In this thesis, we propose a novel architecture for hybrid radio frequency (RF)/free–space optics (FSO) wireless systems. Hybrid RF/FSO systems are attractive since the RF and FSO sub–systems are affected differently by weather and fading phenomena. We give a thorough introduction to the RF and FSO technology, respectively. The state of the art of hybrid RF/FSO systems is reviewed. We show that a hybrid system robust to different weather conditions is obtained by joint bit–interleaved coded modulation (BICM) of the bit streams transmitted over the RF and FSO sub–channels. An asymptotic performance analysis reveals that a properly designed convolutional code can exploit the diversity offered by the independent sub–channels. Furthermore, we develop code design and power assignment criteria and provide an efficient code search procedure. The cut–off rate of the proposed hybrid system is also derived and compared to that of hybrid systems with perfect channel state information at the transmitter. Simulation results show that hybrid RF/FSO systems with BICM outperform previously proposed hybrid systems employing a simple repetition code and selection diversity. / Applied Science, Faculty of / Electrical and Computer Engineering, Department of / Graduate

La croissance exponentielle du trafic de données sans fils, causée par l'Internet mobile et les smartphones, contraint les futurs systèmes radio à inclure des modulations/formes d'ondes plus avancées offrant un débit plus élevé et une utilisation efficace des ressources spectrales. Les transmissions dites Faster-Than-Nyquist (FTN), introduites en 1975, sont parmi les meilleurs candidates pour répondre à ces besoins. En transmettant les symboles à une cadence plus rapide que celle définie par le critère de Nyquist, FTN peut théoriquement augmenter le débit mais en introduisant des interférences en contrepartie. Dans cette thèse, nous explorons le concept des transmissions FTN à travers un canal AWGN (Additive White Gaussian Noise) dans le contexte des modulations OFDM/OQAM (Orthogonal Frequency Division Multiplexing with Offset Quadrature Amplitude Modulation).L'objectif principal de cette thèse est de présenter un système OFDM/OQAM qui permet de transmettre l'information au-delà de la cadence de Nyquist tout en tenant en compte la complexité globale du système. Tout d'abord, nous proposons une nouvelle implémentation efficace des systèmes OFDM/OQAM appliquant le concept FTN, désignée ici par FTN-OQAM, qui garde la même complexité que les systèmes OFDM/OQAM et qui permet un gain en débit très proche du gain théorique. Vu que la condition de Nyquist n'est plus respectée, le signal transmis est maintenant perturbé par des interférences. Pour remédier à ce problème, nous proposons un récepteur basé sur le principe de l'égalisation linéaire sous le critère minimum erreur quadratique moyenne avec annulation d'interférences appelé MMSE LE-IC. Le but de notre système est d'augmenter le débit de transmission, ce qui signifie que des constellations d'ordres élevés seront ciblées. Dans ce contexte, le MMSE LE-IC, dont la complexité est indépendante de la constellation, représente un bon compromis entre efficacité et complexité. Puisque la modulation OFDM/OQAM utilise différents types de formes d'ondes, nous proposons pour plusieurs d'entre elles un algorithme pour déterminer la valeur minimale du facteur d'accélération, en fonction de l'ordre de constellation, qui apporte un gain en efficacité spectrale tout en gardant les mêmes performances que les systèmes respectant le critère de Nyquist à un SNR fixé. Ensuite, nous étudions l'amélioration du traitement itératif de l'émetteur-récepteur. La méthode proposée consiste à combiner un précodeur avec le système FTN-OQAM afin de réduire les interférences causées par du FTN à l'émission. Nous proposons un modèle de précodage dispersé, car il est difficile de précoder conjointement tous les symboles transmis. Nous présentons trois familles de précodeurs avec les récepteurs correspondants. En outre, nous modifions différents blocs de l'émetteur FTN-OQAM tels que le codage canal, le mappage des bits et le mappage des symboles afin d'améliorer davantage le transmetteur FTN-OQAM. Les résultats présentés révèlent le potentiel important des systèmes proposés. / The exponential growth of wireless data traffic driven by mobile Internet and smart devices constrains the future radio systems to include advanced modulations/waveforms offering higher data rates with more efficient bandwidth usage. One possibility is to violate the well known Nyquist criterion by transmitting faster than the Nyquist rate, i.e., using a technique also known as Faster-Than-Nyquist (FTN) signaling. Nyquist-based