Control of feedback for assistive listening devices

Acoustic feedback refers to the undesired acoustic coupling between the loudspeaker and microphone in hearing aids. This feedback channel poses limitations to the normal operation of hearing aids under varying acoustic scenarios. This work makes contributions to improve the performance of adaptive feedback cancellation techniques and speech quality in hearing aids. For this purpose a two microphone approach is proposed and analysed; and probe signal injection methods are also investigated and improved upon

multi-band acoustic echo canceller

Thesis (S.B. and M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.Includes bibliographical references (leaves 68-69).by Mingxi Fan.S.B.and M.Eng

Sparseness-controlled adaptive algorithms for supervised and unsupervised system identification

In single-channel hands-free telephony, the acoustic coupling between the loudspeaker and the microphone can be strong and this generates echoes that can degrade user experience. Therefore, effective acoustic echo cancellation (AEC) is necessary to maintain a stable system and hence improve the perceived voice quality of a call. Traditionally, adaptive filters have been deployed in acoustic echo cancellers to estimate the acoustic impulse responses (AIRs) using adaptive algorithms. The performances of a range of well-known algorithms are studied in the context of both AEC and network echo cancellation (NEC). It presents insights into their tracking performances under both time-invariant and time-varying system conditions. In the context of AEC, the level of sparseness in AIRs can vary greatly in a mobile environment. When the response is strongly sparse, convergence of conventional approaches is poor. Drawing on techniques originally developed for NEC, a class of time-domain and a frequency-domain AEC algorithms are proposed that can not only work well in both sparse and dispersive circumstances, but also adapt dynamically to the level of sparseness using a new sparseness-controlled approach. As it will be shown later that the early part of the acoustic echo path is sparse while the late reverberant part of the acoustic path is dispersive, a novel approach to an adaptive filter structure that consists of two time-domain partition blocks is proposed such that different adaptive algorithms can be used for each part. By properly controlling the mixing parameter for the partitioned blocks separately, where the block lengths are controlled adaptively, the proposed partitioned block algorithm works well in both sparse and dispersive time-varying circumstances. A new insight into an analysis on the tracking performance of improved proportionate NLMS (IPNLMS) is presented by deriving the expression for the mean-square error. By employing the framework for both sparse and dispersive time-varying echo paths, this work validates the analytic results in practical simulations for AEC. The time-domain second-order statistic based blind SIMO identification algorithms, which exploit the cross relation method, are investigated and then a technique with proportionate step-size control for both sparse and dispersive system identification is also developed

Theory and Design of Spatial Active Noise Control Systems

The concept of spatial active noise control is to use a number of loudspeakers to generate anti-noise sound waves, which would cancel the undesired acoustic noise over a spatial region. The acoustic noise hazards that exist in a variety of situations provide many potential applications for spatial ANC. However, using existing ANC techniques, it is difficult to achieve satisfying noise reduction for a spatial area, especially using a practical hardware setup. Therefore, this thesis explores various aspects of spatial ANC, and seeks to develop algorithms and techniques to promote the performance and feasibility of spatial ANC in real-life applications. We use the spherical harmonic analysis technique as the basis for our research in this work. This technique provides an accurate representation of the spatial noise field, and enables in-depth analysis of the characteristics of the noise field. Incorporating this technique into the design of spatial ANC systems, we developed a series of algorithms and methods that optimizes the spatial ANC systems, towards both improving noise reduction performance and reducing system complexity. Several contributions of this work are: (i) design of compact planar microphone array structures capable of recording 3D spatial sound fields, so that the noise field can be monitored with minimum physical intrusion to the quiet zone, (ii) derivation of a Direct-to-Reverberant Energy Ratio (DRR) estimation algorithm which can be used for evaluating reverberant characteristics of a noisy environment, (iii) propose a few methods to estimate and optimize spatial noise reduction of an ANC system, including a new metric for measuring spatial noise energy level, and (iv) design of an adaptive spatial ANC algorithm incorporating the spherical harmonic analysis technique. The combination of these contributions enables the design of compact, high performing spatial ANC systems for various applications

