Constant evolution requirements of Wi-Fi and cellular standards to meet the demands of better power efficiency, longer range and higher throughput of wireless networks has drawn attention to multiple antenna transmitters and receivers, i.e., multi-input multi-output(MIMO) systems. This research falls in the larger context of distributed MIMO, or DMIMO systems, wherein groups of cooperating transceivers organize themselves into virtual antenna arrays which can, in principle, emulate any MIMO technique that a centralized array can support. Beamforming and nullforming are techniques that can be employed by centralized or distributed antenna array, and can be building blocks for MIMO communication systems; these impart directionality to the array and can help cater to the demands of today's wireless networks. In beamforming, a set of distributed transmitters in a wireless network cooperatively transmit a common message signal in such a way that their individual transmissions add up to a desired SNR level at the set of designated receivers while in nullforming, cooperative transmission ensures that the individual transmissions cancel each other at the set of designated receivers. The key bottleneck in the practical realization of DMIMO is synchronization. Distributed nullforming specifically poses challenges that call for special attention. Here, we develop a set of scalable algorithms for beamforming and nullforming using distributed transmitters by forming a virtual antenna array and overcome the involved challenges in a purely distributed fashion. Under a per-antenna power constraint and assuming equal-gain channels, an ideal N-antenna beamformer provides an N squared-fold coherent power gain on target. Ideal nullforming on the other hand results in zero power on the target. These properties motivate applications in cooperative jamming or communications, where the goal is to maximize the net transmitted power using multiple transmitters while simultaneously protecting a designated receiver. For example, in a cognitive radio system where the transmit array is a secondary user of licensed spectrum which seeks to communicate with a set of secondary receivers (beam targets) without causing any interference at primary receivers (null targets). Another possible application is a cellular network where adjacent Base Stations form a transmit array and coordinate their transmissions to avoid cochannel interference. Recent algorithms on wireless security critically rely on nodes blanketing a landscape with full power jamming signals while protecting a cooperating receiver through nullforming. So a third application can be electronic warfare where a transmit array broadcasts strong jamming signals that disable an enemy's communication infrastructure while protecting friendly stations (null targets) from interference due to the jamming signal. The joint beam and nullforming specifically can be more generally thought of as a fundamental building block for increased spatial spectrum reuse and toward achieving the full spatial multiplexing gains available from MIMO techniques with distributed antenna arrays.

[ES] La tecnologia en fases de paquets es va convertir en dècades enrere en la indústria del radar. Avui en dia, la matriu de fases o la formació de bigues s'està convertint en una necessitat per a la comunicació digital. L'explotació d'un sistema de transmissió de feixos ajudarà a reduir el consum total d'energia de les estacions base i dels equips d'usuaris. També permetrà al servei oferir dades molt més elevades i millorar la qualitat del servei. La investigació sobre la comunicació digital i la comunicació requereix una antena i un maquinari compatible. El maquinari hauria de ser capaç de gestionar diferents escenaris i enfocaments per a problemes de comunicació mòbil. Hi ha diversos sistemes utilitzats per a la investigació de la formació de bigues, especialment per a la comunicació mòbil. Aquests sistemes pateixen de diverses deficiències. Són cares d'implementar, no adaptatives i fixades a una arquitectura relacionada amb determinat algorisme de generació de feixos o amb un nombre d'elements d'array fix. En aquesta tesi es proposa un nou sistema de matrius per fases. Aquest sistema es podria explotar per a la investigació en problemes de comunicació mòbil o radar. Està compost per una xarxa d'antenes planes de 8x1, canals de conversió de RF a banda base i processador de banda base. Es fa una estimació de la transformació de fases i de la DOA en mostres digitals de banda base. Això proporciona al