A Geometrical-Statistical Approach to Outlier Removal for TDOA Measurements

The curse of outlier measurements in estimation problems is a well-known issue in a variety of fields. Therefore, outlier removal procedures, which enables the identification of spurious measurements within a set, have been developed for many different scenarios and applications. In this paper, we propose a statistically motivated outlier removal algorithm for time differences of arrival (TDOAs), or equivalently range differences (RD), acquired at sensor arrays. The method exploits the TDOA-space formalism and works by only knowing relative sensor positions. As the proposed method is completely independent from the application for which measurements are used, it can be reliably used to identify outliers within a set of TDOA/RD measurements in different fields (e.g., acoustic source localization, sensor synchronization, radar, remote sensing, etc.). The proposed outlier removal algorithm is validated by means of synthetic simulations and real experiments

Localization using Distance Geometry : Minimal Solvers and Robust Methods for Sensor Network Self-Calibration

In this thesis, we focus on the problem of estimating receiver and sender node positions given some form of distance measurements between them. This kind of localization problem has several applications, e.g., global and indoor positioning, sensor network calibration, molecular conformations, data visualization, graph embedding, and robot kinematics. More concretely, this thesis makes contributions in three different areas.First, we present a method for simultaneously registering and merging maps. The merging problem occurs when multiple maps of an area have been constructed and need to be combined into a single representation. If there are no absolute references and the maps are in different coordinate systems, they also need to be registered. In the second part, we construct robust methods for sensor network self-calibration using both Time of Arrival (TOA) and Time Difference of Arrival (TDOA) measurements. One of the difficulties is that corrupt measurements, so-called outliers, are present and should be excluded from the model fitting. To achieve this, we use hypothesis-and-test frameworks together with minimal solvers, resulting in methods that are robust to noise, outliers, and missing data. Several new minimal solvers are introduced to accommodate a range of receiver and sender configurations in 2D and 3D space. These solvers are formulated as polynomial equation systems which are solvedusing methods from algebraic geometry.In the third part, we focus specifically on the problems of trilateration and multilateration, and we present a method that approximates the Maximum Likelihood (ML) estimator for different noise distributions. The proposed approach reduces to an eigendecomposition problem for which there are good solvers. This results in a method that is faster and more numerically stable than the state-of-the-art, while still being easy to implement. Furthermore, we present a robust trilateration method that incorporates a motion model. This enables the removal of outliers in the distance measurements at the same time as drift in the motion model is canceled

Development and Human Factors Evaluation of a Portable Auditory Localization Acclimation Training System

Auditory situation awareness (ASA) is essential for safety and survivability in military operations where many of the hazards are not immediately visible. Unfortunately, the Hearing Protection Devices (HPDs) required to operate in these environments can impede auditory localization performance. Promisingly, recent studies have exhibited the plasticity of the human auditory system by demonstrating that training can improve auditory localization ability while wearing HPDs, including military Tactical Communications and Protective Systems (TCAPS). As a result, the U.S. military identified the need for a portable system capable of imparting auditory localization acquisition skills at similar levels to those demonstrated in laboratory environments. The purpose of this investigation was to develop and validate a Portable Auditory Localization Acclimation Training (PALAT) system equipped with an improved training protocol against a proven laboratory grade system referred to as the DRILCOM system and subsequently evaluate the transfer-of-training benefit in a field environment. In Phase I, a systems decision process was used to develop a prototype PALAT system consisting of an expandable frame housing 32-loudspeakers operated by a user-controlled tablet computer capable of reproducing acoustically accurate localization cues similar to the DRILCOM system. Phase II used a within-subjects human factors experiment to validate whether the PALAT system could impart similar auditory localization training benefits as the DRILCOM system. Results showed no significant difference between the two localization training systems at each stage of training or in training rates for the open ear and with two TCAPS devices. The PALAT system also demonstrated the ability to detect differences in localization accuracy between listening conditions in the same manner as the DRILCOM system. Participant ratings indicated no perceived difference in localization training benefit but significantly preferred the PALAT system user interface which was specifically designed to improve usability features to meet requirements of a user operable system. The Phase III investigation evaluated the transfer-of-training benefit imparted by the PALAT system using a broadband stimulus to a field environment using gunshot stimulus. Training under the open ear and in-the-ear TCAPS resulted in significant differences between the trained and untrained groups from in-office pretest to in-field posttest

Development and Human Factors Evaluation of a Portable Auditory Localization Acclimation Training System

