Made available in DSpace on 2014-06-11T19:30:51Z (GMT). No. of bitstreams: 0 Previous issue date: 2005-02-01Bitstream added on 2014-06-13T20:00:56Z : No. of bitstreams: 1 duarte_maq_dr_ilha.pdf: 2208096 bytes, checksum: 7daf91683010b0f39c715c9cc1ded5d8 (MD5) / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES) / Neste trabalho é feito um estudo sobre os métodos de redução de ruído aditivo em sinais de voz baseados em wavelets e, através deste estudo, propõe-se um novo método de redução de ruído em sinais de voz no domínio wavelet. O princípio básico da maioria dos métodos de redução de ruído baseados em wavelets é a determinação e aplicação de um limiar, que permite bons resultados para sinais contaminados por ruído branco, mas não são eficientes no processamento de sinais contaminados por ruído colorido, que é o tipo de ruído mais comum em situações reais. Nesses métodos, o limiar, geralmente, é calculado nos intervalos de silêncio e aplicado em todo o sinal. Os coeficientes no domínio wavelet são comparados com este limiar e aqueles que estão abaixo deste valor são eliminados, fazendo assim uma aplicação linear deste limiar. Esta eliminação acaba causando descontinuidades no tempo e na freqüência no sinal processado. Além disso, a forma com que o limiar é calculado pode degradar os trechos de voz do sinal processado, principalmente nos casos em que o limiar depende fortemente da última janela do último trecho de silêncio. O método proposto neste trabalho também é baseado em corte por limiar, mas em vez de uma aplicação linear do limiar, ele faz uma aplicação não-linear, o que evita as descontinuidades causadas por outros algoritmos. O limiar é calculado nos trechos de silêncio e não depende apenas da última janela do último trecho de silêncio, mas sim de todas as janelas, já que este limiar é uma média de todos os limiares calculados neste trecho. Isto faz com que a redução do ruído seja mais uniforme e introduza menos distorções no sinal processado. Além disso, nos trechos de voz ainda é calculado um novo limiar que também será usado, em conjunto com o limiar calculado no silêncio. Isto faz com que a energia da janela que... . / In this work a study of additive noise reduction in speech based on wavelets is presented and, based on this study a new noise reduction method in speech in the wavelet domain is proposed. The basic idea of most methods of noise reduction based on wavelets is the determination and application of a threshold, that produces good results for signals contaminated by white noise, but they are not very efficient in processing signals contaminated by colored noise, which is more common in real situations. In those methods, the threshold, generally, is calculated in the silence intervals and applied to the whole signal. The coefficients in the wavelet domain are compared with this threshold and those that are below this value are eliminated, making a linear application of this threshold. This elimination causes discontinuities in time and frequency of the processed signal. Besides, the way that the threshold is computed can degrade the voice segments of the processed signal, principally when the threshold depends strongly on the last window of the last silence segment. The proposed method in this work is also based in thresholding, but, instead of a linear application of the threshold, it makes a non-linear application, which avoids the discontinuities caused by other algorithms. The threshold is calculated in the silence segments and is not dependent only on the last window of the last silence segment, but of all the windows, since this threshold is an average of all thresholds calculated in this segment. It makes noise reduction more uniform and introduces less distortion in the processed signal. Besides, in the voice segments a new threshold is calculated that will be also used with the threshold calculated in the silence. It makes that the energy of the window that is being processed is also considered. This way, it is... (Complete abstract, click electronic address below).

abstract: VCO as a ubiquitous circuit in many systems is highly demanding for the phase noises. Lowering the noise migrated from the power supply has been the trending topics for many years. Considering the Ring Oscillator(RO) based VCO is more sensitive to the supply noise, it is more significant to find out a useful technique to reduce the supply noise. Among the conventional supply noise reduction techniques such as filtering, channel length adjusting for the transistors, and the current noise mutual canceling, the new feature of the 28nm UTBB-FD-SOI process launched by the ST semiconductor offered a new method to reduce the noise, which is realized by allowing the circuit designer to dynamically control the threshold voltage. In this thesis, a new structure of the linear coarse-fine VCO with 1V supply voltage is designed for the ring typed VCO. The structure is also designed to be flexible to tune the frequency coverage by the fine and coarse tunable on-board resistors. The thesis has given the model of the phase noise reduction method. The model has also been proved to be meaningful with the newly designed VCO circuit. For instances, given 1μV/√Hz white noise coupled on the supply, the 3GHz VCO can have a more than 7dBc/Hz phase noise lowering at the 10MHz frequency offset. / Dissertation/Thesis / Masters Thesis Electrical Engineering 2018

