Studies on Implementation of . . . High Throughput and Low Power Consumption

In this thesis we discuss design and implementation of frequency selective digital filters with high throughput and low power consumption. The thesis includes proposed arithmetic transformations of lattice wave digital filters that aim at increasing the throughput and reduce the power consumption of the filter implementation. The thesis also includes two case studies where digital filters with high throughput and low power consumption are required. A method for obtaining high throughput as well as reduced power consumption of digital filters is arithmetic transformation of the filter structure. In this thesis arithmetic transformations of first- and second-order Richards’ allpass sections composed by symmetric two-port adaptors and implemented using carry-save arithmetic are proposed. Such filter sections can be used for implementation of lattice wave digital filters and bireciprocal lattice wave digital filters. The latter structures are efficient for implementation of interpolators and decimators by factors of two. Th

Adaptive Interference Mitigation in GPS Receivers

Satellite navigation systems (GNSS) are among the most complex radio-navigation systems, providing positioning, navigation, and timing (PNT) information. A growing number of public sector and commercial applications rely on the GNSS PNT service to support business growth, technical development, and the day-to-day operation of technology and socioeconomic systems. As GNSS signals have inherent limitations, they are highly vulnerable to intentional and unintentional interference. GNSS signals have spectral power densities far below ambient thermal noise. Consequently, GNSS receivers must meet high standards of reliability and integrity to be used within a broad spectrum of applications. GNSS receivers must employ effective interference mitigation techniques to ensure robust, accurate, and reliable PNT service. This research aims to evaluate the effectiveness of the Adaptive Notch Filter (ANF), a precorrelation mitigation technique that can be used to excise Continuous Wave Interference (CWI), hop-frequency and chirp-type interferences from GPS L1 signals. To mitigate unwanted interference, state-of-the-art ANFs typically adjust a single parameter, the notch centre frequency, and zeros are constrained extremely close to unity. Because of this, the notch centre frequency converges slowly to the target frequency. During this slow converge period, interference leaks into the acquisition block, thus sabotaging the operation of the acquisition block. Furthermore, if the CWI continuously hops within the GPS L1 in-band region, the subsequent interference frequency is locked onto after a delay, which means constant interference occurs in the receiver throughout the delay period. This research contributes to the field of interference mitigation at GNSS's receiver end using adaptive signal processing, predominately for GPS. This research can be divided into three stages. I first designed, modelled and developed a Simulink-based GPS L1 signal simulator, providing a homogenous test signal for existing and proposed interference mitigation algorithms. Simulink-based GPS L1 signal simulator provided great flexibility to change various parameters to generate GPS L1 signal under different conditions, e.g. Doppler Shift, code phase delay and amount of propagation degradation. Furthermore, I modelled three acquisition schemes for GPS signals and tested GPS L1 signals acquisition via coherent and non-coherent integration methods. As a next step, I modelled different types of interference signals precisely and implemented and evaluated existing adaptive notch filters in MATLAB in terms of Carrier to Noise Density (\u1d436/\u1d4410), Signal to Noise Ratio (SNR), Peak Degradation Metric, and Mean Square Error (MSE) at the output of the acquisition module in order to create benchmarks. Finally, I designed, developed and implemented a novel algorithm that simultaneously adapts both coefficients in lattice-based ANF. Mathematically, I derived the full-gradient term for the notch's bandwidth parameter adaptation and developed a framework for simultaneously adapting both coefficients of a lattice-based adaptive notch filter. I evaluated the performance of existing and proposed interference mitigation techniques under different types of interference signals. Moreover, I critically analysed different internal signals within the ANF structure in order to develop a new threshold parameter that resets the notch bandwidth at the start of each subsequent interference frequency. As a result, I further reduce the complexity of the structural implementation of lattice-based ANF, allowing for efficient hardware realisation and lower computational costs. It is concluded from extensive simulation results that the proposed fully adaptive lattice-based provides better interference mitigation performance and superior convergence properties to target frequency compared to traditional ANF algorithms. It is demonstrated that by employing the proposed algorithm, a receiver is able to operate with a higher dynamic range of JNR than is possible with existing methods. This research also presents the design and MATLAB implementation of a parameterisable Complex Adaptive Notch Filer (CANF). Present analysis on higher order CANF for detecting and mitigating various types of interference for complex baseband GPS L1 signals. In the end, further research was conducted to suppress interference in the GPS L1 signal by exploiting autocorrelation properties and discarding some portion of the main lobe of the GPS L1 signal. It is shown that by removing 30% spectrum of the main lobe, either from left, right, or centre, the GPS L1 signal is still acquirable

