This research focuses on a novel integrated approach for computing and representing complex numbers as a single entity without the use of any dedicated multiplier for calculating the fast Fourier transform algorithm (FFT), using the Distributed Arithmetic (DA) technique and Complex Binary Number Systems (CBNS). The FFT algorithm is one of the most used and implemented technique employed in many Digital Signal Processing (DSP) applications in the field of science, engineering, and mathematics. The DA approach is a technique that is used to compute the inner dot product between two vectors without the use of any dedicated multipliers. These dedicated multipliers are fast but they consume a large amount of hardware and are quite costly. The DA multiplier process is accomplished by shifting and adding only without the need of any dedicated multiplier. In today's technology, complex numbers are computed using the divide and conquer approach in which complex numbers are divided into two parts: the real and imaginary. The CBNS technique however, allows for each complex addition and multiplication to be computed in one single step instead of two. With the combined DA-CBNS approach for computing the FFT algorithm, those dedicated multipliers are being replaced with a DA system that utilize a Rom-based memory for storing the twiddle factor 'wn' value and the complex arithmetic operations being represented as a single entity, not two, with the CBNS approach. This architectural design was implemented by coding in a very high speed integrated circuit (VHSIC) hardware description language (VHDL) using Xilinx ISE design suite software program version 14.2. This computer aided tool allows for the design to be synthesized to a logic gate level in order to be further implemented onto a Field Programmable Gate Array (FPGA) device. The VHDL code used to build this architecture was downloaded on a Nexys 4 DDR Artix-7 FPGA board for further testing and analysis. This novel technique resulted in the use of no dedicated multipliers and required half the amount of complex arithmetic computations needed for calculating an FFT structure compared with its current traditional approach. Finally, the results showed that for the proposed architecture design, for a 32 bit, 8-point DA-CBNS FFT structure, the results showed a 32% area reduction, 41% power reduction, 59% reduction in run-time, 42% reduction in logic gate cost, and 66% increase in speed. For a 28 bit, 16-point DA-CBNS FFT structure, its area size, power consumption, run-time, and logic gate, were also found to be reduced at approximately 30%, 37%, 60%, and 39%, respectively, with an increase of speed of approximately 67% when compared to the traditional approach that employs dedicated multipliers and computes its complex arithmetic as two separate entities: the real and imaginary.

abstract: Advancements in mobile technologies have significantly enhanced the capabilities of mobile devices to serve as powerful platforms for sensing, processing, and visualization. Surges in the sensing technology and the abundance of data have enabled the use of these portable devices for real-time data analysis and decision-making in digital signal processing (DSP) applications. Most of the current efforts in DSP education focus on building tools to facilitate understanding of the mathematical principles. However, there is a disconnect between real-world data processing problems and the material presented in a DSP course. Sophisticated mobile interfaces and apps can potentially play a crucial role in providing a hands-on-experience with modern DSP applications to students. In this work, a new paradigm of DSP learning is explored by building an interactive easy-to-use health monitoring application for use in DSP courses. This is motivated by the increasing commercial interest in employing mobile phones for real-time health monitoring tasks. The idea is to exploit the computational abilities of the Android platform to build m-Health modules with sensor interfaces. In particular, appropriate sensing modalities have been identified, and a suite of software functionalities have been developed. Within the existing framework of the AJDSP app, a graphical programming environment, interfaces to on-board and external sensor hardware have also been developed to acquire and process physiological data. The set of sensor signals that can be monitored include electrocardiogram (ECG), photoplethysmogram (PPG), accelerometer signal, and galvanic skin response (GSR). The proposed m-Health modules can be used to estimate parameters such as heart rate, oxygen saturation, step count, and heart rate variability. A set of laboratory exercises have been designed to demonstrate the use of these modules in DSP courses. The app was evaluated through several workshops involving graduate and undergraduate students in signal processing majors at Arizona State University. The usefulness of the software modules in enhancing student understanding of signals, sensors and DSP systems were analyzed. Student opinions about the app and the proposed m-health modules evidenced the merits of integrating tools for mobile sensing and processing in a DSP curriculum, and familiarizing students with challenges in modern data-driven applications. / Dissertation/Thesis / M.S. Electrical Engineering 2013

