System approach to robust acoustic echo cancellation through semi-blind source separation based on independent component analysis

We live in a dynamic world full of noises and interferences. The conventional acoustic echo cancellation (AEC) framework based on the least mean square (LMS) algorithm by itself lacks the ability to handle many secondary signals that interfere with the adaptive filtering process, e.g., local speech and background noise. In this dissertation, we build a foundation for what we refer to as the system approach to signal enhancement as we focus on the AEC problem. We first propose the residual echo enhancement (REE) technique that utilizes the error recovery nonlinearity (ERN) to "enhances" the filter estimation error prior to the filter adaptation. The single-channel AEC problem can be viewed as a special case of semi-blind source separation (SBSS) where one of the source signals is partially known, i.e., the far-end microphone signal that generates the near-end acoustic echo. SBSS optimized via independent component analysis (ICA) leads to the system combination of the LMS algorithm with the ERN that allows for continuous and stable adaptation even during double talk. Second, we extend the system perspective to the decorrelation problem for AEC, where we show that the REE procedure can be applied effectively in a multi-channel AEC (MCAEC) setting to indirectly assist the recovery of lost AEC performance due to inter-channel correlation, known generally as the "non-uniqueness" problem. We develop a novel, computationally efficient technique of frequency-domain resampling (FDR) that effectively alleviates the non-uniqueness problem directly while introducing minimal distortion to signal quality and statistics. We also apply the system approach to the multi-delay filter (MDF) that suffers from the inter-block correlation problem. Finally, we generalize the MCAEC problem in the SBSS framework and discuss many issues related to the implementation of an SBSS system. We propose a constrained batch-online implementation of SBSS that stabilizes the convergence behavior even in the worst case scenario of a single far-end talker along with the non-uniqueness condition on the far-end mixing system. The proposed techniques are developed from a pragmatic standpoint, motivated by real-world problems in acoustic and audio signal processing. Generalization of the orthogonality principle to the system level of an AEC problem allows us to relate AEC to source separation that seeks to maximize the independence, hence implicitly the orthogonality, not only between the error signal and the far-end signal, but rather, among all signals involved. The system approach, for which the REE paradigm is just one realization, enables the encompassing of many traditional signal enhancement techniques in analytically consistent yet practically effective manner for solving the enhancement problem in a very noisy and disruptive acoustic mixing environment.PhDCommittee Chair: Biing-Hwang Juang; Committee Member: Brani Vidakovic; Committee Member: David V. Anderson; Committee Member: Jeff S. Shamma; Committee Member: Xiaoli M

Beiträge zu breitbandigen Freisprechsystemen und ihrer Evaluation

This work deals with the advancement of wideband hands-free systems (HFS’s) for mono- and stereophonic cases of application. Furthermore, innovative contributions to the corr. field of quality evaluation are made. The proposed HFS approaches are based on frequency-domain adaptive filtering for system identification, making use of Kalman theory and state-space modeling. Functional enhancement modules are developed in this work, which improve one or more of key quality aspects, aiming at not to harm others. In so doing, these modules can be combined in a flexible way, dependent on the needs at hand. The enhanced monophonic HFS is evaluated according to automotive ITU-T recommendations, to prove its customized efficacy. Furthermore, a novel methodology and techn. framework are introduced in this work to improve the prototyping and evaluation process of automotive HF and in-car-communication (ICC) systems. The monophonic HFS in several configurations hereby acts as device under test (DUT) and is thoroughly investigated, which will show the DUT’s satisfying performance, as well as the advantages of the proposed development process. As current methods for the evaluation of HFS’s in dynamic conditions oftentimes still lack flexibility, reproducibility, and accuracy, this work introduces “Car in a Box” (CiaB) as a novel, improved system for this demanding task. It is able to enhance the development process by performing high-resolution system identification of dynamic electro-acoustical systems. The extracted dyn. impulse response trajectories are then applicable to arbitrary input signals in a synthesis operation. A realistic dynamic automotive auralization of a car cabin interior is available for HFS evaluation. It is shown that this system improves evaluation flexibility at guaranteed reproducibility. In addition, the accuracy of evaluation methods can be increased by having access to exact, realistic imp. resp. trajectories acting as a so-called “ground truth” reference. If CiaB is included into an automotive evaluation setup, there is no need for an acoustical car interior prototype to be present at this stage of development. Hency, CiaB may ease the HFS development process. Dynamic acoustic replicas may be provided including an arbitrary number of acoustic car cabin interiors for multiple developers simultaneously. With CiaB, speech enh. system developers therefore have an evaluation environment at hand, which can adequately replace the real environment.Diese Arbeit beschäftigt sich mit der Weiterentwicklung breitbandiger Freisprechsysteme für mono-/stereophone Anwendungsfälle und liefert innovative Beiträge zu deren Qualitätsmessung. Die vorgestellten Verfahren basieren auf im Frequenzbereich adaptierenden Algorithmen zur Systemidentifikation gemäß Kalman-Theorie in einer Zustandsraumdarstellung. Es werden funktionale Erweiterungsmodule dahingehend entwickelt, dass mindestens eine Qualitätsanforderung verbessert wird, ohne andere eklatant zu verletzen. Diese nach Anforderung flexibel kombinierbaren algorithmischen Erweiterungen werden gemäß Empfehlungen der ITU-T (Rec. P.1110/P.1130) in vorwiegend automotiven Testszenarien getestet und somit deren zielgerichtete Wirksamkeit bestätigt. Es wird eine Methodensammlung und ein technisches System zur verbesserten Prototypentwicklung/Evaluation von automotiven Freisprech- und Innenraumkommunikationssystemen vorgestellt und beispielhaft mit dem monophonen Freisprechsystem in diversen Ausbaustufen zur Anwendung gebracht. Daraus entstehende Vorteile im Entwicklungs- und Testprozess von Sprachverbesserungssystem werden dargelegt und messtechnisch verifiziert. Bestehende Messverfahren zum Verhalten von Freisprechsystemen in zeitvarianten Umgebungen zeigten bisher oft nur ein unzureichendes Maß an Flexibilität, Reproduzierbarkeit und Genauigkeit. Daher wird hier das „Car in a Box“-Verfahren (CiaB) entwickelt und vorgestellt, mit dem zeitvariante elektro-akustische Systeme technisch identifiziert werden können. So gewonnene dynamische Impulsantworten können im Labor in einer Syntheseoperation auf beliebige Eingangsignale angewandt werden, um realistische Testsignale unter dyn. Bedingungen zu erzeugen. Bei diesem Vorgehen wird ein hohes Maß an Flexibilität bei garantierter Reproduzierbarkeit erlangt. Es wird gezeigt, dass die Genauigkeit von darauf basierenden Evaluationsverfahren zudem gesteigert werden kann, da mit dem Vorliegen von exakten, realen Impulsantworten zu jedem Zeitpunkt der Messung eine sogenannte „ground truth“ als Referenz zur Verfügung steht. Bei der Einbindung von CiaB in einen Messaufbau für automotive Freisprechsysteme ist es bedeutsam, dass zu diesem Zeitpunkt das eigentliche Fahrzeug nicht mehr benötigt wird. Es wird gezeigt, dass eine dyn. Fahrzeugakustikumgebung, wie sie im Entwicklungsprozess von automotiven Sprachverbesserungsalgorithmen benötigt wird, in beliebiger Anzahl vollständig und mind. gleichwertig durch CiaB ersetzt werden kann

