this thesis the problem of maximum likelihood (ML) detection for the linear multiple-input multiple-output (MIMO) channel is considered. The thesis investigates two algorithms previously proposed in the literature for implementing the ML detector, namely semide nite relaxation and sphere decoding. The first algorithm, semide nite relaxation, is a suboptimal implementation of the ML detector meaning that it is not guaranteed to solve the maximum likelihood detection problem. Still, numerical evidence suggests that the performance of the semide nite relaxation detector is close to that of the true ML detector. A contribution made in this thesis is to derive conditions under which the semide nite relaxation estimate can be guaranteed to coincide with the ML estimate. The second algorithm, the sphere decoder, can be used to solve the ML detection problem exactly. Numerical evidence has previously shown that the complexity of the sphere decoder is remarkably low for problems of moderate size. This has led to the widespread belief that the sphere decoder is of polynomial expected complexity. This is however unfortunately not true. Instead, in most scenarios encountered in digital communications, the expected complexity of the algorithm is exponential in the number of symbols jointly detected. However, for high signal to noise ratio the rate of exponential increase is small. In this thesis it is proved that for a large class of detection problems the expected complexity is lower bounded by an exponential function. Also, for the special case of an i.i.d. Rayleigh fading channel, an asymptotic analysis is presented which enables the computation of the expected complexity up to the linear term in the exponent.

Two estimation problems are treated in this thesis. Estimators are suggested and the asymptotical properties of the estimates are investigated analytically. Numerical simulations are used to assess small-sample performance. In addition, performance bounds are calculated. The first problem treated is parameter estimation for the reduced rank linear regression. A new method based on instrumental variable principles is proposed and its asymptotical performance analyzed. In addition, the Cram\'r-Rao bound for the problem is derived for a general Gaussian noise model. The new method is asymptotically efficient (it has the smallest possible covariance) if the noise is temporally white, and outperforms previously suggested algorithms when the noise is temporally correlated. The approximation of a matrix with one of lower rank under a weighted norm is needed as part of the estimation algorithm. Two new, computationally efficient, methods are suggested. While the general matrix approximation problem has no known closed form solution, the proposed methods are asymptotically optimal as part of the estimation procedure in question. A new algorithm is also suggested for the related rank detection problem. The second part of this thesis treats direction of arrival estimation for narrowband signals using an array of sensors. Most algorithms require the noise covariance matrix to be known (up to a scaling factor) or to possess a known structure. In many cases the noise covariance is in fact estimated from a separate batch of signal-free samples. This work addresses the combined effects of finite sample sizes both in the estimated noise covariance matrix and in the data with signals present. No assumption is made on the structure of the noise covariance. The asymptotical covariance of weighted subspace fitting (WSF) is derived for the case in which the data are whitened using the noise covariance estimate. The obtained expression suggests an optimal weighting that improves performance compared to the standard choice. In addition, a new method based on covariance matching is proposed. Both methods are asymptotically statistically efficient. The Cramér-Rao bound for the problem is derived, and the expression becomes surprisingly simple. / <p>QC 20110103</p>

Multi-robot systems have important applications, such as space explorations, underwater missions, and surveillance operations. In most of these cases robots need to exchange data through communication. Limitations in the communication system however impose constraints on the design of coordination strategies. In this thesis we present three papers on cooperative control problems in which different communication constraints are considered. The first paper describes a rendezvous problem for a team of robots that exchanges position information through communication. A local control law for each robot should steer the team to a common meeting point when communicated data are quantized. The robots are not equipped with any sensors so the positions of other teammates are not measured. Two different types of quantized communication are considered: uniform and logarithmic. Logarithmic quantization is often preferable since it requires that fewer bits are communicated compared to when uniform quantization is used. For a class of feasible communication topologies, control laws that solve the rendezvous problem are derived. A hierarchical control structure is proposed in the second paper, for modelling autonomous underwater vehicles employed in finding a minimum of a scalar field. The controller is composed of two layers. The upper layer is the team controller, which is modeled as discrete-event system. It generates waypoints based on the simplex search optimization algorithm. The waypoints are used as target points by the lower control layer, which continuously steers each vehicle from the current to the next waypoint. It is shown that the communication of measurements is needed at each step for the team controller to generate unique waypoints. A protocol is proposed to reduce the amount of data to be exchanged, motivated by that underwater communication is costly in terms of energy. In the third paper, a probabilistic pursuit{evasion game is considered as an example to study constrained communication in multi-robot systems. This system can be used to model search-and-rescue operations and multi-robot exploration. Communication protocols based on time-triggered and event-triggered synchronization schemes are considered. It is shown that by limiting the communication to events when the probabilistic map updated by the individual pursuer contains new information, as measured by a map entropy, the utilization of the communication link can be considerably improved compared to conventional time-triggered communication.

