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Les terminaux mobiles offrent à ce jour des performances de plus en plus élevées (CPU, résolution de l’écran, capteurs optiques, etc.) Cela rehausse la qualité vidéo des services média, que ce soit pour le visionnage de contenu vidéo (streaming, TV, etc.) ou pour des applications interactives telles que le jeu vidéo. Mais cette évolution concernant l'image n'est pas ou peu suivie par l'intégration de systèmes de restitution audio de haute qualité dans ce type de terminal. Or, parallèlement à ces évolutions concernant l'image, des solutions de son spatialisé sur casque, à travers notamment la technique de restitution binaurale basée sur l'utilisation de filtres HRTF (Head Related Transfer Functions) voient le jour.Dans ce travail de thèse, nous nous proposons d’évaluer l’intérêt que peut présenter le son binaural lorsqu'il est utilisé sur une application mobile audiovisuelle. Une partie de notre travail a consisté à déterminer les différents sens que l’on pouvait donner au terme « application mobile audiovisuelle » et parmi ces sens ceux qui d’une part étaient pertinents et d’autre part pouvaient donner lieu à une évaluation comparative avec ou sans son binaural.Le couplage entre son binaural et visuel sur mobile occasionne en premier lieu une question d’ordre perceptive : comment peut-on organiser spatialement une scène virtuelle dont le son peut se déployer tout autour de l’utilisateur, et dont le visuel est restreint à un si petit écran ? La première partie de cette thèse est consacrée à cette question. Nous menons une expérience visant à étudier le découplage spatial possible entre un son binaural et un visuel rendus sur smartphone. Cette expérience révèle une forte tolérance de l’être humain face aux dégradations spatiales pouvant survenir entre les deux modalités. En particulier, l’absence d’individualisation des HRTF, ainsi qu’un très grand découplage en élévation ne semblent pas affecter la perception. Par ailleurs, les sujets semblent envisager la scène « comme si » ils y étaient eux-mêmes directement projetés, à la place de la caméra, et cela indépendamment de leur propre distance à l’écran. Tous ces résultats suggèrent la possibilité d’une association entre son binaural et visuel sur mobile dans des conditions d’utilisation proches du grand public.Dans la seconde partie de la thèse, nous tentons de répondre à la question de l’apport du binaural en déployant une expérience « hors les murs », dans un contexte plausible d’utilisation grand public. Trente sujets jouent dans leur vie quotidienne à un jeu vidéo de type Infinite Runner, développé pour l’occasion en deux versions, une avec du son binaural, et l’autre avec du son monophonique. L’expérience dure cinq semaines, à raison de deux sessions par jour. Ce protocole procède de la méthode dite "Experience Sampling Method", sur l’état de l’art de laquelle nous nous sommes appuyés. Nous calculons à chaque session des notes d’immersion, de mémorisation et de performance, et nous comparons les notes obtenues entre les deux versions sonores. Les résultats indiquent une immersion significativement meilleure pour le binaural. La mémorisation et la performance ne sont en revanche pas soumises à un effet statistiquement significatif du rendu sonore. Au-delà des résultats, cette expérience nous permet de discuter de la question de la validité des données en fonction de la méthode de déploiement, en confrontant notamment bien-fondé théorique et faisabilité pratique. / In recent years, smartphone and tablet global performances have been increased significantly (CPU, screen resolution, webcams, etc.). This can be particularly observed with video quality of mobile media services, such as video streaming applications, or interactive applications (e.g., video games). However, these evolutions barely go with the integration of high quality sound restitution systems. Beside these evolutions though, new technologies related to spatialized sound on headphones have been developed, namely the binaural restitution model, using HRTF (Head Related Transfer Functions) filters.In this thesis, we assess the potential contribution of the binaural technology to enhance the quality of experience of an audiovisual mobile application. A part of our work has been dedicated to define what is an “audiovisual mobile application”, what kind of application could be fruitfully experienced with a binaural sound, and among those applications which one could lead to a comparative experiment with and without binaural.In a first place, the coupling of a binaural sound with a mobile-rendered visual tackles a question related to perception: how to spatially arrange a virtual scene whose sound can be spread all around the user, while its visual is limited to a very small space? We propose an experiment in these conditions to study how far a sound and a visual can be moved apart without breaking their perceptual