abstract: Magnetic Resonance Imaging (MRI) is limited in speed and resolution by the inherently low Signal to Noise Ratio (SNR) of the underlying signal. Advances in sampling efficiency are required to support future improvements in scan time and resolution. SNR efficiency is improved by sampling data for a larger proportion of total imaging time. This is challenging as these acquisitions are typically subject to artifacts such as blurring and distortions. The current work proposes a set of tools to help with the creation of different types of SNR efficient scans. An SNR efficient pulse sequence providing diffusion imaging data with full brain coverage and minimal distortion is first introduced. The proposed method acquires single-shot, low resolution image slabs which are then combined to reconstruct the full volume. An iterative deblurring algorithm allowing the lengthening of spiral SPoiled GRadient echo (SPGR) acquisition windows in the presence of rapidly varying off-resonance fields is then presented. Finally, an efficient and practical way of collecting 3D reformatted data is proposed. This method constitutes a good tradeoff between 2D and 3D neuroimaging in terms of scan time and data presentation. These schemes increased the SNR efficiency of currently existing methods and constitute key enablers for the development of SNR efficient MRI. / Dissertation/Thesis / Ph.D. Electrical Engineering 2011

The purpose of this paper is to study the possibilities of the OFDM system adaptive modulation. The study presents mechanism for acquiring the necessary data of every frequency subchannel using pilot signals and application of these data for system adjustment. The paper introduces the water-filling principle for subchannel power allocation on the basis of their SNR and adaptive modulation mechanisms for variety of OFDM usage. These mechanisms are implemented into a MATLAB model. The code is then transformed into graphical representation and using USRP2, the signal is transmitted by real world channel.

No / A proposed method for cancelling or reducing the effect of the transmitted power within the transceiver MIMO unit is investigated and discussed using a feedback signal process between the elements. Several parameters including the level of feedback power mismatch, the transmitted power and the received power are considered in this work to test the performance of the system in term of the bit error rate (BER) versus signal to noise ratio (SNR). A new software programme using MATLAB is implemented to evaluate the proposed method. The results showed that the performances of the system are heavily dependent on the amount of the transmitted power, the received power, and the mismatch in the feedback component.

No description available.

We are interested in measuring the in situ antenna performance in analog land mobile radio (LMR) FM systems. The gain (efficiency and directivity) and self-impedance of an antenna sufficiently characterize its performance and a number of traditional methods exist to measure these quantities. However it is hard to do antenna gain measurements using these methods. Furthermore, it turns out that volumetric antenna gain measurements are not quite relevant for understanding in situ performance. In this thesis, we present a novel approach for directly measuring the in situ performance of antennas in analog LMR systems. The procedure involves receiving an FM signal simultaneously using the antenna under test (AUT) and a reference antenna. Both received signals are demodulated to audio using separate but identical receivers. Then a convenient method for characterizing the audio signal quality is to analyze the private line (PL) tone. The PL tone signal-to-noise ratio (SNR) is calculated by measuring the power of the tone relative to the sub-audio noise power. The PL tone SNR for both antenna systems is compared as it provides a ``bottom line'' evaluation of the antenna performance. The audio SNR can also be mapped to RF SNR using a theoretical method. From simulation and experimental studies, we conclude that the RF SNR estimated using this technique is within 0.5 dB of the actual value for RF SNR values between +3 and +36 dB. Finally, we demonstrate this procedure in actual in situ LMR antenna measurements. / Master of Science

One of the fundamental hurdles in realizing new spectrum sharing allocation policies is that of reliable spectrum sensing. In this thesis, three research thrusts are presented in order to further research in this critical area. The first of these research thrusts is the development of a novel asynchronous and noncoherent modulation classifier for PSK/QAM modulated signals in flat-fading channels. In developing this classifier, a novel estimator for the unknown channel gain and fractional time delay is proposed which uses a method-of-moments based estimation approach. For the second research thrust of this thesis, the developed method-of-moments based estimation approach is extended to estimate the signal-to-noise ratio of PSK/QAM modulated signals in flat-fading channels, in which no a priori knowledge of the modulation format and channel parameters is assumed. Finally, in the third research thrust, a distributed spectrum sensing approach is proposed in which a network of radios collaboratively detects the presence, as well as the modulation scheme, of a signal through the use of a combination of cyclic spectrum feature-based signal classification and an iterative algorithm for optimal data fusion. / Master of Science