systems have the advantage of simple transmitter and receiver architectures at the detriment of bandwidth efficiency. The idea of signaling beyond the Nyquist rate to trade the interference-free transmission for more throughput goes back to 1975. In this dissertation, we investigate the concept of FTN signaling over Additive White Gaussian Noise (AWGN) channel in the context of Orthogonal Frequency Division Multiplexing with Offset Quadrature Amplitude Modulation OFDM/OQAM modulation.The main objective of our work is to present an OFDM/OQAM system signaling faster than the Nyquist one and explore its potential rate improvement while keeping under consideration the overall system complexity. First, we propose a new efficient FTN implementation of OFDM/OQAM systems, denoted by FTN-OQAM, that has the same complexity as OFDM/OQAM systems, while approaching very closely the FTN theoretical rate improvement. As the Nyquist condition is no longer respected, severe interference impacts the transmitted signals. To deal with the introduced interferences, we propose a turbo-like receiver based on Minimum Mean Square Error Linear Equalization and Interference Cancellation, named MMSE LE-IC. The aim of our system is to boost the transmission rate, which means that high constellation orders will be targeted. In this respect, the MMSE LE-IC, whose complexity is independent of the constellation, turns out to be a good candidate. Since OFDM/OQAM modulation can be equipped with different types of pulse shapes, we propose an algorithm to find, for different constellation orders, the minimum achieved FTN packing factor for various pulse shapes. Then, we aim at improving the iterative processing of the introduced transceiver. The proposed method involves combining a precoder with the FTN-OQAM system in order to remove FTN-induced interference at the transmitter. We also present a sparse precoding pattern as it is difficult to jointly precode all the transmitted symbols. We introduce three families of precoders along with the corresponding receivers. Furthermore, we propose several modifications of the FTN-OQAM transmitter concerning different blocks such as channel coding, bits mapping and symbols mapping to further enhance the FTN-OQAM transceiver design. Presented results reveal the significant potential of the proposed methods.

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / In this master thesis it is proposed a MIMO communication scheme with four transmit antennas and unitary transmission rate. The performance and robustness of the scheme are evaluated and compared with other good proposals recently presented in the literature. The proposed scheme uses a preprocessor based on phase feedback. The preprocessor allows the proposed scheme to obtain full diversity and also coding gain. The pre-processing considers a codebook design, whose size is dependent on the number of feedback bits. Error probability analysis is provided, where the upper and lower bounds for different numbers of feedback bits and antennas are presented, considering two different types of constellations, QAM and PSK. For robustness analysis, channels estimators were used. For the channel estimators model it was added the effects of spatial correlation, enabling the evaluation the losses caused by the spatial correlation among the antennas, and the Doppler effect, for evaluating the loss performance due to the relative mobilitys between trnamitter and receiver. / Nesta dissertação propõe-se um esquema de comunicação MIMO para quatro antenas transmissoras com taxa de transmissão unitária. O desempenho e a robustez do esquema proposto são avaliados e comparados com outras boas propostas recentemente apresentadas na literatura. O esquema proposto utiliza um pré-processador baseado na realimentação de fase provinda do receptor. O pré-processamento permite que o esquema proposto alcance um grau de diversidade completo bem como um ganho de codificação. A pré-codificação considera o uso de um codebook, cujo comprimento depende do número de bits de realimentação disponível. Faz-se uma análise de desempenho da probabilidade de erro, mostrando os limitantes superior e inferior do esquema para diferentes quantidades de bits de realimentação, diferentes números de antenas e para as constelações do tipo PSK e QAM. Para a análise de robustez, considerou-se o emprego de estimadores de canais clássicos. No modelo utilizado para os estimadores de canais foram acrescentados os efeitos da correlação espacial, para se verificar as perdas decorrentes da correlação espacial entre as antenas, e do efeito Doppler, para se avaliar a perda de desempenho decorrente da mobilidade relativa entre transmissor e receptor.