Recently, smart dust or wireless sensor networks are gaining more attention.These autonomous, ultra-low power sensor-based electronic devices sense and process burst-type environmental variations and pass the data from one node (mote) to another in an ad-hoc network. Subsystems for smart dust are typically the analog interface (AI), analog-to-digital converter (ADC), digital signal processor (DSP), digital-to-analog converter (DAC), power management, and transceiver for communication. This thesis project describes an event-driven (ED) digital signal processing system (ADC, DSP and DAC) operating in continuous-time (CT) with smart dust as the target application. The benefits of the CT system compared to its conventional counterpart are lower in-band quantization noise and no requirement of a clock generator and anti-aliasing filter, which makes it suitable for processing burst-type data signals. A clockless EDADC system based on a CT delta modulation (DM) technique is presented. The ADC output is digital data, continuous in time, known as “data token”. The ADC employs an unbuffered, area efficient, segmented resistor-string (R-string) feedback DAC. A study of different segmented R-string DAC architectures is presented. A comparison in component reduction with prior art shows nearly 87.5% reduction of resistors and switches in the DAC and the D flip-flops in the bidirectional shift registers for an 8-bit ADC, utilizing the proposed segmented DAC architecture. The obtained SNDR for the 3-bit, 4-bit and 8-bit ADC system is 22.696 dB, 30.435 dB and 55.73 dB, respectively, with the band of interest as 220.5 kHz. The CTDSP operates asynchronously and process the data token obtained from the EDADC. A clockless transversal direct-form finite impulse response (FIR) low-pass filter (LPF) is designed. Systematic top-down test-driven methodology is employed through out the project. Initially, MATLAB models are used to compare the CT systems with the sampled systems. The complete CTDSP system is implemented in Cadence design environment. The thesis has resulted in two conference contributions. One for the 20th European Conference on Circuit Theory and Design, ECCTD’11 and the other for the 19th IFIP/IEEE International Conference on Very Large Scale Integration, VLSI-SoC’11. We obtained the second-best student paper award at the ECCTD.

Σκοπός της παρούσας διπλωματικής εργασίας είναι η δημιουργία συστήματος που θα διευρύνει τις δυνατότητες των εξ’ αποστάσεως εργαστηρίων προσανατολισμένα σε θέματα ψηφιακής επεξεργασίας σήματος και εικόνας. Η υλοποίηση πραγματοποιήθηκε με το LabVIEW v 8.6 και ονομάστηκε R-DSP Server. Αξιοποιώντας τις δυνατότητες που προσφέρει ο R-DSP Server οι χρήστες μπορούν να αναπτύξουν τα δικά τους γραφικά περιβάλλοντα (Graphical User Interfaces -GUIs) τα οποία ονομάζονται προσαρμοζόμενα γραφικά περιβάλλοντα (Customized GUIs,) για τον απομακρυσμένο έλεγχο DSP εφαρμογών. Για την εύκολη και γρήγορη ανάπτυξη τέτοιων γραφικών εφαρμογών στο περιβάλλων του LabVIEW, αναπτύχθηκε μια σειρά εργαλείων που ονομάστηκε R-DSP LabVIEW Toolkit. Η εργασία ολοκληρώνεται με την παρουσίαση της λειτουργιάς του R-DSP Server αλλά και της χρήσης του R-DSP Toolkit. / The purpose of this work is to present an approach which could expand the features of Remote Laboratories focused on embedded Digital Signal Processing (DSP) systems. The proposed approach is based on a system which is designed and developed with LabVIEW and is called R-DSP Server. Exploiting this system, users are able to develop their own Graphical User Interfaces (GUIs), named Customized GUIs, for the remote control and validation of real-time DSP applications. These GUIs are tailored to the needs of each DSP application and can be implemented in any programming language. The rapid design of Customized GUIs using LabVIEW for the communication with the R-DSP Server is achieved using an implemented set of functions, called R-DSP LabVIEW Toolkit.

