This thesis presents a bio-signal recording system with offset cancellation and a low power comparator. The recording of bio-signal requires high-gain amplification before recording, to match the input to the range of the analog to digital converter (ADC); interference could be a problem if it causes the amplifier to reach saturation, leaving the recording inoperable (i.e., blank) until it returns to its normal state. The proposed system can monitor the amplifier output, and reset the amplifier output to a point near the center of its dynamic range before the amplifier output leaves its dynamic range. The proposed system provides discrete compensation voltages to cancel the offset voltage, and thereby avoids the shortcomings of conventional filters. Furthermore, a low power and low offset voltage comparator for low current operation is proposed. It is suitable for the clock controller in a sampled bio-signal acquisition system. The measured current consumption of the comparator is less than 130 nA, and the offset voltage is 2 mV. The proposed recording system and comparator have been implemented in the TSMC (Taiwan Semiconductor Manufacturing Company) 0.35£gm 2P4M CMOS process technology to verify the simulation results as well as the correctness of the proposed architecture.

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In this thesis, I conduct the hardware prototyping of a two-way relay system using the National Instruments FlexRIO hardware platform. First of all, I develop several practical mechanisms to solve the critical synchronization issues of the systems, including Orthogonal Frequency-Division Multiplexing (OFDM) frame synchronization at the receiver, source to source node synchronization, and handshaking between the sources and relay nodes. Those synchronization methods control the behavior of the two source nodes and the relay node, which play critical roles in the two-way relay systems. Secondly, I develop a pilot-based channel estimation scheme and validate it by showing the successful self-interference cancellation for the two-way relay systems. In particular, I experiment the self-interference cancellation technique by using several channel estimation schemes to estimate both source to relay channels and relay to source channels. Moreover, I implement the physical layer of a 5 MHz OFDM scheme for the two-way relay system. Both the transmitter and receiver are designed to mimic the Long Term Evolution (LTE) downlink scenario. The physical layer of the transmitter has been implemented in Field-Programmable Gate Arrays (FPGAs) and executed on the hardware board, which provides high throughput and fundamental building blocks for the two-way relay system. The physical layer of receiver is implemented in the real-time controller, which provides the ?exibility to rapidly recon?gure the system. Finally, I demonstrate that the 5MHz OFDM based two-way relay system can achieve reliable communications, when the channel estimation and system synchronization can be correctly executed.

