Toward Early-Warning Detection of Gravitational Waves from Compact Binary Coalescence

Rapid detection of compact binary coalescence (CBC) with a network of advanced gravitational-wave detectors will offer a unique opportunity for multi-messenger astronomy. Prompt detection alerts for the astronomical community might make it possible to observe the onset of electromagnetic emission from (CBC). We demonstrate a computationally practical filtering strategy that could produce early-warning triggers before gravitational radiation from the final merger has arrived at the detectors.Comment: 16 pages, 7 figures, published in ApJ. Reformatted preprint with emulateap

Effects of Multirate Systems on the Statistical Properties of Random Signals

In multirate digital signal processing, we often encounter time-varying linear systems such as decimators, interpolators, and modulators. In many applications, these building blocks are interconnected with linear filters to form more complicated systems. It is often necessary to understand the way in which the statistical behavior of a signal changes as it passes through such systems. While some issues in this context have an obvious answer, the analysis becomes more involved with complicated interconnections. For example, consider this question: if we pass a cyclostationary signal with period K through a fractional sampling rate-changing device (implemented with an interpolator, a nonideal low-pass filter and a decimator), what can we say about the statistical properties of the output? How does the behavior change if the filter is replaced by an ideal low-pass filter? In this paper, we answer questions of this nature. As an application, we consider a new adaptive filtering structure, which is well suited for the identification of band-limited channels. This structure exploits the band-limited nature of the channel, and embeds the adaptive filter into a multirate system. The advantages are that the adaptive filter has a smaller length, and the adaptation as well as the filtering are performed at a lower rate. Using the theory developed in this paper, we show that a matrix adaptive filter (dimension determined by the decimator and interpolator) gives better performance in terms of lower error energy at convergence than a traditional adaptive filter. Even though matrix adaptive filters are, in general, computationally more expensive, they offer a performance bound that can be used as a yardstick to judge more practical "scalar multirate adaptation" schemes

Image interpolation using Shearlet based iterative refinement

This paper proposes an image interpolation algorithm exploiting sparse representation for natural images. It involves three main steps: (a) obtaining an initial estimate of the high resolution image using linear methods like FIR filtering, (b) promoting sparsity in a selected dictionary through iterative thresholding, and (c) extracting high frequency information from the approximation to refine the initial estimate. For the sparse modeling, a shearlet dictionary is chosen to yield a multiscale directional representation. The proposed algorithm is compared to several state-of-the-art methods to assess its objective as well as subjective performance. Compared to the cubic spline interpolation method, an average PSNR gain of around 0.8 dB is observed over a dataset of 200 images

Time-delay interferometric ranging for LISA: Statistical analysis of bias-free ranging using laser noise minimization