Dans l'instrumentation nucléaire, les architectures de traitement numérique du signal doivent faire face à la nature poissonienne du signal, composée d'impulsions d'arrivées aléatoires qui imposent aux architectures actuelles de travailler en flux de données. En effet, si le débit d'impulsion est trop élevé, les besoins en temps réel impliquent de paralyser l'acquisition du signal durant le traitement d'une impulsion. Durant ce délai, appelé temps mort, des impulsions peuvent être perdues. Cette contrainte conduit les architectures actuelles à utiliser des solutions dédiées à base de FPGA. Les utilisateurs finaux doivent cependant pouvoir mettre en oeuvre un large éventail d'applications sur un nombre de canaux d'acquisition qui varie. Ce besoin en flexibilité conduit à proposer une architecture programmable (C, C ++). Cette thèse présente une architecture numérique « dirigée par les impulsions » qui répond à ces contraintes. En premier lieu, cette architecture se compose d'extracteurs d'impulsions capables d'extraire de façon dynamique les impulsions en fonction de leur taille pour n'importe quel type de détecteur délivrant des impulsions. Ces impulsions sont ensuite distribuées sur des unités fonctionnelles programmables (FU) indépendante. Ces FUs gèrent l'arrivée d'événements aléatoires et des durées d'exécution de programme non-déterministes. Le simulateur de l'architecture est développé en SystemC au cycle d'horloge près. Il montre des résultats prometteurs en termes de passage à l'échelle, tout en maintenant le zéro-temps mort. Cette architecture permet d'embarquer de nouveaux algorithmes de traitement des impulsions traditionnellement utilisés hors ligne. / In the field of nuclear instrumentation, digital signal processing architectures have to deal with the poissonian characteristic of the signal, composed of random arrival pulses which requires current architectures to work in dataflow. Thus, the real-time needs implies losing pulses when the pulse rate is too high. Current architectures paralyze the acquisition of the signal during the pulse processing inducing a time during no signal can be processed, this is called the dead time. These issue have led current architectures to use dedicated solutions based on reconfigurable components such as FPGAs. The requirement of end users to implement a wide range of applications on a large number of channels leads to propose an easily programmable architecture platform (C, C++). This thesis present presents a digital “pulse-driven” architecture that meets these constraints. This architecture is first composed of pulse extractors. They are capable of dynamically extracting the pulses according to their size for any type of detector that delivering pulses. These pulses are then distributed on a set of programmable and independent Functional Units (FU) which are "pulses driven". These FUs are able to handle the arrival of non-deterministic events and variable program execution times and indeterminate in advance. The virtual prototype of the architecture is developed in cycle accurate SystemC and shows promising results in terms of scalability while maintaining zero dead time. This architecture paves the way for novel real time pulse processing by reducing the gap between embedded real time processing and offline processing.

Contemporary research in digital signal processing (DSP) is focused on issues of computational complexity, very high data rate and large quantum of data. Thus, the success in newer applications and areas hinge on handling these issues. Conventional ways to address these challenges are to develop newer structures like Multirate signal processing, Multiple Input Multiple Output(MIMO), bandpass sampling, compressed domain sensing etc. In the implementation domain, the approach is to look at floating point over fixed point representation and / or longer wordlength etc., related to number representations and computations. Of these, a simple approach is to look at number representation, perhaps with a simple integer. This automatically guarantees accuracy and zero quantization error as well as longer wordlength. Thus, it is necessary and interesting to explore viable DSP alternatives that can reduce complexity and yet match the required performance. The main aim of this work is to highlight the importance, use and analysis of integer sequences. Firstly, the thesis explores the use of integer sequences as windowing functions. The results of these investigations show that integer sequences and their convolution, indeed, outperform many of the classical real valued window functions in terms of mainlobe width, sidelobe attenuation etc. Secondly, the thesis proposes techniques to approximate discrete Gaussian distribution using integer sequences. The key idea is to convolve symmetrized integer sequences and examine the resulting profiles. These profiles are found to approximate discrete Gaussian distribution with a mean square error of the order of 10−8 or less. While looking at integer sequences to approximate discrete Gaussian, Fibonacci sequence was found to exhibit some interesting properties. The third part of the thesis proves certain fascinating optimal probabilistic limit properties (mean and variance) of Fibonacci sequence. The thesis also provides complete generalization of these properties to probability distributions generated by second order linear recurrence relation with integer coefficients and any kth order linear recurrence relation with unit coefficients. In addition to the above, the thesis also throws light on possible architectural implications of using integer sequences in DSP applications and ideas for further exploration.

