One of the mixing engineers most important tool when listening, analyzing and taking decisions is the physical space in which the reproduction system and the monitors are placed. The space needs to fulfill certain aspects to be able to qualify as a control room considered good for taking these decisions. In this study an active listening test was conducted to investigate if the RT60 of the natural reverberation in the room affected a mixing engineers’ decisions when adjusting parameters on an artificial reverb. 16 subjects participated, the subjects adjusted reverb level and reverb time of an artificial reverb in two different acoustical environments, with two different values of RT60. The environments were based on a professional control room made for mixing and mastering. The two values for RT60 was achieved through manipulating the room with diffusors and absorbing material. It was found that the subjects were able to adapt to the different acoustical environments, although other differences were found. The order of the which of the environments the subjects started in and which parameter the subject started adjusting, showed a considerable impact on the result. As well as what kind of factors the subjects considered when adjusting the artificial reverb.

This thesis presents novel signal processing algorithms for speech and music dereverberation. The proposed algorithms focus on blind single-channel suppression of late reverberation; however binaural and semi-blind methods have also been introduced. Late reverberation is a particularly harmful distortion, since it significantly decreases the perceived quality of the reverberant signals but also degrades the performance of Automatic Speech Recognition (ASR) systems and other speech and music processing algorithms. Hence, the proposed deverberation methods can be either used as standalone enhancing techniques or implemented as preprocessing schemes prior to ASR or other applied systems. The main dereverberation method proposed here is a blind dereverberation technique based on perceptual reverberation modeling has been developed. This technique employs a computational auditory masking model and locates the signal regions where late reverberation is audible, i.e. where it is unmasked from the clean signal components. Following a selective signal processing approach, only such signal regions are further processed through sub-band gain filtering. The above technique has been evaluated for both speech and music signals and for a wide range of reverberation conditions. In all cases it was found to minimize the processing artifacts and to produce perceptually superior clean signal estimations than any other tested technique. Moreover, extensive ASR tests have shown that it significantly improves the recognition performance, especially in highly reverberant environments. / Η διατριβή αποτελείται από εννιά κεφάλαια, δύο παραρτήματα καθώς και την σχετική βιβλιογραφία. Είναι γραμμένη στα αγγλικά ενώ περιλαμβάνει και ελληνική περίληψη. Στην παρούσα διατριβή, αναπτύσσονται μεθόδοι ψηφιακής επεξεργασίας σήματος για την αφαίρεση αντήχησης από σήματα ομιλίας και μουσικής. Οι προτεινόμενοι αλγόριθμοι καλύπτουν ένα μεγάλο εύρος εφαρμογών αρχικά εστιάζοντας στην τυφλή (“blind”) αφαίρεση για μονοκαναλικά σήματα. Στοχεύοντας σε πιο ειδικά σενάρια χρήσης προτείνονται επίσης αμφιωτικοί αλγόριθμοι αλλά και τεχνικές που προϋποθέτουν την πραγματοποίηση κάποιας ακουστικής μέτρησης. Οι αλγόριθμοι επικεντρώνουν στην αφαίρεση της καθυστερημένης αντήχησης που είναι ιδιαίτερα επιβλαβής για την ποιότητα σημάτων ομιλίας και μουσικής και μειώνει την καταληπτότητα της ομιλίας. Επίσης, επειδή αλλοιώνει σημαντικά τα στατιστικά των σημάτων, μειώνει σημαντικά την απόδοση συστημάτων αυτόματης αναγνώρισης ομιλίας καθώς και άλλων αλγορίθμων ψηφιακής επεξεργασίας ομιλίας και μουσικής. Έτσι οι προτεινόμενοι αλγόριθμοι μπορούν είτε να χρησιμοποιηθούν σαν αυτόνομες τεχνικές βελτίωσης της ποιότητας των ακουστικών σημάτων είτε να ενσωματωθούν σαν στάδια προ-επεξεργασίας σε άλλες εφαρμογές. Η κύρια μέθοδος αφαίρεσης αντήχησης που προτείνεται στην διατριβή, είναι βασισμένη στην αντιληπτική μοντελοποίηση και χρησιμοποιεί ένα σύγχρονο ψυχοακουστικό μοντέλο. Με βάση αυτό το μοντέλο γίνεται μία εκτίμηση των σημείων του σήματος που η αντήχηση είναι ακουστή δηλαδή που δεν επικαλύπτεται από το ισχυρότερο σε ένταση καθαρό από αντήχηση σήμα. Η συγκεκριμένη εκτίμηση οδηγεί σε μία επιλεκτική επεξεργασία σήματος όπου η αφαίρεση πραγματοποιείται σε αυτά και μόνο τα σημεία, μέσω πρωτότυπων υβριδικών συναρτήσεων κέρδους που βασίζονται σε δείκτες αντικειμενικής και υποκειμενικής αλλοίωσης. Εκτεταμένα αντικειμενικά και υποκειμενικά πειράματα δείχνουν ότι η προτεινόμενη τεχνική δίνει βέλτιστες ποιοτικά ανηχωικές εκτιμήσεις ανεξάρτητα από το μέγεθος του χώρου.

