3 A Bound on the Achievable Rate 118 C Multiuser MIMO Downlink without Spatial Modulation 121 Bibliography

Depuis plusieurs années, les opérateurs de téléphonie mobile sont confrontés à une croissance considérable du trafic de données mobiles. Dans un tel contexte, la technologie Cloud Radio Access Network (CRAN) qui intègre les solutions de Cloud Computing aux réseaux d’accès radio est considérée comme une nouvelle architecture pour les futures générations de réseaux 5G. L’approche CRAN permet une optimisation globale des fonctions de traitement en bande de base du signal et de la gestion des ressources radio pour l’ensemble des RRH et des utilisateurs. Parallèlement, les réseaux hétérogènes (HetNets) ont été proposés pour augmenter efficacement la capacité et la couverture du réseau 5G tout en réduisant la consommation énergétique. En combinant les avantages du Cloud avec ceux des réseaux HetNets, le concept de réseaux H-CRAN (Heterogeneous Cloud Radio Access Networks) est né et est considéré comme l’une des architectures les plus prometteuses pour répondre aux exigences des futurs systèmes. Plus particulièrement, nous abordons le problème important de l’optimisation jointe de l’association utilisateur-RRH et de la solution de beamforming sur la liaison descendante d’un système H-CRAN. Nous formulons un problème de maximisation du débit total du système sous des contraintes de mobilité et d’imperfection de CSI (Channel State Information). Notre principal défi consiste à concevoir une solution capable de maximiser le débit tout en permettant, contrairement aux autres solutions de référence, de réduire la complexité de calcul, et les coûts de signalisation et de feedback CSI dans divers environnements. Notre étude commence par proposer un algorithme Hybride, qui active périodiquement des schémas de clustering dynamiques et statiques pour aboutir à un compromis satisfaisant entre optimalité et le coût en complexité et signalisation CSI et réassociation. L’originalité de l’algorithme Hybride réside aussi dans sa prise en compte de la dimension temporelle du processus d’allocation sur plusieurs trames successives plutôt que son optimalité (ou sous-optimalité) pour la seule trame d’ordonnancement courante. De plus, nous développons une analyse des coûts de l’algorithme en fonction de plusieurs critères afin de mieux appréhender le compromis entre les nombreux paramètres impliqués. La deuxième contribution de la thèse s’intéresse au problème sous la perspective de la mobilité utilisateur. Deux variantes améliorées de l’algorithme Hybride sont proposées : ABUC (Adaptive Beamforming et User Clustering), une version adaptée à la mobilité des utilisateurs et aux variations du canal radio, et MABUC (Mobility-Aware Beamforming et User Clustering), une version améliorée qui règle dynamiquement les paramètres de feedback du CSI (périodicité et type de CSI) en fonction de la vitesse de l’utilisateur. L’algorithme MABUC offre de très bonnes performances en termes de débit cible tout en réduisant efficacement la complexité et les coûts de signalisation CSI. Dans la dernière contribution de la thèse, nous approfondissons l’étude en explorant l’optimisation automatique des paramètres d’ordonnancement du CSI. Pour ce faire, nous exploitons l’outil de l’apprentissage par renforcement afin d’optimiser les paramètres de feedback CSI en fonction du profil de mobilité individuelle des utilisateurs. Plus spécifiquement, nous proposons deux modèles d’apprentissage. Le premier modèle basé sur un algorithme de type Q-learning a permis de démontrer l’efficacité de l’approche dans un scénario à taille réduite. Le second modèle, plus scalable car basé sur une approche Deep Q-learning, a été formulé sous la forme d’un processus de type POMDP (Partially observable Markov decision process). Les résultats montrent l’efficacité des solutions qui permettent de sélectionner les paramètres de feedback les plus adaptés à chaque profil de mobilité, même dans le cas complexe où chaque utilisateur possède un profil de mobilité différent et variable dans le temps. / Recently, mobile operators have been challenged by a tremendous growth in mobile data traffic. In such a context, Cloud Radio Access Network (CRAN) has been considered as a novel architecture for future wireless networks. The radio frequency signals from geographically distributed antennas are collected by Remote Radio Heads (RRHs) and transmitted to the cloud-centralized Baseband Units (BBUs) pool through fronthaul links. This centralized architecture enables a global optimization of joint baseband signal processing and radio resource management functions for all RRHs and users. At the same time, Heterogeneous Networks (HetNets) have emerged as another core feature for 5G network to enhance the capacity/coverage while saving energy consumption. Small cells deployment helps to shorten the wireless links to end-users and thereby improving the link quality in terms of spectrum efficiency (SE) as well as energy efficiency (EE). Therefore, combining both cloud computing and HetNet advantages results in the so-called Heterogeneous-Cloud Radio Access Networks (H-CRAN) which is regarded as one of the most promising network architectures to meet 5G and beyond system requirements. In this context, we address the crucial issue of beamforming and user-to-RRH association (user clustering) in the downlink of H-CRANs. We formulate this problem as a sum-rate maximization problem under the assumption of mobility and CSI (Channel State Information) imperfectness. Our main challenge is to design a framework that can achieve sum-rate maximization while, unlike other traditional reference solutions, being able to alleviate the computational complexity, CSI feedback and reassociation signaling costs under various mobility environments. Such gain helps in reducing the control and feedback overhead and in turn improve the uplink throughput. Our study begins by proposing a simple yet effective algorithm baptized Hybrid algorithm that periodically activates dynamic and static clustering schemes to balance between the optimality of the beamforming and association solutions while being aware of practical system constraints (complexity and signaling overhead). Hybrid algorithm considers time dimension of the allocation and scheduling process rather than its optimality (or suboptimality) for the sole current scheduling frame. Moreover, we provide a cost analysis of the algorithm in terms of several parameters to better comprehend the trade-off among the numerous dimensions involved in the allocation process. The second key contribution of our thesis is to tackle the beamforming and clustering problem from a mobility perspective. Two enhanced variants of the Hybrid algorithm are proposed: ABUC (Adaptve Beamforming and User Clustering), a mobility-aware version that is fit to the distinctive features of channel variations, and MABUC (Mobility-Aware Beamforming and User Clustering), an advanced version of the algorithm that tunes dynamically the feedback scheduling parameters (CSI feedback type and periodicity) in accordance with individual user velocity. MABUC algorithm achieves a targeted sum-rate performance while supporting the complexity and CSI signaling costs to a minimum. In our last contribution, we propose to go further in the optimization of the CSI feedback scheduling parameters. To do so, we take leverage of reinforcement learning (RL) tool to optimize on-the-fly the feedback scheduling parameters according to each user mobility profile. More specifically, we propose two RL models, one based on Q-learning and a second based on Deep Q-learning algorithm formulated as a POMDP (Partially observable Markov decision process). Simulation results show the effectiveness of our proposed framework, as it enables to select the best feedback parameters tailored to each user mobility profile, even in the difficult case where each user has a different mobility profile.

