78 research outputs found

    High Fidelity Satellite Navigation Receiver Front-End for Advanced Signal Quality Monitoring and Authentication

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    Over the last several years, interest in utilizing foreign satellite timing and navigation (satnav) signals to augment GPS has grown. Doing so is not without risks; foreign satnav signals must be vetted and determined to be trustworthy before use in military applications. Advanced signal quality monitoring methods can help to ensure that only authentic and reliable satnav signals are utilized. To effectively monitor and authenticate signals, the front-end must impress as little distortions upon the received signal as possible. The purpose of this study is to design, fabricate, and test the performance of a high-fidelity satnav receiver front-end for advanced monitoring of foreign and domestic space vehicle signals

    Development and Calibration of New 3-D Vector VSP Imaging Technology: Vinton Salt Dome, LA

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    Novel Digital Alias-Free Signal Processing Approaches to FIR Filtering Estimation

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    This thesis aims at developing a new methodology of filtering continuous-time bandlimited signals and piecewise-continuous signals from their discrete-time samples. Unlike the existing state-of-the-art filters, my filters are not adversely affected by aliasing, allowing the designers to flexibly select the sampling rates of the processed signal to reach the required accuracy of signal filtering rather than meeting stiff and often demanding constraints imposed by the classical theory of digital signal processing (DSP). The impact of this thesis is cost reduction of alias-free sampling, filtering and other digital processing blocks, particularly when the processed signals have sparse and unknown spectral support. Novel approaches are proposed which can mitigate the negative effects of aliasing, thanks to the use of nonuniform random/pseudorandom sampling and processing algorithms. As such, the proposed approaches belong to the family of digital alias-free signal processing (DASP). Namely, three main approaches are considered: total random (ToRa), stratified (StSa) and antithetical stratified (AnSt) random sampling techniques. First, I introduce a finite impulse response (FIR) filter estimator for each of the three considered techniques. In addition, a generalised estimator that encompasses the three filter estimators is also proposed. Then, statistical properties of all estimators are investigated to assess their quality. Properties such as expected value, bias, variance, convergence rate, and consistency are all inspected and unveiled. Moreover, closed-form mathematical expression is devised for the variance of each single estimator. Furthermore, quality assessment of the proposed estimators is examined in two main cases related to the smoothness status of the filter convolution’s integrand function, \u1d454(\u1d461,\u1d70f)∶=\u1d465(\u1d70f)ℎ(\u1d461−\u1d70f), and its first two derivatives. The first main case is continuous and differentiable functions \u1d454(\u1d461,\u1d70f), \u1d454′(\u1d461,\u1d70f), and \u1d454′′(\u1d461,\u1d70f). Whereas in the second main case, I cover all possible instances where some/all of such functions are piecewise-continuous and involving a finite number of bounded discontinuities. Primarily obtained results prove that all considered filter estimators are unbiassed and consistent. Hence, variances of the estimators converge to zero after certain number of sample points. However, the convergence rate depends on the selected estimator and which case of smoothness is being considered. In the first case (i.e. continuous \u1d454(\u1d461,\u1d70f) and its derivatives), ToRa, StSa and AnSt filter estimators converge uniformly at rates of \u1d441−1, \u1d441−3, and \u1d441−5 respectively, where 2\u1d441 is the total number of sample points. More interestingly, in the second main case, the convergence rates of StSa and AnSt estimators are maintained even if there are some discontinuities in the first-order derivative (FOD) with respect to \u1d70f of \u1d454(\u1d461,\u1d70f) (for StSa estimator) or in the second-order derivative (SOD) with respect to \u1d70f of \u1d454(\u1d461,\u1d70f) (for AnSt). Whereas these rates drop to \u1d441−2 and \u1d441−4 (for StSa and AnSt, respectively) if the zero-order derivative (ZOD) (for StSa) and FOD (for AnSt) are piecewise-continuous. Finally, if the ZOD of \u1d454(\u1d461,\u1d70f) is piecewise-continuous, then the uniform convergence rate of the AnSt estimator further drops to \u1d441−2. For practical reasons, I also introduce the utilisation of the three estimators in a special situation where the input signal is pseudorandomly sampled from otherwise uniform and dense grid. An FIR filter model with an oversampled finite-duration impulse response, timely aligned with the grid, is proposed and meant to be stored in a lookup table of the implemented filter’s memory to save processing time. Then, a synchronised convolution sum operation is conducted to estimate the filter output. Finally, a new unequally spaced Lagrange interpolation-based rule is proposed. The so-called composite 3-nonuniform-sample (C3NS) rule is employed to estimate area under the curve (AUC) of an integrand function rather than the simple Rectangular rule. I then carry out comparisons for the convergence rates of different estimators based on the two interpolation rules. The proposed C3NS estimator outperforms other Rectangular rule estimators on the expense of higher computational complexity. Of course, this extra cost could only be justifiable for some specific applications where more accurate estimation is required

    Potential use of the Undersampling Technique in the Acquisition of Nuclear Magnetic Resonance Signals