Porttelefoner används ofta i högljudda miljöer. Ett exempel på en sådan miljö är vindutsatta områden, där operatören i ett rum kan uppleva svårigheter att uppfatta tal från användaren som talar i en porttelefon på grund av den omgivande höga ljudnivån. Många porttelefoner och andra liknande enheter stöter vanligtvis på utmaningar och begränsningar, särskilt när det gäller snabbhet, storlek, resurshantering och hantering av dynamiska signaler.  Detta projekt genomfördes vid ett svenskt företag inom nätverksbaserade lösningar för videoövervakning och fysisk säkerhet. Projektet syftar till att utforska och implementera ett adaptiv filter med en adaptiv algoritm i C-programmering för att komplettera ett digitalt signalbehandlingssystem som en strategi för att förbättra ljudkvaliteten genom att reducera bruset hos porttelefoner i utmanande miljöer. Genom att tillämpa ett lämpligt adaptiv filter i en Raspberry Pi för att simulera en porttelefon, strävar projektet efter att reducera brus och optimera talet. Några av de vanligaste filtreringsalgoritmerna som använts i tidigare forskning för att förbättra ljudkvaliteten är Least mean square, Normalized least mean square och Recursive least square som även utvärderas i denna studie. Efter noggranna studier valdes algoritmen Normalized least mean square för implementering i detta projekt. Algoritmens prestanda utvärderas med hjälp av beräkningstiden, medelkvadratfelet och signal-till-brus-förhållandet i Matlab samt användartester för att säkerställa kvaliteten. Detta projekt uppnådde målen genom att utveckla ett fungerande adaptivt filter. Det rekommenderas att implementera filtret i en porttelefon där mikrofonerna inte är placerade nära varandra för att förhindra upptagning av dubbla liknande signaler. Under projektets gång hanterade systemet kontinuerligt dataströmmar effektivt i praktiska tester, vilket bekräftade att det fungerade utan fördröjningar. Detta bevisade det adaptiva filtrets effektivitet i verkliga applikationer, särskilt i högljudda miljöer. / Intercoms are often used in noisy environments. An example of such an environment is windy areas, where the operator inside a room may find it difficult to perceive speech from a user speaking through an intercom due to the surrounding high noise levels. Many intercoms and other similar devices typically encounter challenges and limitations, especially in terms of speed, size, resource management, and handling of dynamic signals. This project was carried out at a Swedish company specializing in network-based solutions for video surveillance and physical security. The project’s objective was to study and implement an adaptive filter with an adaptive algorithm in C programming to complement a digital signal processing system, as a strategy to enhance sound quality by reducing noise in intercoms in challenging environments. By applying a suitable adaptive filter in a Raspberry Pi to simulate an intercom, the goal of the project is to reduce noise and optimize speech clarity. Some of the most common filtering algorithms used in previous research to improve sound quality include Least mean square, Normalized least mean square och Recursive least square, which are evaluated in this study. After thorough studies, the Normalized least mean square algorithm was selected for implementation in this project. The performance of the algorithm is assessed using computation time, mean squared error, and signal-to-noise ratio in Matlab, along with user testing to ensure quality. This project achieved its goals by developing a functional adaptive filter. It is recommended to implement the filter in an intercom where the microphones are not placed close to each other to prevent the capture of similar duplicate signals. Throughout the project, the system continuously handled data streams effectively in practical tests, confirming that it operated without delays. This demonstrated the adaptive filter's effectiveness in real applications, particularly in noisy environments.