sistema dinàmica quant als algorismes provats. Amb aquesta finalitat, es fan servir juntes SDR àgils per adquirir senyals de la matriu d'antenes i convertir-les en fluxos de dades digitals. Els fluxos de dades es processen després en un processador de banda base basat en FPGA. A més de ser baixos en costos i assequibles per part de petits instituts d'investigació i investigacions independents, el sistema es pot ajustar per portar més elements de matriu d'antenes. La matriu monopola plana de 8x1 està dissenyada, simulada i mesurada. Es combinen i descriuen les característiques d'impedància i de radiació. Els SDR s'introdueixen i es calibren per al funcionament de diversos elements i s'introdueixen els mètodes de calibratge per incerteses de fase i amplitud. El rendiment global del sistema es prova mitjançant diferents algorismes de formulació de feixos i algorismes de direcció d'estimació d'arribada. Els resultats de la mesura mostren que el sistema és fiable. S'aconsegueix un model de beamformació amb bona resolució i un rebuig elevat de la interferència. La estimació de la direcció d'arribada és precisa. / [CA] La tecnología de matriz en fase hizo una rotación en la industria del radar hace décadas. Hoy en día, la matriz en fase, o formación de haz, se está convirtiendo en una necesidad para la comunicación digital. La explotación de la formación de haz ayudaría a reducir el consumo de energía general de las estaciones base y el equipo del usuario. También permitirá que el servicio brinde datos mucho más altos y mejore la calidad del servicio. La investigación sobre la formación y comunicación de haces digitales requiere un conjunto de antenas y hardware compatible. El hardware debe ser capaz de manejar diferentes escenarios y enfoques para problemas de comunicación móvil. Hay varios sistemas utilizados para la investigación de conformación de haz, especialmente para la comunicación móvil. Estos sistemas sufren de varias deficiencias. Son costosos de implementar, no adaptativos y fijos a una arquitectura relacionada con cierto algoritmo de conformación de haz o con un número de elementos de arreglo fijo. En esta tesis, se propone un nuevo sistema matricial por fases. Este sistema podría ser explotado para la investigación en comunicaciones móviles o problemas de radar. Está compuesto por un conjunto de antenas planas de 8x1, canales de conversión de RF a banda base y procesador de banda base. La formación de haz y la estimación de DOA se realizan en muestras digitales de banda base. Esto proporciona al sistema dinamismo con respecto a los algoritmos probados. Para ese propósito, las tarjetas SDR ágiles se utilizan para adquirir señales de la red de antenas y convertirlas en flujos de datos digitales. Los flujos de datos se procesan en un procesador de banda base basado en FPGA. Además de ser de bajo costo y asequible para los pequeños institutos de investigación e investigaciones independientes, el sistema se puede ajustar para llevar más elementos de la red de antenas. El conjunto monopolo plano 8x1 está diseñado, simulado y medido. La correspondencia de impedancia y las características de radiación se representan y describen. Los SDR se introducen y se calibran para la operación de elementos múltiples y se introducen los métodos de calibración para las incertidumbres de fase y amplitud. El rendimiento general del sistema se prueba mediante diferentes algoritmos de conformación de haz y algoritmos de estimación de la dirección de llegada. Los resultados de las mediciones muestran que el sistema es confiable. Se logra una conformación de haz con buena resolución y alto rechazo de interferencia. Dirección de estimación de la llegada es precisa. / [EN] Phased array technology made a turnover in radar industry decades ago. Nowadays, phased array, or beamforming, is becoming a necessity for digital communication. Exploiting beamforming would help in reducing the overall power consumption of base stations and user equipment. It will also enables the service to provide much higher datarates and enhance the quality of service. Research on digital beamforming and communication requires antenna array and compatible hardware. The hardware should be capable of handling different scenarios and approaches for mobile communication problems. There are several systems used for beamforming research especially for mobile communication. These systems suffer from several deficiencies. They are either expensive to implement, not adaptive and fixed to an architecture related to certain beamforming algorithm or