Die Laser Interferometer Space Antenna (LISA) ist eine Mission der europäischen Weltraumagentur (ESA) zur Detektion von Gravitationswellen im Frequenzbereich zwischen 10^-4 Hz und 1 Hz. Gravitationswellen induzieren relative Abstandsänderungen, die LISA mithilfe von Laserinterferometrie mit Picometerpräzision misst. Ein großes Problem hierbei ist das Frequenzrauschen der Laser. Um dieses zu unterdrücken, ist es notwendig, mithilfe eines Algorithmus namens TDI (engl. time-delay interferometry), virtuelle Interferometer mit gleichlangen Armen zu konstruieren, wie z.B. das klassische Michelson-Interferometer. In dieser Arbeit untersuchen wir die Performanz von TDI unter realistischen Bedingungen und identifizieren verschiedene Kopplungsmechanismen des Laserfrequenzrauschens. Als erstes betrachten wir die Datenverarbeitung an Bord der Satelliten, die benötigt wird, um die Abtastrate der interferometrischen Messungen zu reduzieren. Hierfür sind Anti-Alias-Filter vorgesehen, die der Faltung von Laserrauschleistung in das Beobachtungsband vorbeugen. Außerdem wirkt sich die Ebenheit der Filter auf die Effektivität von TDI aus (engl. flexing-filtering-effect). Dieser Effekt ist bereits in der Literatur beschrieben und wir demonstrieren in dieser Arbeit die Möglichkeit, ihn mithilfe von Kompensationsfiltern effektiv zu reduzieren. Als zweites betrachten wir Kopplungsmechanismen von Laserfrequenzrauschen im TDI-Algorithmus selbst. Fehler in der Interpolation der interferometrischen Messungen und Ungenauigkeiten in den absoluten Abstandsmessungen zwischen den Satelliten führen ebenfalls zu einer unzureichenden Reduzierung des Laserfrequenzrauschens. Wir beschreiben die oben genannten Kopplungsmechanismen analytisch und validieren die zugrundeliegenden Modelle mithilfe von numerischen Simulationen. Das tiefere Verständnis dieser Residuen ermöglicht es uns, geeignete instrumentelle Parameter zu wählen, die von hoher Relevanz für das Missionsdesign von LISA sind. Des Weiteren beschäftigen wir uns in dieser Arbeit mit der möglichst genauen Bestimmung der absoluten Abständen zwischen den Satelliten, die für den TDI Algorithmus erforderlich sind. Hierfür werden die Abstandsinformationen aus den Seitenbändern und der PRN-Modulation (engl. pseudo-random noise) kombiniert. Wir zeigen, dass die PRN-Messung von systematischen Verzerrungen betroffen ist, die zu Laserrauschresiduen in den TDI-Variablen führen. Um diesen Fehler zu korrigieren, schlagen wir als zusätzliche Abstandsmessung TDI-Ranging (TDI-R) vor. TDI-R ist zwar ungenauer, aber frei von systematischen Verzerrungen und kann daher zur Kalibrierung der PRN-Messungen herangezogen werden. Wir präsentieren in dieser Arbeit eine ausführliche statistische Studie, um die Performanz von TDI-R zu charakterisieren. Dafür formulieren wir die Likelihood-Funktion der interferometrischen Messungen und berechnen die Fisher-Informationsmatrix, um die theoretisch mögliche untere Grenze der Schätzvarianz zu finden. Diese verhält sich invers proportional zur Integrationszeit und dem Verhältnis von Sekundärrauschleistung, die die interferometrische Messung fundamental limitiert, und Laserrauschleistung. Zusätzlich validieren wir die analytische untere Grenze der Schätzvarianz mithilfe von numerischen Simulationen und zeigen damit, dass unsere Implementierung von TDI-R optimal ist. Der entwickelte TDI-R-Algorithmus wird Teil der Datenverarbeitungspipeline sein und Konsistenzprüfungen und Kalibrierung der primären Abstandsmessmethoden ermöglichen.The Laser Interferometer Space Antenna (LISA) is a future ESA-led space-based observatory to explore the gravitational universe in the frequency band between 10^-4 Hz and 1 Hz. LISA implements picometer-precise inter-satellite ranging to measure tiny ripples in spacetime induced by gravitational waves (GWs). However, the single-link measurements are dominated by laser frequency noise, which is about nine orders of magnitude larger than the GW signals. Therefore, in post-processing, the time-delay interferometry (TDI) algorithm is used to synthesize virtual equal-arm interferometers to suppress laser frequency noise. In this work we identify several laser frequency noise coupling channels that limit the performance of TDI. First, the on-board processing, which is used to decimate the sampling rate from tens of megahertz down to the telemetry rate of a few hertz, requires careful design. Appropriate anti-aliasing filters must be implemented to mitigate folding of laser noise power into the observation band. Furthermore, the flatness of these filters is important to limit the impact of the flexing-filtering effect. We demonstrate that this effect can be effectively reduced by using compensation filters on ground. Second, the post-processing delays applied in TDI are subject to interpolation and ranging errors. We study these laser and timing noise residuals analytically and perform simulations to validate the models numerically. Our findings have direct implications for the design of the LISA instrument as we identify the instrumental parameters that are essential for successful laser noise suppression and provide methods for designing appropriate filters for the on-board processing. In addition, we discuss a dedicated ranging processing pipeline that produces high-precision range estimates that are the input for TDI by combining the sideband and pseudo-random noise (PRN) ranges. We show in this thesis that biases in the PRN measurements limit the laser noise suppression performance. Therefore, we propose time-delay interferometric ranging (TDI-R) as a third ranging sensor to estimate bias-free ranges that can be used to calibrate the biases in the PRN measurements. We present a thorough statistical study of TDI-R to evaluate its performance. Therefore, we formulate the likelihood function of the interferometric data and use the Fisher information formalism to find a lower bound on the estimation variance of the inter-satellite ranges. We find that the ranging uncertainty is proportional to the inverse of the integration time and the ratio of secondary noise power, that limits the interferometric readout, to the laser noise power. To validate our findings we implement prototype TDI-R pipelines and perform numerical simulations. We show that we are able to formulate optimal estimators of the unbiased range that reach the Cramér-Rao lower bound previously expressed analytically. The developed TDI-R pipeline will be integrated into the ranging processing pipeline to perform consistency checks and ensure well-calibrated inter-satellite ranges

