In high speed applications of visual servoing, latency from the recognition algorithm can cause significant degradation of in response time. Hardware acceleration allows for recognition algorithms to be applied directly during the raster scan from the image sensor, thereby removing virtually all video processing latency. This paper examines one such method, along with an analysis of design decisions made to optimize for use during high speed airborne object tracking tests for the US military. Designing test equipment for defense use involves working around unique challenges that arise from having many details being deemed classified or highly sensitive information. Designing tracking system without knowing any exact numbers for speeds, mass, distance or nature of the objects being tracked requires a flexible control system that can be easily tuned after installation. To further improve accuracy and allow rapid tuning to a yet undisclosed set of parameters, a machine learning powered auto-tuner is developed and implemented as a control loop optimizer.

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Real-time multimedia streaming is an extensively researched topic. The possibility of streaming video over the internet in real time requires smart solutions on many levels at the player and streamer side, as well as along the intermediate network. There are many different methods used to achieve this, but not all of them are suitable for the low latency real-time streaming needed for remote operations of vehicles. This thesis focuses on the bit-rate control at the streamer side to achieve low latency, meaning how the video quality is changed to adapt to the changes in the network. A literature study was conducted, in order to find what algorithms are currently being used for real-time streaming. It investigated both what control methods are used, as well as what feedback metrics are feed to these controllers. These approaches where then evaluated from a theoretical standpoint for real-time low latency streaming on 4G networks together with the rest of the assumed system. Using these discovered methods, two new algorithms were created. They were tested against an already existing benchmark controller, both in simulation and on a real network. As the benchmark algorithm proved to already be using all suitable feedback metrics, only small control alterations where done to the existing benchmark algorithm. The goal for the new algorithms was to increase the total throughput of the video stream, without decreasing the robustness and causing a higher latency.  Simulation and real network tests proved that the new algorithms are unable to provide a higher throughput without increasing the latency. The conclusion is that the benchmark controller is well designed and explicitly configured to work for the goal of low latency video streaming. This being the case with many controllers in the industry, as they are well designed and extensively trimmed for their specific task.

<p dir="ltr">The advancement of terrestrial networks has improved communication services for users in densely-populated areas, outpacing improvements in rural regions. The projected surge in connected devices in upcoming networks entails that the lack of rural and remote connectivity is limiting emerging applications like digital agriculture and intelligent transportation. Thus, expanding rural and remote wireless connectivity requires addressing the limitations of existing terrestrial infrastructure. In this work, we explore two emerging solutions aimed at enhancing wireless connectivity in rural and remote regions. The first approach considers non-terrestrial networks as an alternative to existing terrestrial technology. Specifically, a vertically-integrated, multi-layered architecture involving unmanned aerial vehicles, high-altitude platforms, and satellites serves as complementary elements, offering diverse pathloss, delay, data rates, and network backbone proximity. We address issues such as multi-hop performance degradation, node mobility, placement, and power distribution to optimize network design. The second approach focuses on wireless-powered communication networks, particularly backscatter communications, to overcome challenges associated with the timely data collection of emerging rural applications such as precision agriculture. We utilize ambient orthogonal frequency division multiplexing (OFDM) signals from cellular base stations to facilitate low-power, low-cost, and real-time data collection while eliminating the need for dedicated radio-frequency emitters. Non-coherent detection and modulation schemes are employed to obviate the necessity for accurate channel state information at the power-limited sensors and reader devices. Moreover, we introduce techniques for simultaneous sensor multiplexing by leveraging OFDM signal structure. Our approaches demonstrate substantial improvements in communication performance, offering versatile, scalable, and cost-effective solutions for rural and remote areas.</p>

Numerous power electronics building blocks (PEBB) based power conversion systems have been developed to explore modular design, scalable voltage and current ratings, low-cost operations, etc. This paper further extends the modular concept from the power stage to the control system. The communication network in SiC-based modular power converters is becoming significant for distributed control architecture, with the requirements of tight synchronization and low latency. The influence of the synchronization accuracy on harmonics under the phase-shifted carrier pulse width modulation (PSC-PWM) is evaluated. When the synchronization is accurate, the influence of an increase in harmonics can be ignored. Thus, a synchronous distributed control and communication protocol with well-performed synchronization of 25 ns accuracy is proposed and verified for a 120 kHz SiC-based impedance measurement unit (IMU) with cascaded H-bridge PEBBs. An improved synchronization method with additional analog circuits is further implemented and verified with sub-ns synchronization accuracy. / The power electronics building block (PEBB) concept is proposed for medium-voltage converter applications in order to realize the modular design of the power stage. Traditionally, the central control architecture is popular in converter systems. The voltage and current are sensed and then processed in one central controller. The control hardware interfaces and software have to be customized for a specified number of power cells, and the scalability of controller is lost. In stead, in the distributed control architecture, a local controller in each PEBB can communicate with the sensors, gate drivers, etc. A high-level controller collects the information from each PEBB and conducts the control algorithm. In this way, the design can be more modular, and the local controller can share the computation burden with the high-level controller, which is good for scalability. In such distributed control architecture, a synchronous communication system is required to transmit data and command between the high-level controller and local controllers. A power converter always requires a highly synchronized operation to turn on or turn off the devices. In this work, a synchronous communication protocol is proposed and experimentally validated on a SiC-based modular power converter.