This thesis begins with a survey of current state-of-the art in-car navigation systems. The pros and cons of the four commonly used information sources — GNSS/RF-based positioning, vehicle motion sensors, vehicle models and map information — are described. Common filters to combine the information from the various sources are discussed. Next, a GNSS-aided inertial navigation platform is presented, into which further sensors such as a camera and wheel-speed encoder can be incorporated. The construction of the hardware platform, together with an extended Kalman filter for a closed-loop integration between the GNSS receiver and the inertial navigation system (INS), is described. Results from a field test are presented. Thereafter, an approach is studied for calibrating a low-cost inertial measurement unit (IMU), requiring no mechanical platform for the accelerometer calibration and only a simple rotating table for the gyro calibration. The performance of the calibration algorithm is compared with the Cramér-Rao bound for cases where a mechanical platform is used to rotate the IMU into different precisely controlled orientations. Finally, the effects of time synchronization errors in a GNSS-aided INS are studied in terms of the increased error covariance of the state vector. Expressions for evaluating the error covariance of the navigation state vector are derived. Two different cases are studied in some detail. The first considers a navigation system in which the timing error is not taken into account by the integration filter. This leads to a system with an increased error covariance and a bias in the estimated forward acceleration. In the second case, a parameterization of the timing error is included as part of the estimation problem in the data integration. The estimated timing error is fed back to control an adjustable fractional delay filter, synchronizing the IMU and GNSS-receiver data. / QC 20101117

Measuring construction tool activity has a potential to improve tool productivity, reduce down-time and give insights to various construction processes. Today, a lot of data is being recordedfrom a construction site. This research aims to explore the technical feasibility of handheld powertool activity recognition with real-time tri-axial accelerometer data. The present study has threefocus areas: 1) Data collection using real-time accelerometer data from two tools: a combihammerand a screwdriver. 2) Hand-craft time and frequency domain features from the collected data. 3)Develop two classification algorithms, namely decision trees and random forest, with hand craftedfeatures to detect tool usage activities. The hand-crafted features provide an understanding of themechanical properties of the tools. For the combihammer, the activities recognized were hammerdrilling, chiseling and motor stop. The activity recognition accuracy was 79% with a decision treeand 80.8% with a random forest. For the screwdriver, the activities recognized were screwing,unscrewing and motor stop. The activity recognition accuracy was 87.7% for a decision tree and94.5% for a random forest algorithm. Variance from time domain and energy from frequencydomain were detected as the high importance features by both the classification algorithms forboth the tools.

No description available.

Real-time imaging can be used to monitor, analyze and control medical treatments. In this thesis, we want to explain the spatiotemporal motion and thus enable more advanced procedures, especially real-time adaptation in radiation therapy. The motion occurring between image acquisitions can be quantified by image registration, which generates a mapping between the images. The contribution of the thesis consists of three papers, where we have used different approaches to estimate the motion between images. In Paper I, we combine a state-of-the-art method in real-time tracking with a learned sparse-to-dense interpolation scheme. For this, we track an arbitrary number of regions in a sequence of medical images. We estimated a sparse displacement field, based on the tracking positions and used the interpolation network to achieve its dense representation. Paper II was a contribution to a challenge in learnable image registration where we finished at 2nd place. Here we train a deep learning method to estimate the dense displacement field between two images. For this, we used a network architecture inspired by both conventional medical image registration methods and optical flow in computer vision. For Paper III, we estimate the dynamics of spatiotemporal images by training a generative network. We use nonlinear dimensional reduction techniques and assume a linear dynamic in a low-dimensional latent space. In comparison with conventional image registration methods, we provide a method more suitable for real-world scenarios, with the possibility of imputation and extrapolation. Although the problem is challenging and several questions are left unanswered we believe a combination of conventional, learnable, and dynamic modeling of the motion is the way forward.