Design Of Polynomial-based Filters For Continuously Variable Sample Rate Conversion With Applications In Synthetic Instrumentati

In this work, the design and application of Polynomial-Based Filters (PBF) for continuously variable Sample Rate Conversion (SRC) is studied. The major contributions of this work are summarized as follows. First, an explicit formula for the Fourier Transform of both a symmetrical and nonsymmetrical PBF impulse response with variable basis function coefficients is derived. In the literature only one explicit formula is given, and that for a symmetrical even length filter with fixed basis function coefficients. The frequency domain optimization of PBFs via linear programming has been proposed in the literature, however, the algorithm was not detailed nor were explicit formulas derived. In this contribution, a minimax optimization procedure is derived for the frequency domain optimization of a PBF with time-domain constraints. Explicit formulas are given for direct input to a linear programming routine. Additionally, accompanying Matlab code implementing this optimization in terms of the derived formulas is given in the appendix. In the literature, it has been pointed out that the frequency response of the Continuous-Time (CT) filter decays as frequency goes to infinity. It has also been observed that when implemented in SRC, the CT filter is sampled resulting in CT frequency response aliasing. Thus, for example, the stopband sidelobes of the Discrete-Time (DT) implementation rise above the CT designed level. Building on these observations, it is shown how the rolloff rate of the frequency response of a PBF can be adjusted by adding continuous derivatives to the impulse response. This is of great advantage, especially when the PBF is used for decimation as the aliasing band attenuation can be made to increase with frequency. It is shown how this technique can be used to dramatically reduce the effect of alias build up in the passband. In addition, it is shown that as the number of continuous derivatives of the PBF increases the resulting DT implementation more closely matches the Continuous-Time (CT) design. When implemented for SRC, samples from a PBF impulse response are computed by evaluating the polynomials using a so-called fractional interval, µ. In the literature, the effect of quantizing µ on the frequency response of the PBF has been studied. Formulas have been derived to determine the number of bits required to keep frequency response distortion below prescribed bounds. Elsewhere, a formula has been given to compute the number of bits required to represent µ to obtain a given SRC accuracy for rational factor SRC. In this contribution, it is shown how these two apparently competing requirements are quite independent. In fact, it is shown that the wordlength required for SRC accuracy need only be kept in the µ generator which is a single accumulator. The output of the µ generator may then be truncated prior to polynomial evaluation. This results in significant computational savings, as polynomial evaluation can require several multiplications and additions. Under the heading of applications, a new Wideband Digital Downconverter (WDDC) for Synthetic Instruments (SI) is introduced. DDCs first tune to a signal\u27s center frequency using a numerically controlled oscillator and mixer, and then zoom-in to the bandwidth of interest using SRC. The SRC is required to produce continuously variable output sample rates from a fixed input sample rate over a large range. Current implementations accomplish this using a pre-filter, an arbitrary factor resampler, and integer decimation filters. In this contribution, the SRC of the WDDC is simplified reducing the computational requirements to a factor of three or more. In addition to this, it is shown how this system can be used to develop a novel computationally efficient FFT-based spectrum analyzer with continuously variable frequency spans. Finally, after giving the theoretical foundation, a real Field Programmable Gate Array (FPGA) implementation of a novel Arbitrary Waveform Generator (AWG) is presented. The new approach uses a fixed Digital-to-Analog Converter (DAC) sample clock in combination with an arbitrary factor interpolator. Waveforms created at any sample rate are interpolated to the fixed DAC sample rate in real-time. As a result, the additional lower performance analog hardware required in current approaches, namely, multiple reconstruction filters and/or additional sample clocks, is avoided. Measured results are given confirming the performance of the system predicted by the theoretical design and simulation