Ce mémoire de thèse de doctorat présente, discute et propose des outils de fouille automatique de mégadonnées dans un contexte de classification supervisée musical.L'application principale concerne la classification automatique des thèmes musicaux afin de générer des listes de lecture thématiques.Le premier chapitre introduit les différents contextes et concepts autour des mégadonnées musicales et de leur consommation.Le deuxième chapitre s'attelle à la description des bases de données musicales existantes dans le cadre d'expériences académiques d'analyse audio.Ce chapitre introduit notamment les problématiques concernant la variété et les proportions inégales des thèmes contenus dans une base, qui demeurent complexes à prendre en compte dans une classification supervisée.Le troisième chapitre explique l'importance de l'extraction et du développement de caractéristiques audio et musicales pertinentes afin de mieux décrire le contenu des éléments contenus dans ces bases de données.Ce chapitre explique plusieurs phénomènes psychoacoustiques et utilise des techniques de traitement du signal sonore afin de calculer des caractéristiques audio.De nouvelles méthodes d'agrégation de caractéristiques audio locales sont proposées afin d'améliorer la classification des morceaux.Le quatrième chapitre décrit l'utilisation des caractéristiques musicales extraites afin de trier les morceaux par thèmes et donc de permettre les recommandations musicales et la génération automatique de listes de lecture thématiques homogènes.Cette partie implique l'utilisation d'algorithmes d'apprentissage automatique afin de réaliser des tâches de classification musicale.Les contributions de ce mémoire sont résumées dans le cinquième chapitre qui propose également des perspectives de recherche dans l'apprentissage automatique et l'extraction de caractéristiques audio multi-échelles. / This doctoral dissertation presents, discusses and proposes tools for the automatic information retrieval in big musical databases.The main application is the supervised classification of musical themes to generate thematic playlists.The first chapter introduces the different contexts and concepts around big musical databases and their consumption.The second chapter focuses on the description of existing music databases as part of academic experiments in audio analysis.This chapter notably introduces issues concerning the variety and unequal proportions of the themes contained in a database, which remain complex to take into account in supervised classification.The third chapter explains the importance of extracting and developing relevant audio features in order to better describe the content of music tracks in these databases.This chapter explains several psychoacoustic phenomena and uses sound signal processing techniques to compute audio features.New methods of aggregating local audio features are proposed to improve song classification.The fourth chapter describes the use of the extracted audio features in order to sort the songs by themes and thus to allow the musical recommendations and the automatic generation of homogeneous thematic playlists.This part involves the use of machine learning algorithms to perform music classification tasks.The contributions of this dissertation are summarized in the fifth chapter which also proposes research perspectives in machine learning and extraction of multi-scale audio features.

No description available.

This thesis deals with optimization of sound projectors, based on knowledge on the auditory perception. In sound projection it is desired that the lagging projected sound dominates the localization. One of the most limiting factors here is the leading direct sound, which, however, can only be reduced to a limited extent since the focusing capabilities of sound projectors are physically limited. In order to enable the perception-based optimization, it was therefore essential to gain an understanding of the perceptual role of the direct sound in achieving localization dominance of the projected sound, and which perception-based requirements for sound projection result from this role. A review of existing literature on the perception in scenarios with leading and lagging sound revealed that further insights into lag localization dominance were needed to this end. These insights were gained by conducting several psychoacoustic investigations in an anechoic chamber, reproducing the sounds via individual loudspeakers. Lag localization dominance seemed to be strongly influenced by the temporal characteristics of the playback signal. Afterwards, comprehensive perception-based requirements for sound projection were derived and their consequences for the design of sound projectors were discussed. On this basis, a method for the perception-based optimization was developed with the goal to reduce the influence of the direct sound on localization. This method was named localization masking. Localization masking is based on the additional generation of one or more sounds arriving earlier and from another direction than the direct sound at the position of the listener. An investigation under laboratory conditions, using cascaded lead-lag pairs representing the sounds involved, suggested that localization masking has the potential to achieve that goal. Localization masking enabled the initial lag, representing the projected sound, to dominate the localization up to a 7 dB higher level of the initial lead, representing the direct sound. Finally, localization masking was investigated under realistic conditions. Localization masking was applied to real sound projectors in a real room and proved to work. Localization masking enabled a given projector to be effectively used with a playback signal that requires stronger focusing capabilities. Furthermore, localization masking enabled a projector with less strong focusing capabilities to be effectively used with a given playback signal.

