This thesis investigates the performance differences between Java code compiled by Android Runtime and C++ code compiled by Clang on Android. For testing the differences, the Fast Fourier Transform (FFT) algorithm was chosen to demonstrate examples of when it is relevant to have high performance computing on a mobile device. Different aspects that could affect the execution time of a program were examined. One test measured the overhead related to the Java Native Interface (JNI).  The results showed that the overhead was insignificant for FFT sizes larger than 64.  Another test compared matching implementations of FFTs between Java and native code. The conclusion drawn from this test was that, of the converted algorithms, Columbia Iterative FFT performed the best in both Java and C++. A third test, evaluating the performance of vectorization, proved to be an efficient option for native optimization. Finally, tests examining the effect of using single-point precision (float) versus double-point precision (double) data types were covered. Choosing float could improve performance by using the cache in an efficient manner. / I denna studie undersöktes prestandaskillnader mellan Java-kod kompilerad av Android Runtime och C++-kod kompilerad av Clang på Android. En snabb Fourier Transform (FFT) användes under experimenten för att visa vilka användningsområden som kräver hög prestanda på en mobil enhet. Olika påverkande aspekter vid användningen av en FFT undersöktes. Ett test undersökte hur mycket påverkan Java Native Interface (JNI) hade på ett program i helhet. Resultaten från dessa tester visade att påverkan inte var signifikant för FFT-storlekar större än 64. Ett annat test undersökte prestandaskillnader mellan FFT-algoritmer översatta från Java till C++. Slutsatsen kring dessa tester var att av de översatta algoritmerna var Columbia Iterative FFT den som presterade bäst, både i Java och i C++. Vektorisering visade sig vara en effektiv optimeringsteknik för arkitekturspecifik kod skriven i C++. Slutligen utfördes tester som undersökte prestandaskillnader mellan flyttalsprecision för datatyperna float och double. float kunde förbättra prestandan genom att på ett effektivt sätt utnyttja processorns cache.

Today's world consists of many ways to communicate information. One of the most effective ways to communicate is through the use of speech. Unfortunately many lose the ability to converse. This in turn leads to a large negative psychological impact. In addition, skills such as lecturing and singing must now be restored via other methods. The usage of text-to-speech synthesis has been a popular resolution of restoring the capability to use oral speech. Text to speech synthesizers convert text into speech. Although text to speech systems are useful, they only allow for few default voice selections that do not represent that of the user. In order to achieve total restoration, voice conversion must be introduced. Voice conversion is a method that adjusts a source voice to sound like a target voice. Voice conversion consists of a training and converting process. The training process is conducted by composing a speech corpus to be spoken by both source and target voice. The speech corpus should encompass a variety of speech sounds. Once training is finished, the conversion function is employed to transform the source voice into the target voice. Effectively, voice conversion allows for a speaker to sound like any other person. Therefore, voice conversion can be applied to alter the voice output of a text to speech system to produce the target voice. The thesis investigates how one approach, specifically the usage of voice conversion using Gaussian mixture modeling, can be applied to alter the voice output of a text to speech synthesis system. Researchers found that acceptable results can be obtained from using these methods. Although voice conversion and text to speech synthesis are effective in restoring voice, a sample of the speaker before voice loss must be used during the training process. Therefore it is vital that voice samples are made to combat voice loss.

As wireless protocols become easier to implement, more products come with wireless connectivity. This latest push for wireless connectivity has left a gap in the development of the security and the reliability of some protocols. These wireless protocols can be used in the growing field of IoT where wireless sensors are used to share information throughout a network. IoT is being implemented in homes, agriculture, manufactory, and in the medical field. Disrupting a wireless device from proper communication could potentially result in production loss, security issues, and bodily harm. The 802.15.4/ZigBee protocol is used in low power, low data rate, and low cost wireless applications such as medical devices and home automation devices. This protocol uses CSMA-CA (Carrier Sense Multiple Access w/ Collision Avoidance) which allows for multiple ZigBee devices to transmit simultaneousness and allows for wireless coexistence with the existing protocols at the same frequency band. The CSMA-CA MAC layer seems to introduce an unintentional gap in the reliability of the protocol. By creating a 16-tone signal with center frequencies located in the center of the multiple access channels, all channels will appear to be in use and the ZigBee device will be unable to transmit data. The jamming device will be created using the following hardware implementation. An FPGA connected to a high-speed Digital to Analog Converter will be used to create a digital signal synthesizer device that will create the 16-tone signal. The 16-tone signal will then be mixed up to the 2.4 GHz band, amplified, and radiated using a 2.4 GHz up-converter device. The transmitted jamming signal will cause the ZigBee MAC layer to wait indefinitely for the channel to clear. Since the channel will not clear, the MAC layer will not allow any transmission and the ZigBee devices will not communicate.