In binaural hearing aids, several beamforming algorithms can be used. These beamformers aim to enhance the target speech signal and preserve the binaural cues of the target source (e.g. with constraints on the target). However, the binaural cues of the other directional sources as well the background noise are often lost after processing. This affects the global impression of the acoustic scene, and it limits the perceptual separation of the sources by the hearing aids users. To help the hearing aids users to localize all the sound sources, it is important to keep the binaural cues of all directional sources and the background noise. Therefore, this work is devoted to find the best trade-off between the noise/interferers reduction and the cues preservations not only for the directional interferers but also for the background noise based on selection and mixing processes. In this thesis, some classification decision algorithms, which are based on different criteria such as the power, the power difference, and the coherence, are proposed to complete the selection and mixing processes. Simulations are completed using recorded signals provided by a hearing aid manufacturer to validate the performance of the proposed algorithm under different realistic acoustic scenarios. After detailed testing using different complex acoustic scenarios and different beamforming configurations, the results indicate that some of the proposed classification decision algorithms show good promise, in particular the classification decision algorithm based on coherence.

The master's thesis focuses on the study of algorithms dealing with noise separation from musical signal. The first chapter is an introduction into methods which are used for noise removal of the musical signal. Furthermore, this chapter describes theory to the issue, specifically a description of transformations for converting from time to frequency domain, and finally thresholding method of spectral coefficients is explained in detail. The aim of the second chapter is an analysis of the proposed algorithm, which is engaged in testing. From the beginning fast algorithms of gradual transformation are described and then a detailed description of the algorithm as a whole. Later, this chapter deals with the selection of audio recordings and with preparation of these recordings for the actual testing. Finally, testing of audio samples is presented in the third chapter of this thesis. This chapter also concludes comparison of individual transformations, achieved results and review of algorithm.