Σκοπός αυτής της εργασίας είναι η μελέτη τεχνικών κατανεμημένου προσανατολισμού σε πραγματικές συνθήκες. Πιο συγκεκριμένα σε αυτά στα συστήματα θεωρείται ότι ο κόμβος-πομπός δεν έχει καλή σύνδεση με το δέκτη και κατά συνέπεια δεν μπορεί να επικοινωνήσει απευθείας με τον κόμβο-δέκτη χωρίς δραματική αύξηση της ενέργειας μετάδοσης. Παρόλα αυτά η χρήση κατανεμημένου προσανατολισμού δίνει τη δυνατότητα να βελτιωθεί σημαντικά η κατανάλωση ενέργειας. Το σχήμα που θα χρησιμοποιηθεί είναι ενίσχυση και προώθηση (AF) 2 βημάτων, με το οποίο οι συνεργατικοί κόμβοι απλώς ενισχύουν και στην συνέχεια επαναμεταδίδουν το μήνυμα. Συνεπώς, ζητούμενο είναι η εύρεση των μιγαδικών βαρών με τα οποία πρέπει ο κάθε συνεργαζόμενος κόμβος χωριστά να ενισχύσει το σήμα. Οι τεχνικές που θα χρησιμοποιηθούν έχουν ως κριτήρια την ελαχιστοποίηση της ενέργειας μετάδοσης με ταυτόχρονη ικανοποίηση του SNR, μεγιστοποίηση του SNR με περιορισμένη ολική ενέργεια μετάδοσης και μεγιστοποίηση του SNR με περιορισμένη ενέργεια μετάδοσης ανά συνεργαζόμενο κόμβο. Το πρώτο κριτήριο θα εξεταστεί, επίσης, και σε συστήματα με πολλαπλούς πομπούς και δέκτες. Λόγω της φύσης του προβλήματος, ο κατανεμημένος προσανατολισμός αναμένεται να έχει μεγάλη απήχηση σε συστήματα με πολλούς διασκορπιστές και εμπόδια, όπως σε ένα αστικό περιβάλλον, και, επομένως, είναι λογικό να θεωρηθεί ότι τα κανάλια του συστήματος είναι Rayleigh, δηλαδή ασυσχέτιστα χωρίς οπτική επαφή (LOS). Για να προσομοιωθεί το σύστημα σε πραγματικές συνθήκες οι μέθοδοι που θα υλοποιήσουμε στην εργασία χρησιμοποιούν τα στατιστικά του καναλιού. Επιπλέον, η εκτίμηση καναλιού εφόσον θεωρούμε ότι έχουμε Gaussian λευκό θόρυβο θα γίνει με την χρήση του βέλτιστου γραμμικού εκτιμητή (BLUE). Η επίδραση της εκτίμησης του καναλιού θα μελετηθεί για δύο περιπτώσεις: με αμοιβαία και χωρίς αμοιβαία κανάλια. / The purpose of this thesis is the study of methods of distributed beamforming under real circumstances. More specifically, these systems are considered that the transmitter must increase tremendously the required transmit energy to communicate with the receiver. However the use of the distributed beamforming allows the system to improve the energy consumption. The scheme that is used from relays is amplify and forward of two steps, where the relays only amplify and then forward the message to the destination. That is, the purpose is to find the complex weights to be used by the corresponding relay so as to amplify the message of the transmitter. The methods that are implemented have as criterions the minimization of transmit energy while satisfying the SNR, maximization of SNR while limiting the system's transmit energy and maximization of SNR while limiting transmit energy of each relay individually. The first criterion is also studied at systems with more than one pair transmitter-receiver. Due to the nature of the problem, distributed beamforming is expected to be used at environments with many obstacles and scatterers, like urban environment, and so it is rationale to suppose that the channels should be Rayleigh, meaning uncorrelated without line of sight. To simulate the system under real circumstances the methods that we will implement shall use the second order statistics of the channels. Moreover, due to Gaussian white noise, channels are estimated using the Best Linear Unbiased Estimator. The impact of channel estimation is studied in two cases: "reciprocal" and "not reciprocal".

Cette thèse apporte des solutions théoriques et pratiques à deux problèmes soulevés par la photographie numérique en présence de mouvement, et par la photographie infrarouge. La photographie d'objets en mouvement semblait ne pouvoir se faire qu'avec des temps d'exposition très courts, jusqu'à ce que deux travaux révolutionnaires proposent deux nouveaux types de caméra permettant un temps d'exposition arbitraire. Le flutter shutter de Agrawal et al. crée en effet un flou inversible, grâce à un obturateur aux séquences d'ouverture-fermeture bie{\it n choisies. Le motion invariant photography de Levin et al. obtient ce même effet avec une accélération constante de la caméra. Les deux méthodes suivent ainsi un nouveau paradigme, la computational photography, selon lequel les caméras sont repensées, car elles incluent un traitement numérique sophistiqué. Cette thèse propose une méthode pour évaluer la qualité image des nouvelles caméras. Le fil conducteur de l'analyse est donc l'évaluation du SNR (signal to noise ratio) de l'image obtenue après déconvolution. La théorie fournit des formules explicites pour le SNR, soulève deux paradoxes de ces caméras, et les résout. Elle permet d'obtenir le modèle de mouvement sous-jacent à chaque flutter shutter, notamment tous ceux qui sont brevetés. Une seconde partie plus brève aborde le problème de qualité principal en imagerie vidéo infrarouge, la non-uniformité. Il s'agit d'un bruit évolutif et structuré en colonnes causé par le capteur. La conclusion des travaux est qu'il est non seulement possible mais également efficace et robuste d'effectuer la correction sur une seule image. Cela permet de contourner le problème récurrent des "ghost artifacts"résultant d'une incohérence du traitement par rapport au modèle d'acquisition. / This thesis provides theoretical and practical solutions to two problems raised by digital photography of moving scenes, and infrared photography. Until recently photographing moving objects could only be done using short exposure times. Yet, two recent groundbreaking works have proposed two new designs of camera allowing arbitrary exposure times. The flutter shutter of Agrawal et al. creates an invertible motion blur by using a clever shutter technique to interrupt the photon flux during the exposure time according to a well chosen binary sequence. The motion-invariant photography of Levin et al. gets the same result by accelerating the camera at a constant rate. Both methods follow computational photography as a new paradigm. The conception of cameras is rethought to include sophisticated digital processing. This thesis proposes a method for evaluating the image quality of these new cameras. The leitmotiv of the analysis is the SNR (signal to noise ratio) of the image after deconvolution. It gives the efficiency of these new camera design in terms of image quality. The theory provides explicit formulas for the SNR. It raises two paradoxes of these cameras, and resolves them. It provides the underlying motion model of each flutter shutter, including patented ones. A shorter second part addresses the the main quality problem in infrared video imaging, the non-uniformity. This perturbation is a time-dependent noise caused by the infrared sensor, structured in columns. The conclusion of this work is that it is not only possible but also efficient and robust to perform the correction on a single image. This permits to ensure the absence of ``ghost artifacts'', a classic of the literature on the subject, coming from inadequate processing relative to the acquisition model.