Auditory situation awareness (ASA) is essential for safety and survivability in military operations where many of the hazards are not immediately visible. Unfortunately, the Hearing Protection Devices (HPDs) required to operate in these environments can impede auditory localization performance. Promisingly, recent studies have exhibited the plasticity of the human auditory system by demonstrating that training can improve auditory localization ability while wearing HPDs, including military Tactical Communications and Protective Systems (TCAPS). As a result, the U.S. military identified the need for a portable system capable of imparting auditory localization acquisition skills at similar levels to those demonstrated in laboratory environments. The purpose of this investigation was to develop and validate a Portable Auditory Localization Acclimation Training (PALAT) system equipped with an improved training protocol against a proven laboratory grade system referred to as the DRILCOM system and subsequently evaluate the transfer-of-training benefit in a field environment. In Phase I, a systems decision process was used to develop a prototype PALAT system consisting of an expandable frame housing 32-loudspeakers operated by a user-controlled tablet computer capable of reproducing acoustically accurate localization cues similar to the DRILCOM system. Phase II used a within-subjects human factors experiment to validate whether the PALAT system could impart similar auditory localization training benefits as the DRILCOM system. Results showed no significant difference between the two localization training systems at each stage of training or in training rates for the open ear and with two TCAPS devices. The PALAT system also demonstrated the ability to detect differences in localization accuracy between listening conditions in the same manner as the DRILCOM system. Participant ratings indicated no perceived difference in localization training benefit but significantly preferred the PALAT system user interface which was specifically designed to improve usability features to meet requirements of a user operable system. The Phase III investigation evaluated the transfer-of-training benefit imparted by the PALAT system using a broadband stimulus to a field environment using gunshot stimulus. Training under the open ear and in-the-ear TCAPS resulted in significant differences between the trained and untrained groups from in-office pretest to in-field posttest

Development and Human Factors Evaluation of a Portable Auditory Localization Acclimation Training System

Auditory situation awareness (ASA) is essential for safety and survivability in military operations where many of the hazards are not immediately visible. Unfortunately, the Hearing Protection Devices (HPDs) required to operate in these environments can impede auditory localization performance. Promisingly, recent studies have exhibited the plasticity of the human auditory system by demonstrating that training can improve auditory localization ability while wearing HPDs, including military Tactical Communications and Protective Systems (TCAPS). As a result, the U.S. military identified the need for a portable system capable of imparting auditory localization acquisition skills at similar levels to those demonstrated in laboratory environments. The purpose of this investigation was to develop and validate a Portable Auditory Localization Acclimation Training (PALAT) system equipped with an improved training protocol against a proven laboratory grade system referred to as the DRILCOM system and subsequently evaluate the transfer-of-training benefit in a field environment. In Phase I, a systems decision process was used to develop a prototype PALAT system consisting of an expandable frame housing 32-loudspeakers operated by a user-controlled tablet computer capable of reproducing acoustically accurate localization cues similar to the DRILCOM system. Phase II used a within-subjects human factors experiment to validate whether the PALAT system could impart similar auditory localization training benefits as the DRILCOM system. Results showed no significant difference between the two localization training systems at each stage of training or in training rates for the open ear and with two TCAPS devices. The PALAT system also demonstrated the ability to detect differences in localization accuracy between listening conditions in the same manner as the DRILCOM system. Participant ratings indicated no perceived difference in localization training benefit but significantly preferred the PALAT system user interface which was specifically designed to improve usability features to meet requirements of a user operable system. The Phase III investigation evaluated the transfer-of-training benefit imparted by the PALAT system using a broadband stimulus to a field environment using gunshot stimulus. Training under the open ear and in-the-ear TCAPS resulted in significant differences between the trained and untrained groups from in-office pretest to in-field posttest

Audio Spatio-Temporal Fingerprints for Cloudless Real-Time Hands-Free Diarization on Mobile Devices

In this paper, we propose a new low bit rate representation of a sound field and a new method for the corresponding cloudless low delay hands-free diarization suitable for low-performance mobile devices, e.g. mobile phones. The proposed audio spatio-temporal fingerprint representation results in low bit rate (500 bytes/second), however contains complete information about continuous audio tracking of multiple acoustic sources in an open, unconstrained environment. The core of the algorithm is based on simultaneous multiple data stream processing using audio spatio-temporal fingerprint representation to cover higher level events relevant for diarization, e.g. turns, interruptions, crosstalk, speech and non-speech segments. Performance levels achieved to date on 5 hours of hand-labelled datasets have shown the feasibility of the approach at the same time as resulting in 7.58% CPU load on 1-core ultra-low-power mobile processor running at 1 GHz and low algorithmic delay of 112 ms

