<div>In recent years, intelligent devices such as smartphones and self-driving cars are becoming ubiquitous in daily life, and thus, wireless communication is turning out to be increasingly omnipresent. To efficiently utilize the electromagnetic spectrum, automatically reconfigurable software-controlled radio transceivers are drawing an extensive amount of attention. In order to implement a reconfigurable radio transceiver, automatically tunable RF front-end components such as tunable filters are indispensable. Over the last decade, tunable filters have shown promising performance with high-quality factor (Q), a wide tuning range, and high-power handling. However, most of the existing tunable filters are manually adjusted. In this regard, this research work focuses on developing a novel automatic software-driven tuning technique for continuously tunable microwave and millimeter-wave filters.</div><div><br></div><div><br></div><div>First, a K-band continuously tunable bandpass filter has been demonstrated with contactless printed circuit board (PCB) tuners. Then, an automatic tuning technique based on deep-Q learning has been proposed and realized to tune a filter with contactless tuners automatically. Two-pole, three-pole, and four-pole bandpass filters are experimentally tested as examples without any human intervention to prove the feasibility of the tuning technique. For the first time, unlike a look-up table, the filters can be continuously tuned at a practically infinite number of frequencies inside the tuning range. </div><div><br></div><div>Next, a K/Ka-band tunable absorptive bandstop filter (ABSF) has been designed and fabricated in low-cost PCB technology. Contrary to a reflective bandstop filter, an ABSF filter is preferred for interference mitigation due to its deeper notch and lower reflection. However, the absorbed power may limit the filter's power handling. Therefore, lastly, a comparative analysis of power handling capability (PHC) between a reflective bandstop filter and an absorptive bandstop filter has been studied theoretically and experimentally in this dissertation.</div>

The target of the research is to employ the autoresonant control technique in order to maintain the nonlinear oscillation mode at resonance (i.e. ultrasonic vibration at the tip of a drill bit at a constant level) during vibro-impact process. Numerical simulations and experiments have been executed. A simplified Matlab-Simulink model which simulates the ultrasonically assisted machining process consists of two parts. The first part represents an ultrasonic transducer that contains a piezoelectric transducer and a 2-step concentrator (waveguide). The second part reflects the applied load to the ultrasonic transducer due to the vibro-impact process. Parameters of the numerical models have been established based on experimental measurements and the model validity has been confirmed through experiments performed on an electromechanical ultrasonic transducer. The model of the ultrasonic transducer together with the model of the applied load was supplemented with a model of the autoresonant control system. The autoresonant control intends to provide the possibility of self-tuning and self-adaptation mechanism for an ultrasonic transducer to maintain its resonant regime of oscillations automatically by means of positive feedback. This is done through a signal to be controlled (please refer to Figure 7.2 and Figure 7.3) transformation and amplification. In order to examine the effectiveness and the efficiency of the autoresonant control system, three control strategies have been employed depending on the attributes of the signals to be controlled . Mechanical feedback control uses a displacement signal at the end of the 2nd step of the ultrasonic transducer. The other two control strategies are current feedback control and power feedback control. Current feedback control employs the electrical current flowing through the piezoceramic rings (piezoelectric transducer) as the signal to be controlled while power feedback control takes into account both the electrical current and the power of the ultrasonic transducer. Comparison of the results of the ultrasonic vibrating system excitation with different control strategies is presented. It should be noted that during numerical simulation the tool effect is not considered due to the complexity of a drill bit creates during the Ultrasonically Assisted Drilling (UAD) process. An effective autoresonant control system was developed and manufactured for machining experiments. Experiments on Ultrasonically Assisted Drilling (UAD) have been performed to validate and compare with the numerical results. Two sizes of drill bits with diameters 3mm and 6mm were applied in combination with three autoresonant control strategies. These were executed during drilling aluminium alloys with one fixed rotational speed associated with several different feed rates. Vibration levels, control efforts, feed force reduction were monitored during experiments. Holes quality and surface finish examinations supplement analysis of the autoresonant control results. In addition, another interesting research on the investigation of the universal matchbox (transformer) has been carried out. Introducing a varying air gap between two ferrite cores allows the optimization of the ultrasonic vibrating system, in terms of the vibration level, effective matchbox inductance, voltage and current level, phase difference between voltage and current, supplied active power etc (more details please refer to Appendix I).

