Eigenvector-based multidimensional frequency estimation : identifiability, performance, and applications.

Multidimensional frequency estimation is a classic signal processing problem that has versatile applications in sensor array processing and wireless communications. Eigenvalue-based two-dimensional (2-D) and N -dimensional ( N -D) frequency estimation algorithms have been well documented, however, these algorithms suffer from limited identifiability and demanding computations. This dissertation develops a framework on eigenvector-based N -D frequency estimation, which contains several novel algorithms that estimate a structural matrix from eigenvectors and then resolve the N -D frequencies by dividing the elements of the structural matrix. Compared to the existing eigenvalue-based algorithms, these eigenvector-based algorithms can achieve automatic pairing without an extra frequency pairing step, and tins the computational complexity is reduced. The identifiability, performance, and complexity of these algorithms are also systematically studied. Based on this study, the most relaxed identifiability condition for the N -D frequency estimation problem is given and an effective approach using optimized weighting factors to improve the performance of frequency estimation is developed. These results are applied in wireless communication for time-varying multipath channel estimation and prediction, as well as for joint 2-D Direction-of-arrival (DOA) tracking of multiple moving targets

R-dimensional ESPRIT-type algorithms for strictly second-order non-circular sources and their performance analysis

High-resolution parameter estimation algorithms designed to exploit the prior knowledge about incident signals from strictly second-order (SO) non-circular (NC) sources allow for a lower estimation error and can resolve twice as many sources. In this paper, we derive the R-D NC Standard ESPRIT and the R-D NC Unitary ESPRIT algorithms that provide a significantly better performance compared to their original versions for arbitrary source signals. They are applicable to shift-invariant R-D antenna arrays and do not require a centrosymmetric array structure. Moreover, we present a first-order asymptotic performance analysis of the proposed algorithms, which is based on the error in the signal subspace estimate arising from the noise perturbation. The derived expressions for the resulting parameter estimation error are explicit in the noise realizations and asymptotic in the effective signal-to-noise ratio (SNR), i.e., the results become exact for either high SNRs or a large sample size. We also provide mean squared error (MSE) expressions, where only the assumptions of a zero mean and finite SO moments of the noise are required, but no assumptions about its statistics are necessary. As a main result, we analytically prove that the asymptotic performance of both R-D NC ESPRIT-type algorithms is identical in the high effective SNR regime. Finally, a case study shows that no improvement from strictly non-circular sources can be achieved in the special case of a single source.Comment: accepted at IEEE Transactions on Signal Processing, 15 pages, 6 figure

State-Space Approaches to Ultra-Wideband Doppler Processing

National security needs dictate the development of new radar systems capable of identifying and tracking exoatmospheric threats to aid our defense. These new radar systems feature reduced noise floors, electronic beam steering, and ultra-wide bandwidths, all of which facilitate threat discrimination. However, in order to identify missile attributes such as RF reflectivity, distance, and velocity, many existing processing algorithms rely upon narrow bandwidth assumptions that break down with increased signal bandwidth. We present a fresh investigation into these algorithms for removing bandwidth limitations and propose novel state-space and direct-data factoring formulations such as * the multidimensional extension to the Eigensystem Realization Algorithm, * employing state-space models in place of interpolation to obtain a form which admits a separation and isolation of solution components, * and side-stepping the joint diagonalization of state transition matrices, which commonly plagues methods like multidimensional ESPRIT. We then benchmark our approaches and relate the outcomes to the Cramer-Rao bound for the case of one and two adjacent reflectors to validate their conceptual design and identify those techniques that compare favorably to or improve upon existing practices

Hybrid Time and Time-Frequency Blind Source Separation Towards Ambient System Identi cation of Structures