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

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Spatial hearing rendering in wireless microphone systems for binaural hearing aids

In 2015, 360 million people, including 32 million children, were suffering from hearing impairment all over the world. This makes hearing disability a major worldwide issue. In the US, the prevalence of hearing loss increased by 160% over the past generations. However, 72% of the 34 million impaired American persons (11% of the population) still have an untreated hearing loss. Among the various current solutions alleviating hearing disability, hearing aid is the only non-invasive and the most widespread medical apparatus. Combined with hearing aids, assisting listening devices are a powerful answer to address the degraded speech understanding observed in hearing-impaired subjects, especially in noisy and reverberant environments. Unfortunately, the conventional devices do not accurately render the spatial hearing property of the human auditory system, weakening their benefits. Spatial hearing is an attribute of the auditory system relying on binaural hearing. With 2 ears, human beings are able to localize sounds in space, to get information about the acoustic surroundings, to feel immersed in environments... Furthermore, it strongly contributes to speech intelligibility. It is hypothesized that recreating an artificial spatial perception through the hearing aids of impaired people might allow for recovering a part of these subjects' hearing performance. This thesis investigates and supports the aforementioned hypothesis with both technological and clinical approaches. It reveals how certain well-established signal processing methods can be integrated in some assisting listening devices. These techniques are related to sound localization and spatialization. Taking into consideration the technical constraints of current hearing aids, as well as the characteristics of the impaired auditory system, the thesis proposes a novel solution to restore a spatial perception for users of certain types of assisting listening devices. The achieved results demonstrate the feasibility and the possible implementation of such a functionality on conventional systems. Additionally, this thesis examines the relevance and the efficiency of the proposed spatialization feature towards the enhancement of speech perception. Via a clinical trial involving a large number of patients, the artificial spatial hearing shows to be well appreciated by disabled persons, while improving or preserving their current hearing abilities. This can be considered as a prominent contribution to the current scientific and technological knowledge in the domain of hearing impairment

The Bird's Ear View: Audification for the Spectral Analysis of Heliospheric Time Series Data.

The sciences are inundated with a tremendous volume of data, and the analysis of rapidly expanding data archives presents a persistent challenge. Previous research in the field of data sonification suggests that auditory display may serve a valuable function in the analysis of complex data sets. This dissertation uses the heliospheric sciences as a case study to empirically evaluate the use of audification (a specific form of sonification) for the spectral analysis of large time series. Three primary research questions guide this investigation, the first of which addresses the comparative capabilities of auditory and visual analysis methods in applied analysis tasks. A number of controlled within-subject studies revealed a strong correlation between auditory and visual observations, and demonstrated that auditory analysis provided a heightened sensitivity and accuracy in the detection of spectral features. The second research question addresses the capability of audification methods to reveal features that may be overlooked through visual analysis of spectrograms. A number of open-ended analysis tasks quantitatively demonstrated that participants using audification regularly discovered a greater percentage of embedded phenomena such as low-frequency wave storms. In addition, four case studies document collaborative research initiatives in which audification contributed to the acquisition of new domain-specific knowledge. The final question explores the potential benefits of audification when introduced into the workflow of a research scientist. A case study is presented in which a heliophysicist incorporated audification into their working practice, and the “Think-Aloud” protocol is applied to gain a sense for how audification augmented the researcher’s analytical abilities. Auditory observations are demonstrated to make significant contributions to ongoing research, including the detection of previously unidentified equipment-induced artifacts. This dissertation provides three primary contributions to the field: 1) an increased understanding of the comparative capabilities of auditory and visual analysis methods, 2) a methodological framework for conducting audification that may be transferred across scientific domains, and 3) a set of well-documented cases in which audification was applied to extract new knowledge from existing data archives. Collectively, this work presents a “bird’s ear view” afforded by audification methods—a macro understanding of time series data that preserves micro-level detail.PhDDesign ScienceUniversity of Michigan, Horace H. Rackham School of Graduate Studieshttp://deepblue.lib.umich.edu/bitstream/2027.42/111561/1/rlalexan_1.pd