On the spectral factor ambiguity of FIR energy compaction filter banks

This paper focuses on the design of signal-adapted finite-impulse response (FIR) paraunitary (PU) filter banks optimized for energy compaction (EC). The design of such filter banks has been shown in the literature to consist of the design of an optimal FIR compaction filter followed by an appropriate Karhunen-Loe/spl grave/ve transform (KLT). Despite this elegant construction, EC optimal filter banks have been shown to perform worse than common nonadapted filter banks for coding gain, contrary to intuition. Here, it is shown that this phenomenon is most likely due to the nonuniqueness of the compaction filter in terms of its spectral factors. This nonuniqueness results in a finite set of EC optimal filter banks. By choosing the spectral factor yielding the largest coding gain, it is shown that the resulting filter bank behaves more and more like the infinite-order principal components filter bank (PCFB) in terms of numerous objectives such as coding gain, multiresolution, noise reduction with zeroth-order Wiener filters in the subbands, and power minimization for discrete multitone (DMT)-type nonredundant transmultiplexers

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

Adaptive Feedback Cancellation in Hearing Aids

Acoustic feedback is a well-known phenomenon in hearing aids and public address systems. Under certain conditions it causes the so-called howling effect, which is highly annoying for the hearing aid user and limits the maximum amplification of the hearing aid. The most common choice to prevent howling is the adaptive feedback cancellation algorithm, which is able to completely eliminate the feedback signal. However, standard adaptive feedback cancellation algorithms suffer from a biased adaptation if the input signal is spectrally colored, as it is for speech and music signals. Due to this bias distortion artifacts (entrainment) are generated and consequently, the sound quality is significantly reduced. Most of the known methods to reduce the bias have focused on speech signals. However, those methods do not cope with music, since the tonality and correlation are much stronger for such signals. This leads to a higher bias and consequently, to stronger entrainment for music than for speech. Other methods, which deal with music signals, work only satisfactorily when using a very slow adaptation speed. This reduces the ability to react fast to feedback path changes. Hence, howling occurs for a longer time when the feedback path is changing. In this thesis, a new sub-band adaptive feedback cancellation system for hearing aid applications is proposed. It combines decorrelation methods with a new realization of a non-parametric variable step size. The adaptation is realized in sub-bands which decreases the computational complexity and increases the adaptation performance of the system simultaneously. The applied decorrelation methods, prediction error filter and frequency shift, are well known approaches to reduce the bias. However, the combination of both is first proposed in this thesis. To apply the proposed step size in the context of adaptive feedback cancellation, a method to estimate the signal power of the desired input signal, i.e., without feedback, also referred to as source signal power is necessary. This estimate is theoretically derived and it is demonstrated that it is a reliabe estimate if the decorrelation methods are additionally applied. In order to further improve the performance of the system three additional control methods are derived: The first one is an impulse detection to detect wideband impulses, which could lead to misadaptation. Secondly, a modified estimate of the source signal power to stabilize the system in case of jarring voices is proposed. Lastly, a correlation detection, which is applied to improve the trade-off between adaptation stability and tracking behavior, is developed. The complete system is optimized and evaluated for several speech and music signals as well as for different feedback scenarios in simulations with feedback paths measured under realistic situations. Additionally, the system is tested by real-time simulations with hearing aid dummies and a torso and head simulator. For both simulation setups hearing loss compensation methods as applied in realistic hearing aids are used. The performance is measured in terms of being able to prevent entrainment (adaptation stability) and reacting to feedback path changes (tracking behavior). The complete adaptive feedback cancellation system shows an excellent performance. Furthermore, the system relies only on few parameters, shows a low computational complexity, and therefore has a strong practical relevance

Filtrado adaptativo multicanal para control local de campo sonoro basado en algoritmos de proyección afín

Esta Tesis doctoral se ha centrado en el desarrollo e implementación de algoritmos eficientes multicanal, basados en el algoritmo de proyección afín, aplicados al control activo de ruido. Para abordar esta cuestión primeramente se han estudiado diferentes algoritmos eficientes de proyección afín que han sido analizados y validados mediante simulación, finalizando con la implementación, en un recinto, de un sistema real de control activo de ruido multicanal ejecutado en un DSP controlado por dichos algoritmos. En los últimos años, los algoritmos de proyección afín han sido propuestos como algoritmos de control en sistemas adaptativos, que pretenden mejorar la velocidad de convergencia de los algoritmos basados en el LMS, siendo una alternativa eficiente, robusta y estable frente a estos algoritmos, cuya limitación principal es, precisamente, la velocidad de convergencia. Los algoritmos de proyección afín pueden ser considerados como una extensión natural del algoritmo NLMS, ya que éste actualiza sus coeficientes basándose en un único vector de datos de la señal de entrada mientras que los algoritmos de proyección afín actualizan los coeficientes de los filtros adaptativos usando N vectores de datos de la señal de entrada (siendo N el orden de proyección). Se han dedicado muchos esfuerzos para tratar de optimizar la eficiencia computacional de estos algoritmos aplicados al problema de la cancelación de eco, surgiendo diferentes versiones eficientes del algoritmo de proyección afín. Sin embargo, al aplicarlo al control activo de ruido, es necesario reducir aún más la complejidad computacional, teniendo en cuenta que, por lo general, la eficiencia computacional se consigue a costa de la degradación de alguna otra característica del algoritmo (generalmente la velocidad de convergencia). En este trabajo se presentan algunas alternativas a versiones eficientes existentes, que no degradan significativamente las prestaciones de dicho algoritmo, y se analiza cómo reducir aúnFerrer Contreras, M. (2008). Filtrado adaptativo multicanal para control local de campo sonoro basado en algoritmos de proyección afín [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/3796Palanci