Acoustic event detection and localization using distributed microphone arrays

Automatic acoustic scene analysis is a complex task that involves several functionalities: detection (time), localization (space), separation, recognition, etc. This thesis focuses on both acoustic event detection (AED) and acoustic source localization (ASL), when several sources may be simultaneously present in a room. In particular, the experimentation work is carried out with a meeting-room scenario. Unlike previous works that either employed models of all possible sound combinations or additionally used video signals, in this thesis, the time overlapping sound problem is tackled by exploiting the signal diversity that results from the usage of multiple microphone array beamformers. The core of this thesis work is a rather computationally efficient approach that consists of three processing stages. In the first, a set of (null) steering beamformers is used to carry out diverse partial signal separations, by using multiple arbitrarily located linear microphone arrays, each of them composed of a small number of microphones. In the second stage, each of the beamformer output goes through a classification step, which uses models for all the targeted sound classes (HMM-GMM, in the experiments). Then, in a third stage, the classifier scores, either being intra- or inter-array, are combined using a probabilistic criterion (like MAP) or a machine learning fusion technique (fuzzy integral (FI), in the experiments). The above-mentioned processing scheme is applied in this thesis to a set of complexity-increasing problems, which are defined by the assumptions made regarding identities (plus time endpoints) and/or positions of sounds. In fact, the thesis report starts with the problem of unambiguously mapping the identities to the positions, continues with AED (positions assumed) and ASL (identities assumed), and ends with the integration of AED and ASL in a single system, which does not need any assumption about identities or positions. The evaluation experiments are carried out in a meeting-room scenario, where two sources are temporally overlapped; one of them is always speech and the other is an acoustic event from a pre-defined set. Two different databases are used, one that is produced by merging signals actually recorded in the UPC¿s department smart-room, and the other consists of overlapping sound signals directly recorded in the same room and in a rather spontaneous way. From the experimental results with a single array, it can be observed that the proposed detection system performs better than either the model based system or a blind source separation based system. Moreover, the product rule based combination and the FI based fusion of the scores resulting from the multiple arrays improve the accuracies further. On the other hand, the posterior position assignment is performed with a very small error rate. Regarding ASL and assuming an accurate AED system output, the 1-source localization performance of the proposed system is slightly better than that of the widely-used SRP-PHAT system, working in an event-based mode, and it even performs significantly better than the latter one in the more complex 2-source scenario. Finally, though the joint system suffers from a slight degradation in terms of classification accuracy with respect to the case where the source positions are known, it shows the advantage of carrying out the two tasks, recognition and localization, with a single system, and it allows the inclusion of information about the prior probabilities of the source positions. It is worth noticing also that, although the acoustic scenario used for experimentation is rather limited, the approach and its formalism were developed for a general case, where the number and identities of sources are not constrained

Optimization and improvements in spatial sound reproduction systems through perceptual considerations