fusion. The results reveal a strong tolerance of subjects to spatial discrepancies between the two modalities. Notably, the absence or presence of individualization for the HRTF filters, and a large separation in elevation between sound and visual don’t seem to affect the perception. Besides, subjects consider the virtual scene as if they were projected inside, at the camera’s position, no matter what distance to the phone they sit. All these results suggest that an association between a binaural sound and a visual on a smartphone could be used by the general public.In the second part, we address the main question of the thesis, i.e., the contribution of binaural, and we conduct an experiment in a realistic context of use. Thirty subjects play an Infinite Runner video game in their daily lives. The game was developed for the occasion in two versions, a monophonic one and a binaural one. The experiment lasts five weeks, at a rate of two sessions per day, which relates to a protocol known as the “Experience Sampling Method”. We collect at each session notes of immersion, memorization and performance, and compare the notes between the monophonic sessions and the binaural ones. Results indicate a significantly better immersion in the binaural sessions. No effect of sound rendering was found for memorization and performance. Beyond the contribution of the binaural, we discuss about the protocol, the validity of the collected data, and oppose theoretical considerations to practical feasibility.

Orientador: João Antônio Pereira / Resumo: O presente trabalho trata do estudo de uma fonte acústica com direcionamento sonoro controlado. A proposta envolve o desenvolvimento do projeto de um equipamento de teste para uso biomédico na aplicação e análises fonoaudiológicas, em que é proposto o dimensionamento de uma fonte acústica direcional, com adequada isolação e direcionamento do som emitido em relação ao posicionamento do ouvido do paciente, para ser usada dentro de um ambiente acusticamente isolado, dimensionado para permitir os testes, treinamentos e análise dos exames fonoaudiológicos relativos às percepções espaciais binaurais. / Abstract: The present work deals with the study of an acoustic source with controlled sound direction. The proposal involves the development of design a test equipment for biomedical use in the application and phonoaudiological analyzes, in which it is proposed dimensioning of a directional acoustic source, with adequate isolation and directness of the of the emitted sound in relation to the positioning of patient's ear, To be used within an acoustically isolated environment, sized to allow the testing, training and analysis of speech-language pathology examinations related to binaural spatial perceptions. / Mestre

An interpretative methodology for understanding meaning in cinema since the 1950s, auteur analysis is an approach to film studies in which an individual, usually the director, is studied as the author of her or his films. The principal argument of this thesis is that proponents of auteurism have privileged examination of the visual components in a film-maker’s body of work, neglecting the potentially significant role played by sound. The thesis seeks to address this problematic imbalance by interrogating the creative use of sound in the films written and directed by Rolf de Heer, asking the question, “Does his use of sound make Rolf de Heer an aural auteur?” In so far as the term ‘aural’ encompasses everything in the film that is heard by the audience, the analysis seeks to discover if de Heer has, as Peter Wollen suggests of the auteur and her or his directing of the visual components (1968, 1972 and 1998), unconsciously left a detectable aural signature on his films. The thesis delivers an innovative outcome by demonstrating that auteur analysis that goes beyond the mise-en-scène (i.e. visuals) is productive and worthwhile as an interpretive response to film. De Heer’s use of the aural point of view and binaural sound recording, his interest in providing a ‘voice’ for marginalised people, his self-penned song lyrics, his close and early collaboration with composer Graham Tardif and sound designer Jim Currie, his ‘hands-on’ approach to sound recording and sound editing and his predilection for making films about sound are all shown to be examples of de Heer’s aural auteurism. As well as the three published (or accepted for publication) interviews with de Heer, Tardif and Currie, the dissertation consists of seven papers refereed and published (or accepted for publication) in journals and international conference proceedings, a literature review and a unifying essay. The papers presented are close textual analyses of de Heer’s films which, when considered as a whole, support the thesis’ overall argument and serve as a comprehensive auteur analysis, the first such sustained study of his work, and the first with an emphasis on the aural.

