Tensor Network alternating linear scheme for MIMO Volterra system identification

This article introduces two Tensor Network-based iterative algorithms for the identification of high-order discrete-time nonlinear multiple-input multiple-output (MIMO) Volterra systems. The system identification problem is rewritten in terms of a Volterra tensor, which is never explicitly constructed, thus avoiding the curse of dimensionality. It is shown how each iteration of the two identification algorithms involves solving a linear system of low computational complexity. The proposed algorithms are guaranteed to monotonically converge and numerical stability is ensured through the use of orthogonal matrix factorizations. The performance and accuracy of the two identification algorithms are illustrated by numerical experiments, where accurate degree-10 MIMO Volterra models are identified in about 1 second in Matlab on a standard desktop pc

ADAPTIVE AND NONLINEAR SIGNAL PROCESSING

1996/1997X Ciclo1967Versione digitalizzata della tesi di dottorato cartacea

Study on Reduction of Computational Complexity for Volterra Filters by Using Multi-rate Signal Processing and Its Application to Identification of Loudspeaker Systems

関西大

Adaptive Algorithms for Intelligent Acoustic Interfaces

Modern speech communications are evolving towards a new direction which involves users in a more perceptive way. That is the immersive experience, which may be considered as the “last-mile” problem of telecommunications. One of the main feature of immersive communications is the distant-talking, i.e. the hands-free (in the broad sense) speech communications without bodyworn or tethered microphones that takes place in a multisource environment where interfering signals may degrade the communication quality and the intelligibility of the desired speech source. In order to preserve speech quality intelligent acoustic interfaces may be used. An intelligent acoustic interface may comprise multiple microphones and loudspeakers and its peculiarity is to model the acoustic channel in order to adapt to user requirements and to environment conditions. This is the reason why intelligent acoustic interfaces are based on adaptive filtering algorithms. The acoustic path modelling entails a set of problems which have to be taken into account in designing an adaptive filtering algorithm. Such problems may be basically generated by a linear or a nonlinear process and can be tackled respectively by linear or nonlinear adaptive algorithms. In this work we consider such modelling problems and we propose novel effective adaptive algorithms that allow acoustic interfaces to be robust against any interfering signals, thus preserving the perceived quality of desired speech signals. As regards linear adaptive algorithms, a class of adaptive filters based on the sparse nature of the acoustic impulse response has been recently proposed. We adopt such class of adaptive filters, named proportionate adaptive filters, and derive a general framework from which it is possible to derive any linear adaptive algorithm. Using such framework we also propose some efficient proportionate adaptive algorithms, expressly designed to tackle problems of a linear nature. On the other side, in order to address problems deriving from a nonlinear process, we propose a novel filtering model which performs a nonlinear transformations by means of functional links. Using such nonlinear model, we propose functional link adaptive filters which provide an efficient solution to the modelling of a nonlinear acoustic channel. Finally, we introduce robust filtering architectures based on adaptive combinations of filters that allow acoustic interfaces to more effectively adapt to environment conditions, thus providing a powerful mean to immersive speech communications

Adaptive Algorithms for Intelligent Acoustic Interfaces

Modern speech communications are evolving towards a new direction which involves users in a more perceptive way. That is the immersive experience, which may be considered as the “last-mile” problem of telecommunications. One of the main feature of immersive communications is the distant-talking, i.e. the hands-free (in the broad sense) speech communications without bodyworn or tethered microphones that takes place in a multisource environment where interfering signals may degrade the communication quality and the intelligibility of the desired speech source. In order to preserve speech quality intelligent acoustic interfaces may be used. An intelligent acoustic interface may comprise multiple microphones and loudspeakers and its peculiarity is to model the acoustic channel in order to adapt to user requirements and to environment conditions. This is the reason why intelligent acoustic interfaces are based on adaptive filtering algorithms. The acoustic path modelling entails a set of problems which have to be taken into account in designing an adaptive filtering algorithm. Such problems may be basically generated by a linear or a nonlinear process and can be tackled respectively by linear or nonlinear adaptive algorithms. In this work we consider such modelling problems and we propose novel effective adaptive algorithms that allow acoustic interfaces to be robust against any interfering signals, thus preserving the perceived quality of desired speech signals. As regards linear adaptive algorithms, a class of adaptive filters based on the sparse nature of the acoustic impulse response has been recently proposed. We adopt such class of adaptive filters, named proportionate adaptive filters, and derive a general framework from which it is possible to derive any linear adaptive algorithm. Using such framework we also propose some efficient proportionate adaptive algorithms, expressly designed to tackle problems of a linear nature. On the other side, in order to address problems deriving from a nonlinear process, we propose a novel filtering model which performs a nonlinear transformations by means of functional links. Using such nonlinear model, we propose functional link adaptive filters which provide an efficient solution to the modelling of a nonlinear acoustic channel. Finally, we introduce robust filtering architectures based on adaptive combinations of filters that allow acoustic interfaces to more effectively adapt to environment conditions, thus providing a powerful mean to immersive speech communications