We present hereafter the development of the Liebherr Intelligent Hydraulic Cylinder, in which the hydraulic component is used as smart sensing element providing useful information for the system in which the cylinder is operated. The piston position and velocity are the most important signals derived from this new measuring approach. The performance under various load and temperature conditions (measured both on dedicated test facilities and in field in a real machine) will be presented. An integrated control electronics, which is performing the cylinder state processing, additionally allows the synchronized acquisition of external sensors. Providing comprehensive state information, such as temperature and system pressure, advanced control techniques or monitoring functions can be realized with a monolithic device. Further developments, trends and benefits for the system architecture will be briefly analyzed and discussed.

Les techniques adaptatives de transmission vidéo s’appuient sur un contenu qui est encodé à différents niveaux de qualité et divisé en segments temporels. Avant de télécharger un segment, le client exécute un algorithme d’adaptation pour décider le meilleur niveau de qualité à considérer. Selon les services, ce niveau de qualité doit correspondre aux ressources réseaux disponibles, mais aussi à d’autres éléments comme le mouvement de tête d’un utilisateur regardant une vidéo immersive (à 360°) afin de maximiser la qualité de la portion de la vidéo qui est regardée. L’efficacité de l’algorithme d’adaptation a un impact direct sur la qualité de l’expérience finale. En cas de mauvaise sélection de segment, un client HTTP/1 doit attendre le téléchargement du prochain segment afin de choisir une qualité appropriée. Dans cette thèse, nous proposons d’utiliser le protocole HTTP/2 pour remédier à ce problème. Tout d’abord, nous nous focalisons sur le service de vidéo en direct. Nous concevons une stratégie de rejet d’images vidéo quand la bande passante est très variable afin d’éviter les arrêts fréquents de la lecture vidéo et l’accumulation des retards. Le client doit demander chaque image vidéo dans un flux HTTP/2 dédié pour contrôler la livraison des images par appel aux fonctionnalités HTTP/2 au niveau des flux concernées. Ensuite, nous optimisons la livraison des vidéos immersives en bénéficiant de l’amélioration de la prédiction des mouvements de têtes de l’utilisateur grâce aux fonctionnalités d’initialisation et de priorité de HTTP/2. Les résultats montrent que HTTP/2 permet d’optimiser l’utilisation des ressources réseaux et de s’adapter aux latences exigées par chaque service. / Adaptive video streaming techniques enable the delivery of content that is encoded at various levels of quality and split into temporal segments. Before downloading a segment, the client runs an adaptation algorithm to determine the level of quality that best matches the network resources. For immersive video streaming this adaptation mechanism should also consider the head movement of a user watching the 360° video to maximize the quality of the viewed portion. However, this adaptation may suffer from errors, which impact the end user’s quality of experience. In this case, an HTTP/1 client must wait for the download of the next segment to choose a suitable quality. In this thesis, we propose to use the HTTP/2 protocol instead to address this problem. First, we focus live streaming video. We design a strategy to discard video frames when the band width is very variable in order so as to avoid the rebuffering events and the accumulation of delays. The customer requests each video frame in an HTTP/2 stream which allows to control the delivery of frames by leveraging the HTTP/2 features at the level of the dedicated stream. Besides, we use the priority and reset stream features of HTTP/2 to optimize the delivery of immersive videos. We propose a strategy to benefit from the improvement of the user’s head movements prediction overtime. The results show that HTTP/2 allows to optimize the use of network resources and to adapt to the latencies required by each service.

<div>Remote Direct Memory Access, or RDMA, is a technology that allows one computer server to direct access the memory of another server without involving its CPU. Compared with traditional network technologies, RDMA offers several benefits including low latency, high throughput, and low CPU utilization. These features are especially attractive to datacenters, and because of this, datacenters have started to adopt RDMA in production scale in recent years.</div><div>However, RDMA was designed for confined, single-tenant, High-Performance-Computing (HPC) environments. Many of its design choices do not fit datacenters well, and it cannot be readily used by datacenter applications. To use RDMA, current datacenter applications have to build customized software stacks and fine-tune their performance. In addition, RDMA offers limited scalability and does not have good support for resource sharing or protection across different applications.</div><div>This dissertation sets out to seek solutions that can solve issues of RDMA in a systematic way and makes it more suitable for a wide range of datacenter applications.</div><div>Our first task is to make RDMA more scalable, easier to use, and have better support for safe resource sharing in datacenters. For this purpose, we propose to add an indirection layer on top of native RDMA to virtualize its low-level abstraction into a high-level one. This indirection layer safely manages RDMA resources for different datacenter applications and also provide a means for better scalability.</div><div>After making RDMA more suitable for datacenter environments, our next task is to build applications that can exploit all the benefits from (our improved) RDMA. We designed a set of systems that store data in remote persistent memory and let client machines access these data through pure one-sided RDMA communication. These systems lower monetary and energy cost compared to traditional datacenter data stores (because no processor is needed at remote persistent memory), while achieving good performance and reliability.</div><div>Our final task focuses on a completely different and so far largely overlooked one — security implications of RDMA. We discovered several key vulnerabilities in the one-sided communication pattern and in RDMA hardware. We exploited one of them to create a novel set of remote side-channel attacks, which we are able to launch on a widely used RDMA system with real RDMA hardware.</div><div>This dissertation is one of the initial efforts in making RDMA more suitable for datacenter environments from scalability, usability, cost, and security aspects. We hope that the systems we built as well as the lessons we learned can be helpful to future networking and systems researchers and practitioners.</div>