Blind source separation methods such as independent component analysis (ICA) and second order blind identification (SOBI) have shown considerable potential in the area of ambient vibration system identification. The objective of these methods is to separate the modal responses, or sources, from the measured output responses, without the knowledge of excitation. Several frequency domain and time domain methods have been proposed and successfully implemented in the literature. Whereas frequency-domain methods pose several challenges typical of dealing with signals in the frequency-domain, popular time-domain methods such as NExT/ERA and SSI pose limitations in dealing with noise, low sensor density, modes having low energy content, or in dealing with systems having closely-spaced modes, such as those found in structures with passive energy dissipation devices, for example, tuned mass dampers.Motivated by these challenges, the current research focuses on developing methods to address the problem of separability of sources with low energy content, closely-spaced modes, and under-determined blind identification, that is, when the number of response measurements is less than the number of sources. These methods, requiring the time and frequency diversities of the measured outputs, are referred to as hybrid time and time-frequency source separation methods. The hybrid methods are classified into two categories. In the first one, the basic principles of modified SOBI are extended using the stationary wavelet transform (SWT) in order to improve the separability of sources, thereby improving the quality of identification. In the second category, empirical mode decomposition is employed to extract the intrinsic mode functions from measurements, followed by an estimation of the mode shape matrix using iterative and/or non iterative procedures within the framework of modified-SOBI. Both experimental and large-scale structural simulation results are included to demonstrate the applicability of these hybrid approaches to structural system identification problems

Algorithm and architecture for simultaneous diagonalization of matrices applied to subspace-based speech enhancement

This thesis presents algorithm and architecture for simultaneous diagonalization of matrices. As an example, a subspace-based speech enhancement problem is considered, where in the covariance matrices of the speech and noise are diagonalized simultaneously. In order to compare the system performance of the proposed algorithm, objective measurements of speech enhancement is shown in terms of the signal to noise ratio and mean bark spectral distortion at various noise levels. In addition, an innovative subband analysis technique for subspace-based time-domain constrained speech enhancement technique is proposed. The proposed technique analyses the signal in its subbands to build accurate estimates of the covariance matrices of speech and noise, exploiting the inherent low varying characteristics of speech and noise signals in narrow bands. The subband approach also decreases the computation time by reducing the order of the matrices to be simultaneously diagonalized. Simulation results indicate that the proposed technique performs well under extreme low signal-to-noise-ratio conditions. Further, an architecture is proposed to implement the simultaneous diagonalization scheme. The architecture is implemented on an FPGA primarily to compare the performance measures on hardware and the feasibility of the speech enhancement algorithm in terms of resource utilization, throughput, etc. A Xilinx FPGA is targeted for implementation. FPGA resource utilization re-enforces on the practicability of the design. Also a projection of the design feasibility for an ASIC implementation in terms of transistor count only is include

Learning, Inference, and Unmixing of Weak, Structured Signals in Noise

In this thesis, we study two methods that can be used to learn, infer, and unmix weak, structured signals in noise: the Dynamic Mode Decomposition algorithm and the sparse Principal Component Analysis problem. Both problems take as input samples of a multivariate signal that is corrupted by noise, and produce a set of structured signals. We present performance guarantees for each algorithm and validate our findings with numerical simulations. First, we study the Dynamic Mode Decomposition (DMD) algorithm. We demonstrate that DMD can be used to solve the source separation problem. That is, we apply DMD to a data matrix whose rows are linearly independent, additive mixtures of latent time series. We show that when the latent time series are uncorrelated at a lag of one time-step then the recovered dynamic modes will approximate the columns of the mixing matrix. That is, DMD unmixes linearly mixed sources that have a particular correlation structure. We next broaden our analysis beyond the noise-free, fully observed data setting. We study the DMD algorithm with a truncated-SVD denoising step, and present recovery guarantees for both the noisy data and missing data settings. We also present some preliminary characterizations of DMD performed directly on noisy data. We end with some complementary perspectives on DMD, including an optimization-based formulation. Second, we study the sparse Principal Component Analysis (PCA) problem. We demonstrate that the sparse inference problem can be viewed in a variable selection framework and analyze the performance of various decision statistics. A major contribution of this work is the introduction of False Discovery Rate (FDR) control for the principal component estimation problem, made possible by the sparse structure. We derive lower bounds on the size of detectable coordinates of the principal component vectors, and utilize these lower bounds to derive lower bounds on the worst-case risk.PHDElectrical Engineering: SystemsUniversity of Michigan, Horace H. Rackham School of Graduate Studieshttps://deepblue.lib.umich.edu/bitstream/2027.42/155061/1/prasadan_1.pd