Study on Compensating Nonlinear Distortions of Electro-dynamic Loudspeaker Systems

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Aplicación de algoritmos combinados de filtrado adaptativo a acústica de salas

Las aplicaciones de procesamiento de señales acústicas están cobrando una importancia creciente. La mayoría de aplicaciones de este tipo (como la cancelación de eco acústico, la cancelación de ruido, la dereverberación, la separación y el seguimiento de fuentes acústicas, etc.) requieren la identificación de una (o varias) respuestas al impulso del recinto (RIRs). Estas respuestas pueden variar con el tiempo, por lo que se precisa de esquemas adaptativos para su identificación. La utilización de esquemas adaptativos en escenarios de identificación de respuestas acústicas se ve sujeta a diferentes compromisos, como, p. ej., la conocida relación entre velocidad de convergencia y precisión en estacionario. Varios de estos compromisos se comparten con otras aplicaciones, mientras que otros son específicos del procesamiento de señales acústicas. Entre los diferentes métodos que tratan de aliviar estas limitaciones, destaca la combinación adaptativa de filtros adaptativos debido fundamentalmente a su sencillez, versatilidad y eficacia. En esta Tesis Doctoral se aborda el estudio, diseño, implementación y adecuación de los esquemas de combinación adaptativa para que resulten provechosos y convenientes en aplicaciones de procesamiento de señales acústicas. Para ello, se proponen y analizan esquemas de combinación que ofrecen robustez y un comportamiento adecuado con respecto a las particularidades que presentan las señales acústicas involucradas y las RIRs. De entre los posibles condicionantes y sus potenciales soluciones, en esta Tesis Doctoral se contemplan: - La relación señal a ruido es normalmente desconocida a priori y puede variar. Se han desarrollado dos esquemas de combinación de filtros robustos frente a cambios en dicha relación. - El espectro de las señales acústicas (música y voz) no es plano en frecuencia, lo que ralentiza la convergencia de los filtros adaptativos. Se presenta un algoritmo de combinación en el dominio frecuencial que permite combinar de forma independiente diferentes bandas de frecuencia, obteniendo ganancias debido a que, por lo general, la relación señal a ruido es diferente en cada subbanda, y los cambios producidos en la RIR no afectan de igual forma a todo el margen frecuencial. En algunos casos, la relación entre la señal a reproducir por los altavoces y la captada por los transductores receptores es no lineal. La solución estándar para este problema de identificación no lineal se basa normalmente en los filtros de Volterra, y esta Tesis Doctoral presenta dos novedosas estrategias de combinación ad-hoc para su utilización en este contexto, las cuales obtienen ventajas de las particularidades de este tipo de filtros. Además, se propone un esquema que presenta una gran robustez con respecto a la ausencia o presencia de distorsión no lineal, e incluso con respecto a variaciones en la potencia de esta distorsión, con un modesto incremento de coste computacional con respecto al de un filtro de Volterra clásico. En muchas ocasiones, la longitud de la RIR es grande y la distribución de su energía no uniforme. Se propone un esquema que, explotando el compromiso entre sesgo y varianza, permite ganancias en esta situación, principalmente cuando la relación señal a ruido es baja. Para mostrar las ventajas del uso de los esquemas de combinación propuestos, se han llevado a cabo una serie de experimentos utilizando un escenario de cancelación de eco acústico monocanal. En todos los casos, las soluciones presentadas han obtenido resultados satisfactorios, demostrando la versatilidad y el potencial de estos algoritmos, y permitiendo mejorar el funcionamiento de los filtros adaptativos ante los condicionantes anteriormente citados. ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------Acoustic signal processing applications are becoming increasingly important. Most of these applications, such as acoustic echo cancellation, noise cancellation, dereverberation, separation and tracking of acoustic sources, etc., requires the identification of a (or several) room impulse response (RIR). This response is usually time-varying, what justifies the use of adaptive algorithms to carry out the identification task. The use of adaptive schemes in RIR identification scenarios is subject to different compromises, such as the well-known compromise between speed of convergence and steady-state precision. Several of these tradeoffs are shared by other applications, while others are specific to acoustic signal processing. Among the different methods available to alleviate these limitations, adaptive combination of adaptive filters has been recently receiving a lot of attention, mainly because of its simplicity, versatility, and effectiveness. In this Ph. D. Thesis, we deal with the development, study and implementation of adaptive combination schemes that are especially suited to acoustic signal processing applications. For this purpose, we propose and analyze combination schemes that offer robustness and a suitable behavior with respect to the peculiarities of the involved signals and RIRs. Among all possible determining factors and their potential solutions, in this Ph. D. Thesis we consider: The signal to noise ratio is usually unknown a priori and it can be time-varying. In order to deal with this situation, two new different schemes are proposed. The spectrum of acoustic signals (music and speech) is not flat, what slows down the convergence of adaptive filters. We present a combination algorithm in the frequency domain that allows to mix different frequency bands independently, offering gains that exploit the frequency dependent signal to noise power ratio and the fact that RIR changes can also take place in a frequency-localized manner. Occasionally, the relationship between the signal to be reproduced by the loudspeakers and the signal received by the microphones is nonlinear. The standard solution for this nonlinear identification problem is frequently based on Volterra filters. The Thesis presents two novel ad-hoc combinations strategies to be used in this context, which take advantage of the particularities of this kind of filters. In addition, we propose an additional algorithm that shows great robustness with respect to the presence or absence of nonlinear distortion, and even with respect to changes in the power of nonlinear distortion, with a very modest increment in terms of computational cost. In many cases, very large RIRs are present, and their energies are typically distributed in a non-uniform manner. We propose a scheme that, exploiting the tradeoff between bias and variance, permits important gains in this situation, mainly for low signal to noise power ratios. In order to illustrate the advantages of the proposed combinations schemes, several experiments have been carried out considering a single-channel acoustic echo cancellation scenario. The satisfactory results obtained by the presented solutions demonstrate the versatility and potential of these algorithms, allowing to improve the performance of adaptive filters in the presence of the aforementioned conditions