The reduced speech intelligibility caused by feed feedback oscillation is a major problem for hearing aid users. The demand for improved signal quality has led researchers to look for feedback reduction techniques. In this Thesis, we studied several feedback reduction schemes with emphasis in adaptive feedback cancellation algorithms. The main goal was to develop a system for feedback cancellation that is able to adapt to non-stationary environments while having reasonable computational complexity. This requirement is imposed by the need to implement the feedback cancellation scheme in low power DSP systems. In Chapter 1, we briefly introduced hearing aid systems. We examined the parts that are made of and the types of hearing aids that are available in the market. Then, we described the mechanism that causes feedback oscillation in hearing aids and the adverse effects it has on signal quality. Chapter 2 contains some theoretical results on the field of adaptive linear system identification algorithms and simulation results that support this theory. The chapter begins by giving a derivation of the popular LMS algorithm. A theoretical analysis of LMS using the independence assumption is also provided. Then we are concerned with the least squares filter. We described the RLS algorithm and a linear complexity version of it, the FAEST algorithm. Subsequently, we discussed the FNTF algorithm that trades computational complexity for performance in solving the system identification problem. Next, we developed a new algorithm, the FLMS, by making simplifications to FNTF. We also proved that the proposed algorithm outperforms LMS at least when the input signal is an AR process. Finally, we provided simulation results which prove the superiority of FLMS over LMS. Chapter 3 is devoted in using some algorithms described in Chapter 2 for feedback cancellation in hearing aids. The chapter begins with a hearing aid model that includes an acoustic feedback mechanism. On this system, a linear filter is added that estimates the acoustic feedback so that it can be removed from he signal captured by the microphone. The feedback estimation is performed with LMS and FLMS. Using simulation results, we saw that FLMS can be successfully used in feedback systems and continues to outperform LMS. We also saw that, contrary to the open loop case, when feedback is present, the stochastic approximation theory does not satisfactorily predict the mean learning curves of LMS. / Ένα από τα σημαντικότερα προβλήματα που πρέπει να αντιμετωπιστούν κατά το σχεδιασμό ενός ακουστικού βαρηκοΐας είναι αυτό της ακουστικής ανάδρασης. Με τον όρο ακουστική ανάδραση αναφερόμαστε στο φαινόμενο κατά το οποίο ένα μέρος της εξόδου του ακουστικού επιστρέφει στην είσοδο και ενισχύεται εκ νέου. Γνωστό και ως μικροφωνισμός, το φαινόμενο αυτό γίνεται αντιληπτό από τους ασθενείς ως ένα συνεχές σφύριγμα και είναι ιδιαίτερα ενοχλητικό. Για την αντιμετώπιση του φαινομένου έχουν προταθεί διάφορες τεχνικές. Για παράδειγμα, ο περιορισμός του κέρδους ενίσχυσης στις συχνότητες όπου εμφανίζεται ο μικροφωνισμός είναι μια λύση που συναντάται συχνά σε αναλογικά ακουστικά βαρηκοΐας. Η μέθοδος αυτή, όμως, απαιτεί τον προσδιορισμό των επικίνδυνων συχνοτήτων κατά τη διαδικασία προσαρμογής του ακουστικού στον εκάστοτε ασθενή. Ακόμα και αν ο προσδιορισμός γίνει με μεγάλη ακρίβεια, οι συχνότητες στις οποίες εμφανίζεται ο μικροφωνισμός αλλάζουν κατά τη διάρκεια χρήσης του ακουστικού, περιορίζοντας έτσι την αποτελεσματικότητα της μεθόδου. Με την καθιέρωση της ψηφιακής τεχνολογίας στα ακουστικά βαρηκοΐας, εμφανίζονται νέες δυνατότητες για την αντιμετώπιση του μικροφωνισμού. Είμαστε σε θέση, πλέον, να μοντελοποιήσουμε το σύστημα της ακουστικής ανάδρασης και να χρησιμοποιήσουμε το μοντέλο αυτό για εξαλείψουμε το μικροφωνισμό. Για την μοντελοποίηση αυτή χρησιμοποιείται κατά κόρον ο αλγόριθμος LMS. Η χαμηλή υπολογιστική πολυπλοκότητα που τον χαρακτηρίζει τον κάνει ιδανικό για ακουστικά βαρηκοΐας. Στην εργασία αυτή παρουσιάζουμε έναν νέο αλγόριθμο, επίσης χαμηλής πολυπλοκότητας, για το πρόβλημα της αναγνώρισης γραμμικών συστημάτων. Αποδεικνύουμε με μαθηματικό τρόπο ότι είναι πιο αποτελεσματικός από τον LMS για συγκεκριμένα μοντέλα σημάτων εισόδου, ενώ με εξομοιώσεις ότι υπερτερεί του LMS και για πολύ πιο γενικές εισόδους. Επιπρόσθετα, δείχνουμε ότι ο νέος αλγόριθμος μπορεί να χρησιμοποιηθεί για την ακύρωση της ανάδρασης σε ακουστικά βαρηκοΐας, όπου παραμένει πιο αποτελεσματικός από τον LMS.

BACKGROUND: There is a need to identify the populations at high risk of nonadherence to outpatient lung cancer appointments in order to reduce the delay from diagnosis to treatment. The patient navigation system, which helps patients with barriers navigate the health care system, was examined to see if the correct high-risk groups were being addressed. METHODS: A case-control study with 195 subjects from the lung cancer clinics at Boston Medical Center (BMC) was conducted examining three nonadherence case groups: no-shows (n=40), cancelations (n=64) and combined (n=20). Nonadherence was defined as any patient who was a no-show for at least one appointment or who canceled more than one appointment over the three month study period. The combined group incorporated both of these factors. The patients were stratified by 10 patient characteristics, including patient navigation. Odds ratios (ORs) and 95% confidence intervals (CIs) were used for the analysis. A second analysis was done on patients in the patient navigation program (n=33) to determine if the high risk groups identified were being addressed. This was done using ORs and 95% CIs. RESULTS: This study has shown that there are certain patient groups in the lung cancer clinics at BMC that are at higher risk of being nonadherent to lung cancer outpatient appointments. Among those are Hispanic/Latino patients, Spanish and Haitian Creole speaking patients, small cell lung cancer (SCLC) patients, and those patients who have Medicaid, and with late stage lung cancer patients at significantly higher risk (no-shows: OR-5.26 (1.85, 14.95), cancelations: OR-2.49 (1.12, 5.54), combined: OR-12.49 (1.48, 105.46)). Patients in the patient navigation system were also found to be at significantly higher risk of nonadherence (no-shows: OR-3.85 (1.72, 8.65), cancelations: OR-4.13 (1.89, 9.00), combined: OR-5.15 (1.93, 13.72)) than those not in the program. Some patients were also found to be at significantly decreased odds of nonadherence, including those who were: 1000-1999 days post diagnosis (no-shows: OR-0.14 (0.03, 0.59), cancelations: OR-0.20 (0.06, 0.65), combined: OR-0.07 (0.01, 0.64)); 2000-2999 days post diagnosis (no-shows: OR-0.09 (0.01, 0.80), cancelations: OR-0.06 (0.01, 0.50)); aged 71-75 (cancelations: OR-0.25 (0.08, 0.79)). The subset analysis with the patient navigation data yielded no statistically significant results. CONCLUSIONS: The study identified high-risk populations within the total lung cancer population at BMC that should be addressed by the patient navigation program. This study demonstrated that while the program does have its flaws, it is decreasing the odds of nonadherence of many of the high-risk populations.