[ES] Los sistemas de audio han experimentado un gran desarrollo en los últimos años gracias al aumento de dispositivos con procesadores de alto rendimiento capaces de realizar un procesamiento cada vez más eficiente. Además, las comunicaciones inalámbricas permiten a los dispositivos de una red estar ubicados en diferentes lugares sin limitaciones físicas. La combinación de estas tecnologías ha dado lugar a la aparición de las redes de sensores acústicos (ASN). Una ASN está compuesta por nodos equipados con transductores de audio, como micrófonos o altavoces. En el caso de la monitorización acústica del campo, sólo es necesario incorporar sensores acústicos a los nodos ASN. Sin embargo, en el caso de las aplicaciones de control, los nodos deben interactuar con el campo acústico a través de altavoces. La ASN puede implementarse mediante dispositivos de bajo coste, como Raspberry Pi o dispositivos móviles, capaces de gestionar varios micrófonos y altavoces y de ofrecer una buena capacidad de cálculo. Además, estos dispositivos pueden comunicarse mediante conexiones inalámbricas, como Wi-Fi o Bluetooth. Por lo tanto, en esta tesis, se propone una ASN compuesta por dispositivos móviles conectados a altavoces inalámbricos mediante un enlace Bluetooth. Además, el problema de la sincronización entre los dispositivos de una ASN es uno de los principales retos a abordar, ya que el rendimiento del procesamiento de audio es muy sensible a la falta de sincronismo. Por lo tanto, también se lleva a cabo un análisis del problema de sincronización entre dispositivos conectados a altavoces inalámbricos en una ASN. En este sentido, una de las principales aportaciones es el análisis de la latencia de audio cuando los nodos acústicos de la ASN están formados por dispositivos móviles que se comunican altavoces mediante enlaces Bluetooth. Una segunda contribución significativa de esta tesis es la implementación de un método para sincronizar los diferentes dispositivos de una ASN, junto con un estudio de sus limitaciones. Por último, se ha introducido el método propuesto para implementar aplicaciones de zonas sonoras personales (PSZ). Por lo tanto, la implementación y el análisis del rendimiento de diferentes aplicaciones de audio sobre una ASN compuesta por dispositivos móviles y altavoces inalámbricos es también una contribución significativa en el área de las ASN. Cuando el entorno acústico afecta negativamente a la percepción de la señal de audio emitida por los altavoces de la ASN, se uti­lizan técnicas de ecualización para mejorar la percepción de la señal de audio. Para ello, en esta tesis se implementa un sistema de ecualización inteligente. Para ello, se emplean algoritmos psicoacústicos para implementar un procesamiento inteligente basado en el sis­tema auditivo humano capaz de adaptarse a los cambios del entorno. Por ello, otra contribución importante de esta tesis es el análisis del enmas­caramiento espectral entre dos sonidos complejos. Este análisis permitirá calcular el umbral de enmascaramiento de un sonido con más precisión que los métodos utilizados actualmente. Este método se utiliza para implementar una aplicación de ecualización perceptiva que pretende mejorar la percepción de la señal de audio en presencia de un ruido ambien­tal. Para ello, esta tesis propone dos algoritmos de ecualización diferentes: 1) la pre-ecualización de la señal de audio para que se perciba por encima del umbral de enmascaramiento del ruido ambiental y 2) diseñar un con­trol de ruido ambiental perceptivo en los sistemas de ecualización activa de ruido (ANE), para que el nivel de ruido ambiental percibido esté por debajo del umbral de enmascaramiento de la señal de audio. Por lo tanto, la ultima aportación de esta tesis es la implementación de una aplicación de ecualización perceptiva con los dos diferentes algorit­mos de ecualización embebidos y el análisis de su rendimiento a través del banco de pruebas realizado en el laboratorio GTAC-iTEAM. / [CA] El sistemes de so han experimentat un gran desenvolupament en els últims anys gràcies a l'augment de dispositius amb processadors d'alt rendiment capaços de realitzar un processament d'àudio cada vegada més eficient. D'altra banda, l'expansió de les comunicacions inalàmbriques ha permès implementar xarxes en les quals els dispositius poden estar situats a difer­ents llocs sense limitacions físiques. La combinació d'aquestes tecnologies ha donat lloc a l'aparició de les xarxes de sensors acústics (ASN). Una ASN està composta per nodes equipats amb transductors d'àudio, com micr`ofons o altaveus. En el cas del monitoratge del camp acústic, només cal incorporar sensors acústics als nodes de l'ASN. No obstant això, en el cas de les aplicacions de control, els nodes han d'interactuar amb el camp acústic a través d'altaveus. Una ASN pot implementar-se mitjant¿cant dispositius de baix cost, com ara Raspberry Pi o dispositius mòbils, capaços de gestionar di­versos micròfons i altaveus i d'oferir una bona capacitat computacional. A més, aquests dispositius poden comunicar-se a través de connexions inalàmbriques, com Wi-Fi o Bluetooth. Per això, en aquesta tesi es proposa una ASN composta per dispositius mòbils connectats a altaveus inalàmbrics a través d'un enllaç Bluetooth. El problema de la sincronització entre els dispositius d'una ASN és un dels principals reptes a abordar ja que el rendiment del processament d'àudio és molt sensible a la falta de sincronisme. Per tant, també es duu a terme una anàlisi profunda del problema de la sincronització entre els dispositius comercials connectats als altaveus inalàmbrics en una ASN. En aquest sentit, una de les principals contribucions és l'anàlisi de la latència d'àudio quan els nodes acústics en l'ASN estan compostos per dispositius mòbils que es comuniquen amb els altaveus corresponents mitjançant enllaços Bluetooth. Una segona contribuciò sig­nificativa d'aquesta tesi és la implementació d'un mètode per sincronitzar els diferents dispositius d'una ASN, juntament amb un estudi de les seves limitacions. Finalment, s'ha introduït el mètode proposat per implemen­tar aplicacions de zones de so personal. Per tant, la implementació i