Coherent detection together with multilevel modulation formats has the potential to significantly increase the capacity of existing optical communication systems at no extra cost in signal bandwidth. However, these modulation formats are more susceptible to the impact of different noise sources and distortions as the distance between its constellation points in the complex plane reduces with the modulation index. In this context, digital signal processing (DSP) plays a key role as it allows compensating for the impairments occurring during signal generation, transmission and/or detection relaxing the complexity of the overall system. The transition towards pluggable optical transceivers, offers flexibility for network design/upgrade but sets strict requirements on the power consumption of the DSP thus limiting its complexity. The DSP module complexity however, scales with the modulation order and, in this scenario, low complex yet high performance DSP algorithms are highly desired. In this thesis, we mainly focus on the impact of laser phase noise arising from the transmitter and local oscillator (LO) lasers in coherent optical communication systems employing high order modulation formats. In these systems, the phase noise of the transmitting and LO lasers translate into phase noise in the received constellation impeding the proper recovery of the transmitted data. In order to increase the system phase noise tolerance, we firstly explore the possibility of re-arranging the constellation points in a circularly shaped mQAM (C-mQAM) constellation shape to exploit its inherent phase noise tolerance. Different low-complex carrier phase recovery (CPR) schemes applicable to these constellations are proposed along with a discussion on its performance and implementation complexity. Secondly, the design guidelines of high performance and low complex CPR schemes for conventional square mQAM constellations are presented. We identify the inherent limitation of the state-of-the-art blind phase search (BPS) carrier phase recovery algorithm which hinders its achievable performance and implementation complexity and present a low complex solution to overcome it. The design guidelines of multi-stage CPR schemes for high order modulation formats, where the BPS algorithm is employed at any of the stages, are also provided and discussed. Finally, the interplay between the received dispersed signal and the LO phase noise is analytically investigated to characterize the origin of the equalization enhanced phase noise phenomena. / <p>QC 20170516</p> / EU project ICONE, gr. #608099

Die Grundlage vieler zukünftiger Flüssigszintillator-Neutrinoexperimente (SNO+, Daya Bay, LENA) ist das Lösungsmittel Lineare-Alkyl-Benzene (LAB, C6H5CnH2n+1, n = 10 - 13). Zusammen mit dem weit verbreiteten Szintillator 2,5-Diphenyloxazole (PPO) ist es ein farb- und geruchsloses Detektormaterial mit hohem Flammpunkt. Im Vergleich zu toluol- oder xylolbasierten Szintillatoren ist LAB+PPO preiswert und nicht gesundheitsschädlich. Die Eigenschaften von LAB machen es ebenfalls interessant für die Anwendung an nELBE, die Neutronenfugzeitanlage im Helmholtz-Zentrum Dresden - Rossendorf. Ein neuer Ansatz zur Bestimmung der Lichtausbeute im niederenergetischen Bereich (bis 2 MeV) wird vorgestellt. Kombiniert wurden Messungen mit (quasi) monoenergetischen Gammastrahlungs-Prüfstrahlern und einem in dieser Arbeit aufgebauten Compton-Spektrometer. Letzteres ermöglicht die Bestimmung der Lichtausbeute bis zu 5 keVee. Der Birks-Parameter wurde für eine Lösung von LAB + 3 g/l PPO sowie für den Flüssigszintillator NE-213 bestimmt. Die relative Lichtausbeute in Bezug auf letzteren konnte mit diesen Messmethoden ebenfalls ermittelt werden. Zur spektralen Analyse des Lumineszenzlichtes wurden Messungen an Fluoreszenz- und UV/VIS- Spektrometern durchgeführt. Die Pulsformdiskriminationsfähigkeit auf LAB basierenden Szintillatoren wurde während eines Flugzeitexperiments in einem gemischten n-gamma-Feld eines Cf(252)-Prüfstrahlers ermittelt. Dabei kamen unterschiedliche Algorithmen der semi-analogen und digitalen Pulsformdiskrimination zum Einsatz. / Linear alkyl benzene (LAB, C6H5CnH2n+1, n = 10 - 13) is the proposed solvent for the SNO+, the Daya Bay Neutrino and LENA experiment. In solution with the commonly used scintillator PPO it is a colourless, odourless and cheap liquid scintillator with a high fash point and low health hazard compared to toluene based ones. The properties of LAB make this scintillator interesting also for nELBE, the neutron time-of-fight facility at Helmholtz-Zentrum Dresden - Rossendorf. A new approach to measure the light yield in the low-energy range using a combination of quasi-monoenergetic photon sources and a Compton-spectrometer is described. The latter allows the measurement of the light yield down to 5 keVee (electron equivalent). The Birks- Parameter was determined for a homemade solution (LAB + 3 g/l PPO) and for NE-213. The light yield (relative to this standard scintillator) was confrmed by measurements using a fuorescence spectrometer. The ability of pulse-shape-discrimination in a mixed n-gamma- field of a Cf(252) source was tested using different digital and semi-analogue techniques.

