Thesis (M.Eng.)--Memorial University of Newfoundland, 2003. / Includes bibliographical references.

Mobile terminals are arguably the most popular telecommunications device of the present day. With the expectation of use anytime, anywhere, mobile terminals are increasingly used in adverse scenarios such as in hands-free mode and in noisy environments. Speech quality is commonly degraded in such cases by the presence of acoustic echo and ambient noise. In consequence, mobile terminals are generally equipped with speech signal processing algorithms in order to assure acceptable speech quality. Classical approaches to speech signal processing involve independent acoustic echo cancellation, noise suppression and post-filtering. While performance is generally acceptable, degradations are noticeable at low signal-to-echo ratios (hands-free scenarios) and computational complexity can be high. Furthermore, while mobile terminals are increasingly equipped with multiple microphones, they are generally exploited for noise suppression alone, even if there is natural potential for combined noise suppression and echo control. This thesis presents new combination and synchronization architecture for acoustic echo cancellation for single- and dual-microphone devices. It moves beyond the current state-of-the-art by reducing computational complexity while improving performance in low signal-to-echo conditions. The thesis also presents the first dual-microphone solution to double-talk detection. These contributions pave the way for further applied research in speech processing; the novel architecture is readily extendible to multiple-microphone scenarios while respecting levels of computational efficiency required for integration in current mobile terminals. / Les téléphones mobiles sont sans aucun doute les terminaux de télécommunication le plus populaire de nos jours. Le besoin de mobilité étant toujours croissant, les téléphones mobiles sont parfois utilisés dans des conditions très adverses : mains-libres ou environnements bruités. Dans ces conditions, la qualité de la parole est perturbée par la présence de l'écho acoustique et du bruit ambiant. Les terminaux sont généralement équipés d'algorithmes de traitement de la parole afin de garantir une qualité de la parole acceptable. Composés d’un annuleur d’écho adaptatif, d’une réduction de bruit et d’une suppression d’écho résiduel, les chaines de traitement de parole classiques fournissent en général une qualité de la parole acceptable moyennant une complexité de calcul importante. Néanmoins, lorsque le rapport signal à écho est faible on peut noter des dégradations du signal utile. Les terminaux mobiles récents sont de plus en plus équipés de plusieurs microphones qui ne sont alors utilisés que pour la réduction de bruit bien qu’ils présentent un indéniable intérêt pour les systèmes de réduction conjointe de bruit et d’écho résiduel. Cette thèse présente une nouvelle architecture combinée d’annulation d’écho pour terminaux mobiles à un ou deux microphones. L’architecture proposée réduit efficacement la complexité de calcul tout en améliorant la qualité de la parole dans les scénarios défavorables. Nous présentons également la première solution bi-microphones de détection de double parole. Enfin, nos techniques bi-microphones peuvent facilement être appliquées aux terminaux multi-microphones et tout en ayant une capacité calculatoire acceptable pour les téléphones mobiles.

Heart disease is one of the primary causes of morbidity and mortality for the adult population over the age of 65. Furthermore, ailments such as hypertension can affect as many as 50% of the adult population over the age of 45. If left untreated, these ailments eventually precipitate the onset of diastolic dysfunction and heart failure. Diastolic dysfunction is the alteration or impairment of performance in either the left or right ventricle of the heart. Although there has been a marked increase in study of this disease, there is still an apparent difficulty to diagnose patients. Flow visualization techniques have been commonly employed to study the development of these diseases as they relate to the filling process of the ventricles. One method, Echo Particle Image Velocimetry (Echo-PIV) is a relatively new method for cardiac flow chamber visualization, with the potential to provide physicians with a cost-effective and safe method for obtaining high temporal resolution recordings for extending knowledge on the filling processes in cardiac chamber flow. This work presents a new approach to extending the capabilities of Echo-PIV for more accurate measurement of cardiac flows for patients with poor quality recordings. Currently, much of the literature notes that temporal resolution and poor acoustic windows results in exclusion from study. These recordings are more representative of the contrast-enhancement studies used by physicians to better identify chamber walls. When applying standard PIV cross-correlation techniques, measurements tend to fail due to image noise and artifacts. By implementing a Moving Ensemble (MWE) with Product of Correlation (PoC) processing scheme, measurement accuracy, reliability, and robustness can be obtained for measurement in left ventricular filling assessment. / Master of Science