Les kits pour téléphonie mains-libres en voiture sont un équipement qui devient de plus en plus standard dans les véhicules actuels. Ces accessoires répondent à un besoin de communiquer tout en conduisant, pour des raisons professionnelles ou personnelles. Or, la voiture s’avère un environnement acoustique particulièrement difficile. En effet, un habitacle de voiture présente une forte réverbération, du fait de la présence de nombreuses surfaces vitrées, et il est aussi très bruyant. Ce bruit vient de multiples sources, comme le moteur, le roulement du pneu sur la route, le vent, la circulation environnante... et ces sources varient fortement d’une condition de conduite à l’autre. La réduction de bruit ambiant au niveau de la prise de son dans l’habitacle constitue donc un élément majeur dans le confort des utilisateurs de ce type d’équipement. L’objectif des travaux effectués dans le cadre de cette thèse est de fournir une solution efficace de réduction de bruit pour cette application, en utilisant plusieurs microphones. Nous nous intéressons ici principalement à la situation d’autoroute. Une grande campagne de mesures dans un véritable habitacle automobile en roulant a permis d’observer des caractéristiques spatiales et spectrales du champ de bruit ambiant présent dans cette situation. Ces mesures nous ont permis de mettre en évidence de fortes différences en termes de Rapport Signal à Bruit d’entrée (RSB) et de cohérence spatiale du bruit capté selon la fréquence considérée. Ces observations nous amènent à concevoir des systèmes hybrides : nous cherchons à appliquer des traitements différents en basses et hautes fréquences. Nous avons développé une implémentation adaptative du beamforming Minimum Variance Distortionless Response, qui est efficace dans des conditions de fort RSB d’entrée, et quand la cohérence inter-capteur du bruit est faible. Nous avons également étudié le placement des capteurs pour cette approche de façon à maximiser ses performances, qui seront bonnes en hautes fréquences. Pour les basses fréquences, nous avons étudié deux systèmes :- L’un est basé sur de l’Annulation de Bruit Adaptative, exploitant un bruit fortement cohérent d’un capteur sur l’autre - L’autre est un dérivé du Filtre de Wiener Multicanal . A chaque fois, une étude des performances en fonction de l’antenne de capteurs utilisée a été menée, pour utiliser la stratégie acoustique la plus appropriée. Les systèmes hybrides ainsi conçus ont été évalués de façon subjective, en faisant passer un test d’écoute à un panel d’individus. Ce test montre que le système hybride utilisant le filtrage de Wiener permet de réduire de façon significative la gêne liée au bruit ambiant, sans montrer de contrepartie sur la qualité de la parole transmise. / Hands-free car kits have gained a lot of popularity among drivers over the last years. These equipments fill the need for the users to communicate while driving their car, whether it is for professional or personal use. Having a phone conversation in a car is really challenging, as the inside of an automobile is a strongly adverse acoustic environment. Indeed, this compartment is strongly reverberant, because of the presence of important glass surface (such as the windbreaker and windows), and it is also very noisy. The noise comes from various sources, such as the engine, the contact of the tire on the road, the wind..., and those sources show different characteristics from one situation to another. Hence, the noise reduction for in-car voice pickup is a major element for the user’s comfort. The main objective of the work reported in this thesis is to build an efficient noise reduction solution for in-car telephony, using a plurality of microphones. We are in this work mostly interested in the freeway situation. Hence, a database of measurements has been recorded in a real car interior, to understand what are the spectral and spatial characteristics of the noise field in this situation. These measurements showed that the noise field has different characteristics in high and low frequencies. Indeed, the noise has more energy, and is more spatially coherent in the low frequency range. This leads us to propose hybrid subband systems, in order to use different algorithmic approaches in high and low frequencies. We made an adaptive implementation of the well-known Minimum Variance Distortionless Response beamforming, which is efficient when the input Signal-to-Noise Ratio is high, and the noise field shows a low spatial coherence. We also conducted an analysis on the impact of sensors’ positions, in order to build a microphone array which will allow this method to be efficient in the high frequency range. We also considered two different processings for the low frequency range : — one is based on Adaptive Noise Cancellation, which uses the high spatial coherence of recorded noises, — the other is based on Multichannel Wiener Filter. For both methods, an analysis of the impact of sensors’ positions has been made, in order to build an efficient microphone array. To assess the performance of these hybrid systems, a subjective evaluation has been conducted, through a listening test. This evaluation shows that the hybrid system using a Multichannel Wiener Filter in the low frequency range supresses a significant amount of noise, while keeping the voice distortion to a minimum, perceptually unnoticed.

Electromagnetic interference (EMI) is a major concern in inverter motor drive systems. The sources of EMI have been commonly identified as high switching dv/dt and di/dt rates interacting with inverter parasitic components. The reduction of parasitic components relies on highly integrated circuit layout and packaging. This is the way to deal with noise path. On the other hand, switching dv/dt and di/dt can be potentially reduced by soft-switching techniques; thus the intensity of noise source is reduced. In this paper, the relation between the dv/dt di/dt and the EMI generation are discussed. The EMI sources of a hard-switching single-phase PWM inverter are identified and measured with separation of common-mode and differential-mode noises. The noise reduction in an auxiliary resonant snubber inverter (RSI) is presented. The observation of voltage ringing and current ringing and the methods to suppress these ringing in the implementation of RSI are also discussed. The test condition and circuit layout are described as the basis of the study. And the experimental EMI spectra of both hard- and soft-switching inverter are compared. The effectiveness and limitation of the EMI reduction of the ZVT-RSI are also discussed and concluded. The control interface circuit and gate driver design are described in the appendix. The implementation of variable charging time control of the resonant inductor current is also explained in the appendix. / Master of Science