This thesis analyses if the change of auditory feedback can improve the effectiveness of performance in the interaction with a non-visual system, or with a system used by individuals with visual impairment. Two prototypes were developed, one with binaural audio and the other with stereo audio. The interaction was evaluated in an experiment where 22 participants, divided into two groups, performed a number of interaction tasks. A post-interview were conducted together with the experiment. The result of the experiment displayed that there were no great difference between binaural audio and stereo regarding the speed and accuracy of the interaction. The post-interviews displayed interesting differences in the way participants visualized the virtual environment that affected the interaction. This opened up interesting questions for future studies.

The main objective of this study is to learn about the audiovisual stimulator and to create hardware resolution of stimulation LED glasses and in environment of the program LabView application, which operate this stimulation LED glasses and in the same time create sound of stimulation. Use environment of the program LabView. Application, which is create in environment of the program LabView, enable operate stimulation LED glasses and arrange sound from three source with two different method, which use modern AVS. Application contains a lot of security, informative and agreement components.

The main focus of this thesis is to analyse signals (signal-dependent analysis) and room responses (system-dependent analysis) from a statistical point of view, attempt to determine the underlying statistical relationships between the reverberant signals and the room responses and propose relevant statistical models. Based on such a statistical framework, this thesis aims to propose novel methodologies for the extraction of room acoustical information and parameters from reverberant signals. Schroeder's theory is experimentally evaluated for various Room Transfer Functions (RTFs) measured in many source/receiver positions in various enclosures and several related aspects are discussed. Using a statistical approach, the effects of reverberant energy on the histograms and statistical measures are discussed and models describing the relationship of statistical measures between the reverberant signal and the RTFs are extracted. Then, the statistical properties of Binaural Room Transfer Functions (BRTFs) and binaural cues are examined. The well-known property of the spectral standard deviation of the magnitude of RTFs, that is its convergence to 5.6 dB for diffuse fields, is examined for the case of BRTFs, using a similar approach and a generic model for the relationship of the spectral standard deviation of RTFs and BRTFs. This thesis is also concerned with the distance estimation problem from a perceptual and computational point of view. Two novel methods for the estimation of the source/receiver distance using speech signals are proposed. The first method is able to detect the distance between the speaker and the microphone in a room environment using single-channel signals. The distance-dependent variation of several temporal and spectral statistical features of single-channel signals is studied and a novel sound source distance detector, based on these features is developed. The second method estimates distance from binaural speech signals (two-channel signals). This method does not require a priori knowledge of the room impulse response, the reverberation time or any other acoustical parameter and relies on a set of novel features extracted from the reverberant binaural signals. For this method, a novel distance estimation feature is introduced exploiting the standard deviation of the difference of the magnitude spectra of the left and right binaural signals (termed here as Binaural Spectral Magnitude Difference Standard Deviation (BSMD STD)). Moreover, an extended and novel set of additional features based on the statistical properties of binaural cues (ILDs, ITDs, ICs) is extracted from an auditory front-end which models the peripheral processing of the human auditory system. Both methods rely on novel distance-dependent features, related to statistical parameters of speech signals. Finally, a novel method for the estimation of the direct-to-reverberant-ratio (DRR) from dual-channel microphone recordings without having knowledge of the source signal is presented. / Η παρούσα διατριβή ασχολείται με τη μελέτη και ανάλυση των στατιστικών χαρακτηριστικών ηχητικών σημάτων και των ακουστικών αποκρίσεων χώρου, έχοντας ως πρωταρχικό σκοπό να προτείνει σχέσεις που περιγράφουν τη συσχέτιση των στατιστικών χαρακτηριστικών των σημάτων με αντήχηση με τις ακουστικές αποκρίσεις χώρων. Βάσει ενός τέτοιου θεωρητικού πλαισίου, η διατριβή αυτή αποσκοπεί στο να προτείνει νέες μεθοδολογίες για την εξαγωγή πληροφορίας που σχετίζεται με τα ακουστικά χαρακτηριστικά των χώρων, κάνοντας χρήση ηχογραφημένων ηχητικών σημάτων (π.χ. σήματα ομιλίας) στους εκάστοτε κλειστούς χώρους. Το θεωρητικό υπόβαθρο αυτής της διατριβής βασίζεται σε υπάρχοντα θεωρητικά μοντέλα για το ηχητικό πεδίο μέσα σε ένα κλειστό χώρο, όπως, για παράδειγμα, το στατιστικό μοντέλο του Schroeder. Το μοντέλο του Schroeder επιβεβαιώνεται πειραματικά για ακουστικές αποκρίσεις που έχουν μετρηθεί σε διάφορες θέσεις, μέσα σε κλειστούς χώρους, οι οποίοι διαφέρουν στα ακουστικά χαρακτηριστικά τους. Βάσει στατιστικής ανάλυσης, εξάγονται στατιστικά μοντέλα, τα οποία περιγράφουν την επίδραση της αντήχησης στα ηχητικά σήματα, όταν αυτά αναπαραχθούν μέσα σε ένα κλειστό χώρο. Στη συνέχεια, λαμβάνοντας υπόψη αντιληπτικά μοντέλα ακοής, τα οποία προϋποθέτουν την ύπαρξη δυο ηχητικών σημάτων (δυο αυτιά, αμφιωτική ακοή) σε αυτή τη διατριβή, μελετώνται κάποιες παράμετροι οι οποίες εξάγονται από αμφιωτικές ακουστικές αποκρίσεις χώρου. Η ιδιότητα της φασματικής τυπικής απόκλισης συναρτήσεων μεταφοράς χώρων να συγκλίνει στην τιμή των 5.6~dB για διάχυτα ηχητικά πεδία, επεκτείνεται στην περίπτωση των αμφιωτικών αποκρίσεων χώρου και προτείνεται ένα γενικευμένο μοντέλο που συσχετίζει τη φασματική τυπική απόκλιση μονοφωνικών και αμφιωτικών συναρτήσεων μεταφοράς χώρου. Η διατριβή αυτή, επίσης, ασχολείται με το πρόβλημα της εκτίμησης της απόστασης μεταξύ πηγής και δέκτη. Προτείνονται δυο νέες μέθοδοι για την εκτίμηση της απόστασης μεταξύ πηγής και δέκτη, κάνοντας χρήση ηχητικών σημάτων ομιλίας. Η προτεινόμενη μέθοδος βασίζεται σε μια σειρά από στατιστικές παραμέτρους των οποίων οι τιμές μεταβάλλονται είτε στο πεδίο του χρόνου είτε στο πεδίο της συχνότητας. Η δεύτερη προτεινόμενη μέθοδος αφορά, επίσης, στην εκτίμηση της απόστασης πηγής/δέκτη, αλλά από αμφιωτικά σήματα. Η μέθοδος αυτή δεν προαπαιτεί γνώση της ακουστικής απόκρισης του χώρου, του χρόνου αντήχησης ή άλλης ακουστικής παραμέτρου και βασίζεται σε μια σειρά από νέες παραμέτρους, οι οποίες μπορούν να υπολογισθούν από τα αμφιωτικά σήματα με αντήχηση. Οι παράμετροι συνδυάζονται με δυο διαφορετικές τεχνικές αναγνώρισης προτύπων των οποίων τα μειονεκτήματα και πλεονεκτήματα συζητώνται. Στα πλαίσια αυτής της μεθόδου, προτείνεται μια νέα παράμετρος, η οποία βασίζεται στη διαφορά της φασματικής τυπικής απόκλισης του αριστερού και του δεξιού αμφιωτικού ηχητικού σήματος, η οποία αποδεικνύεται ότι σχετίζεται με τα στατιστικά της αντίστοιχης μονοφωνικής ακουστικής απόκρισης. Τέλος, προτείνεται μια σειρά από παραμέτρους οι οποίες βασίζονται στα στατιστικά χαρακτηριστικά αμφιωτικών παραμέτρων και σχετίζονται με το αντιληπτικό μοντέλο της ανθρώπινης ακοής. Τέλος, προτείνεται μια νέα μέθοδος για την εκτίμηση της στάθμης λόγου κατευθείαν προς ανακλώμενου ήχου από στερεοφωνικά σήματα.

