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DI, CROCE NICOLA. "Sonic acknowledgments. La territorialità delle politiche nella pratica dell'ascolto." Doctoral thesis, Università IUAV di Venezia, 2016. http://hdl.handle.net/11578/278716.
Повний текст джерелаАнотація:
The research aims at investigating the relationship between urban and regional planning, policy analysis and design, and Sound Studies. It seeks to acknowledge the urban and regional issues through the raise of sonic awareness, thus discussing the epistemology of specific urban “problems” and proposing an alternative research tools. The analytical approach is preliminary to policy design and to the empowerment of vulnerable communities. Listening becomes then a research device through which it is possible to explore those problematic issues that go unnoticed by the institutions.
The thesis is an invitation to read alternatively public policies, and to overcome the “problem solving” approach towards a sonic articulation of the “problem setting” approach. By investigating the sonic environment the research intends to show how knowledge is produced and used by policy makers. Thus sound and sonic environment turn to shape a new model of understanding full of potential although scarcely used by urban planning and policy design; therefore the sonic environment defines its deep political dimension. Within this framework the research will introduce and clarify the notion of soundscape, sonic environment, acoustic atmosphere, and will explore the set of cultural and political studies that are facing the perception, emotion and social behavior related to sound production and fruition.La ricerca si propone di indagare le possibili interazioni tra i campi della pianificazione territoriale, delle politiche pubbliche, e dei Sound Studies. L’obiettivo è esplicitare le connessioni tra ambiente sonoro e questioni urbane e territoriali attraverso la pratica dell’ascolto; tematizzare dunque le modalità di produzione di conoscenza per affrontare l’inquadramento di particolari “problemi” e l’elaborazione di possibili approcci alternativi di ricerca. Se l’attitudine è sicuramente progettuale, lo scopo dichiarato è analitico, propedeutico all’azione: al disegno di politiche come all’empowerment di comunità. L’ascolto assume quindi le caratteristiche di uno strumento di ricerca; se ne vogliono delineare le potenzialità, esplorare le tracce attraverso cui approcciare le questioni urbane e territoriali che pur “facendo problema” sembrano passare spesso “inosservate”, soprattutto su un piano istituzionale. Si è invitati a seguire un percorso inedito di lettura delle politiche pubbliche, che rimanda al superamento del problem solving e procede verso un’articolazione sonora del problem setting. Attraverso la lettura dell’ambiente sonoro si vuole infatti esplicitare i termini su cui si articola la produzione di conoscenza a supporto dell’azione; conoscenza che costituisce un aiuto essenziale per il processo decisionale, dunque per il disegno di politiche. Il suono, e l’ambiente acustico, intervengono come messaggeri di un modello interpretativo finora troppo poco praticato dalla pianificazione e dal policy design. Se il suono è il centro della ricerca, il suo potenziale evocativo e rappresentativo si esplicita e riverbera nella dimensione politica. Sono allora da introdurre e chiarire i cardini del discorso: definire il paesaggio sonoro, l’ambiente acustico, ed esplorare gli studi culturali che ne stanno decodificando i segnali, dunque il quadro percettivo consapevole e inconsapevole cui l’ascolto fa costantemente riferimento.
2
Liao, Wei-Hsiang. "Modelling and transformation of sound textures and environmental sounds." Thesis, Paris 6, 2015. http://www.theses.fr/2015PA066725/document.