Network connectivity is a fundamental property affecting network performance. Given the reliability of each link, network connectivity determines the probability that a message can be delivered from the source to the destination. In this thesis, we study the directed network connectivity where the message will be forwarded toward the destination hop by hop, so long as the neighbor(s) is (are) closer to the destination. Directed connectivity, closely related to directed percolation, is very complicated to calculate. The existing state-of-the-art can only calculate directed connectivity for a lattice network up-to-the size of 10 × 10. In this thesis, we devise a new approach that is simpler and more scalable and can handle general network topology and heterogeneous links. The proposed approach uses an unambiguous hop count to divide the networks into hops and gives two steps of pre-process to transform hop-count ambiguous networks into unambiguous ones, and derive the end-to-end connectivity. Then, using the Markov property to obtain the state transition probability hop by hop. Second, with tens of thousands of Low Earth Orbit (LEO) satellites covering the Earth, LEO satellite networks can provide coverage and services that are otherwise not possible using terrestrial communication systems. The regular and dense LEO satellite constellation also provides new opportunities and challenges for network protocol design. In this thesis, we apply the directed connectivity analytical model on LEO satellite backbone networks to ensure ultra-reliable and low-latency (URLL) services using LEO networks, and propose a directed percolation routing (DPR) algorithm to lower the cost of transmission without sacrificing speed. Using Starlink constellation (with 1,584 satellites) as an example, the proposed DPR can achieve a few to tens of milliseconds latency reduction for inter-continental transmissions compared to the Internet backbone, while maintaining high reliability without link-layer retransmissions. Finally, considering the link redundancy overhead and delay/reliability tradeoff, DPR can control the size of percolation. In other words, we can choose a part of links to be active links considering the reliability and cost tradeoff. / Graduate

Sistemas computadorizados para o processamento de multimídia em tempo real demandam alta capacidade de processamento. Problemas que exigem grandes capacidades de processamento são comumente abordados através do uso de sistemas paralelos ou distribuídos; no entanto, a conjunção das dificuldades inerentes tanto aos sistemas de tempo real quanto aos sistemas paralelos e distribuídos tem levado o desenvolvimento com vistas ao processamento de multimídia em tempo real por sistemas computacionais de uso geral a ser baseado em equipamentos centralizados e monoprocessados. Em diversos sistemas para multimídia há a necessidade de baixa latência durante a interação com o usuário, o que reforça ainda mais essa tendência para o processamento em um único nó. Neste trabalho, implementamos um mecanismo para o processamento síncrono e distribuído de áudio com características de baixa latência em uma rede local, permitindo o uso de um sistema distribuído de baixo custo para esse processamento. O objetivo primário é viabilizar o uso de sistemas computacionais distribuídos para a gravação e edição de material musical em estúdios domésticos ou de pequeno porte, contornando a necessidade de hardware dedicado de alto custo. O sistema implementado consiste em duas partes: uma, genérica, implementada sob a forma de um middleware para o processamento síncrono e distribuído de mídias contínuas com baixa latência; outra, específica, baseada na primeira, voltada para o processamento de áudio e compatível com aplicações legadas através da interface padronizada LADSPA. É de se esperar que pesquisas e aplicações futuras em que necessidades semelhantes se apresentem possam utilizar o middleware aqui descrito para outros tipos de processamento de áudio bem como para o processamento de outras mídias, como vídeo. / Computer systems for real-time multimedia processing require high processing power. Problems that depend on high processing power are usually solved by using parallel or distributed computing techniques; however, the combination of the difficulties of both real-time and parallel programming has led the development of applications for real-time multimedia processing for general purpose computer systems to be based on centralized and single-processor systems. In several systems for multimedia processing, there is a need for low latency during the interaction with the user, which reinforces the tendency towards single-processor development. In this work, we implemented a mechanism for synchronous and distributed audio processing with low latency on a local area network which makes the use of a low cost distributed system for this kind of processing possible. The main goal is to allow the use of distributed systems for recording and editing of musical material in home and small studios, bypassing the need for high-cost equipment. The system we implemented is made of two parts: the first, generic, implemented as a middleware for synchronous and distributed processing of continuous media with low latency; and the second, based on the first, geared towards audio processing and compatible with legacy applications based on the standard LADSPA interface. We expect that future research and applications that share the needs of the system developed here make use of the middleware we developed, both for other kinds of audio processing as well as for the processing of other media forms, such as video.