Muitas pesquisas tem sido feitas em processamento digital de sinais (PDS) na tentativa de se avaliar o sinal de fala para diagnosticar doenças da laringe. Medidas acústicas têm sido propostas de forma a avaliar indiretamente o trato glotal por meio do sinal de voz coletado através de microfone convencional. Para isso, o modelo paramétrico Liljencrants-Fant (LF) foi desenvolvido para representar o sinal glotal em condições normais e patológicas. Tais parâmetros apresentam vantagens sobre medidas acústicas por possuírem características fisiológicas reais das pregas vocais. Assim, podendo ser empregados para identificação de doenças da laringe. Além da estimação dos parâmetros LF, no domínio do tempo (parâmetros T), a forma de onda da derivativa glotal também pôde ser quantificada através dos parâmetros identificados na literatura por parâmetros R (Rd, Ra, Rk e Rg), parâmetros quocientes Q (SQ, OQ, CQ, AQ e NAQ), parâmetros B1 e B2 que são as extensões de bandas do pulso derivativo LF, e o parâmetro ece, que relaciona os parâmetros &#946 e Ta. Os parâmetros B1 e B2 e ece apesar de serem propostos na literatura, não são encontrados resultados diferentes a essas duas medidas. Os resultados mostraram que os parâmetros B não foram confiáveis na discriminação entre as vozes, por outro lado, o parâmetro ece mostrou-se ser opção na discriminação entre as vozes normais, nódulo e Reinke. O objetivo deste trabalho é direcionar a atenção sobre o sinal glotal, estimando-o automaticamente mediante técnicas de PDS aplicadas ao sinal de fala, visando extrair parâmetros que identifiquem as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições fisiológicas do movimento das pregas vocais. Com um nível de confiança de 95%, o parâmetro de primeira ordem (TRp) é efetivo na discriminação do Edema de Reinke, porém mostrou-se ineficaz na detecção do nódulo. Em relação ao parâmetro de ordem superior, conclui-se que o TRs é um excelente detetor de vozes patológicas (nódulo e Edema de Reinke), porém não é capaz de discriminar as patologias. / Many researches has been conducted in digital signal processing (DSP) atempting to evaluate the physiological conditions of larynx. Acoustical parameters have been proposed to evaluate the glotal tract from voice signal. One technique proposed is the Liljencrants-Fant model (LF) developed to represent normal and pathologic conditions of the larynx. Those parameters compare favourably as far as real physiologic characteristic of vocal folds is concerned. So, a primary use of the model is the larynx pathologic identification. Beyond LF parameters estimation, (T parameters in the time domain), the waveform of glotal pulse derivative also can be quantified through, R parameters (Rd, Ra, Rk and Rg), quocient parameters (SQ, OQ, CQ, AQ and NAQ), B parameters (B1 and B2) that are band extension of the LF glotal pulse derivative and the ece parameter that in fact, is a relationship between &#946 and Ta. Although proposed in the literature, no results are found, related to B and ece parameters. Our founds show that B parameters do not present good results in voice discrimination, however, ece parameter seems to be good option to discriminate normal voice, nodulo and Reinke edema. The main purpose of this work is to estimate the glotal signal from the voice signal using DSP techniques in order to obtain parameters that identifies the physiological larynx condition. In order to estimate the shape of return phase of glotal pulse, twoparameters have been proposed in this work. The first one evaluates the pulse (TRp, in other words, the first order component of the return phase. The second is responsible to evaluate superior orders components of the return phase (TRs), i.e, the non-linear component of the glotal pulse. With 95% of confidence level, TRp is effective in Reinke edema discrimination however it is inefficient for nodule e dection. By the other hand, the TRs parameter works well to detect pathologic voice however is unable to discriminated them.