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / In this master s thesis, a wireless communication scheme with a single transmit and single receive antenna, SISO case, with use of the OFDM multicarrier technique and application of LDPC coding was considered. It is well known that one of the biggest challenge related to OFDM system refers to the reduction of the peak-to-average power ratio (PAPR) factor. Thus this master s thesis proposes the use of 16-APSK and 16-QAM modulations by aiming to mitigate the PAPR factor. It is also considered the use of LDPC codes to evaluate the performance of bit error rate (BER) versus signal-to-noise ratio (SNR) performance for the 16-QAM, A16- QAM and 16-APSK constellations. The LDPC codes were adopted since they are one of the most effective methods of channel coding presented in the literature. Monte Carlo simulations were performed in AWGN and Rayleigh fading channels in order to evaluate the system performance in terms of BER × SNR and the PAPR factor analysis from the perspective of CCDF curves. Results reveal that with the use of LDPC codes associated with the use of 16-APSK constellations in multicarrier systems promotes BER performance gain and also reduction of the PAPR factor. / Nesta dissertação foi considerado um esquema de comunicação sem fio, com uma antena transmissora e uma antena receptora, caso SISO, com uso da técnica multiportadora OFDM e aplicação de codificação LDPC. Sabe-se que um dos maiores desafios na utilização do sistema OFDM refere-se à redução do fator razão de potência de pico e potência média (PAPR). Assim, nesta dissertação de mestrado propõe-se o emprego de modulações 16-APSK e A16-QAM com intuito de mitigar esse fator. Considera-se também o emprego de códigos LDPC para melhorar o desempenho de taxa de erro de bit (BER) versus razão sinal-ruído (SNR) para as constelações 16-QAM, A16-QAM e 16-APSK. Os códigos LDPC foram adotados por serem um dos mais eficazes métodos de codificação de canal apresentados na literatura. Foram realizadas simulações do tipo Monte Carlo em canais AWGN e com desvanecimento do tipo Rayleigh para se avaliar o desempenho do sistema em termos de curvas de BER × SNR e o fator PAPR sob a perspectiva de curvas de CCDF. Os resultados revelam que o emprego de códigos LDPC associados ao uso de constelacões 16-APSK em sistemas multiportadora promovem ganhos na taxa de erros e redução no fator PAPR.

This thesis deals with Matlab application developed for simulation of the DVB-T channel coder and decoder. The first part of this thesis includes description of terrestrial digital video broadcasting system and comparison with analogue television. Channel coding and OFDM modulation, used in the DVB-T standard, is described in detail. Application developed in the Matlab environment is described in the second part. The application simulates data transfer of the DVB-T system. Results of the simulated transmission, using developed application are presented in the last part. Namely dependence of the BER on the S/N ratio, using various coder settings, was examined. Maximal possible data rate was determined for these various setting. All obtained values are graphically represented.

The paper deals with the channel coding and modulation in DVB-H and DVB-SH and with the simulation of the components in MATLAB. In the case of DVB-H the document also discuss the influence of different types of transmission channels on the transmission process. The first part of this work describes the channel coding and inner modulation in DVB-H and DVB-SH. The next part describes applications created in MATLAB for settings and simulation of the transmission in the standards DVB-H and DVB-SH. The last part of this document graphically presents selected results of the simulation of the transmission and discuss them.

LTE (Long Term Evolution) a été développé et normalisé par le 3GPP (3rd Generation Partnership Project). C’est un réseau à commutation de paquets. Cela signifie que la voix sur LTE (VoLTE) est un service de VoIP avec les exigences de qualité de service garantis au lieu de transmettre dans un réseau à commutation de circuits tels que les systèmes existants (2G/3G). VoLTE est déployé dans un réseau entièrement IP combinée avec IMS (IP Sous-système Multimédia). De ce fait, le déploiement de VoLTE est assez complexe et comment assurer la qualité de transmission de la voix sur les réseaux LTE est un très grand défi. Ainsi, il faut plusieurs solutions différentes pour renforcer et améliorer la qualité de transmission de la voix sur les réseaux LTE. Dans cette thèse, nous présentons des solutions en vue d’améliorer la qualité de transmission de la voix sur les réseaux LTE pour les services audio à bandes étroites et larges. Pour cela, il nous faudra différents facteurs complets en solutions. L’un d’eux est QoE (Qualité de l’Expérience) qui est une nouvelle tendance. Et afin de déterminer la perception des utilisateurs pour le service en temps réel tel que VoLTE, nous utilisons le E-model étendu et le WB (large bande) E-model pour des services audio à bandes étroites et larges respectivement. Les solutions proposées ici portent principalement sur des éléments clés dans les réseaux LTE, tels que le codage par chaine, MAC (Contrôle d’Accès Moyen) des systèmes de planification et la qualité de voix du moniteur décrits