[ES] La reproducción de las propiedades espaciales del sonido es una cuestión cada vez más importante en muchas aplicaciones inmersivas emergentes. Ya sea en la reproducción de contenido audiovisual en entornos domésticos o en cines, en sistemas de videoconferencia inmersiva o en sistemas de realidad virtual o aumentada, el sonido espacial es crucial para una sensación de inmersión realista. La audición, más allá de la física del sonido, es un fenómeno perceptual influenciado por procesos cognitivos. El objetivo de esta tesis es contribuir con nuevos métodos y conocimiento a la optimización y simplificación de los sistemas de sonido espacial, desde un enfoque perceptual de la experiencia auditiva. Este trabajo trata en una primera parte algunos aspectos particulares relacionados con la reproducción espacial binaural del sonido, como son la escucha con auriculares y la personalización de la Función de Transferencia Relacionada con la Cabeza (Head Related Transfer Function - HRTF). Se ha realizado un estudio sobre la influencia de los auriculares en la percepción de la impresión espacial y la calidad, con especial atención a los efectos de la ecualización y la consiguiente distorsión no lineal. Con respecto a la individualización de la HRTF se presenta una implementación completa de un sistema de medida de HRTF y se introduce un nuevo método para la medida de HRTF en salas no anecoicas. Además, se han realizado dos experimentos diferentes y complementarios que han dado como resultado dos herramientas que pueden ser utilizadas en procesos de individualización de la HRTF, un modelo paramétrico del módulo de la HRTF y un ajuste por escalado de la Diferencia de Tiempo Interaural (Interaural Time Difference - ITD). En una segunda parte sobre reproducción con altavoces, se han evaluado distintas técnicas como la Síntesis de Campo de Ondas (Wave-Field Synthesis - WFS) o la panoramización por amplitud. Con experimentos perceptuales se han estudiado la capacidad de estos sistemas para producir sensación de distancia y la agudeza espacial con la que podemos percibir las fuentes sonoras si se dividen espectralmente y se reproducen en diferentes posiciones. Las aportaciones de esta investigación pretenden hacer más accesibles estas tecnologías al público en general, dada la demanda de experiencias y dispositivos audiovisuales que proporcionen mayor inmersión.[CA] La reproducció de les propietats espacials del so és una qüestió cada vegada més important en moltes aplicacions immersives emergents. Ja siga en la reproducció de contingut audiovisual en entorns domèstics o en cines, en sistemes de videoconferència immersius o en sistemes de realitat virtual o augmentada, el so espacial és crucial per a una sensació d'immersió realista. L'audició, més enllà de la física del so, és un fenomen perceptual influenciat per processos cognitius. L'objectiu d'aquesta tesi és contribuir a l'optimització i simplificació dels sistemes de so espacial amb nous mètodes i coneixement, des d'un criteri perceptual de l'experiència auditiva. Aquest treball tracta, en una primera part, alguns aspectes particulars relacionats amb la reproducció espacial binaural del so, com són l'audició amb auriculars i la personalització de la Funció de Transferència Relacionada amb el Cap (Head Related Transfer Function - HRTF). S'ha realitzat un estudi relacionat amb la influència dels auriculars en la percepció de la impressió espacial i la qualitat, dedicant especial atenció als efectes de l'equalització i la consegüent distorsió no lineal. Respecte a la individualització de la HRTF, es presenta una implementació completa d'un sistema de mesura de HRTF i s'inclou un nou mètode per a la mesura de HRTF en sales no anecoiques. A mès, s'han realitzat dos experiments diferents i complementaris que han donat com a resultat dues eines que poden ser utilitzades en processos d'individualització de la HRTF, un model paramètric del mòdul de la HRTF i un ajustament per escala de la Diferencià del Temps Interaural (Interaural Time Difference - ITD). En una segona part relacionada amb la reproducció amb altaveus, s'han avaluat distintes tècniques com la Síntesi de Camp d'Ones (Wave-Field Synthesis - WFS) o la panoramització per amplitud. Amb experiments perceptuals, s'ha estudiat la capacitat d'aquests sistemes per a produir una sensació de distància i l'agudesa espacial amb que podem percebre les fonts sonores, si es divideixen espectralment i es reprodueixen en diferents posicions. Les aportacions d'aquesta investigació volen fer més accessibles aquestes tecnologies al públic en general, degut a la demanda d'experiències i dispositius audiovisuals que proporcionen major immersió.[EN] The reproduction of the spatial properties of sound is an increasingly important concern in many emerging immersive applications. Whether it is the reproduction of audiovisual content in home environments or in cinemas, immersive video conferencing systems or virtual or augmented reality systems, spatial sound is crucial for a realistic sense of immersion. Hearing, beyond the physics of sound, is a perceptual phenomenon influenced by cognitive processes. The objective of this thesis is to contribute with new methods and knowledge to the optimization and simplification of spatial sound systems, from a perceptual approach to the hearing experience. This dissertation deals in a first part with some particular aspects related to the binaural spatial reproduction of sound, such as listening with headphones and the customization of the Head Related Transfer Function (HRTF). A study has been carried out on the influence of headphones on the perception of spatial impression and quality, with particular attention to the effects of equalization and subsequent non-linear distortion. With regard to the individualization of the HRTF a complete implementation of a HRTF measurement system is presented, and a new method for the measurement of HRTF in non-anechoic conditions is introduced. In addition, two different and complementary experiments have been carried out resulting in two tools that can be used in HRTF individualization processes, a parametric model of the HRTF magnitude and an Interaural Time Difference (ITD) scaling adjustment. In a second part concerning loudspeaker reproduction, different techniques such as Wave-Field Synthesis (WFS) or amplitude panning have been evaluated. With perceptual experiments it has been studied the capacity of these systems to produce a sensation of distance, and the spatial acuity with which we can perceive the sound sources if they are spectrally split and reproduced in different positions. The contributions of this research are intended to make these technologies more accessible to the general public, given the demand for audiovisual experiences and devices with increasing immersion.Gutiérrez Parera, P. (2020). Optimization and improvements in spatial sound reproduction systems through perceptual considerations [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/142696TESI