A Study on Efficient Receiver Design for UWA Communication System

Underwater Acoustic Channels are fast varying channel according to environmental conditions and exhibit strong random fluctuations in amplitude as well as phase due to reflection, refraction, and diffraction. Due to these highly space, time and frequency dependent channel characteristics, it is very difficult to establish reliable and long-range underwater acoustic communication. In this project, channel modeling has been done showing the different channel characteristics of underwater and their dependencies on frequency, temperature, pressure, salinity etc. Also, it has been shown through some theoretical and practical results that the nakagami fading is the best suitable generalized fading to be used in underwater. In this research work various techniques such as equalization, pilot based OFDM and LDPC Coding has also been done to mitigate the channel fading effect and to improve the performance. An adaptive equalizer has been implemented through three different algorithms LMS, NLMS and RLS for linear as well as non-linear channels to mitigate ISI and, their convergence characteristics along with bit error rate performance has been compared. Two types of pilot insertion, block and Comb type has also been done while implementing OFDM. Block type pilot based OFDM is suitable for slow fading and comb type pilot based OFDM is suitable for a fast fading channel. As in underwater, both types of fading exist, hence, lattice type pilot based OFDM is the best suitable for underwater acoustic communication. LDPC channel coding through which almost Shannon capacity performance can be achieved; has also been implemented taking nakagami channel fading. Bit error rate performance has been compared for different LDPC decoding techniques and for different code rate

Modular decomposition techniques for stored-logic digital filters

Digital filtering is an important signal processing technique whose theory is now well established. At present, however, there are no well-defined and systematic methods available for realising digital filters in hardware. This project aims to develop such methods which are general and technology independent, and adopts a systems and sub-systems design philosophy. The realisation problem is approached in a new way using concepts from finite-automata theory and implementing complete digital filter sections as stored-logic units. Two methods are introduced and developed. [Continues.

FPGA Implementation of Data Flow Graphs for Digital Signal Processing Applications

A rapid growth in digital signal processing applications has increased the requirement for high-speed digital systems. Multiprocessor systems are the best choice for these applications. A prior sequence of operations should be applied to the operations that described the nature of these applications before hardware implementation is produced. These operations should be scheduled and hardware allocated. This paper proposes a new scheduling technique for digital signal processing (DSP) applications has been represented by data flow graphs (DFGs). In addition, hardware allocation is implemented in the form of embedded system. A proposed scheduling technique also achieves the optimal scheduling of a DFG at design time. The optimality criteria considered in this algorithm are the maximum throughput within the available hardware resources. The maximum throughput is achieved by arranging the DFG nodes according to their inter-related data dependencies. Then, two nodes can be clustered into one compound task to reduce the overall execution time by minimizing the number of tasks to be executed that minimizing the number of cycles to execute them. Then each task is presented in form of instruction to be executed in the hardware system. A hardware system is composed of one or multiple homogenous pipelined processing elements and it is designed to meet the maximum-rate schedule.  Two implementations are proposed of the system architecture according to the number of the processing elements, namely:  the serial system and the parallel system. The serial system comprises one processing element where all tasks are processed sequentially, whilst the parallel system has four processing elements to execute tasks concurrently. These systems consist mainly of seven units: central shared memory, state table, multiway function unit buffer, execution array, processing element/s, instruction buffer and the address generation unit. The hardware components were built on an FPGA chip using Verilog HDL. In synthesis results, the parallel system has better system performance by 25.5% than the serial system. While the serial system requires smaller area size, which described by the number of slice registers and the number of the slice lookup tables (LUTs) than the parallel one. The relationship between the number of instructions that are executed in both systems, and the system area and the system performance that presented by system frequency, are studied. By increasing memories size in both systems, the system performance isn’t affected as in a serial system, and it is slightly decreased as the parallel system by 1.5% to 4.5%. In terms of the systems area, both serial system area and parallel system area are increased and in some cases are doubled. The proposed scheduling technique is shown to outperform the retaining technique, which we have chosen to compare with.  The serial system has better performance by 19.3% higher system frequency than a retiming technique. And the parallel system also outperforms the retaining technique by 51.2% higher system frequency in synthesis results