comme suit. Tout d’abord, des algorithmes améliorés pour renforcer le codec de la chaine LTE (codeur et décodeur) ont été proposés. Pour améliorer le codeur de chaine LTE, un algorithme d’adaptation conjointe a été déployé. Le but de cet algorithme est de minimiser la redondance générée par codage en chaine avec une légère réduction de la perception de l’utilisateur. Ensuite, afin d’améliorer le décodeur par chaine LTE, un algorithme amélioré Log-MAP a été présenté. Cet algorithme vise à obtenir la performance BER (Bit Error Rate) qui est le plus proche du Log-MAP avec une complexité de calcul réduite par rapport à l’état de l’art. Deuxièmement, la chaine et les systèmes QoS de planification améliorés de la perception de l’utilisateur et du mode de priorité VoIP ont été proposés. Ces planificateurs sont déployés à la fois pour les utilisateurs d’audio à larges et à étroites bandes. Les résultats numériques montrent qu’ils surpassent plusieurs planificateurs en vedette tels que FLS, M-LWDF et EXP/PF en termes de retard, de taux de perte de paquets, de débit cellulaire, d’indice et de l’équité et d’efficacité spectrale dans presque tous les cas. Enfin, pour assurer la qualité vocale de transmission sur le réseau LTE, la prédiction de la satisfaction des utilisateurs est essentielle. Pour cette raison, nous présentons deux modèles non intrusifs pour mesurer la qualité de la voix sur les réseaux LTE. Ces modèles sont utilisés pour les utilisateurs d’audio à bandes étroites et larges bandes. Les modèles proposés ne se réfèrent pas au signal original. Par conséquent, ils sont très appropriés pour prédire la qualité de l’appel vocal sur les réseaux LTE / LTE (Long Term Evolution) has been developed and standardized by 3GPP (3rd Generation Partnership Project). It is a packet-switched network. This means voice over LTE (VoLTE) is a VoIP service with the guaranteed QoS requirements instead of transmitted in a circuit-switched network such as the legacy system (2G/3G). Since VoLTE is deployed in an All-IP network combined with IMS (IP Multimedia Subsytem), thus, the VoLTE deployment is quite complex and how to ensure voice transmission quality over LTE networks is a very big challenge. Therefore, there needs to be many different solutions to enhance and improve voice transmission quality over LTE networks. In this dissertation, we present solutions to enhance and improve voice transmission quality over LTE networks for both narrowband and wideband audio services. In order to do that, there needs to be many different factors complemented in solutions. One of them is QoE (Quality of Experience) which is a new trend. And in order to determine user perception for real-time service such as VoLTE, we use extended E-model and WB (Wideband) E-model for narrowband and wideband audio services, respectively. The proposed solutions in this thesis mainly focus on key elements in LTE networks such as channel coding, MAC (Medium Access Control) scheduling schemes and monitor voice quality described as follows. First, enhanced/improved algorithms for enhancing LTE channel codec (coder and decoder) have been proposed. In order to enhance LTE channel coder, a joint source-channel code rate adaption algorithm has been deployed. The goal of this algorithm is to minimize redundancy generated by channel coding with a slight reduction of user perception. Next, in order to enhance LTE channel decoder, an improved Log-MAP algorithm has been presented. This algorithm aims at obtaining BER performance that is closest to the LOP-MAP with the computational complexity reduced in comparison with state-of-the-art. Second, channel- and QoS-Aware scheduling schemes with the enhancement of user perception and VoIP priority mode have been proposed. These schedulers are deployed for both narrowband and wideband audio users. The numerical results show that they outperform several featured schedulers such as FLS, M-LWDF, and EXP/PF in terms of delay, packet loss rate, cell throughput, fairness index, and spectral efficiency in almost cases. Last, in order to ensure voice transmission quality over LTE network, prediction of user satisfaction is essential. For this reason, we present two object non-intrusive models for measuring voice quality in LTE networks. These models are used for narrowband and wideband audio users. The proposed models do not refer to the original signal, thus, they are very suitable for predicting voice call quality in LTE networks

Avec la croissance rapide des appareils et des applications mobiles, les besoins en débit et en connectivité dans les réseaux sans fil augmentent rapidement. Il est prouvé que les communications coopératives peuvent augmenter significativement l’efficacité spectrale et la fiabilité des transmissions entre les nœuds extrémaux. Le concept de coopération dans un réseau sans fil compte parmi les sujets de recherche les plus actifs en télécommunications, le but étant d'identifier les stratégies de coopération qui maximiseraient les gains en efficacité spectrale et en puissance d'émission. Pour coopérer, les nœuds du réseau partagent leurs ressources (énergie, bande de fréquence, etc. ...) pour améliorer mutuellement leurs transmissions et leurs réceptions. Dans les réseaux sans fil avec relais, les relais sont des nœuds dédiés à améliorer la qualité de la communication entre les