Повний текст джерелаАнотація:
Le traitement et la synthèse des sons environnementaux sont devenue un sujet important. Une classe des sons, qui est très important pour la constitution d'environnements sonore, est la classe des textures sonores. Les textures sonores sont décrit par des relations stochastiques et qui contient des composantes non-sinusoïdales à caractère fortement bruité. Il a été montré récemment que la reconnaissance de textures sonores est basée sur des mesures statistiques caractérisant les enveloppes dans les bandes critiques. Il y actuellement très peu d'algorithmes qui permettent à imposer des propriétés statistiques de façon explicite lors de la synthèse de sons. L'algorithme qui impose l'ensemble de statistique qui est perceptivement relevant pour les textures sonore est très couteuse en temps de calcul. Nous proposons une nouvelle approche d'analyse-synthèse qui permet une analyse des statistiques relevant et un mécanisme efficace d'imposer ces statistiques dans le domaine temps-fréquence. La représentation temps-fréquence étudié dans cette thèse est la transformée de Fourier à court terme. Les méthodes proposées par contre sont plus générale et peuvent être généralisé à d'autres représentations temps-fréquence reposant sur des banques de filtres si certaines contraintes sont respectées. L'algorithme proposé dans cette thèse ouvre plusieurs perspectives. Il pourrait être utilisé pour générer des textures sonores à partir d'une description statistique créée artificiellement. Il pourrait servir de base pour des transformations avancées comme le morphing, et on pourrait aussi imaginer à utiliser le modèle pour développer un contrôle sémantique de textures sonoresThe processing of environmental sounds has become an important topic in various areas. Environmental sounds are mostly constituted of a kind of sounds called sound textures. Sound textures are usually non-sinusoidal, noisy and stochastic. Several researches have stated that human recognizes sound textures with statistics that characterizing the envelopes of auditory critical bands. Existing synthesis algorithms can impose some statistical properties to a certain extent, but most of them are computational intensive. We propose a new analysis-synthesis framework that contains a statistical description that consists of perceptually important statistics and an efficient mechanism to adapt statistics in the time-frequency domain. The quality of resynthesised sound is at least as good as state-of-the-art but more efficient in terms of computation time. The statistic description is based on the STFT. If certain conditions are met, it can also adapt to other filter bank based time-frequency representations (TFR). The adaptation of statistics is achieved by using the connection between the statistics on TFR and the spectra of time-frequency domain coefficients. It is possible to adapt only a part of cross-correlation functions. This allows the synthesis process to focus on important statistics and ignore the irrelevant parts, which provides extra flexibility. The proposed algorithm has several perspectives. It could possibly be used to generate unseen sound textures from artificially created statistical descriptions. It could also serve as a basis for transformations like stretching or morphing. One could also expect to use the model to explore semantic control of sound textures
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Chatterley, James J. "Sound Quality Analysis of Sewing Machines." BYU ScholarsArchive, 2005. https://scholarsarchive.byu.edu/etd/424.
Повний текст джерелаАнотація:
Sound quality analysis is a tool designed to help determine customer preferences, which can be used to help the designer improve product quality. Many industries desire to know how the consuming public perceives their product, as this affects the product life and success. This research investigates which of the six sewing machines provided by Viking Sewing Machine Group (VSM group) consumers find most acoustically appealing. The sound quality analysis methods used include both jury based listening tests and quantitative sound quality metrics from empirical equations. The results from both methods are completely independent and are shown to have a very strong correlation. The procedures and results of both methods, jury listening tests and mathematical metrics, are presented. Near field sound intensity scans identified acoustic hot spots and give direction for possible design modifications to improve the acoustic signature of the two top tier machines, the Designer 1 and Creative 2144 (Husqvarna Viking and Pfaff respectively). This research determined that the entry level Pfaff Select 1530 has the most acoustically appealing sound of the six machines evaluated. In addition, it was also determined that a reduction in the higher frequency sounds produced by the machines is preferred over a reduction in the lower frequency sounds. Further investigations, including an evaluation of machine isolation and startup sounds, were also performed. The machine isolation results are highly dependant on the individual machine being evaluated and would require independent evaluation. In the machine startup sound assessment, it was discovered that again the Pfaff Select 1530 has the preferred sound. Near field acoustic intensity scans provide additional information on locations of strong acoustic radiation. The near field scans provided valuable design information. The acoustic "hot" spots were discovered to exist in the lower portions of the machines near the main stepper motor in the Designer 1, and radiating from the bottom plate of the machine in the Pfaff Creative 2144. This analysis has led to various design modifications that could be implemented to improve the sound quality of the machines, specifically the Designer 1 and the Creative 2144.
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TERENZI, Alessandro. "Innovative Digital Signal Processing Methodologies for Identification and Analysis of Real Audio Systems." Doctoral thesis, Università Politecnica delle Marche, 2021. http://hdl.handle.net/11566/287822.
Повний текст джерелаАнотація:
Esistono molti sistemi audio reali e ciascuno ha le proprie caratteristiche ma tutti sono accomunati dal fatto che sono sistemi in grado di generare o modificare un suono. Se un sistema naturale o artificiale può essere definito come sistema sonoro, allora è possibile applicare le tecniche del digital signal processing per studiare il sistema ed emularne il comportamento. In questo lavoro di tesi si propone di introdurre delle metodologie innovative di processamento del segnale applicate ad alcuni sistemi sonori reali.