Design and Implementation of a Re-Configurable Arbitrary Signal Generator and Radio Frequency Spectrum Analyser

This research is focused on the design, simulation and implementation of a reconfigurable arbitrary signal generator and the design, simulation and implementation of a radio frequency spectrum analyser based on digital signal processing. Until recently, Application Specific Integrated Circuits (ASICs) were used to produce high performance re-configurable function and arbitrary waveform generators with comprehensive modulation capabilities. However, that situation is now changing with the availability of advanced but low cost Field Programmable Gate Arrays (FPGAs), which could be used as an alternative to ASICs in these applications. The availability of high performance FPGA families opens up the opportunity to compete with ASIC solutions at a fraction of the development cost of an ASIC solution. A fast digital signal processing algorithm for digital waveform generation, using primarily but not limited to Direct Digital Synthesis (DDS) technologies, developed and implemented in a field-configurable logic, with control provided by an embedded microprocessor replacing a high cost ASIC design appeared to be a very attractive concept. This research demonstrates that such a concept is feasible in its entirety. A fully functional, low-complexity, low cost, pulse, Gaussian white noise and DDS based function and arbitrary waveform generator, capable of being amplitude, frequency and phase modulated by an internally generated or external modulating signal was implemented in a low-cost FPGA. The FPGA also included the capabilities to perform pulse width modulation and pulse delay modulation on pulse waveforms. Algorithms to up-convert the sampling rate of the external modulating signal using Cascaded Integrator Comb (CIC) filters and using interpolation method were analysed. Both solutions were implemented to compare their hardware complexities. Analysis of generating noise with user-defined distribution is presented. The ability of triggering the generator by an internally generated or an external event to generate a burst of waveforms where the time between the trigger signal and waveform output is fixed was also implemented in the FPGA. Finally, design of interface to a microprocessor to provide control of the versatile waveform generator was also included in the FPGA. This thesis summarises the literature, design considerations, simulation and implementation of the generator design. The second part of the research is focused on radio frequency spectrum analysis based on digital signal processing. Most existing spectrum analysers are analogue in nature and their complexity increases with frequency. Therefore, the possibility of using digital techniques for spectrum analysis was considered. The aim was to come up with digital system architecture for spectrum analysis and to develop and implement the new approach on a suitable digital platform. This thesis analyses the current literature on shifting algorithms to remove spurious responses and highlights its drawbacks. This thesis also analyses existing literature on quadrature receivers and presents novel adaptation of the existing architectures for application in spectrum analysis. A wide band spectrum analyser receiver with compensation for gain and phase imbalances in the Radio Frequency (RF) input range, as well as compensation for gain and phase imbalances within the Intermediate Frequency (IF) pass band complete with Resolution Band Width (RBW) filtering, Video Band Width (VBW) filtering and amplitude detection was implemented in a low cost FPGA. The ability to extract the modulating signal from a frequency or amplitude modulated RF signal was also implemented. The same family of FPGA used in the generator design was chosen to be the digital platform for this design. This research makes arguments for the new architecture and then summarises the literature, design considerations, simulation and implementation of the new digital algorithm for the radio frequency spectrum analyser