<p> This thesis investigates the processing of emergency locator transmitter (ELT) signals which are used in search and rescue satellite-aided tracking (SARSAT) systems. Essentially, the system relies on the transmission of ELT signals from a distressed platform being relayed through an orbiting satellite to an earth station where signal processing can be performed. </p> <p> The methods of signal processing investigated here include both linear and nonlinear. The linear methods include the window function, the autocorrelation function, the digital filtering and the Fast Fourier Transform (FFT). The nonlinear processing is based on the Maximum Entropy Method (MEM) . In addition, additive white Gaussian noise has been added to simulate the performance under different carrier-to-noise density ratio conditions. </p> <p> For a single ELT signal, it is shown in the thesis that the MEM processor gives good spectral performance as compared to the FFT when applied to all types of modulation. When multiple ELT signals are present, the MEM also provides certain benefits in improving the spectral performance as compared to the FFT. </p> / Thesis / Master of Engineering (ME)

<p dir="ltr">The industry's interest in electrified powertrain-equipped vehicles has increased due to environmental and economic reasons. Electrified powertrains, in general, produce lower sound and vibration level than those equipped with internal combustion engines, making noise and vibration (N&V) from other non-engine powertrain components more perceptible. One such N&V type that arouses concern to both vehicle manufacturers and passengers is gear growl, but the signal characteristics of gear growl noise and vibration and the threshold of those characteristics that can be used to determine whether a gear growl requires attention are not yet well understood. This study focuses on developing a method to detect gear-growl based on the N\&V measurements and determining thresholds on various severities of gear-growl using supervised machine learning classification. In general, a machine learning classifier requires sufficient high-quality training data with strong information independence to ensure accurate classification performance. In industrial practices, acquiring high-quality vehicle NVH data is expensive in terms of finance, time, and effort. A physically informed data augmentation method is, thus, proposed to generate realistic powertrain NVH signals based on high-quality measurements which not only provides a larger training data set but also enriches the signal feature variations included in the data set. More specifically, this method extracts physical information such as angular speed, tonal amplitudes distribution, and broadband spectrum shape from the measurement data. Then, it recreates a synthetic signal that mimics the measurement data. The measured and simulated (via data augmentation) are transformed into feature matrix representation so that the N\&V signals can be used in the classification model training process. Features describing signal characteristics are studied, extracted, and selected. While the root-mean-square (RMS) of the vibration signal and spectral entropy were sufficient for detecting gear-growl with a test accuracy of 0.9828, the acoustic signal required more features due to background noise, making data linearly inseparable. The minimum Redundancy Maximum Relevance (mRMR) feature scoring method was used to assess the importance of acoustic signal features in classification. The five most important features based on the importance score were the angular acceleration of the driveshaft, the time derivative of RMS, the tone-to-noise ratio (TNR), the time derivative of the spectral spread of the tonal component of the acoustic signal, and the time derivative of the spectral spread of the original acoustic signal (before tonal and broadband separation). A supervised classification model is developed using a support vector machine from the extracted acoustic signal features. Data used in training and testing consists of steady-state vehicle operations of 25, 35, 45, and 55 mph, with two vehicles with two different powertrain specs: axles with 4.56 and 6.14 gear ratios. The dataset includes powertrains with swapped axles (four different configurations). Techniques such as cost weighting, median filter, and hyperparameter tuning are implemented to improve the classification performance where the model classifies if a segment in the signal represents a gear-growl event or no gear-growl event. The average accuracy of test data was 0.918. A multi-class classification model is further implemented to classify different severities based on preliminary subjective listening studies. Data augmentation using signal simulation showed improvement in binary classification applications. In this study, only gear-growl was used as a fault type. Still, data augmentation, feature extraction and selection, and classification methods can be generalized for NVH signal-based fault diagnosis applications. Further listening studies are suggested for improved classification of multi-class classification applications.</p>