We live in a dynamic world full of noises and interferences. The conventional acoustic echo cancellation (AEC) framework based on the least mean square (LMS) algorithm by itself lacks the ability to handle many secondary signals that interfere with the adaptive filtering process, e.g., local speech and background noise. In this dissertation, we build a foundation for what we refer to as the system approach to signal enhancement as we focus on the AEC problem. We first propose the residual echo enhancement (REE) technique that utilizes the error recovery nonlinearity (ERN) to "enhances" the filter estimation error prior to the filter adaptation. The single-channel AEC problem can be viewed as a special case of semi-blind source separation (SBSS) where one of the source signals is partially known, i.e., the far-end microphone signal that generates the near-end acoustic echo. SBSS optimized via independent component analysis (ICA) leads to the system combination of the LMS algorithm with the ERN that allows for continuous and stable adaptation even during double talk. Second, we extend the system perspective to the decorrelation problem for AEC, where we show that the REE procedure can be applied effectively in a multi-channel AEC (MCAEC) setting to indirectly assist the recovery of lost AEC performance due to inter-channel correlation, known generally as the "non-uniqueness" problem. We develop a novel, computationally efficient technique of frequency-domain resampling (FDR) that effectively alleviates the non-uniqueness problem directly while introducing minimal distortion to signal quality and statistics. We also apply the system approach to the multi-delay filter (MDF) that suffers from the inter-block correlation problem. Finally, we generalize the MCAEC problem in the SBSS framework and discuss many issues related to the implementation of an SBSS system. We propose a constrained batch-online implementation of SBSS that stabilizes the convergence behavior even in the worst case scenario of a single far-end talker along with the non-uniqueness condition on the far-end mixing system. The proposed techniques are developed from a pragmatic standpoint, motivated by real-world problems in acoustic and audio signal processing. Generalization of the orthogonality principle to the system level of an AEC problem allows us to relate AEC to source separation that seeks to maximize the independence, hence implicitly the orthogonality, not only between the error signal and the far-end signal, but rather, among all signals involved. The system approach, for which the REE paradigm is just one realization, enables the encompassing of many traditional signal enhancement techniques in analytically consistent yet practically effective manner for solving the enhancement problem in a very noisy and disruptive acoustic mixing environment.

The objective of this research is to presents new acoustic echo cancellation design methods that can effectively work in the nonlinear environment. Acoustic echo is an annoying issue for voice communication systems. Because of room acoustics and delay in the transmission path, echoes affect the sound quality and may hamper communications. Acoustic echo cancellers (AECs) are employed to remove the acoustic echo while keeping full-duplex communications. AEC designs face a variety of challenges, including long room impulse response, acoustic path nonlinearity, ambient noise, and double-talk situation. We investigate two parts of echo canceller design: echo cancellation algorithm design and control logic algorithm design. In the first part, our work focuses on the nonlinear adaptive and fast-convergence algorithms. We investigate three different structures: predistortion linearization, cascade structure, and nonlinear residual echo suppressor. Specifically, we are interested in the coherence function, since it provides a means for quantifying linear association between two stationary random processes. By using the coherence as a criterion to design the nonlinear echo canceller in the system, our method guarantees the algorithm stability and leads to a faster convergence rate. In the second part, our work focuses on the robustness of AECs in the presence of interference. With regard to the near-end speech, we investigate the double-talk detector (DTD) design in conjunction with nonlinear AECs. Specifically, we propose to design a DTD based on the mutual information (MI). We show that the advantage of the MI-based method, when compared with the existing methods, is that it is applicable to both the linear and nonlinear scenarios. With respect to the background noise, we propose a variable step-size and variable tap-length least mean square (LMS) algorithm. Based on the fact that the room impulse response usually exhibits an exponential decay power profile in acoustic echo cancellation applications, the proposed method finds optimal step size and tap length at each iteration. Thus, it achieves faster convergence rate and better steady-state performance. We show a number of experimental results to illustrate the performance of the proposed algorithms.

No description available.

Electron spin echo envelope modulation (ESEEM) spectroscopy is widely used to study the radical pairs created during the primary steps of photosynthesis. In this thesis the analysis of ESEEM spectra is improved, and some new applications and variations of this experiment suggested. Experimental spectra from species such as P⁺Q^-_A, the secondary radical pair formed in the reaction centre of the bacterium Rhodobacter sphaeroides, give information about the exchange and dipolar couplings between the radicals. The model used to analyse the data affects the results; this thesis suggests two improvements. First, the effect of anisotropic hyperfine couplings in the radicals is considered by the addition of a single spin-1/2 nucleus to the model. This approach suggests that previous models neglecting the effect of nuclei may have been slightly in error. Secondly, several model fittings are performed in the time domain. This approach avoids the Fourier transformation to the frequency domain so that experimental dead-time does not corrupt the data. An excellent fit to experimental data is found with a model containing one spin-1/2 nucleus on each radical. The hyperfine coupling parameters resulting from the fit are consistent with independent experimental results. Use is made of the method of Cramér-Rao lower bounds to assess the precision to which experimental parameters are determined from a time domain curve fitting. It is shown that the lower bounds may also be used to determine the optimum sampling strategy for the experiment. An example is given of the novel use of ESEEM to determine the distance between the radicals in the strongly coupled, uncorrelated radical pair Q^-_AQ^-_B ESEEM has not yet been used for this purpose, and the simulated spectra produced here indicate that the experiment could be used to evaluate the dipolar coupling and hence the inter-radical distance. This thesis considers the possibility of performing ESEEM at higher frequencies than are usually considered. Calculations show that the increased resolution of the g-tensors allow an experiment performed at the W-band frequency of 95 GHz to make a correlation between the relative orientations of the radicals and the dipolar axis, information which has previously been unavailable from a single experiment.