Design and implementation of a downlink MC-CDMA receiver

Cette thèse présente une étude d'un système complet de transmission en liaison descendante utilisant la technologie multi-porteuse avec l'accès multiple par division de code (Multi-Carrier Code Division Multiple Access, MC-CDMA). L'étude inclut la synchronisation et l'estimation du canal pour un système MC-CDMA en liaison descendante ainsi que l'implémentation sur puce FPGA d'un récepteur MC-CDMA en liaison descendante en bande de base. Le MC-CDMA est une combinaison de la technique de multiplexage par fréquence orthogonale (Orthogonal Frequency Division Multiplexing, OFDM) et de l'accès multiple par répartition de code (CDMA), et ce dans le but d'intégrer les deux technologies. Le système MC-CDMA est conçu pour fonctionner à l'intérieur de la contrainte d'une bande de fréquence de 5 MHz pour les modèles de canaux intérieur/extérieur pédestre et véhiculaire tel que décrit par le "Third Genaration Partnership Project" (3GPP). La composante OFDM du système MC-CDMA a été simulée en utilisant le logiciel MATLAB dans le but d'obtenir des paramètres de base. Des codes orthogonaux à facteur d'étalement variable (OVSF) de longueur 8 ont été choisis comme codes d'étalement pour notre système MC-CDMA. Ceci permet de supporter des taux de transmission maximum jusquà 20.6 Mbps et 22.875 Mbps (données non codées, pleine charge de 8 utilisateurs) pour les canaux intérieur/extérieur pédestre et véhiculaire, respectivement. Une étude analytique des expressions de taux d'erreur binaire pour le MC-CDMA dans un canal multivoies de Rayleigh a été réalisée dans le but d'évaluer rapidement et de façon précise les performances. Des techniques d'estimation de canal basées sur les décisions antérieures ont été étudiées afin d'améliorer encore plus les performances de taux d'erreur binaire du système MC-CDMA en liaison descendante. L'estimateur de canal basé sur les décisions antérieures et utilisant le critère de l'erreur quadratique minimale linéaire avec une matrice' de corrélation du canal de taille 64 x 64 a été choisi comme étant un bon compromis entre la performance et la complexité pour une implementation sur puce FPGA. Une nouvelle séquence d'apprentissage a été conçue pour le récepteur dans la configuration intérieur/extérieur pédestre dans le but d'estimer de façon grossière le temps de synchronisation et le décalage fréquentiel fractionnaire de la porteuse dans le domaine du temps. Les estimations fines du temps de synchronisation et du décalage fréquentiel de la porteuse ont été effectués dans le domaine des fréquences à l'aide de sous-porteuses pilotes. Un récepteur en liaison descendante MC-CDMA complet pour le canal intérieur /extérieur pédestre avec les synchronisations en temps et en fréquence en boucle fermée a été simulé avant de procéder à l'implémentation matérielle. Le récepteur en liaison descendante en bande de base pour le canal intérieur/extérieur pédestre a été implémenté sur un système de développement fabriqué par la compagnie Nallatech et utilisant le circuit XtremeDSP de Xilinx. Un transmetteur compatible avec le système de réception a également été réalisé. Des tests fonctionnels du récepteur ont été effectués dans un environnement sans fil statique de laboratoire. Un environnement de test plus dynamique, incluant la mobilité du transmetteur, du récepteur ou des éléments dispersifs, aurait été souhaitable, mais n'a pu être réalisé étant donné les difficultés logistiques inhérentes. Les taux d'erreur binaire mesurés avec différents nombres d'usagers actifs et différentes modulations sont proches des simulations sur ordinateurs pour un canal avec bruit blanc gaussien additif

Biorthogonal partners and applications

Two digital filters H(z) and F(z) are said to be biorthogonal partners of each other if their cascade H(z)F(z) satisfies the Nyquist or zero-crossing property. Biorthogonal partners arise in many different contexts such as filterbank theory, exact and least squares digital interpolation, and multiresolution theory. They also play a central role in the theory of equalization, especially, fractionally spaced equalizers in digital communications. We first develop several theoretical properties of biorthogonal partners. We also develop conditions for the existence of biorthogonal partners and FIR biorthogonal pairs and establish the connections to the Riesz basis property. We then explain how these results play a role in many of the above-mentioned applications

Causal Instrument Corrections for Short-Period and Broadband Seismometers

Of all the filters applied to recordings of seismic waves, which include source, path, and site effects, the one we know most precisely is the instrument filter. Therefore, it behooves seismologists to accurately remove the effect of the instrument from raw seismograms. Applying instrument corrections allows analysis of the seismogram in terms of physical units (e.g., displacement or particle velocity of the Earth’s surface) instead of the output of the instrument (e.g., digital counts). The instrument correction can be considered the most fundamental processing step in seismology since it relates the raw data to an observable quantity of interest to seismologists. Complicating matters is the fact that, in practice, the term “instrument correction” refers to more than simply the seismometer. The instrument correction compensates for the complete recording system including the seismometer, telemetry, digitizer, and any anti‐alias filters