Fundação Carlos Chagas Filho de Amparo a Pesquisa do Estado do Rio de Janeiro / Neste trabalho é apresentada uma nova abordagem para obter as respostas impulsivas biauriculares (BIRs) para um sistema de aurilização utilizando um conjunto de redes neurais artificiais (RNAs). O método proposto é capaz de reconstruir as respostas impulsivas associadas à cabeça humana (HRIRs) por meio de modificação espectral e de interpolação espacial. A fim de cobrir todo o espaço auditivo de recepção, sem aumentar a complexidade da arquitetura da rede, uma estrutura com múltiplas RNAs (conjunto) foi adotada, onde cada rede opera uma região específica do espaço (gomo). Os três principais fatores que influenciam na precisão do modelo arquitetura da rede, ângulos de abertura da área de recepção e atrasos das HRIRs são investigados e uma configuração ideal é apresentada. O erro de modelagem no domínio da frequência é investigado considerando a natureza logarítmica da audição humana. Mais ainda, são propostos novos parâmetros para avaliação do erro, definidos em analogia com alguns dos bem conhecidos parâmetros de qualidade acústica de salas. Através da metodologia proposta obteve-se um ganho computacional, em redução do tempo de processamento, de aproximadamente 62% em relação ao método tradicional de processamento de sinais utilizado para aurilização. A aplicabilidade do novo método em sistemas de aurilização é reforçada mediante uma análise comparativa dos resultados, que incluem a geração das BIRs e o cálculo dos parâmetros acústicos biauriculares (IACF e IACC), os quais mostram erros de magnitudes reduzidas. / This work presents a new approach to obtain the Binaural Impulse Responses (BIRs) for an auralization system by using a committee of artificial neural networks (ANNs). The proposed method is capable to reconstruct the desired modified Head Related Impulse Responses (HRIRs) by means of spectral modification and spatial interpolation. In order to cover the entire auditory reception space, without increasing the networks architecture complexity, a structure with multiple RNAs (group) was adopted, where each network operates in a specific reception region (bud). The three major parameters that affect the models accuracy the networks architecture, the reception regions aperture angles and the HRIRs time shifts are investigated and an optimal setup is presented. The modeling error, in the frequency domain, is investigated considering the logarithmic nature of the human hearing. Moreover, new parameters are proposed for error evaluation in the time domain, defined in analogy with some of the well known acoustical quality parameters in rooms. It was observed that the proposed methodology obtained a computational gain of approximately 62%, in terms of processing time reduction, compared to the classical signal processing method used to obtain auralizations. The applicability of the new method in auralizations systems is validated by comparative analysis, which includes the BIRs generation and calculation of the binaural acoustic parameters (IACF and IACC), showing very low magnitude errors.