Multivariate and 2D Extensions of Singular Spectrum Analysis with the Rssa Package

Implementation of multivariate and 2D extensions of singular spectrum analysis (SSA) by means of the R package Rssa is considered. The extensions include MSSA for simultaneous analysis and forecasting of several time series and 2D-SSA for analysis of digital images. A new extension of 2D-SSA analysis called shaped 2D-SSA is introduced for analysis of images of arbitrary shape, not necessary rectangular. It is shown that implementation of shaped 2D-SSA can serve as a basis for implementation of MSSA and other generalizations. Efficient implementation of operations with Hankel and Hankel-block-Hankel matrices through the fast Fourier transform is suggested. Examples with code fragments in R, which explain the methodology and demonstrate the proper use of Rssa, are presented

Advanced array signal processing algorithms for multi-dimensional parameter estimation

Multi-dimensional high-resolution parameter estimation is a fundamental problem in a variety of array signal processing applications, including radar, mobile communications, multiple-input multiple-output (MIMO) channel estimation, and biomedical imaging. The objective is to estimate the frequency parameters of noise-corrupted multi-dimensional harmonics that are sampled on a multi-dimensional grid. Among the proposed parameter estimation algorithms to solve this problem, multi-dimensional (R-D) ESPRIT-type algorithms have been widely used due to their computational efficiency and their simplicity. Their performance in various scenarios has been objectively evaluated by means of an analytical performance assessment framework. Recently, a relatively new class of parameter estimators based on sparse signal reconstruction has gained popularity due to their robustness under challenging conditions such as a small sample size or strong signal correlation. A common approach towards further improving the performance of parameter estimation algorithms is to exploit prior knowledge on the structure of the signals. In this thesis, we develop enhanced versions of R-D ESPRIT-type algorithms and the relatively new class of sparsity-based parameter estimation algorithms by exploiting the multi-dimensional structure of the signals and the statistical properties of strictly non-circular (NC) signals. First, we derive analytical expressions for the gain from forward-backward averaging and tensor-based processing in R-D ESPRIT-type and R-D Tensor-ESPRIT-type algorithms for the special case of two sources. This is accomplished by simplifying the generic analytical MSE expressions from the performance analysis of R-D ESPRIT-type algorithms. The derived expressions allow us to identify the parameter settings, e.g., the number of sensors, the signal correlation, and the source separation, for which both gains are most pronounced or no gain is achieved. Second, we propose the generalized least squares (GLS) algorithm to solve the overdetermined shift invariance equation in R-D ESPRIT-type algorithms. GLS directly incorporates the statistics of the subspace estimation error into the shift invariance solution through its covariance matrix, which is found via a first-order perturbation expansion. To objectively assess the estimation accuracy, we derive performance analysis expressions for the mean square error (MSE) of GLS-based ESPRIT-type algorithms, which are asymptotic in the effective SNR, i.e., the results become exact for a high SNR or a small sample size. Based on the performance analysis, we show that the simplified MSE expressions of GLS-based 1-D ESPRIT-type algorithms for a single source and two sources can be transformed into the corresponding Cramer-Rao bound (CRB) expressions, which provide a lower limit on the estimation error. Thereby, ESPRIT-type algorithms can become asymptotically efficient, i.e., they asymptotically achieve the CRB. Numerical simulations show that this can also be the case for more than two sources. In the third contribution, we derive matrix-based and tensor-based R-D NC ESPRIT-type algorithms for multi-dimensional strictly non-circular signals, where