<div>Electrical nerve stimulation (ENS) is an emerging therapy for many neurological disorders. Compared with conventional one-way stimulations, closed-loop ENS approaches increase the stimulation efficacy and minimize patient's discomfort by constantly adjusting the stimulation parameters according to the feedback biomarkers from patients. Wireless neurostimulation devices capable of both stimulation and telemetry of recorded physiological signals are welcome for closed-loop ENS systems to improve the quality and reduce the costs of treatments, and real-time digital signal processing (DSP) engines processing and extracting features from recorded signals can reduce the data transmission rate and the resulting power consumption of wireless devices. Electrically-evoked compound action potential (ECAP) is an objective measure of nerve activity and has been used as the feedback biomarker in closed-loop ENS systems including neural response telemetry (NRT) systems and a newly proposed autonomous nerve control (ANC) platform. It's desirable to design a DSP engine for real-time processing of ECAP in closed-loop ENS systems. </div><div><br></div><div>This thesis focuses on developing the DSP architecture for real-time processing of ECAP, including stimulus artifact rejection (SAR), denoising, and extraction of nerve fiber responses as biomedical features, and its VLSI implementation for optimal hardware costs. The first part presents the DSP architecture for real-time SAR and denoising of ECAP in NRT systems. A bidirectional-filtered coherent averaging (BFCA) method is proposed, which enables the configurable linear-phase filter to be realized hardware efficiently for distortion-free filtering of ECAPs and can be easily combined with the alternating-polarity (AP) stimulation method for SAR. Design techniques including folded-IIR filter and division-free averaging are incorporated to reduce the computation cost. The second part presents the fiber-response extraction engine (FREE), a dedicated DSP engine for nerve activation control in the ANC platform. FREE employs the DSP architecture of the BFCA method combined with the AP stimulation, and the architecture of computationally efficient peak detection and classification algorithms for fiber response extraction from ECAP. FREE is mapped onto a custom-made and battery-powered wearable wireless device incorporating a low-power FPGA, a Bluetooth transceiver, a stimulation and recording analog front-end and a power-management unit. In comparison with previous software-based signal processing, FREE not only reduces the data rate of wireless devices but also improves the precision of fiber response classification in noisy environments, which contributes to the construction of high-accuracy nerve activation profile in the ANC platform. An application-specific integrated circuit (ASIC) version of FREE is implemented in 180-nm CMOS technology, with total chip area and core power consumption of 19.98 mm² and 1.95 mW, respectively. </div><div><br></div>

La majeure partie des communications mondiales est transportée par des systèmes transocéaniques à fibre optique. Il est estimé que d'ici 2020 le trafic de données atteindra 4.3 ZB par an. Afin de faire face à cette demande, différentes technologies sont actuellement étudiées pour augmenter la capacité des systèmes de transmission très longue distance. Avec l'avènement des circuits intégrés à haute vitesse, des formats de modulation avancés et des techniques de traitement de signal numérique (DSP) peuvent être utilisés pour maximiser l'efficacité spectrale de transmission. Par ailleurs, la capacité des systèmes modernes est fortement limitée par les effets non-linéaires de type Kerr dans la fibre. Ainsi, la première partie de ce travail est axée sur l’étude de la performance et des gains réalisables par des techniques DSP à faible complexité pour mitiger les effets non-linéaires monocanal. En outre, l’utilisation des formats de modulation multiniveaux à haute efficacité spectrale au-delà de 16QAM a pris de l'ampleur pour augmenter le débit de transmission des systèmes, notamment avec l’introduction des formats QAM avec mise en forme probabiliste (PCS-QAM), plus performants que les formats QAM classiques. La deuxième partie de ce travail présente donc une comparaison théorique ainsi qu’expérimentale du format PCS-64QAM avec d’autres formats à haute efficacité spectrale pour les distances transatlantiques. La mise en œuvre d’un format PCS-64QAM conçu pour les distances transpacifiques est également abordée. Enfin, la dernière partie de ce travail concrétise les résultats des travaux menés dans les deux sections précédentes en présentant plusieurs records de transmission / Global data traffic is expected to reach up to 4.3 ZB per year by 2020. With the majority of the global communications being transported on submarine point-to-point fiber-optic systems, different cutting-edge technologies