Effizientes binaurales Rendering von virtuellen akustischen Realitäten : technische und wahrnehmungsbezogene Konzepte

Binaural rendering aims to immerse the listener in a virtual acoustic scene, making it an essential method for spatial audio reproduction in virtual or augmented reality (VR/AR) applications. The growing interest and research in VR/AR solutions yielded many different methods for the binaural rendering of virtual acoustic realities, yet all of them share the fundamental idea that the auditory experience of any sound field can be reproduced by reconstructing its sound pressure at the listener's eardrums. This thesis addresses various state-of-the-art methods for 3 or 6 degrees of freedom (DoF) binaural rendering, technical approaches applied in the context of headphone-based virtual acoustic realities, and recent technical and psychoacoustic research questions in the field of binaural technology. The publications collected in this dissertation focus on technical or perceptual concepts and methods for efficient binaural rendering, which has become increasingly important in research and development due to the rising popularity of mobile consumer VR/AR devices and applications. The thesis is organized into five research topics: Head-Related Transfer Function Processing and Interpolation, Parametric Spatial Audio, Auditory Distance Perception of Nearby Sound Sources, Binaural Rendering of Spherical Microphone Array Data, and Voice Directivity. The results of the studies included in this dissertation extend the current state of research in the respective research topic, answer specific psychoacoustic research questions and thereby yield a better understanding of basic spatial hearing processes, and provide concepts, methods, and design parameters for the future implementation of technically and perceptually efficient binaural rendering.Binaurales Rendering zielt darauf ab, dass der Hörer in eine virtuelle akustische Szene eintaucht, und ist somit eine wesentliche Methode für die räumliche Audiowiedergabe in Anwendungen der virtuellen Realität (VR) oder der erweiterten Realität (AR – aus dem Englischen Augmented Reality). Das wachsende Interesse und die zunehmende Forschung an VR/AR-Lösungen führte zu vielen verschiedenen Methoden für das binaurale Rendering virtueller akustischer Realitäten, die jedoch alle die grundlegende Idee teilen, dass das Hörerlebnis eines beliebigen Schallfeldes durch die Rekonstruktion seines Schalldrucks am Trommelfell des Hörers reproduziert werden kann. Diese Arbeit befasst sich mit verschiedenen modernsten Methoden zur binauralen Wiedergabe mit 3 oder 6 Freiheitsgraden (DoF – aus dem Englischen Degree of Freedom), mit technischen Ansätzen, die im Kontext kopfhörerbasierter virtueller akustischer Realitäten angewandt werden, und mit aktuellen technischen und psychoakustischen Forschungsfragen auf dem Gebiet der Binauraltechnik. Die in dieser Dissertation gesammelten Publikationen befassen sich mit technischen oder wahrnehmungsbezogenen Konzepten und Methoden für effizientes binaurales Rendering, was in der Forschung und Entwicklung aufgrund der zunehmenden Beliebtheit von mobilen Verbraucher-VR/AR-Geräten und -Anwendungen zunehmend an Relevanz gewonnen hat. Die Arbeit ist in fünf Forschungsthemen gegliedert: Verarbeitung und Interpolation von Außenohrübertragungsfunktionen, parametrisches räumliches Audio, auditive Entfernungswahrnehmung ohrnaher Schallquellen, binaurales Rendering von sphärischen Mikrofonarraydaten und Richtcharakteristik der Stimme. Die Ergebnisse der in dieser Dissertation enthaltenen Studien erweitern den aktuellen Forschungsstand im jeweiligen Forschungsfeld, beantworten spezifische psychoakustische Forschungsfragen und führen damit zu einem besseren Verständnis grundlegender räumlicher Hörprozesse, und liefern Konzepte, Methoden und Gestaltungsparameter für die zukünftige Umsetzung eines technisch und wahrnehmungsbezogen effizienten binauralen Renderings.BMBF, 03FH014IX5, Natürliche raumbezogene Darbietung selbsterzeugter Schallereignisse in virtuellen auditiven Umgebungen (NarDasS