l'anàlisi del rendiment de diferents aplicacions d'àudio sobre una ASN composta per dispositius mòbils i al­taveus inalàmbrics és també una contribució significativa a l'àrea de les ASN. Quan l'entorn acústic afecta negativament a la percepció del senyal d'àudio emesa pels altaveus de l'ASN, es fan servir tècniques d'equalització per a millorar la percepció del senyal d'àudio. En consequència, en aquesta tesi s'implementa un sistema d'equalització intel·ligent. Per això, s'utilitzen algoritmes psicoacústics per implementar un processament intel·ligent basat en el sistema audi­tiu humà capaç d'adaptar-se als canvis de l'entorn. Per aquest motiu, una altra contribució important d'aquesta tesi és l'anàlisi de l'emmascarament espectral entre dos sons complexos. Aquesta anàlisi permetrà calcular el llindar d'emmascarament d'un so sobre amb més precisió que els mètodes utilitzats actualment. Aquest mètode s'utilitza per a imple­mentar una aplicació d'equalització perceptual que pretén millorar la per­cepció del senyal d'àudio en presència d'un soroll ambiental. Per això, aquesta tesi proposa dos algoritmes d'equalització diferents: 1) la pree­qualització del senyal d'àudio perquè es percebi per damunt del llindar d'emmascarament del soroll ambiental i 2) dissenyar un control de soroll ambiental perceptiu en els sistemes d'equalització activa de soroll (ANE) de manera que el nivell de soroll ambiental percebut estiga per davall del llindar d'emmascarament del senyal d'àudio. Per tant, l'última aportació d'aquesta tesi és la implementació d'una aplicació d'equalització perceptiva amb els dos algoritmes d'equalització embeguts i l'anàlisi del seu rendiment a través del banc de proves realitzat al laboratori GTAC-iTEAM. / [EN] Audio systems have been extensively developed in recent years thanks to the increase of devices with high-performance processors able to per­form more efficient processing. In addition, wireless communications allow devices in a network to be located in different places without physical limitations. The combination of these technologies has led to the emergence of Acoustic Sensor Networks (ASN). An ASN is com­posed of nodes equipped with audio transducers, such as microphones or speakers. In the case of acoustic field monitoring, only acoustic sensors need to be incorporated into the ASN nodes. However, in the case of control applications, the nodes must interact with the acoustic field through loudspeakers. ASN can be implemented through low-cost devices, such as Rasp­berry Pi or mobile devices, capable of managing multiple mi­crophones and loudspeakers and offering good computational capacity. In addition, these devices can communicate through wireless connections, such as Wi-Fi or Bluetooth. Therefore, in this dissertation, an ASN composed of mobile devices connected to wireless speak­ers through a Bluetooth link is proposed. Additionally, the problem of syn­chronization between the devices in an ASN is one of the main challenges to be addressed since the audio processing performance is very sensitive to the lack of synchronism. Therefore, an analysis of the synchroniza­tion problem between devices connected to wireless speakers in an ASN is also carried out. In this regard, one of the main contributions is the analysis of the audio latency of mobile devices when the acoustic nodes in the ASN are comprised of mobile devices communicating with the corresponding loudspeakers through Bluetooth links. A second significant contribution of this dissertation is the implementation of a method to synchronize the different devices of an ASN, together with a study of its limitations. Finally, the proposed method has been introduced in order to implement personal sound zones (PSZ) applications. Therefore, the imple­mentation and analysis of the performance of different audio applications over an ASN composed of mobile devices and wireless speakers is also a significant contribution in the area of ASN. In cases where the acoustic environment negatively affects the percep­tion of the audio signal emitted by the ASN loudspeakers, equalization techniques are used with the objective of enhancing the perception thresh­old of the audio signal. For this purpose, a smart equalization system is implemented in this dissertation. In this regard, psychoacous­tic algorithms are employed to implement a smart processing based on the human hearing system capable of adapting to changes in the envi­ronment. Therefore, another important contribution of this thesis focuses on the analysis of the spectral masking between two complex sounds. This analysis will allow to calculate the masking threshold of one sound over the other in a more accurate way than the currently used methods. This method is used to implement a perceptual equalization application that aims to improve the perception threshold of the audio signal in presence of ambient noise. To this end, this thesis proposes two different equalization algorithms: 1) pre-equalizing the audio signal so that it is perceived above the ambient noise masking threshold and 2) designing a perceptual control of ambient noise in active noise equalization (ANE) systems, so that the perceived ambient noise level is below the masking threshold of the audio signal. Therefore, the last contribution of this dissertation is the imple­mentation of a perceptual equalization application with the two different embedded equalization algorithms and the analysis of their performance through the testbed carried out in the GTAC-iTEAM laboratory. / This work has received financial support of the following projects: • SSPRESING: Smart Sound Processing for the Digital Living (Reference: TEC2015-67387-C4-1-R. Entity: Ministerio de Economia y Empresa. Spain). • FPI: Ayudas para contratos predoctorales para la formación de doctores (Reference: BES-2016-077899. Entity: Agencia Estatal de Investigación. Spain). DANCE: Dynamic Acoustic Networks for Changing Environments (Reference: RTI2018-098085-B-C41-AR. Entity: Agencia Estatal de Investigación. Spain). • DNOISE: Distributed Network of Active Noise Equalizers for Multi-User Sound Control (Reference: H2020-FETOPEN-4-2016-2017. Entity: I+D Colaborativa competitiva. Comisión de las comunidades europea). / Estreder Campos, J. (2022). Smart Sound Control in Acoustic Sensor Networks: a Perceptual Perspective [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/181597