with fixed array elements number. In this thesis, a new phased array system is proposed. This system could be exploited for research in mobile communication or radar problems. It is composed of 8x1 planar antenna array, RF to baseband conversion channels and base band processor. Beamforming and DOA estimation is done on base band digital samples. This provides the system with dynamicity regarding tested algorithms. For that purpose, agile SDR boards are used to acquire signals from antenna array and convert them to digital data streams. Data streams are then processed in an FPGA based base band processor. In addition to being low in cost and affordable by small research institutes and freelancing researches, the system can be adjusted to carry more antenna array elements. The 8x1 planar monopole array is designed, simulated and measured. Impedance matching and radiation characteristics are plotted and described. SDRs are introduced and calibrated for multi-element operation and calibration method for phase and amplitude uncertainties are introduced. Overall system performance is tested by different beamforming algorithms and direction of arrival estimation algorithms. Measurement results show that the system is reliable. Beamforming with good resolution and high interference rejection is achieved. Direction of arrival estimation is accurate. / Haroun, MH. (2019). 8×1 Antenna Array System for Uplink Beamforming in LTE-A and 5G NR [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/129852 / TESIS

Multiple-input-multiple-output (MIMO) systems of current LTE releases are capable of adaptation in the azimuth only. More recently, the trend is to enhance the system performance by exploiting the channel's degrees of freedom in the elevation through the dynamic adaptation of the vertical antenna beam pattern. This necessitates the derivation and characterization of three-dimensional (3D) channels. Over the years, channel models have evolved to address the challenges of wireless communication technologies. In parallel to theoretical studies on channel modeling, many standardized channel models like COST-based models, 3GPP SCM, WINNER, ITU have emerged that act as references for industries and telecommunication companies to assess system-level and link-level performances of advanced signal processing techniques over real-like channels. Given the existing channels are only two dimensional (2D) in nature; a large effort in channel modeling is needed to study the impact of the channel component in the elevation direction. The first part of this work sheds light on the current 3GPP activity around 3D channel modeling and beamforming, an aspect that to our knowledge has not been extensively covered by a research publication. The standardized MIMO channel model is presented, that incorporates both the propagation effects of the environment and the radio effects of the antennas. In order to facilitate future studies on the use of 3D beamforming, the main features of the proposed 3D channel model are discussed. A brief overview of the future 3GPP 3D channel model being outlined for the next generation of wireless networks is also provided. In the subsequent part of this work, we present an information-theoretic channel model for MIMO systems that supports the elevation dimension. The model is based on the principle of maximum entropy, which enables us to determine the distribution of the channel matrix consistent with the prior information on the angles of departure and angles of arrival of the propagation paths. Based on this model, an analytical expression for the cumulative density function (CDF) of the mutual information (MI) for systems with a single receive and finite number of transmit antennas in the general signal-to-interference-plus-noise-ratio (SINR) regime is provided. The result is extended to systems with multiple receive antennas in the low SINR regime. A Gaussian approximation to the asymptotic behavior of the MI distribution is derived for the large number of transmit antennas and paths regime. Simulation results study the performance gains realizable through meticulous selection of the transmit antenna down tilt angles, confirming the potential of elevation beamforming to enhance system performance. The results validate the proposed analytical expressions and elucidate the dependence of system performance on azimuth and elevation angular spreads and antenna patterns. We believe that the derived expressions will help evaluate the performance of 3D 5G massive MIMO systems in the future.