Ever since the birth of the telephony system, the problem with echoes, arising from impedance mismatch in 2/4-wire hybrids, or acoustic echoes where a loudspeaker signal is picked up by a closely located microphone, has been ever present. The removal of these echoes is crucial in order to achieve an acceptable audio quality for conversation. Today, the perhaps most common way for echo removal is through cancellation, where an adaptive filter is used to produce an estimated replica of the echo which is then subtracted from the echo-infested signal. Echo cancellation in practice requires extensive control of the filter adaptation process in order to obtain as rapid convergence as possible while also achieving robustness towards disturbances. Moreover, despite the rapid advancement in the computational capabilities of modern digital signal processors there is a constant demand for low-complexity solutions that can be implemented using low power and low cost hardware. This thesis presents low-complexity solutions for echo cancellation related to both the actual filter adaptation process itself as well as for controlling the adaptation process in order to obtain a robust system. Extensive simulations and evaluations using real world recorded signals are used to demonstrate the performance of the proposed solutions.

We  are increasingly turn to social media for our news consumption two related media phenomenon that influence media consumption are the “Echo chamber” and “Filter Bubble”. Echo chamber this the phenomenon that we tend to have conversation only with those that has the same likeminded as we do while Filter bubble is created by Social media and information retrieval technology that tends to priorities showing us things it already know we like. The aim of this thesis is to suggest design solution for social media that may counter the effect of “Echo chamber” and “Filter Bubble”. The precise method used on this thesis is play centric design method and both intermediate and final evaluations were done through qualitative evaluation. At the end a design solution of Viewlette game were presented. It can be concluded that the suggested design solution may have the tendency to counter the effect of Filter bubble and Echo chamber on social networking site by enabling people that has conflicting points of view to still listen to each other and understand an argument from different perspective. / Som vi i allt högre grad vänder  oss till sociala medier för vår nyhetskonsumtion, två besläktade medier fenomen som påverkar mediekonsumtion är "Ekokammare" och "Filter Bubble". Ekokammare är fenomen där vi tenderar att ha samtal endast med dem som har samma intresse som vi själva, medan Filter bubbla skapas av sociala medier och informationssöknings teknik som tenderar att prioriteringar och visar oss saker man redan vet att vi gillar. Syftet med denna avhandling är att föreslå konstruktionslösningen för sociala medier som kan motverka effekten av "Echo kammare" och "Filter Bubble". Den exakta metod som används på denna avhandling är att spela centrerade designmetod och både den mellanliggande och slutliga utvärderingen som görs genom kvalitativ bedömning. I slutet presenterades en designlösning av ”Viewlette”- spelet. Man kan dra slutsatsen att den föreslagna designlösningen kan ha en tendens att motverka effekten av “Filter bubble” och “Echo chamber” på nätgemenskap, genom att människor som har motstridiga synpunkter fortfarande lyssna på varandra och förstå ett argument från olika perspektiv.

The accurate prediction of warm-season convective systems, and the heavy rainfall and severe weather associated with them, remains a challenge for numerical weather prediction models. This study looks at one such circumstance in which back-building convection forms perpendicular to, and above the cold pool region behind strong bow echoes. We refer to this phenomenon as a "bow and arrow" because, on radar imagery, the two convective lines resemble an archer's bow and arrow. The "arrow" can extend over hundreds of kilometers and can cause damage from high winds, hail and flooding. Events of this nature pose a particular challenge to forecast because they require an accurate forecast of the earlier convection and the effects of that convection on the environment. In this study, radar and surface observations of four events are presented to identify common environmental conditions prior to the development of the back-building convection. Additionally, simulations of three cases using the Weather Research and Forecasting (WRF) model are analyzed in an attempt to understand the mechanisms responsible for initiating and maintaining the convective line. Due to coarse resolution, observational analyses are only useful for inspection of the synoptic-scale. Model output from numerical simulations is utilized to examine the mesoscale in the vicinity of the convective arrow. Several environmental characteristics are evident in each of the studied cases. Strong southwesterly flow (inducing warm air advection and gradual isentropic lifting), in addition to directional and speed convergence into the convective arrow region possibly contribute to convection initiation. Horizontal wind speed shear and increased wind speed in the area surrounding the arrow may be associated with the linear orientation of the arrow. It seems as though when these ingredients are combined with thermodynamic instability, there is a greater possibility of formation and maintenance of a convective arrow behind a bow echo.