INTRODUÇÃO: O Autismo é um distúrbio que tem início na infância, cujas principais características são a presença de um desenvolvimento anormal ou prejudicado na interação social e comunicação, e um repertório restrito de atividades e interesses. Algumas teorias consideram o autismo como um distúrbio do desenvolvimento causado por uma alteração do sistema nervoso central, e salientam a presença do déficit cognitivo nessa população. Estudos demonstram também a presença de anormalidades eletrofisiológicas nos potenciais evocados auditivos de curta, média e longa latências. Considerando a importância da integridade do sistema auditivo periférico e central na aquisição e desenvolvimento de fala, linguagem e aprendizado, mostra-se imprescindível que anormalidades auditivas tanto periféricas como centrais sejam identificadas e tratadas em indivíduos autistas. OBJETIVO: caracterizar os achados das avaliações comportamentais, eletroacústicas e eletrofisiológicas da audição em indivíduos com autismo, bem como compará-los aos obtidos em indivíduos normais da mesma faixa etária. MÉTODOS: foram realizadas anamnese, audiometria tonal, logoaudiometria, medidas de imitância acústica, potencial evocado auditivo de tronco encefálico, potencial evocado auditivo de média latência e potencial cognitivo em 16 indivíduos com autismo (grupo pesquisa) e 25 normais (grupo controle), com idades entre oito e 20 anos. RESULTADOS: Na comparação entre os resultados normais e alterados (análise qualitativa), não foram encontradas alterações na avaliação comportamental da audição para os dois grupos. Na comparação dos resultados das avaliações comportamentais e eletroacústicas entre os grupos, não ocorreram diferenças estatisticamente significantes. O grupo controle apresentou alterações apenas no resultado do potencial evocado auditivo de média latência, sendo que o tipo de alteração mais freqüentemente encontrada foi ambas (efeito eletrodo e efeito orelha ocorrendo concomitantemente). O grupo pesquisa apresentou resultados alterados em todos os potenciais evocados auditivos, havendo diferença estatisticamente significante quando comparado ao grupo controle. Com relação aos tipos de alterações encontradas no grupo pesquisa, foi observada uma maior ocorrência de alteração em tronco encefálico baixo no potencial evocado auditivo de tronco encefálico, alteração do tipo ambas (efeito eletrodo e efeito orelha ocorrendo concomitantemente) no potencial evocado auditivo de média latência e ausência de resposta no potencial cognitivo. Na análise quantitativa dos resultados dos potenciais evocados auditivos, verificou-se que apenas para o potencial evocado auditivo de tronco encefálico ocorreu diferença estatisticamente significante entre os grupos, com relação às latências das ondas III e V e interpicos I-III e I-V. CONCLUSÃO: Indivíduos com autismo não apresentam alterações nas avaliações comportamentais e eletroacústicas da audição, e apresentam alterações nos potenciais evocados auditivos de tronco encefálico e potencial cognitivo, sugerindo comprometimento da via auditiva em tronco encefálico e regiões corticais. / INTRODUCTION: Autism is a disorder, which begins in the infancy, and the main characteristics are the presence of an abnormal or impaired development of social interaction and communication, and restrict range of activities and interest. Some theories consider autism as a developmental disorder caused by a central nervous system alteration, and stress the presence of a cognitive deficit in this population. Studies also demonstrate the presence of electrophysiological abnormalities in the auditory evoked potentials of short middle and long latencies. Considering the importance of the peripheral and central auditory system integrity for the speech and language acquisition and development and for learning, it becomes important to identify and treat hearing abnormalities, either peripheral or central, in autistic individuals. AIM: to characterize the findings of behavioral, electroacoustic and electrophysiological assessments of autistic individuals, as well as to compare those findings with the ones of normal individuals of the same age. METHOD: 16 individuals with autism (study group) and 25 normal ones (control group), ranging in age from eight and 20 years underwent anamnesis, pure tone audiometry, speech audiometry, acoustic immitance measures, brainstem auditory evoked potential, middle latency response and cognitive potential. RESULTS: Comparing the normal and altered results (qualitative analysis), no alterations were found in the behavioral assessment of hearing in both groups. Comparing the results of the behavioral and electroacoustic evaluations between the two groups, there were no statistical differences. The control group presented altered results only in the middle latency auditory evoked potential and the most common type of alteration was both electrode effect and ear effect occurring simultaneously. The study group presented altered results in all auditory evoked potentials with a significant statistical difference when compared to the control group. Concerning the types of alterations found in the study group it was verified higher occurrence of lower brainstem alteration in the brainstem auditory evoked potential, both electrode and ear effect occurring simultaneously in the middle latency auditory evoked potential, and absence of response in the cognitive potential. The quantitative analysis of the auditory evoked potentials results showed a significant statistical difference between the groups only in the brainstem auditory evoked potential, concerning the latencies of waves III and V and interpeaks I-III and I-V. CONCLUSION: autistic individuals do not present altered behavioral and electroacoustic evaluations, and present altered brainstem auditory evoked potential and cognitive potential, suggesting prejudice in the brainstem auditory pathway and cortical regions.