In particolare, vengono studiati e discussi tre diversi sistemi: il mondo dei dispositivi non lineari basati su valvole, con particolare attenzione agli amplificatori per chitarra e hi-fi, l'ambiente acustico di una stanza ed il suo effetto sulla propagazione del suono ed infine il suono emesso dalle api in un alveare.
Per quanto riguarda il primo sistema, vengono proposti dei contributi innovativi per l'identificazione di modelli come la serie di Volterra ed il modello di Hammerstein; in particolare viene discusso un approccio per superare alcune limitazioni dell'identificazione tramite serie di Volterra e l'applicazione di una struttura in sottobande per ridurre il costo computazionale e incrementare la velocità di convergenza di un algoritmo adattativo per l'identificazione del modello di Hammerstein. In ultima analisi, viene proposto un approccio innovativo in grado di stimare con una singola misura vari parametri di distorsione sfruttando un modello di Hammerstein generallizato.
Per quanto riguarda il secondo ambito, vengono proposti i risultati relativi a due applicazioni di equalizzazione multipunto: nel primo caso si mostrerà come l'equalizzazione può essere usata non solo per compensare le anomalie sonore generate all'interno di una stanza, ma anche per migliorare la risposta in frequenza di particolari trasduttori a vibrazione ancorati ad un pannello rigido; nel secondo caso si illustra come un approccio in sottobande può migliorare l'efficienza computazionale e la velocità di un algoritmo di equalizzazione adattativo multipunto e multicanale.
Infine, viene presentato un sistema sonoro naturale, ovvero quello generato da un alveare. In questo caso si illustrerà un sistema di acquisizione innovativo sviluppato per monitorare gli alveari con particolare attenzione al suono; succesivamente si mostrano gli approcci messi a punto per analizzare il suono registrato in due condizioni reali ed infine verranno si illustrano i risultati ottenuti grazie allo studio del suono usando algoritmi di classificazione.
Inoltre, nella parte finale dell'elaborato sono presenti dei contributi secondari ma che hanno comunque come focus principale il signal processing applicato ad ambienti acustici reali, in particolare si discute di un'implementazione di un algoritmo di cancellazione attiva del rumore e di due algoritmi per effetti digitali in cui il primo migliora le performance sonore di altoparlanti compatti, ed il secondo genera un effetto stereofonico per chitarra elettrica.Many real word audio systems exist, each has its own characteristics but almost all of them can be identified from the fact that they are able to generate or modify a sound. If a natural or artificial system can be defined as a sound system, then it is possible to apply the techniques of digital signal processing for the studying and the emulation of the system. In this thesis, innovative methodologies for digital signal processing applied to real audio systems will be discussed. In particular, three different audio systems will be considered: the world of vacuum-based non linear audio devices with particular attention to guitar and hi-fi amplifiers; the room acoustic environment and its effect on the sound propagation; and finally the sound emitted by honey bees in a beehive. Regarding the first system, innovative approaches for the identification of the Volterra series and Hammerstein models will be proposed, in particular an approach to overcome some limitation of Volterra series identification. The application of a sub-band structure to reduce the computational cost and increase the convergence speed of an adaptive Hammerstein model identification will be proposed as well. Finally, an innovative approach for the measurement of several distortion parameters using a single measure, exploiting a generalized Hammerstein model, will be presented. For the second system, the results of the application of a multi-point equalizer to two different situations will be exposed. In particular, in the first case, it will be shown how a multi-point equalization can be used not only to compensate the acoustical anomalies of a room, but also to improve the frequency response of vibrating transducers mounted on a rigid surface. The second contribution will show how a sub-band approach can be used to improve the computational cost and the speed of an adaptive algorithm for a multi-point and multi channel equalizer. At the end, the focus will be on a natural sound system, i.e., a honey bees colony. In this case, an innovative acquisition system for honey bees sound monitoring will be presented. Then, the approaches developed for sound analysis will be exposed and applied to the recorded sounds in two different situations. Finally, the obtained results, achieved with the application of classification algorithms, will be exposed. In the final part of the work some minor contributions still related to signal processing applied to real sound systems are presented. In particular, an implementation of an active noise control system is discussed, and two algorithms for digital effects where the former improves the sound performances of compact loudspeakers and the latter generates a stereophonic effect for electric guitars are exposed.
