Local sound field synthesis

This thesis investigates the physical and perceptual properties of selected methods for (Local) Sound Field Synthesis ((L)SFS). In agreement with numerical sound field simulations, a specifically developed geometric model shows an increase of synthesis accuracy for LSFS compared to conventional SFS approaches. Different (L)SFS approaches are assessed within listening experiments, where LSFS performs at least as good as conventional methods for azimuthal sound source localisation and achieves a significant increase of timbral fidelity for distinct parametrisations.Die Arbeit untersucht die physikalischen und perzeptiven Eigenschaften von ausgewählten Verfahren zur (lokalen) Schallfeldsynthese ((L)SFS). Zusammen mit numerischen Simulationen zeigt ein eigens entwickeltes geometrisches Modell, dass LSFS gegenüber konventioneller SFS zu einer genauere Synthese führt. Die Verfahren werden in Hörversuchen evaluiert, wobei LSFS bei der horizontalen Lokalisierung von Schallquellen eine Genauigkeit erreicht, welche mindestens gleich der von konventionellen Methoden ist. Für bestimmte Parametrierung wird eine signifikant verbesserte klangliche Treue erreicht

Wave Field Synthesis in a listening room

This thesis investigates the influence of the listening room on sound fields synthesised by Wave Field Synthesis. Methods are developed that allow for investigation of the spatial and timbral perception of Wave Field Synthesis in a reverberant environment using listening experiments based on simulation by binaural synthesis and room acoustical simulation. The results can serve as guidelines for the design of listening rooms for Wave Field Synthesis.Diese Dissertation untersucht den Einfluss des Wiedergaberaums auf Schallfelder, die mit Wellenfeldsynthese synthetisiert werden. Es werden Methoden zur Untersuchung von räumlicher und klangfarblicher Wahrnehmung von Wellenfeldsynthese in einer reflektierenden Umgebung mittels Hörversuchen entwickelt, die auf Simulation mit Binauralsynthese und raumakustischer Simulation beruhen. Die Ergebnisse können als Richtlinien zur Gestaltung von Wiedergaberäumen für Wellenfeldsynthese dienen

Bayesian Models for Multimodal Perception of 3D Structure and Motion

In this text we will formalise a novel solution, the Bayesian Volumetric Map (BVM), as a framework for a metric, short-term, egocentric spatial memory for multimodal perception of 3D structure and motion. This solution will enable the implementation of top-down mechanisms of attention guidance of perception towards areas of high entropy/uncertainty, so as to promote active exploration of the environment by the robotic perceptual system. In the process, we will to try address the inherent challenges of visual, auditory and vestibular sensor fusion through the BVM. In fact, it is our belief that perceptual systems are unable to yield truly useful descriptions of their environment without resorting to a temporal series of sensory fusion processed on a short-term memory such as the BVM

Vision-Guided Robot Hearing

International audienceNatural human-robot interaction (HRI) in complex and unpredictable environments is important with many potential applicatons. While vision-based HRI has been thoroughly investigated, robot hearing and audio-based HRI are emerging research topics in robotics. In typical real-world scenarios, humans are at some distance from the robot and hence the sensory (microphone) data are strongly impaired by background noise, reverberations and competing auditory sources. In this context, the detection and localization of speakers plays a key role that enables several tasks, such as improving the signal-to-noise ratio for speech recognition, speaker recognition, speaker tracking, etc. In this paper we address the problem of how to detect and localize people that are both seen and heard. We introduce a hybrid deterministic/probabilistic model. The deterministic component allows us to map 3D visual data onto an 1D auditory space. The probabilistic component of the model enables the visual features to guide the grouping of the auditory features in order to form audiovisual (AV) objects. The proposed model and the associated algorithms are implemented in real-time (17 FPS) using a stereoscopic camera pair and two microphones embedded into the head of the humanoid robot NAO. We perform experiments with (i)~synthetic data, (ii)~publicly available data gathered with an audiovisual robotic head, and (iii)~data acquired using the NAO robot. The results validate the approach and are an encouragement to investigate how vision and hearing could be further combined for robust HRI

Système d'audition artificielle embarqué optimisé pour robot mobile muni d'une matrice de microphones