abstract: Switching regulator has several advantages over linear regulator, but the drawback of switching regulator is ripple voltage on output. Previously people use LDO following a buck converter and multi-phase buck converter to reduce the output voltage ripple. However, these two solutions also have obvious drawbacks and limitations. In this thesis, a novel mixed signal adaptive ripple cancellation technique is presented. The idea is to generate an artificial ripple current with the same amplitude as inductor current ripple but opposite phase that has high linearity tracking behavior. To generate the artificial triangular current, duty cycle information and inductor current ripple amplitude information are needed. By sensing switching node SW, the duty cycle information can be obtained; by using feedback the amplitude of the artificial ripple current can be regulated. The artificial ripple current cancels out the inductor current, and results in a very low ripple output current flowing to load. In top level simulation, 19.3dB ripple rejection can be achieved. / Dissertation/Thesis / Masters Thesis Electrical Engineering 2016

Abstract Advanced receiver structures capable of suppressing multiple-access interference in code-division multiple-access (CDMA) systems operating in frequency-selective fading channels are considered in this thesis. The aim of the thesis is to develop and validate novel receiver concepts suitable for future wideband cellular CDMA systems. Data detection and synchronization both for downlink and uplink receivers are studied. The linear minimum mean squared error (LMMSE) receivers are derived and analyzed in frequency-selective fading channels. Different versions of the LMMSE receivers are shown to be suitable for different data rates. The precombining LMMSE receiver, whichis also suitable for relatively fast fading channels, is shown to improve the performance of the conventional RAKE receivers signicantly in the FRAMES wideband CDMA concept. It is observed that the performance of the conventional RAKE receivers is degraded signicantly with highest data rates due to multiple-access interference (MAI) as well as due to inter-path interference. Based on a general convergence analysis, it is observed that the postcombining LMMSE receivers are mainly suited to the high data rate indoor systems. The blind adaptive LMMSE-RAKE receiverdeveloped for relatively fast fading frequency-selective channels gives superior rate of convergence and bit error rate (BER) performance in comparison to other blind adaptive receivers based on least mean squares algorithms. The minimum variance method based delay estimation in blind adaptive receivers is shown to result in improved delay acquisition performance in comparison to the conventional matched filter and subspace based acquisition schemes. A novel delay tracking algorithm suitable to blind least squares receivers is also proposed. The analysis shows improved tracking performance in comparison to the standard delay-locked loops. Parallel interference cancellation (PIC) receivers are developed for the uplink. Data detection, channel estimation, delay acquisition, delay tracking, inter-cell interference suppression, and array processing in PIC receivers are considered. A multistage data detector with the tentative data decision and the channel estimate feedback from the last stage is developed. Adaptive channel estimation filters are used to improve the channel estimation accuracy. The PIC method is also applied to the timing synchronization of the receiver. It is shown that the PIC based delay acquisition and tracking methods can be used to improve the performance of the conventional synchronization schemes. Although the overall performance of the PIC receiver is relatively good in the single-cell case, its performance is signicantly degraded in a multi-cell environment due to unknown signal components which degrade the MAI estimates and subsequently the cancellation efficiency. The blind receiver concepts developed for the downlink are integrated into the PIC receivers for inter-cell interference suppression. The resulting LMMSE-PIC receiver is capable of suppressing residual interference and results in good BER performance in the presence of unknown signal components.