Since the birth of charge coupled devices (CCD) and the complementary metal-oxide-semiconductor (CMOS) active pixel sensors, pixel pitch of digital image sensors has been continuously shrinking to meet the resolution and size requirements of the cameras. However, shrinking pixels reduces the maximum number of photons a sensor can hold, a phenomenon broadly known as the full-well capacity limit. The drop in full-well capacity causes drop in signal-to-noise ratio and dynamic range.<div><br></div><div>The Quanta Image Sensor (QIS) is a class of solid-state image sensors proposed by Eric Fossum in 2005 as a potential solution for the limited full-well capacity problem. QIS is envisioned to be the next generation image sensor after CCD and CMOS since it enables sub-diffraction-limit pixels without the inherited problems of pixel shrinking. Equipped with a massive number of detectors that have single-photon sensitivity, the sensor counts the incoming photons and triggers a binary response “1” if the photon count exceeds a threshold, or “0” otherwise. To acquire an image, the sensor oversamples the space and time to generate a sequence of binary bit maps. Because of this binary sensing mechanism, the full-well capacity, signal-to-noise ratio and the dynamic range can all be improved using an appropriate image reconstruction algorithm. The contribution of this thesis is to address three image processing problems in QIS: 1) Image reconstruction, 2) Threshold design and 3) Color filter array design.</div><div><br></div><div>Part 1 of the thesis focuses on reconstructing the latent grayscale image from the QIS binary measurements. Image reconstruction is a necessary step for QIS because the raw binary measurements are not images. Previous methods in the literature use iterative algorithms which are computationally expensive. By modeling the QIS binary measurements as quantized Poisson random variables, a new non-iterative image reconstruction method based on the Transform-Denoise framework is proposed. Experimental results show that the new method produces better quality images while requiring less computing time.</div><div><br></div><div>Part 2 of the thesis considers the threshold design problem of a QIS. A spatially-varying threshold can significantly improve the reconstruction quality and the dynamic range. However, no known method of how to achieve this can be found in the literature. The theoretical analysis of this part shows that the optimal threshold should match with the underlying pixel intensity. In addition, the analysis proves the existence of a set of thresholds around the optimal threshold that give asymptotically unbiased reconstructions. The asymptotic unbiasedness has a phase transition behavior. A new threshold update scheme based on this idea is proposed. Experimentally, the new method can provide good estimates of the thresholds with less computing budget compared to existing methods.</div><div><br></div><div>Part 3 of the thesis extends QIS capabilities to color imaging by studying how a color filter array should be designed. Because of the small pixel pitch of QIS, crosstalk between neighboring pixels is inevitable and should be considered when designing the color filter arrays. However, optimizing the light efficiency while suppressing aliasing and crosstalk in a color filter array are conflicting tasks. A new optimization framework is proposed to solve the problem. The new framework unifies several mainstream design criteria while offering generality and flexibility. Extensive experimental comparisons demonstrate the effectiveness of the framework.</div>

<p>Nonuniform sampling occurs in many practical applications either intentionally or unintentionally. This thesis deals with the reconstruction of two-periodic nonuniform signals which is of great importance in two-channel time-interleaved analog-to-digital converters. In a two-channel time-interleaved ADC, aperture delay mismatch between the channels gives rise to a two-periodic nonuniform sampling pattern, resulting in distortion and severely affecting the linearity of the converter. The problem is solved by digitally recovering a uniformly sampled sequence from a two-periodic nonuniformly sampled set. For this purpose, a time-varying FIR filter is employed. If the sampling pattern is known and fixed, this filter can be designed in an optimal way using least-squares or minimax design. When the sampling pattern changes now and then as during the normal operation of time-interleaved ADC, these filters have to be redesigned. This has implications on the implementation cost as general on-line design is cumbersome. To overcome this problem, a novel time-varying FIR filter with polynomial impulse response is developed and characterized in this thesis. The main advantage with these filters is that on-line design is no longer needed. It now suffices to perform only one design before implementation and in the implementation it is enough to adjust only one variable parameter when the sampling pattern changes. Thus the high implementation cost is decreased substantially.</p><p>Filter design and the associated performance metrics have been validated using MATLAB. The design space has been explored to limits imposed by machine precision on matrix inversions. Studies related to finite wordlength effects in practical filter realisations have also been carried out. These formulations can also be extended to the general M - periodic nonuniform sampling case.</p>