have been under research to cope with this unprecedented traffic growth. Continuous advances in high-speed integrated circuits have allowed the use of advanced modulation formats and digital signal processing (DSP) techniques to maximize the transmission spectral efficiency. With mitigation of fiber linear effects efficiently carried out by DSP with relative low-complexity, the capacity of modern fiber optic systems rests limited by fiber nonlinearities. To this extent, in the first part of this work, the performance and achievable benefits of low-complexity DSP techniques aiming to mitigate fiber Kerr nonlinear effects are investigated. Besides nonlinear compensation techniques, the use of multi-level modulation formats beyond 16QAM and high symbol rate channels have gained momentum to increase the system spectral efficiency. One of the major breakthroughs in the recent years, has been the introduction of QAM-based probabilistic constellation shaping (PCS-QAM), which has proven to outperform regular QAM formats. In this sense, in the second part of this work, the practical achievable rate increase brought by PCS-QAM for transoceanic distances is investigated. A theoretical and experimental comparison with other high-capacity formats is performed, and the design of a PCS-QAM for trans-Pacific distances is addressed. Finally, in the last section, several transmission records using the two above techniques are reported

Les systèmes de communications optiques jouent un role important pour satisfaire la demande incessante de trafics de données. Cette demande, induite par des applications gourmandes en termes de bande passante et débit, necéssite une augmentation de la capacité des réseaux optiques d’accès et par conséquent une augmentation des capacités de réseaux de transports métropolitains et longues distances. La prochaine génération de systèmes WDM longue distance devrait opérée à des débits de 400Gbps ou 1Tbps. Cette montée en débit s’appuiera sur des nouvelles formes d’ondes avancées de type mono-porteuse (Nyquist-WDM) ou multi-porteuse (OFDM multi-bande). Ces approches sont basées sur le multipléxage de plusieurs porteuses espacées par des intervalles de garde réduits. D’autre part, pour générer ces très haut débits, des modulations multi-états sont utilisées pour chaque porteuse grâce à leur efficacité spectrale élevée. Ces types de systèmes, qui combinent à la fois les approches multi-bande et les modulations multi-états, sont extrêmement vulnérables aux effets nonlinéaires de la fibre optique. En fait, les effets nonlinéaires sont dépendants de la puissance de transmission et inversement proportionels à l’intervalle de garde. Cela rend leur compensation indispensable pour maintenir des bonnes performances des systèmes en terme de distance de transmission. Grâce à l’emploi de récepteurs à détection cohérente, des techniques de traitement du signal numérique sont utlisées pour combattre les effets nonlinéaires. Dans cette thèse, nous avons proposé des nouvelles techniques basées sur les séries de Volterra et les égaliseurs à retour de decision pour compenser respectivement les effets nonlinéaires intrabande et les interférences nonlinéaires inter-bande. / Optical communication systems have evolved since their deployment to meet the growing demand for high-speed communications. Over the past decades, the global demand for communication capacity has increased exponentially and the most of the growth has occurred in the last few years when data started dominating network traffic. In order to meet the increase of traffic demands fueled by the growth of internet services, an increase of access network capacity and consequently metro and long-haul network capacities is required. Next generation of long-haul WDM transmission systems is expected to operate at 400Gbps or 1Tbps bit rate. Superchannel approaches, such as Nyquist WDM and multi-band OFDM, allow both high spectral efficiency and small guardband which makes them promising candidates to generate these high bit rates in combination with multi-level modulations formats. Such transmission systems are strongly disturbed by fiber nonlinear effects which increase with the data rate and the small guard band. Therefore, fiber nonlinearities compensation is required to get the desired performance in terms of transmission reach. DSP based approaches such as digital back propagation and third-order Volterra based nonlinear equalizer have been already proposed to deal with intra-channel or intra-band nonlinear effects. In the context of superchannel systems, we have proposed two new compensation techniques to deal with fiber nonlinear effects. The first one, called fifth-order inverse Volterra based nonlinear equalizer, compensate for intra-band nonlinear effects. The second approach, which is the interband/ subcarrier nonlinear interference canceler, is proposed to combat the nonlinear interference insuperchannel systems.