Comportamento de um classe de algoritmos de cancelamento de eco acústico auxiliado por um arranjo de microfones

Tese (doutorado) - Universidade Federal de Santa Catarina, Centro Tecnológico, Programa de Pós-Graduação em Engenharia Elétrica, Florianópolis, 2014.Esta tese apresenta uma análise estatística de uma classe de canceladores de eco acústico (AEC) auxiliados por um arranjo de microfones otimizados de maneira conjunta. A análise é realizada para sistemas em que o conformador de feixes (BF) é implementado na forma direta utilizando o algoritmo constrained least-mean squares e na forma de cancelador de lóbulos generalizado (GSC) utilizando o algoritmo least-mean squares. Para o BF implementado na forma direta, um modelo analítico é desenvolvido para o comportamento estatístico do sistema quando a convergência de ambos AEC e BF é controlada utilizando o mesmo passo de adaptação. Para o BF implementado na forma GSC, a análise é generalizada para considerar o controle de convergência utilizando uma matriz de passos de adaptação. É proposta uma nova formulação analítica, que demonstra que o problema da otimização conjunta é equivalente a um problema de minimização da variância com restrições lineares. Consequentemente, a nova análise leva a modelos analíticos que podem ser utilizados para prever o comportamento transitório de conformadores de feixe de banda larga tanto na forma direta quanto GSC. A generalização para a adaptação utilizando a matriz de passos leva a um modelo mais versátil que permite o estudo do comportamento do sistema sob uma lógica de controle externa. Essa generalização é especialmente interessante para o projeto de canceladores de eco acústicos reais pois a lógica de controle normalmente requer a operação do AEC e do BF com passos de adaptação diferentes durante diferentes condições de adaptação (presença de fala local, mudanças de canal, rastreamento, etc). Modelos estatísticos são determinados para o comportamento transiente e em regime-permanente da potência de eco residual para sinais de entrada Gaussianos e estacionários. A análise de convergência resulta em limites de estabilidade para o passo de adaptação na forma direta e para a matriz de passos na forma GSC. Diretrizes de projeto são obtidas a partir dos modelos analíticos. Simulações de Monte Carlo mostram a precisão dos modelos teóricos e a utilidade das diretrizes de projeto. Exemplos de simulação incluem a operação sob efeito de não-estacionariedades moderadas. Os novos modelos confirmam teoricamente os resultados experimentais que indicam que o mesmo desempenho em cancelamento de AECs com um único microfone pode ser obtido com um AEC de comprimento menor quando há a possibilidade de uso de filtragem espacial. Finalmente, é mostrado que soluções com alta taxa de convergência podem ser obtidas utilizando o BF na forma GSC por meio de uma adaptação baseada em um algoritmo quase-Newton na qual a matriz de passos é projetada para descorrelacionar o vetor de entrada combinado