H διδακτορική διατριβή μελετά ζητήματα που αφορούν την ενσωμάτωση υπολογιστικών μοντέλων ακοής για την μοντελοποίηση και επεξεργασία ηχητικών σηματών για την βέλτιστη αναπαραγωγή τους σε χώρους με αντήχηση καθώς και την κωδικοποίηση ηχητικών δεδομένων. Το κύριο μέρος της διατριβής επικεντρώθηκε στην μοντελοποίηση των αντιληπτικά σημαντικών αλλοιώσεων λόγω αντήχησης, με την βοήθεια κατάλληλα οριζόμενων μόνο-ωτικών και διαφορικών ενδο-καναλικών παραμέτρων και την απεικόνιση τους με τη βοήθεια χρονο-συχνοτικών 2Δ αναπαραστάσεων. Ο λεπτομερής εντοπισμός των αλλοιώσεων στα ηχητικά σήματα μέσω του προτεινόμενου Δείκτη Επικάλυψης λόγω Αντήχησης (ΔΕΑ) διαμόρφωσε κατάλληλη μεθοδολογία ανάλυσης-σύνθεσης, για την καταστολή της αντήχησης σε συγκεκριμένες χρονο-συχνοτικές περιοχές. Το κύριο πλεονέκτημα της προτεινόμενης, εξαρτώμενης του σήματος, μεθοδολογίας είναι ότι επιτυγχάνεται η καταστολή των, με σχετική καθυστέρηση, παραμορφώσεων λόγω αντήχησης σε μια μεγαλύτερη κλίμακα, δεδομένου ότι μόνο οι αντιληπτικά σημαντικές περιοχές του σήματος επηρεάζονται από την επεξεργασία. Επιπλέον, αναζητήθηκε η δυνατότητα ανάλυσης των ηχητικών δεδομένων με βάση τις εσωτερικές τους αναπαραστάσεις (όπως δηλαδή τις παρέχει το υπολογιστικό μοντέλο ακοής) με εφαρμογή στην περιοχή της κωδικοποίησης σημάτων. Ο προτεινόμενος μη-ομοιόμορφος κβαντιστής πραγματοποιεί τη διαδικασία της κβάντισης χρονο-συχνοτικά με κατάλληλη οδήγηση από το υπολογιστικό μοντέλο ακοής, εξασφαλίζοντας καλύτερη υποκειμενική ηχητική ποιότητα, σε σχέση με ένα ομοιόμορφο PCM κβαντιστή. Χρησιμοποιώντας τη βασική λειτουργία του μη-ομοιόμορφου κβαντιστή, υλοποιήθηκε ενά κριτήριο αξιολόγησης ηχητικών δεδομένων, όπου σε αντίθεση με καθιερώμενα κριτήρια (όπως το Noise to Mask Ration, NMR) επιτελεί τις λειτουργίες του στο πεδίο χρόνου-συχνότητας και παρέχει τη δυνατότητα εντοπισμού της υποκειμενικά σημαντικής παραμόρφωσης με βάση την χρονική εξέλιξη του σήματος. / The dissertation studies issues concerning the integration of computational auditory models for modeling and processing of audio signals for optimal reproduction in reverberant spaces as well as topics related to audio coding. Based on the theoretical framework analysis that was established, the necessity of a signal-dependent approach was underlined for modeling the perceptually-relevant effects of reverberation. The main part of the dissertation thesis was focused on describing the perceptually-relevant alterations due to reverberation, based on appropriate defined monaural and differential inter-channel parameters and also their representation with well-defined time-frequency 2D maps. The detailed localization of alterations due to reverberation in the acoustic signals via the proposed Reverberation Masking Index (RMI) introduced an analysis-synthesis methodology for the compensation of reverberation in perceptually-significant time-frequency regions incorporating also, well-established digital signal processing techniques. The main advantage of the proposed signal-dependent methodology is that the suppression of reverberant tails can be achieved on a larger scale under practical conditions, since only perceptually significant regions of the signal are affected after processing. Additionally, the proposed framework complements the more traditional system-dependent inverse filtering methods, enabling novel and efficient signal processing schemes to evolve for room dereverberation applications. The thesis examines also the feasibility of the acoustic signal analysis based on the internal representations provided by the computational auditory model, applicable in the area of audio coding. The proposed non-uniform quantizer operates in the time-frequency domain, where a novel quantization process is driven by the computational auditory model, thus enabling an overall better perceptual quality with respect to uniform PCM quantizer. Considering the fundamental operation of the novel non-uniform quantizer, a criterion for audio quality evaluation was proposed, where contrary to well-established criteria (i.e., Noise to Mask Ratio, NMR) its potential structure performs in the time-frequency domain and provides the detailed localization of perceptually-important distortions based on the input signal’s evolution.