Full dimension (FD) multiple-input multiple-output (MIMO) technology has recently attracted substantial research attention in the 3rd Generation Partnership Project (3GPP) as a promising technique for the next-generation of wireless communication networks. FD-MIMO scenarios utilize a planar two-dimensional (2D) active antenna system (AAS) that not only allows a large number of antenna elements to be placed within feasible base station (BS) form factors, but also provides the ability of elevation beamforming. This dissertation presents the elevation beamforming analysis for cellular networks utilizing FD massive MIMO antenna arrays. In particular, two architectures are proposed for the AAS - the uniform linear array (ULA) and the uniform circular array (UCA) of antenna ports, where each port is mapped to a group of vertically arranged antenna elements with a corresponding downtilt weight vector. To support FD-MIMO techniques, this dissertation presents two different 3D ray-tracing channel modeling approaches, the ITU based ‘antenna port approach’ and the 3GPP technical report (TR) 36.873 based ‘antenna element approach’. The spatial correlation functions (SCF)s for both FD-MIMO arrays are characterized based on the antenna port approach. The resulting expressions depend on the underlying angular distributions and antenna patterns through the Fourier series coefficients of the power spectra and are therefore valid for any 3D propagation environment. Simulation results investigate the performance patterns of the two arrays as a function of several channel and array parameters. The SCF for the ULA of antenna ports is then characterized in terms of the downtilt weight vectors, based on the more recent antenna element approach. The derived SCFs are used to form the Rayleigh correlated 3D channel model. All these aspects are put together to provide a mathematical framework for the design of elevation beamforming schemes in single-cell and multi-cell scenarios. Finally, this dissertation proposes to use the double scattering channel to model limited scattering in realistic propagation environments and derives deterministic equivalents of the signal-to-interference-plus-noise ratio (SINR) and ergodic rate with regularized zeroforcing (RZF) precoding. The performance of a massive MIMO system is shown to be limited by the number of scatterers. To this end, this dissertation points out future research directions

The topic of multiple input multiple output (MIMO) radar recently gained considerable interest because it can transmit partially correlated or fully independent waveforms. The inherited waveform diversity helps MIMO radars identify more targets and adds flexibility to the beampattern design. The realized advantages come at the expense of enhanced processing requirements and increased system complexity. In this regards, a closed-form method is derived to generate practical finite-alphabet waveforms with specific correlation properties to match the desired beampattern. Next, the performance of adaptive estimation techniques is examined. Indeed, target localization or reflection coefficient estimation usually involves optimizing a given cost-function over a grid of points. The estimation performance is directly affected by the grid resolution. In this work, the cost function of Capon and amplitude and phase estimation (APES) adaptive beamformers are reformulated. The new cost functions can be evaluated using the two-dimensional fast-Fourier-transform (2D-FFT) which reduces the estimation runtime. Generalized expressions of the Cram´er-Rao lower bound are computed to assess the performance of our estimators. Afterward, a novel estimation algorithm based on the monopulse technique is proposed. In comparison with adaptive methods, monopulse requires less number of received pulses. Hence, it is widely used for fast target localization and tracking purposes. This work suggests an approach that localizes two point targets present in the hemisphere using one set of four antennas. To separate targets sharing the same elevation or azimuth angles, a second set of antennas is required. Two solutions are suggested to combine the outputs from the antenna sets and improve the overall detection performance. The last part of the dissertation focuses on the application and implementation side of radars rather than the theoretical aspects. It describes the realized hardware and software design of a compact portable 24 GHz frequency-modulated-continuous-wave (FMCW) radar. The prototype can assist the visually impaired during their outdoor journeys and prevents collisions with their surrounding environment. Moreover, the device performs diverse tasks such as range-direction mapping, velocity estimation, presence detection, and vital sign monitoring. The experimental result section demonstrates the device’s capabilities in different use-cases.

Stethoscopes are the most commonly used medical devices for diagnosing heart conditions because they are inexpensive, noninvasive, and light enough to be carried around by a clinician. Auscultation with a stethoscope requires considerable skill and experience, but the introduction of digital stethoscopes allows for the automation of this task. Auscultation waveform segmentation, which is the process of determining the boundaries of heart sound and murmur segments, is the primary challenge in automating the diagnosis of various heart conditions. The purpose of this thesis is to improve the accuracy and efficiency of established techniques for detecting, segmenting, and classifying heart sounds and murmurs in digitized phonocardiogram audio files. Two separate segmentation techniques based on the discrete wavelet transform (DWT) and the simplicity transform are integrated into a MATLAB software system that is capable of automatically detecting and classifying sound segments. The performance of the two segmentation methods for recognizing normal heart sounds and several different heart murmurs is compared by quantifying the results with clinical and technical metrics. The two clinical metrics are the false negative detection rate (FNDR) and the false positive detection rate (FPDR), which count heart cycles rather than sound segments. The wavelet and simplicity methods have a 4% and 9% respective FNDR, so it is unlikely that either method would not detect a heart condition. However, the 22% and 0% respective FPDR signifies that the wavelet method is likely to detect false heart conditions, while the simplicity method is not. The two technical metrics are the true murmur detection rate (TMDR) and the false murmur detection rate (FMDR), which count sound segments rather than heart cycles. Both methods are equally likely to detect true murmurs given their 83% TMDR. However, the 13% and 0% respective FMDR implies that the wavelet method is susceptible to detecting false murmurs, while the simplicity method is not. Simplicity-based segmentation, therefore, demonstrates superior performance to wavelet-based segmentation, as both are equally likely to detect true murmurs, but only the simplicity method has no chance of detecting false murmurs.