Listeners with normal-hearing often overlook their ability to comprehend speech in noisy environments effortlessly. Our brain’s adeptness at identifying and amplifying attended voices while suppressing unwanted background noise, known as the cocktail party problem, has been extensively researched for decades. Yet, many aspects of this complex puzzle remain unsolved and listeners with hearing-impairment still struggle to focus on a specific speaker in noisy environments. While recent intelligent hearing aids have improved noise suppression, the problem of deciding which speaker to enhance remains unsolved, leading to discomfort for many hearing aid users in noisy environments. In this thesis, we explore the complexities of the human brain in challenging auditory environments. Two datasets are investigated where participants were tasked to selectively attend to one of two competing voices, replicating a cocktail-party scenario. The auditory stimuli trigger neurons to generate electrical signals that propagate in all directions. When a substantial number of neurons fire simultaneously, their collective electrical signal becomes detectable by small electrodes placed on the head. This method of measuring brain activity, known as electroencephalography (EEG), holds potential to provide feedback to the hearing aids, enabling adjustments to enhance attended voice(s). EEG data is often noisy, incorporating neural responses with artifacts such as muscle movements, eye blinks and heartbeats. In the first contribution of this thesis, we focus on comparing different manual and automatic artifact-rejection techniques and assessing their impact on auditory attention decoding (AAD). While EEG measurements offer high temporal accuracy, spatial resolution is inferior compared to alternative tools like magnetoencephalography (MEG). This difference poses a considerable challenge for source localization with EEG data. In the second contribution of this thesis, we demonstrate anticipated activity in the auditory cortex using EEG data from a single listener, employing Neuro-Current Response Functions (NCRFs). This method, previously evaluated only with MEG data, holds significant promise in hearing aid development. EEG data may involve both linear and nonlinear components due to the propagation of the electrical signals through brain tissue, skull, and scalp with varying conductivities. In the third contribution, we aim to enhance source localization by introducing a binning-based nonlinear detection and compensation method. The results suggest that compensating for some nonlinear components produces more precise and synchronized source localization compared to original EEG data. In the fourth contribution, we present a novel domain adaptation framework that improves AAD performances for listeners with initially low classification accuracy. This framework focuses on classifying the direction (left or right) of attended speech and shows a significant accuracy improvement when transporting poor data from one listener to the domain of good data from different listeners. Taken together, the contributions of this thesis hold promise for improving the lives of hearing-impaired individuals by closing the loop between the brain and hearing aids.

Ever wonder why two people can sing the same note with the same loudness, but sound completely different? Middle C is middle C no matter who sings it, yet for some reason Lucianno Pavarotti1s middle C sounds richer and more beautiful than Bob Dylan1s middle C, for example. But then again, what is beauty in singing? It is a completely biased and abstract concept. To some, Bob Dylan1s voice may epitomize tonal beauty, while to others his voice may be comparable to fingernails on a chalk board. Anyway, differences in tone quality, or timbre, are due to differences in the spectral characteristics in different voices. The Music Muse is a computer program designed to help singers train their voices by showing them the individual components of their voices that combine to produce timbre. In paintings, many colors are combined to produce different hues and shades of color. The individual colors that make up the hue are difficult to distinguish. Similarly in music, harmonics with varying amplitudes combine to create voice colors, or timbres. These individual harmonics are difficult to distinguish by the ear alone. The Music Muse splits the voice up into its harmonic components by means of a Fourier transform. The transformed data is then plotted on a harmonic spectrum, from which singers can observe the number of harmonics in their tone, and their amplitudes relative to one another. It is these spectral characteristics that are important to voice timbre. The amplitudes of the harmonics in a voiced tone are determined by the resonant frequencies of the vocal tract. These resonances are called formants. When a harmonic that is produced by the vocal cords has a frequency that is at or near a formant frequency, it is amplified. Formants are determined by the length, size, and shape of the vocal tract. These parameters differ from person to person, and change during articulation. Optimal tonal quality during singing is obtained by placing formants at a desired frequency. The Music Muse calculates the formants of the voice by means of cepstral analysis. The formants are then plotted. With this tool, singers can learn how to place their formants. One of the difficulties of voice training is that singing is rated on a scale of quality, which is difficult to quantify. Also, feedback tends to be biased, and therefore subjective in nature. The Music Muse provides singers with the technology to quantify quality to a degree that makes it less of an abstract concept, and therefore more attainable. / Master of Science

This thesis addresses the issue of GPS satellite spoofing using affordable hardware. The approach involves capturing I/Q-samples from GPS-signals and employing a phase interferometry algorithm for direction finding. By determining the direction of a satellite at a given time and comparing this with decoded navigation messages that reveal the satellite's actual location. This method is able to verify the authenticity of the satellite signals where a discrepancy between these locations suggests potential spoofing. Although the project's theoretical contributions are significant, the practical outcomes fell short of the initial ambitions due to various constraints encountered during the study. Nonetheless, the findings provide valuable insights into the detection of GPS spoofing, highlighting both the potential and the limitations of the proposed method within the allotted timeframe.