5
Satakopan, Hariram. "Time-Frequency Feature Extraction for Impact Sound Quality Analysis with Emphasis on Automobile Applications." University of Cincinnati / OhioLINK, 2011. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1299168649.
Повний текст джерела
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Lee, JungSuk. "Categorization and modeling of sound sources for sound analysis/synthesis." Thesis, McGill University, 2013. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=116954.
Повний текст джерелаАнотація:
In this thesis, various sound analysis/re-synthesis schemes are investigated in a source/filter model framework, with emphasis on the source component. This research provides improved methods and tools for sound designers, composersand musicians to flexibly analyze and synthesize sounds used for gaming, film or computer music, ranging from abstract, complex sounds to those of real musical instruments. First, an analysis-synthesis scheme for the reproduction of a rolling ball sound is presented. The proposed scheme is based on the assumption that the rolling sound is generated by a concatenation of micro-contacts between a ball and a surface, each having associated resonances. Contact timing information is extracted from the rolling sound using an onset detection process, allowing for segmentation of a rolling sound. Segmented sound snippets are presumed to correspond to micro-contacts between a ball and a surface; thus, subband based linear predictions (LP) are performed to model time-varying resonances and anti-resonances. The segments are then resynthesized and overlap-added to form a complete rolling sound. A "granular" analysis/synthesis approach is also applied to various kinds of environmental sounds (rain, fireworks, walking, clapping) as an additional investigation into how the source type influences the strategic choices for the analysis/synthesis of sounds. The proposed granular analysis/synthesis system allows for flexible analysis of complex sounds and re-synthesis with temporal modification. Lastly, a novel approach to extract a pluck excitation from a recorded plucked string sound is proposed within a source / filter context using physical models. A time domain windowing method and an inverse filtering-based method are devised based on the behavior of wave propagation on the string. In addition, a parametric model of the pluck excitation as well as a method to estimate its parameters are addressed.Dans cette thèse, nous avons étudié plusieurs scéhmas d'analyse/synthèse dans le cadre des modèles source/filtre, avec un attention particulière portée sur la composante de source. Cette recherche améliore les méthodes ainsi que les outils fournis créateurs de sons, compositeurs et musiciens désirant analyser et synthétiser avec flexibilité des sons destinés aux jeux vidéos, au cinéma ou à la musique par ordinateur. Ces sons peuvent aller de sons abstraits et complexes à ceux provenant d'instruments de musique existants. En premier lieu, un schéma d'analyse-synthèse est introduit permettant la reproduction du son d'une balle en train de rouler. Ce schéma est fondé sur l'hypothèse que le son de ce roulement est généré par la concaténation de micro-contacts entre balle et surface, chacune d'elles possédant sa proper série de résonances. L'information relative aux temps de contact est extradite du son du roulement que l'on cherche à reproduire au moyen d'une procédure détectant le début du son afin de le segmenter. Les segments de son ainsi isolés sont supposés correspondre aux micro-contacts entre la balle et la surface. Ainsi un algorithme de prédiction linéaire est effectué par sous-bande, préalablement extraites afin de modéliser des résonances et des anti-résonances variants dans le temps. Les segments sont ensuite re-synthétisés, superposés et additionnés pour reproduire le son du roulement dans son entier. Cette approche d'analyse/synthèse "granulaire" est également appliquée à plusieurs sons de types environnementaux (pluie, feux d'artifice, marche, claquement) afin d'explorer plus avant l'influence du type de la source sur l'analyse/synthèse des sons. Le système proposé permet une analyse flexible de sons complexes et leur synthèse, avec la possibilité d'ajouter des modifications temporelles.Enfin, une approche novatrice pour extraire le signal d'excitation d'un son de corde pincée est présentée dans le contexte de schémas source/filtre sur une modèlisation physique. A cet effet, nous introduisons une méthode de type fenêtrage, et une méthode de filtrage inverse fondée sur le type de propagation selon laquelle l'onde se déplace le long de la corde. De plus, un modèle paramétrique de l'excitation par pincement ainsi qu'une méthode d'estimation de ces paramètres sont détaillés.