R-D NC Tensor-ESPRIT-type algorithms exploit both the multi-dimensional structure and the strictly non-circular structure of the signals. Exploiting the NC signal structure by means of a preprocessing step leads to a virtual doubling of the original sensor array, which provides an improved estimation accuracy and doubles the number of resolvable signals. We derive an analytical performance analysis and compute simplified MSE expressions for a single source and two sources. These expressions are used to analytically compute the NC gain for these cases, which has so far only been studied via Monte-Carlo simulations. We additionally consider spatial smoothing preprocessing for R-D ESPRIT-type algorithms, which has been widely used to improve the estimation performance for highly correlated signals or a small sample size. Once more, we derive performance analysis expressions for R-D ESPRIT-type algorithms and their corresponding NC versions with spatial smoothing and derive the optimal number of subarrays for spatial smoothing that minimizes the MSE for a single source. In the next part, we focus on the relatively new concept of parameter estimation via sparse signal reconstruction (SSR), in which the sparsity of the received signal power spectrum in the spatio-temporal domain is exploited. We develop three NC SSR-based parameter estimation algorithms for strictly noncircular sources and show that the benefits of exploiting the signals’ NC structure can also be achieved via sparse reconstruction. We develop two grid-based NC SSR algorithms with a low-complexity off-grid estimation procedure, and a gridless NC SSR algorithm based on atomic norm minimization. As the final contribution of this thesis, we derive the deterministic R-D NC CRB for strictly non-circular sources, which serves as a benchmark for the presented R-D NC ESPRIT-type algorithms and the NC SSR-based parameter estimation algorithms. We show for the special cases of, e.g., full coherence, a single snapshot, or a single strictly non-circular source, that the deterministic R-D NC CRB reduces to the existing deterministic R-D CRB for arbitrary signals. Therefore, no NC gain can be achieved in these cases. For the special case of two closely-spaced NC sources, we simplify the NC CRB expression and compute the NC gain for two closely-spaced NC signals. Finally, its behavior in terms of the physical parameters is studied to determine the parameter settings that provide the largest NC gain.Die hochauflösende Parameterschätzung für mehrdimensionale Signale findet Anwendung in vielen Bereichen der Signalverarbeitung in Mehrantennensystemen. Zu den Anwendungsgebieten zählen beispielsweise Radar, die Mobilkommunikation, die Kanalschätzung in multiple-input multiple-output (MIMO)-Systemen und bildgebende Verfahren in der Biosignalverarbeitung. In letzter Zeit sind eine Vielzahl von Algorithmen zur Parameterschätzung entwickelt worden, deren Schätzgenauigkeit durch eine analytische Beschreibung der Leistungsfähigkeit objektiv bewertet werden kann. Eine verbreitete Methode zur Verbesserung der Schätzgenauigkeit von Parameterschätzverfahren ist die Ausnutzung von Vorwissen bezüglich der Signalstruktur. In dieser Arbeit werden mehrdimensionale ESPRIT-Verfahren als Beispiel für Unterraum-basierte Verfahren entwickelt und analysiert, die explizit die mehrdimensionale Signalstruktur mittels Tensor-Signalverarbeitung ausnutzt und die statistischen Eigenschaften von nicht-zirkulären Signalen einbezieht. Weiterhin werden neuartige auf Signalrekonstruktion basierende Algorithmen vorgestellt, die die nicht-zirkuläre Signalstruktur bei der Rekonstruktion ausnutzen. Die vorgestellten Verfahren ermöglichen eine deutlich verbesserte Schätzgüte und verdoppeln die Anzahl der auflösbaren Signale. Die Vielzahl der Forschungsbeiträge in dieser Arbeit setzt sich aus verschiedenen Teilen zusammen. Im ersten Teil wird die analytische Beschreibung der Leistungsfähigkeit von Matrix-basierten und Tensor-basierten ESPRIT-Algorithmen betrachtet. Die Tensor-basierten Verfahren nutzen explizit die mehrdimensionale Struktur der Daten aus. Es werden für beide Algorithmenarten