No description available.

<div>Pavement drainage systems are one of the key drivers of pavement function and longevity, and effective drain maintenance can significantly extend a pavement's service life. Maintenance of these drains, however, is often hampered by the challenge of locating the drains. Ground Penetrating Radar (GPR) typically offers a rapid and effective method to detect these underground targets. However, typical detection schema that rely upon the observation of the hyperbolic return from a GPR scan of a buried conduit still tend to miss many of the older drains beneath pavements as they may be partially or fully filled with sediment and/or may be fabricated from clay or other earthen materials, yielding a return signal that is convolved with significant background noise. </div><div><br></div><div>To manage this challenge, this work puts forward an improved background noise and clutter reduction method to enhance the target signals in what amounts to a constructed environment that tends to have more consistent subsurface properties than one might encounter in a general setting. Within this technique, two major algorithms are employed. Algorithm 1 is the core of this method, and plays the role of reducing background noise and clutter. Algorithm 2 is supplementary, and helps eliminate anomalous discontinuous returns generated by the equipment itself, which could otherwise lead to false detection indications in the output of Algorithm 1. Instead of traditional 2-D GPR images, the result of the proposed algorithms is a 1-D plot along the survey line, highlighting a set of “points of interest” that could indicate buried drain locations identified at any given GPR operating frequency. Subsurface exploration using two different operating frequencies, 900 MHz and 400 MHz herein, is then employed to further enhance detection confidence. Points of interest are ultimately coded to define the confidence of the detection. Comparing the final result of proposed algorithms with the original GPR images, the improved algorithm is demonstrated to provide significantly improved detection results, and could potentially be applied to similar problems in other contexts.</div><div><br></div><div>Besides the background reduction methods, a group of simulations performed using GPRMAX2D software are examined to explore the influence of road cross-section designs on sub-pavement drainage conduit GPR signatures, and evaluate the effectiveness of alternate GPR antennae configurations in locating these buried conduits in different ground conditions. Two different models were explored to simulate conduit detection. In addition, different pipe and soil conditions were modeled, such as pipe size, pipe material, soil moisture level, and soil type. Four different quantitative measurements are used to analyze GPR performance based on different key factors. The four measurements are 1) signal to background ratio (SBR) in dB; 2) signal to receiver noise ratio (SNR) in dB; 3) signal energy in Volts; and 4) average signal band power in Watts.</div><div><br></div><div>The water and clay content of subsurface soil can significantly influence the detection results obtained from ground penetrating radar (GPR). Due to the variation of the material properties underground, the center frequency of transmitted GPR signals shifts to a lower range as wave attenuation increases. Examination of wave propagation in the subsurface employing an attenuation filter based on a linear system model shows that received GPR signals will be shifted to lower frequencies than those originally transmitted. The amount of the shift is controlled by a wave attenuation factor, which is determined by the dielectric constant, electric conductivity, and magnetic susceptibility of the transmitted medium. This work introduces a receiver-transmitter-receiver dual-frequency configuration for GPR that employs two operational frequencies for a given test - one higher and one slightly lower - to take advantage of this phenomenon to improve subpavement drain detection results. In this configuration, the original signal is transmitted from the higher frequency transmitter. After traveling through underground materials, the signal is received by two receivers with different frequencies. One of the receivers has the same higher center frequency as the transmitter, and the other receiver has a lower center frequency. This configuration can be expressed as Rx(low-frequency)-Tx(high-frequency)-Rx(high-frequency) and was applied in both laboratory experiments and field tests. Results are analyzed in the frequency domain to evaluate and compare the properties of the signal obtained by both receivers. The laboratory experiment used the configuration of Rx(400MHz)-Tx(900MHz)-Rx(900MHz). The field tests, in addition to the configuration used in the lab tests, employed another configuration of Rx(270MHz)-Tx(400MHz)-Rx(400MHz) to obtain more information about this phenomenon. Both lab and field test results illustrate the frequency-shift phenomenon described by theoretical calculations. Based on the power spectrum for each signal, the lower frequency antenna typically received more energy (higher density values) at its peak frequency than the higher frequency antenna.</div>

<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>

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.