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Price, M. A. "Sound propagation in woodland." Thesis, Open University, 1986. http://oro.open.ac.uk/56924/.
Повний текст джерелаАнотація:
A review of past research into sound propagation in woodland is presented. The attenuation of sound in woodland is small between about 800 and 2000Hz and greater at low and high frequencies. Attenuation measurements made in three contrasting woodlands are presented and compared with theoretical models. Propagation models using simple one- and two-parameter impedance models are used to calculate appropriate ground parameters for the prediction of impedance of the woodland soils. The ground parameters varied on different days in a single stand due to differences in moisture content and compaction. The overall differences between the stands are not significant. The woodland soil has a considerably lower impedance than other outdoor ground surfaces such as grassland or sand. A theoretical model for the attenuation of sound by thermoviscous absorption and scattering within an array of cylinders is assessed by means of a model experiment with wooden rods in an anechoic chamber. An input density 60% lower than the actual density gives a good agreement with measured attenuation. This modified model also predicts the attenuation by the cylinders in the presence of a ground surface. The scattering model is compared with the high frequency attenuation measured in the , woodland, using sampled trunk densities and radius, this underpredicts the observed attenuation, particularly in the stands with a dense branch and foliage structure. Addition of a second. dense, array of non rigid scatterers gives a good agreement with the measured data, thus modelling the scattering and absorbing effects of trunks, branches and leaves, in the high frequencies. Finally, a combined model is presented in which the attenuation caused by ground interference effects. at low frequencies. is added to a prediction of attenuation by the scattering model. across the whole frequency range. This model reproduces the frequency dependence of the attenuation of sound in woodland.
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Derton, Riccardo. "Door closing sound quality related to door sealing stiffness." Thesis, KTH, Marcus Wallenberg Laboratoriet MWL, 2021. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-302275.
Повний текст джерелаАнотація:
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.
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Servis, Dimitris C. "Sound transmission at pipe joints." Thesis, Heriot-Watt University, 1991. http://hdl.handle.net/10399/782.
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Song, Guanghan. "Effect of sound in videos on gaze : contribution to audio-visual saliency modelling." Thesis, Grenoble, 2013. http://www.theses.fr/2013GRENT013/document.
Повний текст джерелаАнотація:
Les humains reçoivent grande quantité d'informations de l'environnement avec vue et l'ouïe . Pour nous aider à réagir rapidement et correctement, il existe des mécanismes dans le cerveau à l'attention de polarisation vers des régions particulières , à savoir les régions saillants . Ce biais attentionnel n'est pas seulement influencée par la vision , mais aussi influencée par l'interaction audio - visuelle . Selon la littérature existante , l'attention visuelle peut être étudié à mouvements oculaires , mais l'effet sonore sur le mouvement des yeux dans les vidéos est peu connue . L'objectif de cette thèse est d'étudier l'influence du son dans les vidéos sur le mouvement des yeux et de proposer un modèle de saillance audio - visuel pour prédire les régions saillants dans les vidéos avec plus de précision . A cet effet, nous avons conçu une première expérience audio - visuelle de poursuite oculaire . Nous avons créé une base de données d'extraits vidéo courts choisis dans divers films . Ces extraits ont été consultés par les participants , soit avec leur bande originale (condition AV ) , ou sans bande sonore ( état V) . Nous avons analysé la différence de positions de l'oeil entre les participants des conditions de AV et V . Les résultats montrent qu'il n'existe un effet du bruit sur le mouvement des yeux et l'effet est plus important pour la classe de la parole à l'écran . Ensuite , nous avons conçu une deuxième expérience audiovisuelle avec treize classes de sons. En comparant la différence de positions de l'oeil entre les participants des conditions de AV et V , nous concluons que l'effet du son est différente selon le type de son , et les classes avec la voix humaine ( c'est à dire les classes parole , chanteur , bruit humain et chanteurs ) ont le plus grand effet . Plus précisément , la source sonore a attiré considérablement la position des yeux uniquement lorsque le son a été la voix humaine . En outre , les participants atteints de la maladie de AV avaient une durée moyenne plus courte de fixation que de l'état de V . Enfin , nous avons proposé un modèle de saillance audio- visuel préliminaire sur la base des résultats des expériences ci-dessus . Dans ce modèle , deux stratégies de fusion de l'information audio et visuelle ont été décrits: l'un pour la classe de son discours , et l'autre pour la musique