vereinfachte analytische Ausdrücke für den mittleren quadratischen Schätzfehler für zwei Signalquellen hergeleitet, die lediglich von den physikalischen Parametern, wie zum Beispiel die Anzahl der Antennenelemente, das Signal-zu-Rausch-Verhältnis, oder die Anzahl der Messungen, abhängen. Ein Vergleich dieser Ausdrücke ermöglicht die Berechnung einfacher Ausdrücke für den Schätzgenauigkeitsgewinn durch den forward-backward averaging (FBA)-Vorverarbeitungsschritt und die Tensor-Signalverarbeitung, die die analytische Abhängigkeit von den physikalischen Parametern enthalten. Im zweiten Teil entwickeln wir einen neuartigen general least squares (GLS)-Ansatz zur Lösung der Verschiebungs-Invarianz-Gleichung, die die Grundlage der ESPRIT-Algorithmen darstellt. Der neue Lösungsansatz berücksichtigt die statistische Beschreibung des Fehlers bei der Unterraumschätzung durch dessen Kovarianzmatrix und ermöglicht unter bestimmten Annahmen eine optimale Lösung der Invarianz-Gleichung. Mittels einer Performanzanalyse der GLS-basierten ESPRIT-Verfahren und der Vereinfachung der analytischen Ausdrücke für den Schätzfehler für eine Signalquelle und zwei zeitlich unkorrelierte Signalquellen wird gezeigt, dass die Cramer-Rao-Schranke, eine untere Schranke für die Varianz eines Schätzers, erreicht werden kann. Im nächsten Teil werden Matrix-basierte und Tensor-basierte ESPRIT-Algorithmen für nicht-zirkuläre Signalquellen vorgestellt. Unter Ausnutzung der Signalstruktur gelingt es, die Schätzgenauigkeit zu erhöhen und die doppelte Anzahl an Quellen aufzulösen. Dabei ermöglichen die vorgeschlagenen Tensor-ESPRIT-Verfahren sogar die gleichzeitige Ausnutzung der mehrdimensionalen Signalstruktur und der nicht-zirkuläre Signalstruktur. Die Leistungsfähigkeit dieser Verfahren wird erneut durch eine analytische Beschreibung objektiv bewertet und Spezialfälle für eine und zwei Quellen betrachtet. Es zeigt sich, dass für eine Quelle keinerlei Gewinn durch die nicht-zirkuläre Struktur erzielen lässt. Für zwei nicht-zirkuläre Quellen werden vereinfachte Ausdrücke für den Gewinn sowohl im Matrixfall also auch im Tensorfall hergeleitet und die Abhängigkeit der physikalischen Parameter analysiert. Sind die Signale stark korreliert oder ist die Anzahl der Messdaten sehr gering, kann der spatial smoothing-Vorverarbeitungsschritt mit den verbesserten ESPRIT-Verfahren kombiniert werden. Anhand der Performanzanalyse wird die Anzahl der Mittellungen für das spatial smoothing-Verfahren analytisch für eine Quelle bestimmt, die den Schätzfehler minimiert. Der nächste Teil befasst sich mit einer vergleichsweise neuen Klasse von Parameterschätzverfahren, die auf der Rekonstruktion überlagerter dünnbesetzter Signale basiert. Als Vorteil gegenüber den Algorithmen, die eine Signalunterraumschätzung voraussetzen, sind die Rekonstruktionsverfahren verhältnismäßig robust im Falle einer geringen Anzahl zeitlicher Messungen oder einer starken Korrelation der Signale. In diesem Teil der vorliegenden Arbeit werden drei solcher Verfahren entwickelt, die bei der Rekonstruktion zusätzlich die nicht-zirkuläre Signalstruktur ausnutzen. Dadurch kann auch für diese Art von Verfahren eine höhere Schätzgenauigkeit erreicht werden und eine höhere Anzahl an Signalen rekonstruiert werden. Im letzten Kapitel der Arbeit wird schließlich die Cramer-Rao-Schranke für mehrdimensionale nicht-zirkuläre Signale hergeleitet. Sie stellt eine untere Schranke für den Schätzfehler aller Algorithmen dar, die speziell für die Ausnutzung dieser Signalstruktur entwickelt wurden. Im Vergleich zur bekannten Cramer-Rao-Schranke für beliebige Signale, zeigt sich, dass im Fall von zeitlich kohärenten Signalen, für einen Messvektor oder für eine Quelle, beide Schranken äquivalent sind. In diesen Fällen kann daher keine Verbesserung der Schätzgüte erzielt werden. Zusätzlich wird die Cramer-Rao-Schranke für zwei benachbarte nicht-zirkuläre Signalquellen vereinfacht und der maximal mögliche Gewinn in Abhängigkeit der physikalischen Parameter analytisch ermittelt. Dieser Ausdruck gilt als Maßstab für den erzielbaren Gewinn aller Parameterschätzverfahren für zwei nicht-zirkuläre Signalquellen