Switched reluctance motors (SRMs) possess many desirable qualities for the long-term sustainability of electrified transportation such as cheap production costs and simple, robust configurations. However, high acoustic noise and torque ripple are two performance imperfections that have prevented the widespread implementation of SRMs. This thesis investigates design techniques to reduce the acoustic noise produced by an 8/6 SRM while also analyzing the impact each design has on the motor’s performance. The fundamentals of SRMs are discussed including the operating principles, modelling and control strategies. The multiphysics finite element analysis (FEA) toolchain used to accurately model acoustic noise and vibrations of SRMs is described. Using the network of FEA tools, nodal forces and natural frequencies of a four phase 8/6 SRM are analyzed to study the acoustic noise and vibration behaviours. The FEA process is validated experimentally by matching measured vibration modes and acoustic noise sound pressure level (SPL) with FEA numerical results. Through inspiration from an extensive literature review, various design techniques are applied to a baseline four phase 8/6 SRM and compared for both acoustic noise reduction and EM performance criteria. The investigated designs were split into two categories, stator-housing modifications that aim to increase the stiffness of the assembly and rotor modifications that aim to reduce the magnitude of radial forces while preserving performance. The best design strategies as determined by the comparative analysis were then further optimized to combine the best techniques together for the 8/6 SRM. The proposed structural improvements included the modifications of the stator yoke shape along with increasing the number of fastening components involved in the assembly. Additionally, an iterative procedure for the parametric modelling of windows introduced to the rotor poles is outlined. The best design considerations are combined to create the design of a novel 8/6 SRM which significantly reduces the acoustic noise produced by the motor with little impact to performance. / Thesis / Master of Applied Science (MASc)

Commercial concealed weapon detection systems are large and expensive and are not suitable to be used as a portable system. Currently, new methods of concealed weapon detection are being developed to build small and compact systems. One such method is based upon the natural resonances of objects; however, no such system has made it to the market due to the low quality of the signals used in the detection algorithms. In this thesis, a prototype concealed weapon detection system is developed and tested for operation in a cluttered environment. This system utilizes the late-time portion of a radar return to extract the resonance information of an unknown target. After proper signal processing and clutter suppression, the signals are classified to determine if the object is a threat. Multiple measurements with frequency-sweep and time-domain systems are used to verify the algorithm. Microwave tissue imaging techniques aim to reconstruct the internal dielectric distribution of the tissue and rely on the dielectric contrast between healthy and malignant tissues. This contrast has been shown to be weak, and therefore, the signals are easily susceptible to noise. This thesis proposes and validates a method for signal-to-noise ratio analysis of complex S-parameter data sets that are used for microwave imaging. A study of de-noising and artifact reduction techniques for microwave holographic imaging is also presented. / Thesis / Master of Applied Science (MASc)

No description available.

This work introduces new binaural beamforming algorithms for hearing aids, with a robustness to errors in the estimate of the target speaker direction of arrival (DOA) and a good trade-off between noise reduction and preservation of the noise/interferers spatial impression. Three robust designs are proposed, and their robustness is confirmed by simulation results. These robust designs are a combination of binaural and monaural beamformers using two different microphone configurations: one for low frequency components and one for high frequency components. The robust designs are also found to be robust to mismatch between the anechoic propagation models used for the beamformers designs and the reverberant propagation models used to generate the signals at the microphones in the simulations. To preserve the binaural cues of the noise/interferers in the binaural beamformer outputs, a method based on a mixing/selection of different available binaural signals is proposed, using a classification from the phase and magnitude of a complex coherence function. This method is added as a post processor to the beamforming designs robust to target DOA mismatch. Simulation results show that the resulting mixed binaural output signals have a good binaural cues preservation level that outperform the benchmark design, with significant noise reduction and low target distortion. Since knowledge of source DOAs is important for beamforming noise reduction, a beamformer-based broadband multi-source DOA detection system is also developed in the thesis, using information from different frequencies or sub‐bands to obtain global estimates of sources DOAs. Simulation results shows that using one beamformer on each side is capable of detecting the DOAs of active sources under several acoustic scenarios, including scenarios with one, two, or three sources, and with or without the presence of some level of diffuse noise.