Système d'annulation d'écho pour répéteur iso-fréquence : contribution à l'élaboration d'un répéteur numérique de nouvelle génération

On-frequency repeaters are a cost-effective solution to extend coverage and enhance wireless communications, especially in shadow areas. However, coupling between the receiving antenna and the transmitting antenna, called radio frequency echo, increases modulation errors and creates oscillations in the system when the echo power is high. According to the communication standards, extremely weak echoes decrease the transmission rate, while strong echoes damage electroni ccircuits because of power peaks. This thesis aims at characterizing the echo phenomenon under different modulations, and proposing an optimized solution directly integrated to industry. We have turned to digital solutions especially the adaptive because of their high convergence rate and their simplicity to be implemented. The programmable circuits are chosen for their attractive price and their flexibility. When implementing echo cancellation solution, we proposed several reliable solutions, showing that digital processing is much more beneficial. For this reason, digital solutions are generalized, and the new generation of repeaters is fully digital.Le déploiement des répéteurs iso-fréquence est une solution économique pour étendre la couverture d’un émetteur principal aux zones d’ombre. Cependant, ce mode de déploiement fait apparaître le phénomène des échos radio-fréquence entre antennes d’émission et de réception du répéteur. Selon les standards, un écho aussi faible soit-il réduit le débit de la liaison radio, tandis qu’un écho fort fait courir au répéteur le risque d’endommager ses circuits électroniques, ces risques sont dûs aux ondulations de puissance créées par les échos. L’objectif de cette thèse à caractère industriel est d’étudier ce phénomène naturel en considérant des signaux provenant de différents standards des télécommunications. Cette étude permet une caractérisation des échos radio-fréquence pour mieux s’orienter vers une solution optimisée et industriellement réalisable.Nous nous sommes orientés vers la solution du traitement du signal avancé en favorisant le filtrage adaptatif pour sa rapidité de convergence et sa simplicité relative d’implantation matérielle. Les circuits reconfigurables sont retenus pour leur prix et leur souplesse. L’implantation des solutions est effectuée en virgule fixe afin de satisfaire les exigences de réactivité. Durant la mise en oeuvre de la solution anti-écho, nous avons proposé une multitude de solutions numériques souples et fiables. À partir de ce constat, notre partenaire industriel a décidé de généraliser ce mode de traitement par le développement, la fabrication et la commercialisation de répéteurs de nouvelle génération entièrement numériques