Information is often stored and transmitted through electrical signals. This information may need refinement, which may be done by processing and altering the electrical signals, in which it is transmitted. When refining a signal, a frequency selective filter is often used. It can be implemented through digital signal processing (DSP). DSP is a concept where signals are refined using a digital compute system. Digital systems are designed to replace their analog counterpart, mitigating their flaws in scalability, complexity and cost. A DSP system is typically implemented using software on a small computer, while analog systems are implemented through various electronic components. The objective of this project is to design a DSP system that filters analog input data using automatically synthesised filters from user-defined input specifications. The DSP system is implemented using a microcontroller. The system designed the filters and found the filter coefficients. It then uses analog to digital converter (ADC) to sample an input signal and applies the filter. Lastly, it uses the digital to analog converter (DAC) to reconstruct a filtered, analog result. A user interface is not designed for the system, and only a limited number of filters are implemented. However, the system is successful in designing filters and finding their coefficients.

Location-based services (LBSs) have become pervasive, and the demand for these systems and services is rising. Indoor Positioning Systems (IPSs) are key to extend location-based services indoors where the Global Positioning System (GPS) is not reliable due to low signal strength and complicated signal propagation environment. Most existing IPSs either require the installation of special hardware devices or build a fingerprint map, which is expensive, time-consuming, and labor-intensive. Developments in microelectromechanical systems (MEMS) have resulted in significant advancements in the low-cost compact MARG inertial sensors, making it possible to achieve low-cost and high-accuracy IPSs. This research considers the indoor positioning problem and aims to design and develop an infrastructure-free self-contained indoor positioning and tracking system based on Pedestrian Dead Reckoning (PDR) using MEMS MARG inertial sensors. PDR-based systems rely on MARG inertial sensor measurements to estimate the current position of the object by using a previously determined position without external references. Many issues still exist in developing such systems, such as cumulative errors, high-frequency sensor noises, the gyro drift issue, magnetic distortions, etc. As the MARG sensors are inherently error-prone, the most significant challenge is how to design sensor fusion models and algorithms to accurately extract useful location-based information from individual motion and magnetic sensors. The objective of this thesis is to solve these issues and mitigate the challenges. The proposed positioning system is designed with four main modules at the system level and a dual-mode feature. Specifically, the four main modules are mode detection, step detection and moving distance estimation, heading and orientation estimation, and position estimation. To address the cumulative error issue of using low-cost inertial sensors, signal processing and sensor fusion techniques are utilized for algorithm design. Experimental evaluations show that the proposed position estimation algorithm is able to achieve high positioning accuracy at low costs for the indoor environment. / Thesis / Master of Applied Science (MASc) / With the maturity of microelectromechanical systems (MEMS) technology in recent years, Magnetic, Angular Rate, and Gravity (MARG) sensors are embedded in most smart devices. This research considers the indoor positioning problem and aims to design and develop an infrastructure-free self-contained MEMS MARG inertial sensor-based indoor positioning and tracking system with high precision. The proposed positioning system uses the Pedestrian Dead Reckoning (PDR) approach and includes four main modules at the system level with a dual-mode feature. Specifically, the four main modules are mode detection, step detection and moving distance estimation, heading and orientation estimation, and position estimation. The two modes are static mode and dynamic mode. To address the cumulative error issue of using low-cost inertial sensors, signal processing and sensor fusion techniques are utilized for algorithm design. The detection and estimation algorithms of each module are presented in the system design chapter. Experimental evaluations including trajectory results under five scenarios show that the proposed position estimation algorithm achieves a higher position accuracy than that of conventional estimation methods.

<p>Space-based global navigation and precision timing systems are critical for modern infrastructure. Atomic clock technology has increased the precision of these systems so that they are viable for military operations, navigation, telecommunications, and finance. Advances in optical atomic clocks, based on optical frequencies, provide an opportunity for even more precise timing. Therefore, developments in chip-scale optical atomic clock technologies could lead to increased and more wide-spread application of this precision timing. One component of the optical atomic clock is the optical frequency comb which serves as an interface between optical and microwave frequencies. This thesis will cover experiments related to these optical frequency combs. A 2$\mu$m fiber laser was developed in order to test second harmonic devices required to stabilize an optical frequency comb. The laser was then employed to measure the operating wavelengths and efficiencies of non-linear devices. In addition, an analysis of the radio frequency instruments used to evaluate microwave outputs was conducted to determine whether a digital signal analyzer (oscilloscope) or an analog electronic spectrum analyzer provides more accurate results for optical frequency comb based experiments.</p>

No description available.

No description available.