京都大学 / 0048 / 新制・課程博士 / 博士(情報学) / 甲第20034号 / 情博第629号 / 新制||情||109(附属図書館) / 33130 / 京都大学大学院情報学研究科通信情報システム専攻 / (主査)教授 佐藤 亨, 教授 守倉 正博, 教授 山本 衛 / 学位規則第4条第1項該当 / Doctor of Informatics / Kyoto University / DFAM

While it is difficult for base stations to estimate the millimeter wave (mmWave) channels and find the optimal mmWave beam for user equipments (UEs) quickly, the sub-6 GHz channels which are usually easier to obtain and more robust to blockages could be used to reduce the time before initial access and enhance the reliability of mmWave communication. Considering that the channel information is collected by a massive number of radio base stations and would be sensitive to privacy and security, Federated Learning (FL) is a match for this use case. In practice, the channel vectors are usually subject to Non-Independently Distributed (non-IID) distributions due to the greatly varying wireless communication environments between different radio base stations and their UEs. To achieve satisfying performance for all radio base stations instead of only the majority of them, a useful solution is designing personalized methods for each radio base station. In this thesis, we implement two personalized FL methods including 1) Finetuning FL Model on Private Dataset of Each Client and 2) Adaptive Expert Models for FL to predict the optimal mmWave beamforming vector directly from the non-IID sub-6 GHz channel vectors generated from DeepMIMO. According to our experimental results, Finetuning FL Model on Private Dataset of Each Client achieves higher average mmWave downlink spectral efficiency than the global FL. Besides, in terms of the average Top-1 and Top-3 classification accuracies, its performance improvement over the global FL model even exceeds the improvement of the global FL over the pure local models. / Även om det är svårt för en basstation att uppskatta en kanal för millimetervåg (mmWave) och snabbt hitta den bästa mmWave-strålen för en användarutrustning (UE), kan den dra fördel av kanaler under 6 GHz, som i allmänhet är mer lättillgängliga och mer motståndskraftig mot blockering, för att minska tid för första besök och förbättra tillförlitligheten hos mmWave-kommunikation. Med tanke på att kanalinformation samlas in av ett stort antal radiobasstationer och är känslig för integritet och säkerhet är federated learning (FL) väl lämpat för detta användningsfall. I praktiken, eftersom den trådlösa kommunikationsmiljön varierar mycket mellan olika radiobasstationer och deras UE, följer kanalvektorer vanligtvis en icke-oberoende distribution (icke-IID). För att uppnå tillfredsställande prestanda för alla radiobasstationer, inte bara de flesta radiobasstationer, är en användbar lösning att utforma ett individuellt tillvägagångssätt för varje radiobasstation. I detta dokument implementerar vi två personliga FL-metoder, inklusive 1) finjustering av FL-modellen på varje klients privata datauppsättning och 2) en adaptiv expertmodell av FL för att direkt generera icke-IID sub-6 GHz kanalvektorer förutsäga optimal mmWave beamforming vektorer. Enligt våra experimentella resultat uppnår finjustering av FL-modellen på varje klients privata datauppsättning högre genomsnittlig mmWave-nedlänksspektral effektivitet än global FL. Dessutom överträffar dess prestandaförbättring jämfört med den globala FL-modellen till och med den för den globala FL jämfört med den rent lokala modellen vad gäller genomsnittlig klassificeringsnoggrannhet i topp-1 och topp-3.