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    This communication reviews the use of undersampling techniques to acquire NMR signals. Undersampling transforms bandpass free induction decay (FID) signals, centered at high frequencies, into lowpass signals or bandpass signals centered at much lower frequencies. Consequently, the analog electronic stages that perform the demodulation can be eliminated, gaining in stability and reducing the phase distortion while maintaining an equivalent or better signal to noise ratio when an adequate sampling rate is chosen. The technique has been tested on a BRUKER BIOSPEC BMT 47/40, and the results show that undersampling could be used to process NMR and MRI signals, extending the range of applications of the ‘digital radio’ techniques to NMR and MRI apparatusPublicad

    Bistatic SAR data acquisition and processing using SABRINA-X, with TerraSAR-X as the opportunity transmitter

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    This thesis investigates the acquisition and processing of Bistatic SAR data using SABRINA-X, and with TerraSAR-X as the transmitter of opportunity. SABRINA-X is an X-band receiver system that has been recently designed at the UPC Remote-Sensing Laboratory, while TerraSARX is a German satellite for SAR-based active remote-sensing. Prior to the particular case of acquiring TerraSAR-X signals, the hardware aspects of SABRINAX have been investigated further, and improved as necessary (or suggested for up-gradation in future). Two successful data acquisitions have been carried out, to obtain bistatic SAR images of the Barcelona harbor, with the receiver set-up at the close-by Montjuïc hill. Each acquisition campaign necessitated an accurate prediction of the satellite overpass time and precise orientation of the antennas to acquire the direct signal from the satellite and the backscattered signals off the viewed terrain. The thesis also investigates the characteristics of the acquired signals, which is critical as regards the subsequent processing for imaging and interferometric applications. The hardware limitations, combined with ‘off-nominal’ transmissions of the satellite, necessitate improved range processing of the acquired signals. The thesis expounds the possible range compression techniques, and suggests ways for improved compression, thereby improving the quality of the subsequently processed images

    New Directions in Subband Coding

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    Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processo

    Comparison of Simulation Methods of Single and Multi-Bit Continuous Time Sigma Delta Modulators

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    Continuous time Sigma Delta Modulators (CT ΣΔMs) are a type of analog to digital converter (ADC) that are used in mixed signal systems to convert analog signals into digital signals. ADCs typically require antialiasing filter; however antialiasing filters are inherent in CT ΣΔMs, and therefore they require less circuitry and less power than other ADC architectures that require separate antialiasing filters. As a result, CT ΣΔM ADC architectures are preferred in many mixed signal electronic applications. Because of the mixed signal nature of CT ΣΔMs, they can be difficult to simulate. In this thesis, various methods for simulating single-bit and multi-bit CT ΣΔMs are developed and these simulations include the bilinear transform or trapezoidal integration, impulse invariance transform, midpoint integration, Simpson’s rule, delta transform or Euler’s forward integration rule and Simulink modeling. These methods are compared with respect to speed which is given by the total simulation time, accuracy which is given by the signal to noise ratio (SNR) value and the simplicity of the simulation method. The CT ΣΔMs have been extended from first order up to fifth order with one, two and three bit quantizers. Also, the frequency domain analysis is done for all the orders of CT ΣΔMs. The results show that the numerical integration methods are more accurate and faster than Simulink. However, CT ΣΔM simulations using Simulink are simpler because of the availability of the required blocks in Simulink. The overall comparison shows that the numerical integration methods can perform better than Simulink models. The frequency domain analysis proves the correctness of the use of numerical integration methods for CT ΣΔM simulations

    Contribution to the design of continuous -time Sigma - Delta Modulators based on time delay elements

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    The research carried out in this thesis is focused in the development of a new class of data converters for digital radio. There are two main architectures for communication receivers which perform a digital demodulation. One of them is based on analog demodulation to the base band and digitization of the I/Q components. Another option is to digitize the band pass signal at the output of the IF stage using a bandpass Sigma-Delta modulator. Bandpass Sigma- Delta modulators can be implemented with discrete-time circuits, using switched capacitors or continuous-time circuits. The main innovation introduced in this work is the use of passive transmission lines in the loop filter of a bandpass continuous-time Sigma-Delta modulator instead of the conventional solution with gm-C or LC resonators. As long as transmission lines are used as replacement of a LC resonator in RF technology, it seems compelling that transmission lines could improve bandpass continuous-time Sigma-Delta modulators. The analysis of a Sigma- Delta modulator using distributed resonators has led to a completely new family of Sigma- Delta modulators which possess properties inherited both from continuous-time and discretetime Sigma-Delta modulators. In this thesis we present the basic theory and the practical design trade-offs of this new family of Sigma-Delta modulators. Three demonstration chips have been implemented to validate the theoretical developments. The first two are a proof of concept of the application of transmission lines to build lowpass and bandpass modulators. The third chip summarizes all the contributions of the thesis. It consists of a transmission line Sigma-Delta modulator which combines subsampling techniques, a mismatch insensitive circuitry and a quadrature architecture to implement the IF to digital stage of a receiver
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