"Non-invasive remote physiological monitoring of soldiers on the battlefield has the potential to provide fast, accurate status assessments that are key to improving the survivability of critical injuries. The development of WPI’s wearable wireless pulse oximeter, designed for field-based applications, has allowed for the optimization of important hardware features such as physical size and power management. However, software-based digital signal processing (DSP) methods are still required to perform physiological assessments. This research evaluated DSP methods that were capable of providing arterial oxygen saturation (SpO2), heart rate (HR), heart rate variability (HRV), and respiration rate (RR) measurements derived from data acquired using a single optical sensor. In vivo experiments were conducted to evaluate the accuracies of the processing methods across ranges of physiological conditions. Of the algorithms assessed, 13 SpO2 methods, 1 HR method, 6 HRV indices, and 4 RR methods were identified that provided clinically acceptable measurement accuracies and could potentially be employed in a wearable pulse oximeter."

<p>A channel change in digital television is usually conducted at a reference frame, which are sent at certain intervals. A higher compression ratio could however be obtained by sending reference frames at arbitrary long intervals. This would on the other hand increase the average channel change time for the end user. This thesis investigates various approaches for reducing the average channel change time while using arbitrary long intervals between reference frames, and presents an implementation and evaluation of one of these methods, called Baydec.</p><p>The approach of Baydec for solving the channel switch problem is to statistically estimate what the original image looked like, starting with an incoming P-frame and estimate an image between the original and current image. Baydec gathers statistical data from typical video sequences and calculates expected likelihood for estimation. Further on it uses the Simulated Annealing search method to maximise the likelihood function.</p><p>This method is more general than the requirements of this thesis. It is not only applicable to channel switches between video streams, but can also be used for random access in general. Baydec could also be used if an I-frame is dropped in a video stream.</p><p>However, Baydec has so far shown only theoretical result, but very small visual improvements. Baydec produces images with better PSNR than without the method in some cases, but the visual impression is not better than for the motion compensated residual images. Some examples of future work to improve Baydec is also presented.</p>

No description available.

Las técnicas de estimación de canal y de adaptación de la transmisión a las condiciones del entorno son temas de interés actual al estudiar la aplicación de técnicas de diversidad en transmisión en la tercera y cuarta generación de sistemas inalámbricos. En esta tesis se realiza un análisis del impacto del error de estimación de canal y la correlación en sistemas OFDM con diversidad en transmisión basados en codificación espacio-tiempo por bloques (STBC), se proponen técnicas de estimación de canal para estos sistemas y se propone una técnica de adaptación de la transmisión mediante la selección de código espacio-tiempo. En primer lugar, una técnica sencilla de mínimos cuadrados en el dominio de la frecuencia permite la estimación de canal en sistemas con dos antenas y constelaciones complejas, y con tres o cuatro antenas y constelaciones reales o complejas, utilizando STBCs ortogonales como bloques de entrenamiento. En segundo lugar, una representación 'sobre-completa' permite hacer una estimación diferencial de canal para un sistema con tres antenas transmisoras mediante la selección a partir de un banco de posibles estimadores, basándose en la redundancia provista por la matriz de transmisión no cuadrada del código ortogonal esporádico de tasa 3/4 para tres antenas transmisoras.En el contexto de sistemas con adaptación del transmisor, la técnica propuesta de diversidad por selección adaptativa de código espacio-tiempo se basa en el estado instantáneo del vector de canal y en un conjunto de niveles umbrales hallados fuera de línea en función del período de realimentación. Los resultados indican que esta técnica proporciona buenas prestaciones en canales correlados e incorrelados. Su aplicación a sistemas OFDM ha sido estudiada, superando a técnicas de selección de antena y a otras técnicas de transmisión adaptativa. / Channel estimation and adaptive transmission techniques are areas of increasing interest these days when considering transmit diversity systems for the 3G and 4G wireless communication systems. In this thesis an analysis of the channel estimation and channel correlation impact on transmit diversity OFDM systems based on space-time block coding (STBC) is presented, two channel estimation techniques are outlined and an adaptive space-time code selection technique is proposed. First, a simple frequency domain least square technique allows channel estimation for two transmitter systems with complex constellation, and three or four transmitter systems with real or complex constellation, using orthogonal STBCs as training blocks. Second, an 'overcomplete' representation allows a di.erential channel estimation for three transmitter systems through the instantaneous selection from a bank of estimators, based on the redundacy provided by the non-square transmission matrix of the sporadic 3/4-rate STBC for three transmitters.In the context of transmit adaptive systems, the proposed adaptive space-time code selection technique is based on both the instantaneous channel vector state and a set of predetermined threshold levels found o.-line as a function of the feedback period. Analytical and simulation results show that the proposed technique has a good performance in the presence of correlated and uncorrelated channels. Its application to OFDM systems has been considered, outperforming classical antenna selection techniques and other closed-loop adaptive transmission techniques.