Mes travaux de thèse portent sur l’application de l’imagerie fUS (functional ultrasound imaging) à l’imagerie cérébrale préclinique chez le petit animal. Le but était de transformer cette technique d’imagerie cérébrale récente en un véritable outil de quantification de l’état cérébral. Les objectifs principaux ont été de démontrer la faisabilité de l’imagerie fUS chez le petit animal non anesthésié ainsi que de passer du modèle rat au modèle souris - modèle de choix en imagerie préclinique en neurosciences - de surcroît de façon non invasive. J’ai tout d’abord mis au point une nouvelle séquence d’imagerie ultrasonore ultrarapide (Multiplane Wave imaging), permettant d’améliorer le rapport signal-à-bruit des images grâce à l’augmentation virtuelle de l’amplitude du signal émis, sans diminuer la cadence ultrarapide d’acquisition. Dans un deuxième temps j’ai démontré la possibilité d’imager le cerveau de la souris et du jeune rat anesthésiés par échographie Doppler ultrarapide, de manière transcrânienne et complètement non invasive, sans chirurgie ni injection d’agents de contraste. J’ai ensuite mis au point un montage expérimental, une séquence ultrasonore et un protocole expérimental permettant de réaliser de l’imagerie fUS de manière minimalement invasive chez des souris éveillées et libres de leurs mouvements. Enfin, j’ai démontré la possibilité d’utiliser le fUS pour étudier la connectivité fonctionnelle du cerveau au repos (sans stimulus) chez des souris éveillées ou sédatées. L’imagerie fUS et la combinaison « modèle souris » + « minimalement invasif » + « animal éveillé » + « connectivité fonctionnelle » constituent un outil précieux pour la communauté des neuroscientifiques travaillant sur des modèles animaux pathologiques ou de nouvelles molécules pharmacologiques / My work focuses on the application of fUS (functional ultrasound) imaging to preclinical brain imaging in small animals. The goal of my thesis was to turn this recent vascular brain imaging technique into a quantifying tool for cerebral state. The main objectives were to demonstrate the feasibility of fUS imaging in the non-anaesthetized small rodents and to move from rat model imaging to mouse model imaging –most used model for preclinical studies in neuroscience-, while developing the least invasive imaging protocols. First, I have developed a new ultrafast ultrasonic imaging sequence (Multiplane Wave imaging), improving the image signal-to-noise ratio by virtually increasing emitted signal amplitude, without reducing the ultrafast framerate. Then, I have demonstrated the possibility to use ultrafast Doppler ultrasound imaging to image both the mouse brain and the young rat brain, non-invasively and through the intact skull, without surgery or contrast agents injection. Next, I have developed an experimental setup, an ultrasound sequence and an experimental protocol to perform minimally invasive fUS imaging in awake and freely-moving mice. Finally, I have demonstrated the possibility to use fUS imaging to study the functional connectivity of the brain in a resting state in awake or sedated mice, still in a transcranial and minimally invasive way. fUS imaging and the combination of "mouse model" + "minimally invasive" + "awake animal" + "functional connectivity" represent a very promising tool for the neuroscientist community working on pathological animal models or new pharmacological molecules