Design of filters used in grid-connected inverter applications involves multiple constraints. The filter requirements are driven by tight filtering tolerances of standards such as IEEE 519-1992–IEEE Recommended Practices and Requirements for Harmonic Control in Electrical Power Systems and IEEE 1547.2-2008–IEEE Application Guide for IEEE Std 1547, IEEE Standard for Interconnecting Distributed Resources with Electric Power Systems. Higher order LCL filters are essential to achieve these regulatory standard requirements at compact size and weight. This objective of this thesis report is to evaluate design procedures for such higher order LCL filters. The initial configuration of the third order LCL filter is decided by the frequency response of the filter. The design equations are developed in per-unit basis so results can be generalized for different applications and power levels. The frequency response is decided by IEEE specifications for high frequency current ripple at the point of common coupling. The appropriate values of L and C are then designed and constructed. Power loss in individual filter components is modeled by analytical equations and an iterative process is used to arrive at the most efficient design. Different combinations of magnetic materials (ferrite, amorphous, powder) and winding types (round wire, foil) are designed and tested to determine the most efficient design. The harmonic spectrum, power loss and temperature rise in individual filter components is predicted analytically and verified by actual tests using a 3 phase 10 kW grid connected converter setup. Experimental results of filtering characteristics show a good match with analysis in the frequency range of interconnected inverter applications. The design process is stream-lined for the above specified core and winding types. The output harmonic current spectrum is sampled and it is established that the harmonics are within the IEEE recommended limits. The analytical equations predicting the power loss and temperature rise are verified by experimental results. Based on the findings, new LCL filter combinations are formulated by varying the net Lpu to achieve the highest efficiency while still meeting the recommended IEEE specifications. Thus a design procedure which can enable an engineer to design the most efficient and compact filter that can also meet the recommended guidelines of harmonic filtering for grid-connected converter applications is established.

Nonuniform sampling occurs in many practical applications either intentionally or unintentionally. This thesis deals with the reconstruction of two-periodic nonuniform signals which is of great importance in two-channel time-interleaved analog-to-digital converters. In a two-channel time-interleaved ADC, aperture delay mismatch between the channels gives rise to a two-periodic nonuniform sampling pattern, resulting in distortion and severely affecting the linearity of the converter. The problem is solved by digitally recovering a uniformly sampled sequence from a two-periodic nonuniformly sampled set. For this purpose, a time-varying FIR filter is employed. If the sampling pattern is known and fixed, this filter can be designed in an optimal way using least-squares or minimax design. When the sampling pattern changes now and then as during the normal operation of time-interleaved ADC, these filters have to be redesigned. This has implications on the implementation cost as general on-line design is cumbersome. To overcome this problem, a novel time-varying FIR filter with polynomial impulse response is developed and characterized in this thesis. The main advantage with these filters is that on-line design is no longer needed. It now suffices to perform only one design before implementation and in the implementation it is enough to adjust only one variable parameter when the sampling pattern changes. Thus the high implementation cost is decreased substantially. Filter design and the associated performance metrics have been validated using MATLAB. The design space has been explored to limits imposed by machine precision on matrix inversions. Studies related to finite wordlength effects in practical filter realisations have also been carried out. These formulations can also be extended to the general M - periodic nonuniform sampling case.

French vertical-flow constructed wetlands (VFCW) directly treating raw wastewater are known to perform well on for SS, COD and nitrification. They are also known to robustly cope with hydraulic overloads during rainfall events. Although numerous systems have been installed in areas equipped with a combined sewer, the limits of stormwater acceptance remain ill-defined and need to be improved. Looking at the various VFCW designs and usages reported in the literature, it is difficult to draw any consensus on their hydraulic limits. Consequently, designing VFCW to accept hydraulic overloads is a complex task, as local context strongly impacts inlet flows produced during rainfall events. Dynamic models appear a requisite for filter design in such cases. Numerical CW models have essentially focused on horizontal flow, with few attempting to study VFCW dynamics which are more commonly tackled via mechanistic models. Although mechanistic models are powerful tools for describing processes within the VFCW, they are generally too complicated to be readily used by designers. The choice between detailed description and easy handling will depend on the modelling aims. If the aim is a global design tool, simplified models offer a good alternative. However, the simplified models geared to studying VFCW dynamics are extremely reduced. They are easy-handling for design and well-adapted to specific purposes (combined sewer overflow -CSO- treatment) but not necessarily to VFCW treating combined sewer wastewater, where long-term infiltration rates vary significantly. Consequently, this PhD thesis work focused on developing a simplified hydraulic model of VFCW to guide designers through the process of adapting VFCW systems to treat domestic wastewater in both dry and rain events. The simplified model makes it possible to link (i) hydraulics, by simulation of ponding time variations, (ii) biological performances, by establishing "dysfunction alerts" based on treatment performance assessment and variations in online N forms effluent from the young VFCW. These "dysfunction alerts" plot the maximal hydraulic load that a filter can accept without compromising its biological activity. The simplified model was used to model long-term hydraulics in the VFCW (i) to analyze the impact of local context and filter design on hydraulic overload acceptance (using "dysfunction alerts" and bypass discharges) and (ii) to propose VFCW designs for accepting hydraulic overload in different contexts. The modelling demonstrates that VFCW can limit days with bypass discharges to less than 20 times per year without jeopardizing filter performances. Moreover, the most problematic scenario on stormwater treatment remains a watershed with high imperviousness coefficient and low slope under a Bretagne-type climate, demonstrating that the filter is more sensitive to periodicity and duration than to intensity of rainfall events.