Modern embedded processors with dedicated address generation unit support memory accesses using indirect addressing mode with auto-increment and auto-decrement. The auto-increment/decrement mode, if properly utilized, can save address arithmetic instructions, reduce static and dynamic footprint of the program and speed up the execution as well. We propose an optimization framework for embedded processors based on the auto-increment and decrement addressing modes for address registers. Existing work on this class of optimizations focuses on using an access graph and finding the maximum weight path cover to find an optimized stack variables layout. We take this further by using coalescing, addressing mode selection and offset registers to find further opportunities for reducing the number of load-address instructions required. We also propose an algorithm for building the layout with considerations for memory accesses across basic blocks, because existing work mainly considers intra-basic-block information. We then use the available offset registers to try to further reduce the number of address arithmetic instructions after layout assignment.

In recent years, genetic algorithms (GAs) began to be used in many disciplines such as pattern recognition, robotics, biology, and medicine to name just a few. GAs are based on Darwin's principle of natural selection which happens to be a slow process and, as a result, these algorithms tend to require a large amount of computation. However, they offer certain advantages as well over classical gradient-based optimization algorithms such as steepest-descent and Newton-type algorithms. For example, having located local suboptimal solutions they can discard them in favor of more promising local solutions and, therefore, they are more likely to obtain better solutions in multimodal problems. By contrast, classical optimization algorithms though very efficient, they are not equipped to discard inferior local solutions in favour of more optimal ones. This dissertation is concerned with the design of several types of digital filters by using GAs as detailed bellow. In Chap. 2, two approaches for the design of fractional delay (FD) filters based on a GA are developed. The approaches exploit the advantages of a global search technique to determine the coefficients of FD FIR and allpass-IIR filters based on the so-called Farrow structure. The GA approach was compared with a least-squares approach and was found to lead to improvements in the amplitude response and/or delay characteristic. In Chap. 3, a GA-based approach is developed for the design of delay equalizers. In this approach, the equalizer coefficients are optimized using an objective function based on the passband filter-equalizer group delay. The required equalizer is built by adding new second-order sections until the desired accuracy in terms of the flatness of the group delay with respect to the passband is achieved. With this approach stable delay equalizers satisfying arbitrary prescribed specifications with the desired degree of group-delay flatness can easily be obtained. In Chap. 4, a GA-based approach for the design of multiplierless FIR filters is developed. A recently-introduced GA, called orthogonal GA (OGA) based on the so-called experimental design technique, is exploited to obtain fixed-point implementations of linear-phase FIR filters. In this approach, the effects of finite word length are minimized by considering the filter as a cascade of two sections. The OGA leads to an improved amplitude response relative to that of an equivalent direct-form cascade filter obtained using the Remez exchange algorithm. In Chap. 5, a multiobjective GA for the design of asymmetric FIR filters is proposed. This GA uses a specially tailored elitist nondominated sorting GA (ENSGA) to obtain so-called Pareto-optimal solutions for the problem at hand. Flexibility is introduced in the design by imposing phase-response linearity only in the passband instead of the entire baseband as in conventional designs. Three objective functions based on the amplitude-response error and the flatness of the group-delay characteristic are explored in the design examples considered. When compared with a WLS design method, the ENSGA was found to lead to improvements in the amplitude response and passband group-delay characteristic. In Chap. 6, a hybrid approach for the design of IIR filters using a GA along with a quasi-Newton (QN) algorithm is developed. The hybrid algorithm, referenced to as the genetic quasi-Newton (GQN) algorithm combines the flexibility and reliability inherent in the GA with the fast convergence and precision of the QN algorithm. The GA is used as a global search tool to explore different regions in the parameter space whereas the QN algorithm exploits the efficiency of a gradient-based algorithm in locating local solutions. The GQN algorithm works well with an arbitrary random initialization and filters that would satisfy prescribed amplitude-response specifications can easily be designed