Dans un environnement non contrôlé, un robot doit pouvoir interagir avec les personnes d’une façon autonome. Cette autonomie doit également inclure une interaction grâce à la voix humaine. Lorsque l’interaction s’effectue à une distance de quelques mètres, des phénomènes tels que la réverbération et la présence de bruit ambiant doivent être pris en considération pour effectuer efficacement des tâches comme la reconnaissance de la parole ou de locuteur. En ce sens, le robot doit être en mesure de localiser, suivre et séparer les sources sonores présentes dans son environnement. L’augmentation récente de la puissance de calcul des processeurs et la diminution de leur consommation énergétique permettent dorénavant d’intégrer ces systèmes d’audition articielle sur des systèmes embarqués en temps réel. L’audition robotique est un domaine relativement jeune qui compte deux principales librairies d’audition artificielle : ManyEars et HARK. Jusqu’à présent, le nombre de microphones se limite généralement à huit, en raison de l’augmentation rapide de charge de calculs lorsque des microphones supplémentaires sont ajoutés. De plus, il est parfois difficile d’utiliser ces librairies avec des robots possédant des géométries variées puisqu’il est nécessaire de les calibrer manuellement. Cette thèse présente la librairie ODAS qui apporte des solutions à ces difficultés. Afin d’effectuer une localisation et une séparation plus robuste aux matrices de microphones fermées, ODAS introduit un modèle de directivité pour chaque microphone. Une recherche hiérarchique dans l’espace permet également de réduire la quantité de calculs nécessaires. De plus, une mesure de l’incertitude du délai d’arrivée du son est introduite pour ajuster automatiquement plusieurs paramètres et ainsi éviter une calibration manuelle du système. ODAS propose également un nouveau module de suivi de sources sonores qui emploie des filtres de Kalman plutôt que des filtres particulaires. Les résultats démontrent que les méthodes proposées réduisent la quantité de fausses détections durant la localisation, améliorent la robustesse du suivi pour des sources sonores multiples et augmentent la qualité de la séparation de 2.7 dB dans le cas d’un formateur de faisceau à variance minimale. La quantité de calculs requis diminue par un facteur allant jusqu’à 4 pour la localisation et jusqu’à 30 pour le suivi par rapport à la librairie ManyEars. Le module de séparation des sources sonores exploite plus efficacement la géométrie de la matrice de microphones, sans qu’il soit nécessaire de mesurer et calibrer manuellement le système. Avec les performances observées, la librairie ODAS ouvre aussi la porte à des applications dans le domaine de la détection des drones par le bruit, la localisation de bruits extérieurs pour une navigation plus efficace pour les véhicules autonomes, des assistants main-libre à domicile et l’intégration dans des aides auditives

Three-Dimensional Geometry Inference of Convex and Non-Convex Rooms using Spatial Room Impulse Responses

This thesis presents research focused on the problem of geometry inference for both convex- and non-convex-shaped rooms, through the analysis of spatial room impulse responses. Current geometry inference methods are only applicable to convex-shaped rooms, requiring between 6--78 discretely spaced measurement positions, and are only accurate under certain conditions, such as a first-order reflection for each boundary being identifiable across all, or some subset of, these measurements. This thesis proposes that by using compact microphone arrays capable of capturing spatiotemporal information, boundary locations, and hence room shape for both convex and non-convex cases, can be inferred, using only a sufficient number of measurement positions to ensure each boundary has a first-order reflection attributable to, and identifiable in, at least one measurement. To support this, three research areas are explored. Firstly, the accuracy of direction-of-arrival estimation for reflections in binaural room impulse responses is explored, using a state-of-the-art methodology based on binaural model fronted neural networks. This establishes whether a two-microphone array can produce accurate enough direction-of-arrival estimates for geometry inference. Secondly, a spherical microphone array based spatiotemporal decomposition workflow for analysing reflections in room impulse responses is explored. This establishes that simultaneously arriving reflections can be individually detected, relaxing constraints on measurement positions. Finally, a geometry inference method applicable to both convex and more complex non-convex shaped rooms is proposed. Therefore, this research expands the possible scenarios in which geometry inference can be successfully applied at a level of accuracy comparable to existing work, through the use of commonly used compact microphone arrays. Based on these results, future improvements to this approach are presented and discussed in detail

Proceedings of the EAA Spatial Audio Signal Processing symposium: SASP 2019

International audienc

Acoustic Source Localisation in constrained environments

Acoustic Source Localisation (ASL) is a problem with real-world applications across multiple domains, from smart assistants to acoustic detection and tracking. And yet, despite the level of attention in recent years, a technique for rapid and robust ASL remains elusive – not least in the constrained environments in which such techniques are most likely to be deployed. In this work, we seek to address some of these current limitations by presenting improvements to the ASL method for three commonly encountered constraints: the number and configuration of sensors; the limited signal sampling potentially available; and the nature and volume of training data required to accurately estimate Direction of Arrival (DOA) when deploying a particular supervised machine learning technique. In regard to the number and configuration of sensors, we find that accuracy can be maintained at state-of-the-art levels, Steered Response Power (SRP), while reducing computation sixfold, based on direct optimisation of well known ASL formulations. Moreover, we find that the circular microphone configuration is the least desirable as it yields the highest localisation error. In regard to signal sampling, we demonstrate that the computer vision inspired algorithm presented in this work, which extracts selected keypoints from the signal spectrogram, and uses them to select signal samples, outperforms an audio fingerprinting baseline while maintaining a compression ratio of 40:1. In regard to the training data employed in machine learning ASL techniques, we show that the use of music training data yields an improvement of 19% against a noise data baseline while maintaining accuracy using only 25% of the training data, while training with speech as opposed to noise improves DOA estimation by an average of 17%, outperforming the Generalised Cross-Correlation technique by 125% in scenarios in which the test and training acoustic environments are matched.Heriot-Watt University James Watt Scholarship (JSW) in the School of Engineering & Physical Sciences