nœuds sources et destination.Dans la première partie de la thèse, nous nous concentrons sur un réseau sans fil avec relais spécifique où l'ensemble de sources (mobiles) veulent communiquer leurs messages à une destination commune (station de base) avec l'aide d'un ensemble de relais (contexte cellulaire, sens montant). Nous étudions, sur les plans théorique et pratique, un schéma coopératif dans lequel les relais, après une durée d'écoute fixée a priori, essayent de décoder les messages des sources et commencent à transmettre des signaux utiles pour ceux qui sont décodés correctement. Ces signaux utiles sont le résultat d'un codage canal-réseau conjoint.Une des limitations du système coopératif précédent est précisément que le temps d'écoute des relais est figé et ne peut pas être adapté à la qualité fluctuante (aléatoire) des liens instantanés sources-relais. Pour pallier cette difficulté, nous proposons et analysons, dans une seconde partie de la thèse, un schéma de coopération plus avancé où le temps d'écoute de chaque relais peut être dynamique. Dans ces conditions, un relais bénéficiant d'une meilleure qualité de réception des sources peut commencer à coopérer plus tôt que d'autres relais ayant une qualité de réception moindre.Enfin, dans la troisième et dernière partie de la thèse, nous considérons la présence d'une information de retour limitée (limited feedback) entre la destination et les sources et les relais, et tentons de caractériser l'efficacité spectrale d'un tel système. / With the rapid growth of wireless technologies, devices and mobile applications, the quest of high throughput and omnipresent connectivity in wireless networks increases rapidly as well. It is well known that cooperation increases significantly the spectral efficiency (coding gain) and the reliability (diversity gain) of the transmission between the nodes. The concept of cooperation in wireless relays network is still one of the most active research topics in wireless communication, scientists are still searching for the optimal cooperation strategies that make the possible gains at the maximum. Cooperation results when nodes in a network share their power and/or bandwidth resources to mutually enhance their transmissions and receptions. In wireless relay networks, the relays are special nodes that are used to improve the quality of communication between the source nodes and the destination nodes. In particular, the use of relays guarantees more efficient and reliable networks. In this work, we focus on a special wireless relay network where a set of sources (mobiles) want to communicate their messages to a common destination (base station) with the help of a set of relaysAt the beginning of this work, we focused on the cooperative scheme where the relay, after a fixed portion of time, tries to understand (decode) the source’s messages and forwards helpful signals for the correctly decoded ones. One of the limitations of the previous cooperative scheme is the fixe listening time of the relays, which cannot be adapted to the quality of the instantaneous sources-relays links. To solve this problem we propose a more advanced cooperative scheme where the listening time of each relay can be dynamic and not fixed in advanced. So the relay that has strong links with the sources can start cooperating earlier than the other relays with weak links. Currently, we are investigating other directions of possible improvements, for example, how can we use feedback signals to improve the efficiency of the network.

With the ever-growing demand for higher data rate in next generation communication systems, researchers are pushing the limits of the existing architecture. Due to the stochastic nature of communication channels, most systems use some form of adaptive methods to adjust the transmitting parameters and allocation of resources in order to overcome channel variations and achieve optimal throughput. We will study four cases of adaptive transmission and dynamic resource allocation in collaborative systems that are practically significant. Firstly, we study hybrid automatic repeat request (HARQ) techniques that are widely used to handle transmission failures. We propose HARQ policies that improve system throughput and are suitable for point-to-point, two-hop relay, and multi-user broadcast systems. Secondly, we study the effect of having bits of mixed SNR qualities in finite length codewords. We prove that by grouping bits according to their reliability so that each codeword contains homogeneous bit qualities, the finite blocklength capacity of the system is increased. Thirdly, we study the routing and resource allocation problem in multiple collaborative networks. We propose an algorithm that enables collaboration between networks which needs little to no side information shared across networks, but rather infers necessary information from the transmissions. The collaboration between networks provides a significant gain in overall throughput compared to selfish networks. Lastly, we present an algorithm that allocates disjoint transmission channels for our cognitive radio network in the DARPA Spectrum Collaboration Challenge (SC2). This algorithm uses the real-time spectrogram knowledge perceived by the radios and allocates channels adaptively in a crowded spectrum shared with other collaborative networks.