Data transfer rates of communication systems continue to rise fueled by aggressive demand for voice, video and Internet data. Device scaling enabled by modern lithography has paved way for System-on-Chip solutions integrating compute intensive digital signal processing. This trend coupled with demand for low power, battery-operated consumer devices offers extensive research opportunities in analog and mixed-signal designs that enable modern communication systems. The first part of the research deals with broadband wireless receivers. With an objective to gain insight, we quantify the impact of undesired out-band blockers on analog baseband in a broadband radio. We present a systematic evaluation of the dynamic range requirements at the baseband and A/D conversion boundary. A prototype UHF receiver designed using RFCMOS 0.18[mu]m technology to support this research integrates a hybrid continuous- and discrete-time analog baseband along with the RF front-end. The chip consumes 120mW from a 1.8V/2.5V dual supply and achieves a noise figure of 7.9dB, an IIP3 of -8dBm (+2dbm) at maximum gain (at 9dB RF attenuation). High linearity active RC filters are indispensable in wireless radios. A novel feed-forward OTA applicable to active RC filters in analog baseband is presented. Simulation results from the chip prototype designed in RFCMOS 0.18[mu]m technology show an improvement in the out-band linearity performance that translates to increased dynamic range in the presence of strong adjacent blockers. The second part of the research presents an adaptive clock-recovery system suitable for high-speed wireline transceivers. The main objective is to improve the jitter-tracking and jitter-filtering trade-off in serial link clock-recovery applications. A digital state-machine that enables the proposed mixed-signal adaptation solution to achieve this objective is presented. The advantages of the proposed mixed-signal solution operating at 10Gb/s are supported by experimental results from the prototype in RFCMOS 0.18[mu]m technology.

No description available.

<p>In many practical multi-channel active sound control (ASC) applications, such as active noise control (ANC), various constraints need to be satisfied, such as the robust stability constraint, noise amplification constraint, controller output power constraints, etc. One way to enforce these constraints is to add a regularization term to the Wiener filter formulation, which, by tuning only a single parameter, can over-satisfy many constraints and degrade the ANC performance. Another approach for non-adaptive ANC filter design that can produce better ANC performance is to directly solve the constrained optimization problem formulated based on the <em>H</em>₂/<em>H</em>_inf control framework. However, such a formulation does not result in a convex optimization problem and its practicality can be limited by the significant computation time required in the solving process. In this dissertation, the traditional <em>H</em>₂/<em>H</em>_inf formulation is convexified and a global minimum is guaranteed. It is then further reformulated into a cone programming formulation and simplified by exploiting the problem structure in its dual form to obtain a more numerically efficient and stable formulation. A warmstarting strategy is also proposed to further reduce the required iterations. Results show that, compared with the traditional methods, the proposed method is more reliable and the computation time can be reduced from the order of days to seconds. When the acoustic feedback path is not strong enough to cause instability, then only constraints that prevent noise amplification outside the desired noise control band are needed. A singular vector filtering method is proposed to maintain satisfactory noise control performance in the desired noise reduction bands while mitigating noise amplification.</p> <p><br></p> <p>The proposed convex conic formulation can be used for a wide range of ASC applications. For example, the improvement in numerical efficiency and stability makes it possible to apply the proposed method to adaptive ANC filter design. Results also show that compared with the conventional constrained adaptive ANC method (leaky FxLMS), the proposed method can achieve a faster convergence rate and better steady-state noise control performance. The proposed conic method can also be used to design the room equalization filter for sound field reproduction and the hear-through filter design for earphones.</p> <p><br></p> <p>Besides efficient filter design methods, efficient filter implementation methods are also developed to reduce real-time computations in implementing designed control filters. A polyphase-structure-based filter design and implementation method is developed for ANC systems that can reduce the computation load for high sampling rate real-time filter implementation but does not introduce an additional time delay. Results show that, compared with various traditional low sampling rate implementations, the proposed method can significantly improve the noise control performance. Compared with the non-polyphase high-sampling rate method, the real-time computations that increase with the sampling rate are improved from quadratically to linearly. Another efficient filter implementation method is to use the infinite impulse response (IIR) filter structure instead of the finite impulse response (FIR) filter structure. A stable IIR filter design approach that does not need the computation and relocation of poles is improved to be applicable in the ANC applications. The result demonstrated that the proposed method can achieve better fitting accuracy and noise control performance in high-order applications.</p>