The thesis deals with research of pulse compression of the acoustic signal in terms of applications in civil engineering. Based on the study and analysis of these methods, automated measuring equipment for non-destructive testing with pseudorandom sequence of maximum length and automated signal analysis, have been designed and implemented. In a single test cycle are obtained three parameters that characterize the linear and nonlinear behavior of the sample. A nonlinear parameter, Time of Flight of ultrasonic wave in the sample is further in the work compared with the conventional pulse measuring, and spectral analysis is compared with the method impact-echo. Functionality and optimization of the testing method was performed on a total of three sets of test pieces made of various building materials. The experiments proved simple result interpretation, and high sensitivity to structural damage associated with temperature loading. The results were correlated with conventional nondestructive methods and by destructive testing was measured change in compressive strength and flexural strength. This work also includes continual measurement of fundamental frequency influenced by moisture on a mortar sample. Use of pulse compression signal is in the civil engineering quite unusual. Only in recent years this topic is discussed in scientific articles with increasing frequency. Great potential lies in the association of three test methods into a single. Beneficial is high test speed and measurement reproducibility, but also theoretical possibility of testing massive test elements.

Matched frequency responses are a fundamental starting point for a variety ofimplementations for microphone arrays. In this report, two methods for frequencyresponse-calibration of a pre-assembled microphone array are presented andevaluated. This is done by extracting the deviation in frequency responses of themicrophones in relation to a selected reference microphone, using a swept sine asa stimulus signal and an inverse filter. The swept sine includes all frequencieswithin the bandwidth of human speech. This allows for a full frequency responsemeasurements from all microphones using a single recording.Using the swept sine, the deviation in frequency response between the microphonescan be obtained. This deviation represents the scaling factor that all microphonesmust be calibrated with to match the reference microphone. Applying the scalingfactors on the recorded stimulus signal shows an improvement for both implementedmethods, and where one method matches the frequency response of the microphoneswith high accuracy.Once the scaling factors of the various microphones is obtained, it can be usedto calibrate other recorded signals. This leads to an minor improvement formatching the frequency responses, as it has been shown that the differencesin frequency response between the microphones is signal-dependent and variesbetween recordings. The response differences between the microphones dependson the design of the array, speaker, room and the acoustic frequency dispersionthat occurs with sound waves. This makes it difficult to calibrate the frequencyresponses of the microphones without appropriate equipment because the responseof the microphones is noticeably affected by these other factors. Proposals to addressthese problems are discussed in the report as future work. / Matchade frekvenssvar är en grundläggande utgångspunkt för ett flertal implementationer för mikrofonarrayer. I denna rapport presenteras och utvärderas tvåmetoder för frekvenssvarskalibrering för en förmonterad mikrofonarray. Detta görsgenom att extrahera avvikelsen i frekvenssvar hos alla mikrofoner i förhållandetill en vald referensmikrofon. Frekvenssvaren tas fram med hjälp av ettsinussvep som stimulanssignal och ett inverterat filter. Sinussvepet inkluderar helafrekvensbredden för mänskligt tal och möjliggör att mikrofonernas fulla frekvenssvarkan analyseras från en enda inspelning.Med hjälp av sinussvepet kan avvikelsen i frekvenssvar mellan mikrofonerna erhållas.Denna avvikelse representerar den skalningsfaktor alla mikrofoner måste kalibrerasefter för att matcha referensmikrofonen. Genom att applicera faktorerna på deninspelade stimulussignalen ses en förbättring för båda implementerade metoderna,där en metod matchar mikrofonernas frekvenssvar med hög noggrannhet.När skalningsfaktorn för de olika mikrofonerna har erhållits kan den användas föratt kalibrera andra inspelade signaler. Detta leder till en liten förbättring i att matchafrekvenssvaren, då det har visat sig att skillnader mellan mikrofonernas frekvenssvarär signalberoende och varierar mellan inspelningar. Skillnader i frekvenssvar mellanmikrofonerna beror på ljudets utbredning i rummet, utformningen av arrayen,högtalaren och den akustiska frekvensspridningen som uppstår hos ljudvågor. Dettagör det svårt att kalibrera frekvenssvaren hos mikrofonerna utan lämplig utrustningeftersom mikrofonernas respons märkbart påverkas av dessa andra faktorer. Förslagför att kringgå dessa problem diskuteras i rapporten och tas upp som framtidaarbete.