Audio source separation for music in low-latency and high-latency scenarios

Aquesta tesi proposa mètodes per tractar les limitacions de les tècniques existents de separació de fonts musicals en condicions de baixa i alta latència. En primer lloc, ens centrem en els mètodes amb un baix cost computacional i baixa latència. Proposem l'ús de la regularització de Tikhonov com a mètode de descomposició de l'espectre en el context de baixa latència. El comparem amb les tècniques existents en tasques d'estimació i seguiment dels tons, que són passos crucials en molts mètodes de separació. A continuació utilitzem i avaluem el mètode de descomposició de l'espectre en tasques de separació de veu cantada, baix i percussió. En segon lloc, proposem diversos mètodes d'alta latència que milloren la separació de la veu cantada, gràcies al modelatge de components específics, com la respiració i les consonants. Finalment, explorem l'ús de correlacions temporals i anotacions manuals per millorar la separació dels instruments de percussió i dels senyals musicals polifònics complexes.Esta tesis propone métodos para tratar las limitaciones de las técnicas existentes de separación de fuentes musicales en condiciones de baja y alta latencia. En primer lugar, nos centramos en los métodos con un bajo coste computacional y baja latencia. Proponemos el uso de la regularización de Tikhonov como método de descomposición del espectro en el contexto de baja latencia. Lo comparamos con las técnicas existentes en tareas de estimación y seguimiento de los tonos, que son pasos cruciales en muchos métodos de separación. A continuación utilizamos y evaluamos el método de descomposición del espectro en tareas de separación de voz cantada, bajo y percusión. En segundo lugar, proponemos varios métodos de alta latencia que mejoran la separación de la voz cantada, gracias al modelado de componentes que a menudo no se toman en cuenta, como la respiración y las consonantes. Finalmente, exploramos el uso de correlaciones temporales y anotaciones manuales para mejorar la separación de los instrumentos de percusión y señales musicales polifónicas complejas.This thesis proposes specific methods to address the limitations of current music source separation methods in low-latency and high-latency scenarios. First, we focus on methods with low computational cost and low latency. We propose the use of Tikhonov regularization as a method for spectrum decomposition in the low-latency context. We compare it to existing techniques in pitch estimation and tracking tasks, crucial steps in many separation methods. We then use the proposed spectrum decomposition method in low-latency separation tasks targeting singing voice, bass and drums. Second, we propose several high-latency methods that improve the separation of singing voice by modeling components that are often not accounted for, such as breathiness and consonants. Finally, we explore using temporal correlations and human annotations to enhance the separation of drums and complex polyphonic music signals