Theory and Design of Feasible Active Noise Control Systems for 3D Regions

This thesis advances Active Noise Control (ANC) over three-dimensional (3D) space using feasible loudspeaker and microphone array systems. By definition, ANC reduces unwanted acoustic noise by generating an anti-noise signal(s) from secondary loudspeakers. The concept of spatial ANC aims to reduce unwanted acoustic noise over a continuous 3D region, by utilizing multiple microphones and multiple secondary loudspeakers to create a large-sized quiet zone for listeners in three-dimensional space. However, existing spatial ANC techniques are usually impractical and difficult to implement due to their strict hardware requirements and high computation complexity. Therefore, this thesis explores various aspects of spatial ANC, seeking algorithms and techniques to promote the reliability and feasibility of ANC over space in real-life applications. The spherical harmonic analysis technique is introduced as the basis of conventional spatial ANC systems. This technique provides an accurate representation of a given spatial sound field using higher-order microphone (spherical microphone array) recordings. Hence, the residual noise field in a spatial ANC system can be effectively captured spatially by applying the spherical harmonic technique. Incorporating conventional spatial ANC methods, we developed a series of algorithms and methods that optimize conventional methods regarding array geometries and ANC algorithms, towards improving the feasibility of a conventional spatial ANC system involving the spherical harmonic analysis. Overall, motivated by feasible and realistic designs for spatial ANC systems, work included in this thesis mainly solves the three problems of: (i) the impracticality of realizing spherical microphone and loudspeaker arrays, (ii) achieving secondary channel estimation with microphones remote from their desired locations, and (iii) unreasonable delays inherent to frequency domain spatial ANC methods. Based on our work, we have stepped towards achieving a spatial ANC system in a real-world environment for people to enjoy silence in the control region with the reliable usage of resources and algorithms. Several contributions of this work are: (i) designing a 3D spatial ANC system using multiple circular microphone and loudspeaker arrays instead of spherical arrays, (ii) proposing a 3D spatial ANC method with remote microphone technique such that noise reduction over a region is achieved with microphones remote from the region, (iii) proposing a secondary channel estimation method using a moving higher-order microphone such that usage of an error microphone array is not necessary, (iv) deriving a time domain spherical harmonic analysis method for open spherical microphone array recording with less delay than in the frequency domain, and (v) designing a feed-forward adaptive spatial ANC algorithm incorporating the time domain spherical harmonic analysis technique to better minimize the noise in the region of interest

Transient simulation of complex electronic circuits and systems operating at ultra high frequencies

The electronics industry worldwide faces increasingly difficult challenges in a bid to produce ultra-fast, reliable and inexpensive electronic devices. Electronic manufacturers rely on the Electronic Design Automation (EDA) industry to produce consistent Computer A id e d Design (CAD) simulation tools that w ill enable the design of new high-performance integrated circuits (IC), the key component of a modem electronic device. However, the continuing trend towards increasing operational frequencies and shrinking device sizes raises the question of the capability of existing circuit simulators to accurately and efficiently estimate circuit behaviour. The principle objective of this thesis is to advance the state-of-art in the transient simulation of complex electronic circuits and systems operating at ultra high frequencies. Given a set of excitations and initial conditions, the research problem involves the determination of the transient response o f a high-frequency complex electronic system consisting of linear (interconnects) and non-linear (discrete elements) parts with greatly improved efficien cy compared to existing methods and with the potential for very high accuracy in a way that permits an effective trade-off between accuracy and computational complexity. High-frequency interconnect effects are a major cause of the signal degradation encountered b y a signal propagating through linear interconnect networks in the modem IC. Therefore, the development of an interconnect model that can accurately and efficiently take into account frequency-dependent parameters of modem non-uniform interconnect is of paramount importance for state-of-art circuit simulators. Analytical models and models based on a set of tabulated data are investigated in this thesis. Two novel, h igh ly accurate and efficient interconnect simulation techniques are developed. These techniques combine model order reduction methods with either an analytical resonant model or an interconnect model generated from frequency-dependent sparameters derived from measurements or rigorous full-wave simulation. The latter part o f the thesis is concerned with envelope simulation. The complex mixture of profoundly different analog/digital parts in a modern IC gives rise to multitime signals, where a fast changing signal arising from the digital section is modulated by a slower-changing envelope signal related to the analog part. A transient analysis of such a circuit is in general very time-consuming. Therefore, specialised methods that take into account the multi-time nature o f the signal are required. To address this issue, a novel envelope simulation technique is developed. This technique combines a wavelet-based collocation method with a multi-time approach to result in a novel simulation technique that enables the desired trade-off between the required accuracy and computational efficiency in a simple and intuitive way. Furthermore, this new technique has the potential to greatly reduce the overall design cycle