This thesis presents novel signal processing algorithms for speech and music dereverberation. The proposed algorithms focus on blind single-channel suppression of late reverberation; however binaural and semi-blind methods have also been introduced. Late reverberation is a particularly harmful distortion, since it significantly decreases the perceived quality of the reverberant signals but also degrades the performance of Automatic Speech Recognition (ASR) systems and other speech and music processing algorithms. Hence, the proposed deverberation methods can be either used as standalone enhancing techniques or implemented as preprocessing schemes prior to ASR or other applied systems. The main dereverberation method proposed here is a blind dereverberation technique based on perceptual reverberation modeling has been developed. This technique employs a computational auditory masking model and locates the signal regions where late reverberation is audible, i.e. where it is unmasked from the clean signal components. Following a selective signal processing approach, only such signal regions are further processed through sub-band gain filtering. The above technique has been evaluated for both speech and music signals and for a wide range of reverberation conditions. In all cases it was found to minimize the processing artifacts and to produce perceptually superior clean signal estimations than any other tested technique. Moreover, extensive ASR tests have shown that it significantly improves the recognition performance, especially in highly reverberant environments. / Η διατριβή αποτελείται από εννιά κεφάλαια, δύο παραρτήματα καθώς και την σχετική βιβλιογραφία. Είναι γραμμένη στα αγγλικά ενώ περιλαμβάνει και ελληνική περίληψη. Στην παρούσα διατριβή, αναπτύσσονται μεθόδοι ψηφιακής επεξεργασίας σήματος για την αφαίρεση αντήχησης από σήματα ομιλίας και μουσικής. Οι προτεινόμενοι αλγόριθμοι καλύπτουν ένα μεγάλο εύρος εφαρμογών αρχικά εστιάζοντας στην τυφλή (“blind”) αφαίρεση για μονοκαναλικά σήματα. Στοχεύοντας σε πιο ειδικά σενάρια χρήσης προτείνονται επίσης αμφιωτικοί αλγόριθμοι αλλά και τεχνικές που προϋποθέτουν την πραγματοποίηση κάποιας ακουστικής μέτρησης. Οι αλγόριθμοι επικεντρώνουν στην αφαίρεση της καθυστερημένης αντήχησης που είναι ιδιαίτερα επιβλαβής για την ποιότητα σημάτων ομιλίας και μουσικής και μειώνει την καταληπτότητα της ομιλίας. Επίσης, επειδή αλλοιώνει σημαντικά τα στατιστικά των σημάτων, μειώνει σημαντικά την απόδοση συστημάτων αυτόματης αναγνώρισης ομιλίας καθώς και άλλων αλγορίθμων ψηφιακής επεξεργασίας ομιλίας και μουσικής. Έτσι οι προτεινόμενοι αλγόριθμοι μπορούν είτε να χρησιμοποιηθούν σαν αυτόνομες τεχνικές βελτίωσης της ποιότητας των ακουστικών σημάτων είτε να ενσωματωθούν σαν στάδια προ-επεξεργασίας σε άλλες εφαρμογές. Η κύρια μέθοδος αφαίρεσης αντήχησης που προτείνεται στην διατριβή, είναι βασισμένη στην αντιληπτική μοντελοποίηση και χρησιμοποιεί ένα σύγχρονο ψυχοακουστικό μοντέλο. Με βάση αυτό το μοντέλο γίνεται μία εκτίμηση των σημείων του σήματος που η αντήχηση είναι ακουστή δηλαδή που δεν επικαλύπτεται από το ισχυρότερο σε ένταση καθαρό από αντήχηση σήμα. Η συγκεκριμένη εκτίμηση οδηγεί σε μία επιλεκτική επεξεργασία σήματος όπου η αφαίρεση πραγματοποιείται σε αυτά και μόνο τα σημεία, μέσω πρωτότυπων υβριδικών συναρτήσεων κέρδους που βασίζονται σε δείκτες αντικειμενικής και υποκειμενικής αλλοίωσης. Εκτεταμένα αντικειμενικά και υποκειμενικά πειράματα δείχνουν ότι η προτεινόμενη τεχνική δίνει βέλτιστες ποιοτικά ανηχωικές εκτιμήσεις ανεξάρτητα από το μέγεθος του χώρου.