classe de son instrument . Les stratégies de fusion audio - visuelle définies dans le modèle améliore la prévisibilité à la condition AVHumans receive large quantity of information from the environment with sight and hearing. To help us to react rapidly and properly, there exist mechanisms in the brain to bias attention towards particular regions, namely the salient regions. This attentional bias is not only influenced by vision, but also influenced by audio-visual interaction. According to existing literature, the visual attention can be studied towards eye movements, however the sound effect on eye movement in videos is little known. The aim of this thesis is to investigate the influence of sound in videos on eye movement and to propose an audio-visual saliency model to predict salient regions in videos more accurately. For this purpose, we designed a first audio-visual experiment of eye tracking. We created a database of short video excerpts selected from various films. These excerpts were viewed by participants either with their original soundtrack (AV condition), or without soundtrack (V condition). We analyzed the difference of eye positions between participants with AV and V conditions. The results show that there does exist an effect of sound on eye movement and the effect is greater for the on-screen speech class. Then, we designed a second audio-visual experiment with thirteen classes of sound. Through comparing the difference of eye positions between participants with AV and V conditions, we conclude that the effect of sound is different depending on the type of sound, and the classes with human voice (i.e. speech, singer, human noise and singers classes) have the greatest effect. More precisely, sound source significantly attracted eye position only when the sound was human voice. Moreover, participants with AV condition had a shorter average duration of fixation than with V condition. Finally, we proposed a preliminary audio-visual saliency model based on the findings of the above experiments. In this model, two fusion strategies of audio and visual information were described: one for speech sound class, and one for musical instrument sound class. The audio-visual fusion strategies defined in the model improves its predictability with AV condition
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Marques, Alda Sofia Pires de Dias. "The use of computer aided lung sound analysis to characterise adventitious lung sounds : a potential outcome measure for respiratory therapy." Thesis, University of Southampton, 2008. https://eprints.soton.ac.uk/65639/.
Повний текст джерелаАнотація:
A barrier to assessing the effectiveness of respiratory physiotherapy has been insufficient accurate, reliable and sensitive outcome measures. Lung sounds provide useful, specific information for assessing and monitoring respiratory patients. However, standard auscultation techniques are too subjective to allow them to be used as an outcome measure. In this research, Computer Aided Lung Sound Analysis (CALSA) was used to assess whether adventitious lung sounds’ characteristics could be quantified clinically and used as a new objective, non-invasive, bedside clinical outcome measure for physiotherapy alveolar recruitment and airway clearance techniques. Two experimental studies were conducted incorporating ‘before-and-after’ and ‘repeated measures’ components. Fifty four participants with productive lung disorders (cystic fibrosis and bronchiectasis) were recruited from out-patient clinics. Demographic, anthropometric, lung function, oxygen saturation, breathlessness and lung sound data were collected at baseline and after a single intervention (selfintervention in the first study and intervention applied by a physiotherapist in the second study). The intra-subject reliability of crackle frequency (f) within each session was found to be ‘good’ to ‘excellent’, estimated by the Analysis of Variance, Intraclass Correlation Coefficient, Smallest Real Difference and Bland and Altman 95% limits of agreement. Crackle initial deflection width (IDW) and crackle two cycles deflection width (2CD) were reliable over short time periods. The f of crackles increased in the majority of participants post interventions. Agreement on the number (N) and timing (T) of crackles between CALSA and a physiotherapist‘s auscultatory findings was found to be poor in anterior chest sites, but higher in posterior sites. Conclusion: the use of CALSA to identify the type and f of adventitious lung sounds collected clinically is feasible; crackle IDW and 2CD are both reliable measures but crackle 2CD is more consistent; crackle f was more responsive than the N or T of crackles per breathing cycle to the interventions. In future, CALSA may provide an objective and responsive tool for assessing and monitoring respiratory interventions in clinical settings.
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Mantle, Hood Ki. "The Rasa of Sound." Bärenreiter Verlag, 2000. https://slub.qucosa.de/id/qucosa%3A36628.
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Chuang, Ming-Fei. "Interactive tools for sound signal analysis." Thesis, Monterey, California. Naval Postgraduate School, 1997. http://hdl.handle.net/10945/8550.