Fundação Carlos Chagas Filho de Amparo a Pesquisa do Estado do Rio de Janeiro / Neste trabalho é apresentada uma nova abordagem para obter as respostas impulsivas biauriculares (BIRs) para um sistema de aurilização utilizando um conjunto de redes neurais artificiais (RNAs). O método proposto é capaz de reconstruir as respostas impulsivas associadas à cabeça humana (HRIRs) por meio de modificação espectral e de interpolação espacial. A fim de cobrir todo o espaço auditivo de recepção, sem aumentar a complexidade da arquitetura da rede, uma estrutura com múltiplas RNAs (conjunto) foi adotada, onde cada rede opera uma região específica do espaço (gomo). Os três principais fatores que influenciam na precisão do modelo arquitetura da rede, ângulos de abertura da área de recepção e atrasos das HRIRs são investigados e uma configuração ideal é apresentada. O erro de modelagem no domínio da frequência é investigado considerando a natureza logarítmica da audição humana. Mais ainda, são propostos novos parâmetros para avaliação do erro, definidos em analogia com alguns dos bem conhecidos parâmetros de qualidade acústica de salas. Através da metodologia proposta obteve-se um ganho computacional, em redução do tempo de processamento, de aproximadamente 62% em relação ao método tradicional de processamento de sinais utilizado para aurilização. A aplicabilidade do novo método em sistemas de aurilização é reforçada mediante uma análise comparativa dos resultados, que incluem a geração das BIRs e o cálculo dos parâmetros acústicos biauriculares (IACF e IACC), os quais mostram erros de magnitudes reduzidas. / This work presents a new approach to obtain the Binaural Impulse Responses (BIRs) for an auralization system by using a committee of artificial neural networks (ANNs). The proposed method is capable to reconstruct the desired modified Head Related Impulse Responses (HRIRs) by means of spectral modification and spatial interpolation. In order to cover the entire auditory reception space, without increasing the networks architecture complexity, a structure with multiple RNAs (group) was adopted, where each network operates in a specific reception region (bud). The three major parameters that affect the models accuracy the networks architecture, the reception regions aperture angles and the HRIRs time shifts are investigated and an optimal setup is presented. The modeling error, in the frequency domain, is investigated considering the logarithmic nature of the human hearing. Moreover, new parameters are proposed for error evaluation in the time domain, defined in analogy with some of the well known acoustical quality parameters in rooms. It was observed that the proposed methodology obtained a computational gain of approximately 62%, in terms of processing time reduction, compared to the classical signal processing method used to obtain auralizations. The applicability of the new method in auralizations systems is validated by comparative analysis, which includes the BIRs generation and calculation of the binaural acoustic parameters (IACF and IACC), showing very low magnitude errors.

De nombreuses études ont été menées lors des vingt dernières années dans le domaine de l’auralisation.Elles consistent à rendre audible les résultats d’une simulation acoustique. Ces études se sont majoritairementfocalisées sur les algorithmes de propagation et la restitution du champ acoustique dans desenvironnements complexes. Actuellement, de nombreux travaux portent sur le rendu sonore en tempsréel.Cette thèse aborde la problématique du rendu sonore dynamique d’environnements complexes selonquatre axes : la propagation des ondes sonores, le traitement du signal, la perception spatiale du son etl’optimisation informatique. Dans le domaine de la propagation, une méthode permettant d’analyser lavariété des algorithmes présents dans la bibliographie est proposée. A partir de cette méthode d’analyse,deux algorithmes dédiés à la restitution en temps réel des champs spéculaires et diffus ont été extraits.Dans le domaine du traitement du signal, la restitution est réalisée à l’aide d’un algorithme optimisé despatialisation binaurale pour les chemins spéculaires les plus significatifs et un algorithme de convolutionsur carte graphique pour la restitution du champ diffus. Les chemins les plus significatifs sont extraitsgrace à un modèle perceptif basé sur le masquage temporel et spatial des contributions spéculaires.Finalement, l’implémentation de ces algorithmes sur des architectures parallèles récentes en prenant encompte les nouvelles architectures multi-coeurs et les nouvelles cartes graphiques est présenté. / During the past twenty years many studies have been conducted in the field of auralization, which aimsat rendering audible the results of an acoustic simulation. These studies have mainly focused on thepropagation algorithms and the sound field audio rendering for complex environments. Currently, muchresearch concentrates on real-time audio rendering.This thesis addresses the problematic of real-time audio rendering of complex environments accordingto four axes: sound propagation, Digital Signal Processing (DSP), spatial perception of sound andcomputational optimizations. In the field of propagation, a method that aims at analyzing the varietyof existing algorithms is proposed. This method yields two algorithms dedicated to the real-time propagationof both specular and diffuse information. In the field of DSP, the auralization is performed withan efficient binaural spatialization module for the most significant specular information, and a GPUconvolution algorithm for the diffuse sound field auralization. The most significant paths are extractedthanks to a perceptive model based on temporal and spatial masking of the specular contributions.Finally, the implementation of these algorithms on recent computer architectures, taking advantage ofthe parallel processing of the new CPUs, and the benefits of GPUs for DSP calculations is presented.