The presence of acoustic noise in audio recordings is an ongoing issue that plagues many applications. This ambient background noise is difficult to reduce due to its unpredictable nature. Many single channel noise reduction techniques exist but are limited in that they may distort the desired speech signal due to overlapping spectral content of the speech and noise. It is therefore of interest to investigate the use of multichannel noise reduction algorithms to further attenuate noise while attempting to preserve the speech signal of interest. Specifically, this thesis looks to investigate the use of microphone arrays in conjunction with multichannel noise reduction algorithms to aid aiding in speaker identification. Recording a speaker in the presence of acoustic background noise ultimately limits the performance and confidence of speaker identification algorithms. In situations where it is impossible to control the noise environment where the speech sample is taken, noise reduction algorithms must be developed and applied to clean the speech signal in order to give speaker identification software a chance at a positive identification. Due to the limitations of single channel techniques, it is of interest to see if spatial information provided by microphone arrays can be exploited to aid in speaker identification. This thesis provides an exploration of several time domain multichannel noise reduction techniques including delay sum beamforming, multi-channel Wiener filtering, and Spatial-Temporal Prediction filtering. Each algorithm is prototyped and filter performance is evaluated using various simulations and experiments. A three-dimensional noise model is developed to simulate and compare the performance of the above methods and experimental results of three data collections are presented and analyzed. The algorithms are compared and recommendations are given for the use of each technique. Finally, ideas for future work are discussed to improve performance and implementation of these multichannel algorithms. Possible applications for this technology include audio surveillance, identity verification, video chatting, conference calling and sound source localization.

Systémy rozpoznávání řeči v dnešní době dosahují poměrně vysoké úspěšnosti. V případě řeči, která je snímána vzdáleným mikrofonem a je tak narušena množstvím šumu a dozvukem (reverberací), je ale přesnost rozpoznávání značně zhoršena. Tento problém je možné zmírnit využitím mikrofonních polí. Tato práce se zabývá technikami, které umožňují kombinovat signály z více mikrofonů tak, aby byla zlepšena kvalita výsledného signálu a tedy i přesnost rozpoznávání. Práce nejprve shrnuje teorii rozpoznávání řeči a uvádí nejpoužívanější algoritmy pro zpracování mikrofonních polí. Následně jsou demonstrovány a analyzovány výsledky použití dvou metod pro beamforming a metody dereverberace vícekanálových signálů. Na závěr je vyzkoušen alternativní způsob beamformingu za použití neuronových sítí.