INTRODUÇÃO: O Autismo é um distúrbio que tem início na infância, cujas principais características são a presença de um desenvolvimento anormal ou prejudicado na interação social e comunicação, e um repertório restrito de atividades e interesses. Algumas teorias consideram o autismo como um distúrbio do desenvolvimento causado por uma alteração do sistema nervoso central, e salientam a presença do déficit cognitivo nessa população. Estudos demonstram também a presença de anormalidades eletrofisiológicas nos potenciais evocados auditivos de curta, média e longa latências. Considerando a importância da integridade do sistema auditivo periférico e central na aquisição e desenvolvimento de fala, linguagem e aprendizado, mostra-se imprescindível que anormalidades auditivas tanto periféricas como centrais sejam identificadas e tratadas em indivíduos autistas. OBJETIVO: caracterizar os achados das avaliações comportamentais, eletroacústicas e eletrofisiológicas da audição em indivíduos com autismo, bem como compará-los aos obtidos em indivíduos normais da mesma faixa etária. MÉTODOS: foram realizadas anamnese, audiometria tonal, logoaudiometria, medidas de imitância acústica, potencial evocado auditivo de tronco encefálico, potencial evocado auditivo de média latência e potencial cognitivo em 16 indivíduos com autismo (grupo pesquisa) e 25 normais (grupo controle), com idades entre oito e 20 anos. RESULTADOS: Na comparação entre os resultados normais e alterados (análise qualitativa), não foram encontradas alterações na avaliação comportamental da audição para os dois grupos. Na comparação dos resultados das avaliações comportamentais e eletroacústicas entre os grupos, não ocorreram diferenças estatisticamente significantes. O grupo controle apresentou alterações apenas no resultado do potencial evocado auditivo de média latência, sendo que o tipo de alteração mais freqüentemente encontrada foi ambas (efeito eletrodo e efeito orelha ocorrendo concomitantemente). O grupo pesquisa apresentou resultados alterados em todos os potenciais evocados auditivos, havendo diferença estatisticamente significante quando comparado ao grupo controle. Com relação aos tipos de alterações encontradas no grupo pesquisa, foi observada uma maior ocorrência de alteração em tronco encefálico baixo no potencial evocado auditivo de tronco encefálico, alteração do tipo ambas (efeito eletrodo e efeito orelha ocorrendo concomitantemente) no potencial evocado auditivo de média latência e ausência de resposta no potencial cognitivo. Na análise quantitativa dos resultados dos potenciais evocados auditivos, verificou-se que apenas para o potencial evocado auditivo de tronco encefálico ocorreu diferença estatisticamente significante entre os grupos, com relação às latências das ondas III e V e interpicos I-III e I-V. CONCLUSÃO: Indivíduos com autismo não apresentam alterações nas avaliações comportamentais e eletroacústicas da audição, e apresentam alterações nos potenciais evocados auditivos de tronco encefálico e potencial cognitivo, sugerindo comprometimento da via auditiva em tronco encefálico e regiões corticais. / INTRODUCTION: Autism is a disorder, which begins in the infancy, and the main characteristics are the presence of an abnormal or impaired development of social interaction and communication, and restrict range of activities and interest. Some theories consider autism as a developmental disorder caused by a central nervous system alteration, and stress the presence of a cognitive deficit in this population. Studies also demonstrate the presence of electrophysiological abnormalities in the auditory evoked potentials of short middle and long latencies. Considering the importance of the peripheral and central auditory system integrity for the speech and language acquisition and development and for learning, it becomes important to identify and treat hearing abnormalities, either peripheral or central, in autistic individuals. AIM: to characterize the findings of behavioral, electroacoustic and electrophysiological assessments of autistic individuals, as well as to compare those findings with the ones of normal individuals of the same age. METHOD: 16 individuals with autism (study group) and 25 normal ones (control group), ranging in age from eight and 20 years underwent anamnesis, pure tone audiometry, speech audiometry, acoustic immitance measures, brainstem auditory evoked potential, middle latency response and cognitive potential. RESULTS: Comparing the normal and altered results (qualitative analysis), no alterations were found in the behavioral assessment of hearing in both groups. Comparing the results of the behavioral and electroacoustic evaluations between the two groups, there were no statistical differences. The control group presented altered results only in the middle latency auditory evoked potential and the most common type of alteration was both electrode effect and ear effect occurring simultaneously. The study group presented altered results in all auditory evoked potentials with a significant statistical difference when compared to the control group. Concerning the types of alterations found in the study group it was verified higher occurrence of lower brainstem alteration in the brainstem auditory evoked potential, both electrode and ear effect occurring simultaneously in the middle latency auditory evoked potential, and absence of response in the cognitive potential. The quantitative analysis of the auditory evoked potentials results showed a significant statistical difference between the groups only in the brainstem auditory evoked potential, concerning the latencies of waves III and V and interpeaks I-III and I-V. CONCLUSION: autistic individuals do not present altered behavioral and electroacoustic evaluations, and present altered brainstem auditory evoked potential and cognitive potential, suggesting prejudice in the brainstem auditory pathway and cortical regions.