Abstract The problem of receiving direct sequence (DS) spread spectrum, binary phase shift keyed (BPSK) information in pulsed noise and continuous wave (CW) jamming is studied in additive white noise. An automatic gain control is not modelled. The general system theory of receiver analysis is first presented and previous literature is reviewed. The study treats the problem of decision making after matched filter or integrate and dump demodulation. The decision methods have a great effect on system performance with pulsed jamming. The following receivers are compared: hard, soft, quantized soft, signal level based erasure, and chip combiner receivers. The analysis is done using a channel parameter D, and bit error upper bound. Simulations were done in original papers using a convolutionally coded DS/BPSK system. The simulations confirm that analytical results are valid. Final conclusions are based on analytical results. The analysis is done using a Chernoff upper bound and a union bound. The analysis is presented with pulsed noise and CW jamming. The same kinds of methods can also be used to analyse other jamming signals. The receivers are compared under pulsed noise and CW jamming along with white gaussian noise. The results show that noise jamming is more harmful than CW jamming and that a jammer should use a high pulse duty factor. If the jammer cannot optimise a pulse duty factor, a good robust choice is to use continuous time jamming. The best performance was achieved by the use of the chip combiner receiver. Just slightly worse was the quantized soft and signal level based erasure receivers. The hard decision receiver was clearly worse. The soft decision receiver without jammer state information was shown to be the most vulnerable to pulsed jamming. The chip combiner receiver is 3 dB worse than an optimum receiver (the soft decision receiver with perfect channel state information). If a simple implementation is required, the hard decision receiver should be used. If moderate complex implementation is allowed, the quantized soft decision receiver should be used. The signal level based erasure receiver does not give any remarkable improvement, so that it is not worth using, because it is more complex to implement. If receiver complexity is not limiting factor, the chip combiner receiver should be used. Uncoded DS/BPSK systems are vulnerable to jamming and a channel coding is an essential part of antijam communication system. Detecting the jamming and erasing jammed symbols in a channel decoder can remove the effect of pulsed jamming. The realization of erasure receivers is rather easy using current integrated circuit technology.

Indoor localization with proximity information in ultra-high-frequency (UHF) radio-frequency-identification (RFID) is widely considered as a potential candidate of locating items in Internet-of-Things (IoT) paradigm. First, the proximity-based methods are less affected by multi-path distortion and dynamic changes of the indoor environment compared to the traditional range-based localization methods. The objective of this dissertation is to use tag-to-tag backscattering communication link in augmented UHF RFID system (AURIS) for proximity-based indoor localization solution. Tag-to-tag backscattering communication in AURIS has an obvious advantage over the conventional reader-to-tag link for proximity-based indoor localization by keeping both landmark and mobile tags simple and inexpensive. This work is the very first thesis evaluating proximity-based localization solution using tag-to-tag backscattering communication.Our research makes the contributions in terms of phase cancellation effect, the improved mathematical models and localization algorithm. First, we investigate the phase cancellation effect in the tag-to-tag backscattering communication, which has a significant effect on proximity-based localization. We then present a solution to counter such destructive effect by exploiting the spatial diversity of dual antennas. Second, a novel and realistic detection probability model of ST-to-tag detection is proposed. In AURIS, a large set of passive tags are placed at known locations as landmarks, and STs are attached mobile targets of interest. We identify two technical roadblocks of AURIS and existing localization algorithms as false synchronous detection assumption and state evolution model constraints. With the new and more realistic detection probability model we explore the use of particle filtering methodology for localizing ST, which overcomes the aforementioned roadblocks. Last, we propose a landmark-based sequential localization and mapping framework (SQLAM) for AURIS to locate STs and passive tags with unknown locations, which leverages a set of passive landmark tags to localize ST, and sequentially constructs a geographical map of passive tags with unknown locations while ST is moving in the environment. Mapping passive tags with unknown locations accurately leads to practical advantages. First, the localization capability of AURIS is not confined to the objects carrying STs. Second, the problem of failed landmark tags is addressed by including passive tags with resolved locations into landmark set. Each of the contributions is supported by extensive computer simulation to demonstrate the performance of enhancements.

Passive radar detects and ranges targets by receiving signals which are reflected off targets. Communication transmissions are generally used, however, theoretically any signal with a suitable ambiguity function may be used. The exploitation of an existing transmitter and the removal of emissions allow passive radars to act as a complementary sensor which is useful in environments where conventional active radar is not well suited. Such environments are in covert operations and in situations where a low cost or spectrally efficient solution is required. Most developed passive radars employ intensive signal processing and use application specific equipment to achieve detection. The high-end processors and receiver equipment, however, detract from some of the inherent advantages in the passive radar architecture. These include the lower cost and power requirements achieved by removing transmitter hardware. This study investigates the challenges faced when removing application-specific and high end components from the system and replacing them with low-cost alternatives. Solutions to these challenges are presented and validated by designing and evaluating a radar using these principles. It was found that the major limitation in passive radar is the dynamic range of the receiver. While processing the signals was, and is, a significant challenge, be implemented on a low-cost, low-power embedded processor. This was achieved by asserting a few limitations to the configuration, exploiting the subsequently generated redundancy, and taking advantage of the parallelism by using general purpose graphics processing.. Even on this processor, the system was able to run in real time and able to detect targets up to 91 km (bistatic range of 195 km) from the radar. / Dissertation (MEng)--University of Pretoria, 2016. / Electrical, Electronic and Computer Engineering / MEng / Unrestricted