abstract: Ultrasound B-mode imaging is an increasingly significant medical imaging modality for clinical applications. Compared to other imaging modalities like computed tomography (CT) or magnetic resonance imaging (MRI), ultrasound imaging has the advantage of being safe, inexpensive, and portable. While two dimensional (2-D) ultrasound imaging is very popular, three dimensional (3-D) ultrasound imaging provides distinct advantages over its 2-D counterpart by providing volumetric imaging, which leads to more accurate analysis of tumor and cysts. However, the amount of received data at the front-end of 3-D system is extremely large, making it impractical for power-constrained portable systems. In this thesis, algorithm and hardware design techniques to support a hand-held 3-D ultrasound imaging system are proposed. Synthetic aperture sequential beamforming (SASB) is chosen since its computations can be split into two stages, where the output generated of Stage 1 is significantly smaller in size compared to the input. This characteristic enables Stage 1 to be done in the front end while Stage 2 can be sent out to be processed elsewhere. The contributions of this thesis are as follows. First, 2-D SASB is extended to 3-D. Techniques to increase the volume rate of 3-D SASB through a new multi-line firing scheme and use of linear chirp as the excitation waveform, are presented. A new sparse array design that not only reduces the number of active transducers but also avoids the imaging degradation caused by grating lobes, is proposed. A combination of these techniques increases the volume rate of 3-D SASB by 4\texttimes{} without introducing extra computations at the front end. Next, algorithmic techniques to further reduce the Stage 1 computations in the front end are presented. These include reducing the number of distinct apodization coefficients and operating with narrow-bit-width fixed-point data. A 3-D die stacked architecture is designed for the front end. This highly parallel architecture enables the signals received by 961 active transducers to be digitalized, routed by a network-on-chip, and processed in parallel. The processed data are accumulated through a bus-based structure. This architecture is synthesized using TSMC 28 nm technology node and the estimated power consumption of the front end is less than 2 W. Finally, the Stage 2 computations are mapped onto a reconfigurable multi-core architecture, TRANSFORMER, which supports different types of on-chip memory banks and run-time reconfigurable connections between general processing elements and memory banks. The matched filtering step and the beamforming step in Stage 2 are mapped onto TRANSFORMER with different memory configurations. Gem5 simulations show that the private cache mode generates shorter execution time and higher computation efficiency compared to other cache modes. The overall execution time for Stage 2 is 14.73 ms. The average power consumption and the average Giga-operations-per-second/Watt in 14 nm technology node are 0.14 W and 103.84, respectively. / Dissertation/Thesis / Doctoral Dissertation Engineering 2019

Millimeter wave (mmWave) spectrum above 30 GHz offers us an opportunity to pursue high-data-rate transmission using a channel bandwidth up to several gigahertz. To provide reliable link quality in mmWave frequency bands, hybrid analog-digital beamforming plays a crucial role in overcoming severe path loss and, meanwhile, satisfies the demand for low-power-consumption radio frequency (RF) devices. Implementing hybrid beamforming based on available channel state information (CSI) is a common solution in the hybrid beamforming literature. However, many reference methods underestimate the computational complexity of channel estimation for large antenna arrays or subsequent steps, such as the singular value decomposition of a channel matrix. To this end, we present a low-complexity scheme that exploits implicit channel knowledge to facilitate the design of hybrid beamforming for frequency-selective fading channels. We start from the study of channel estimation using the orthogonal matching pursuit (OMP) algorithm and realize that the problems of channel estimation and analog beam selection are equivalent if the candidates for analog beamforming vectors in the codebooks are mutually orthogonal. This implies that when orthogonal codebooks are employed, the observations used in channel estimation for large antenna arrays can be used to implement hybrid beamforming directly. The above-mentioned observations can be regarded as \textbf{implicit CSI}; they are coupling coefficients of all possible pairs of analog beamforming vectors on both sides of the channel. The idea of using implicit CSI to implement hybrid beamforming is further extended to the cases of non-orthogonal codebooks. Instead of calculating the mutual information between the transmitter and receiver, we focus on small-size coupling matrices between beam patterns selected by using appropriate key parameters as performance indicators. Therefore, the proposed hybrid beamforming method becomes much simpler: it amounts to collecting different sets of large-power coupling coefficients to construct multiple alternatives to an effective channel matrix. Then, the set yielding the largest Frobenius norm (or the largest absolute value of the determinant) of the effective channel provides the solution to the hybrid beamforming problem. The proposed hybrid beamforming approach clearly shows that the performance of hybrid beamforming is dominated by the quality of the coupling coefficients. Considering a fixed-length training sequence, we exploit mmWave channels' sparsity shown in the delay and angular domains to refine the quality of the coupling coefficients as well as to improve the hybrid beamforming performance.