The door closing action is a recurrent situation when using a vehicle, and its sound is therefore a common sensation, which would elicit pleasant feelings. Sensory pleasantness is an important aspect in terms of customer’s perspective, and it can be a contributing factor when deciding to buy or not a specific vehicle. The first contact between a prospective customer and the automobile usually happens in car salons or at the car retailer. The initial impression of the vehicle might be sight-based, and the door may commonly be the first physical contact. Depending on the car brand and type, doors differ in terms of mass, structure, dimension. Furthermore, there are differences regarding the latching system and the door sealing structure, in terms of material and construction. The closing sound produced when slamming the door is related to all these parameters. Auditory pleasantness can be described by characteristics of the sound that are described through psychoacoustics. Loudness, sharpness, roughness, and tonality are important auditory parameters to objectively describe this complex sensation. The aim of car doors would be to generate an enthusiastic, low-pitched, and saturated sound, which would elicit feelings of solidity, robustness, and security. On the other side, a metallic, high-pitched, fragmented sound could be a source of annoyance and produce feelings of insecurity and cheap vehicle.The present work aims to provide a broad picture on the mechanics and acoustics of door closing for automobiles. In specific, the closing sound was evaluated in relation to the door gaskets and their sealing performance over time. The sealing performance was analyzed in energy and force terms. The door closing motion was studied as a quasi-static problem, as well as a dynamic problem, where the former is related to the latching capability of the door, the latter is connected to the slamming action. The measurement results include the sealing performance trend from fresh to aged gaskets. From these measurements, the rubber non-linear behaviour could then be evaluated from a sound quality perspective. The acoustic analysis revealed inconsistencies of the psychoacoustic parameters in the description of the hearing sensations. Spectral analysis was also implemented to capture the door closing phenomenon, and the Wavelet transform emerged as the method with the highest resolution in the description of the sound wave progression.Several measurements were performed in order to assess all the established points, and methods were implemented for the sealing stiffness analysis and the acoustic analysis. The severe transiency of the door closing event was put in evidence. The stiffness analysis method showed also potential in helping to adjust the end of line tuning of the vehicle. Finally, benchmarking was included in the project, which enabled comparisons with competitor cars. / Dörrstängning är en återkommande händelse när ett fordon används, och ljudet bör därför ge ett positivt intryck och korrekt information till brukaren. Ett behagligt intryck är en viktig aspekt ur kundens perspektiv och kan vara ett var flera bidragande faktorer när beslut tas om att köpa eller inte köpa ett fordon. Den första kontakten mellan en potentiell kund och bilen sker vanligtvis i bilsalonger eller hos bilhandlare. Det första intrycket av fordonet kan vara visuellt, och dörren är ofta den första fysiska kontakten. Beroende på biltyp och fabrikat skiljer sig dörrarna åt när det kommer till massa, struktur och dimensioner. Dessutom kan det finnas skillnader i låssystem och dörrtätningskonstruktion såsom i både material och utformning. Stängningsljudet som uppstår när dörren slås igen är relaterat till alla dessa parametrar. Ett ljuds upplevda behaglighet i det beror på ljudets egenskaper, som beskrivs med hjälp av psykoakustik. Ljudstyrka, skärpa, råhet och tonalitet är viktiga auditiva parametrar för att objektivt beskriva detta komplexa intryck. Målet med bildörrar bör vara att generera ett dovt och mättat ljud, för att framkalla känslor av soliditet, robusthet och säkerhet. Å andra sidan kan, ett metalliskt, högfrekvent och skramligt ljud vara en källa till irritation och ge känslor av osäkerhet och låg kvalité.Syftet med detta arbete är att ge en övergripande beskrivning av dörrstängning och akustiken kring detta. I synnerhet utvärderades stängningsljudet i förhållande till dörrpackningarna och deras tätningsprestanda mätt över tiden. Tätningsprestanda analyserades i energi- och krafttermer. Dörrens stängningsrörelse studerades både som ett kvasistatiskt problem och som ett dynamiskt problem. Det förstnämnda är relaterat till dörrens låsningsförmåga, medan det sistnämnda är kopplat till smällar i dörren. Mätresultaten visade hur tätningsprestandan förändras över tiden. Gummits icke-linjära beteende har också utvärderats med ett ljudkvalitetsperspektiv. En spektralanalys genomfördes av ljudet från dörrstängningar och Wavelet-transformen visade sig vara den lösning som gav bäst kvalitet. Flera mätningar utfördes för att bedöma alla fastställda punkter och metoder infördes för analysen av tätningens styvhet och för den akustiska analysen. Den kraftiga transiensen i dörrstängningen kunde ses i resultaten. Styvhetsanalysen visade även hur den utvecklade metoden skulle kunna bidra till att justera fordonets end-of-line inställningar. Slutligen ingick benchmarking i projektet vilket möjliggjorde jämförelser med konkurrentbilar.