The concepts of a digital twin and extended reality have recently emerged, which require a massive amount of sensor data to be transmitted with low latency and high reliability. For low-latency communications, joint source-channel coding (JSCC) is an attractive method for error correction coding and compared to highly complex digital systems that are currently in use. I propose the use of complex-valued and quaternionic neural networks (QNN) to decode JSCC codes, where the complex-valued neural networks show a significant improvement over real-valued networks and the QNNs have an exceptionally high performance. Furthermore, I propose mapping encoded JSCC code words to the baseband of the frequency domain in order to enable time/frequency synchronization as well as to mitigate fading using robust estimation theory. Additionally, I perform robust statistical signal processing on the high-dimensional JSCC code showing significant noise immunity with drastic performance improvements at low signal-to-noise ratio (SNR) levels. The performance of the proposed JSCC codes is within 5 dB of the optimal performance theoretically achievable and outperforms the maximum likelihood decoder at low SNR while exhibiting the smallest possible latency. I designed a Bayesian minimum mean square error estimator for decoding high-dimensional JSCC codes achieving 99.96% accuracy. With the recent introduction of electromagnetic reconfigurable intelligent surfaces (RIS), a paradigm shift is currently taking place in the world of wireless communications. These new technologies have enabled the inclusion of the wireless channel as part of the optimization process. In order to decode polarization-space modulated RIS reflections, robust polarization state decoders are proposed using the Weiszfeld algorithm and an generalized Huber M-estimator. Additionally, QNNs are trained and evaluated for the recovery of the polarization state. Furthermore, I propose a novel 64-ary signal constellation based on scaled and shifted Eisenstein integers and generated using media-based modulation with a RIS. The waveform is received using an antenna array and decoded with complex-valued convolutional neural networks. I employ the circular cross-correlation function and a-priori knowledge of the phase angle distribution of the constellation to blindly resolve phase offsets between the transmitter and the receiver without the need for pilots or reference signals. Furthermore, the channel attenuation is determined using statistical methods exploiting that the constellation has a particular distribution of magnitudes. After resolving the phase and magnitude ambiguities, the noise power of the channel can also be estimated. Finally, I tune an Sq-estimator to robustly decode the Eisenstein waveform. / Doctor of Philosophy / This dissertation covers three novel wireless communications methods; analog coding, communications using the electromagnetic polarization and communications with a novel signal constellation. The concepts of a digital twin and extended reality have recently emerged, which require a massive amount of sensor data to be transmitted with low latency and high reliability. Contemporary digital communication systems are highly complex with high reliability at the expense of high latency. In order to reduce the complexity and hence latency, I propose to use an analog coding scheme that directly maps the sensor data to the wireless channel. Furthermore, I propose the use of neural networks for decoding at the receiver, hence using the name neural receiver. I employ various data types in the neural receivers hence leveraging the mathematical structure of the data in order to achieve exceptionally high performance. Another key contribution here is the mapping of the analog codes to the frequency domain enabling time and frequency synchronization. I also utilize robust estimation theory to significantly improve the performance and reliability of the coding scheme. With the recent introduction of electromagnetic reconfigurable intelligent surfaces (RIS), a paradigm shift is currently taking place in the world of wireless communications. These new technologies have enabled the inclusion of the wireless channel as part of the optimization process. Therefore, I propose to use the polarization state of the electromagnetic wave to convey information over the channel, where the polarization is determined using a RIS. As with the analog codes, I also extensively employ various methods of robust estimation to improve the performance of the recovery of the polarization at the receiver. Finally, I propose a novel communications signal constellation generated by a RIS that allows for equal probability of error at the receiver. Traditional communication systems utilize reference symbols for synchronization. In this work, I utilize statistical methods and the known distributions of the properties of the transmitted signal to synchronize without reference symbols. This is referred to as blind channel estimation. The reliability of the third communications method is enhanced using a state-of-the-art robust estimation method.