Повний текст джерелаАнотація:
Approved for public release; distribution is unlimitedThis thesis develops a series of programs that implement the sinusoidal representation model for speech and sound waveform analysis and synthesis. This sinusoidal representation model can also be used for a variety of sound signal transformations such as time-scale modification and frequency scaling. The above sound analysis/synthesis sinusoidal representations and transformations were developed as two interactive tools-with Graphical User Interface (GUI) using MATLAB. In addition, an interactive tool for signal frequency component editing based on the sinusoidal model is also presented in this thesis.
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Howarth, Craig. "Sound propagation over rigid porous layers." Thesis, Open University, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.293156.
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Smith, R. Sean. "Sound transmission through lightweight parallel plates." Thesis, Heriot-Watt University, 1997. http://hdl.handle.net/10399/1290.
Повний текст джерелаАнотація:
This thesis examines the transmission of sound through lightweight parallel plates, (plasterboard double wall partitions and timber floors). Statistical energy analysis was used to assess the importance of individual transmission paths and to determine the overall performance. Several different theoretical models were developed, the choice depending on the frequency range of interest and method of attachment of the plates, whether point or line, to the structural frame. It was found that for a line connected double wall there was very good agreement between the measured and predicted results, where the dominant transmission path was through the frame and the cavity path was weak. The transition frequency where the coupling changes from a line to a point connection is when the first half wavelength is able to fit between the spacings of the nails. For point connected double walls, where the transmission through the frame was weaker than for line connection, the cavity path was dominant unless there was absorption present. When the cavity was sufficiently deep, such that it behaved more like a room, the agreement between the measured and predicted results was good. As the cavity depth decreases the plates of the double wall are closer together and the agreement between the measured and predicted results were not as good. Detailed experiments were carried out to determine the transmission into the double wall cavities and isolated cavities. It was found that the transmission into an isolated cavity could be predicted well. However, for transmission into double wall cavities the existing theories could not predict transmission accurately when the cavity depth was small. Extensive parametric surveys were undertaken to analyse changes to the sound transmission through these structures when the material or design parameters are altered. The SEA models are able to identify the dominant mechanisms of transmission and will be a useful design tool in the design of lightweight partitions and timber floors.
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Rodeia, José Pedro dos Santos. "Analysis and recognition of similar environmental sounds." Master's thesis, FCT - UNL, 2009. http://hdl.handle.net/10362/2305.
Повний текст джерелаАнотація:
Dissertação apresentada na Faculdade de Ciências e Tecnologia da Universidade Nova de Lisboa para obtenção do grau de Mestre em Engenharia InformáticaHumans have the ability to identify sound sources just by hearing a sound. Adapting the same problem to computers is called (automatic) sound recognition. Several sound recognizers have been developed throughout the years. The accuracy provided by these recognizers is influenced by the features they use and the classification method implemented. While there are many approaches in sound feature extraction and in sound classification, most have been used to classify sounds with very different characteristics. Here, we implemented a similar sound recognizer. This recognizer uses sounds with very similar properties making the recognition process harder. Therefore, we will use both temporal and spectral properties of the sound. These properties will be extracted using the Intrinsic Structures Analysis (ISA) method, which uses Independent Component Analysis and Principal Component Analysis. We will implement the classification method based on k-Nearest Neighbor algorithm. Here we prove that the features extracted in this way are powerful in sound recognition. We tested our recognizer with several sets of features the ISA method retrieves, and achieved great results. We, finally, did a user study to compare human performance distinguishing similar sounds against our recognizer. The study allowed us to conclude the sounds are in fact really similar and difficult to distinguish and that our recognizer has much more ability than humans to identify them.
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Picard, Limpens Cécile. "Expressive sound synthesis for animation." Nice, 2009. http://www.theses.fr/2009NICE4075.
Повний текст джерелаАнотація:
The main objective of this thesis is to provide tools for an expressive and real-time synthesis of sounds resulting from physical interactions of various objects in a 3D virtual environment. Indeed, these sounds, such as collisions sounds or sounds from continuous interaction between surfaces, are difficult to create in a pre-production process since they are highly dynamic and vary drastically depending on the interaction and objects. To achieve this goal, two approaches are proposed; the first one is based on simulation of physical phenomena responsible for sound production, the second one based on the processing of a recordings database. According to a physically based point of view, the sound source is modelled as the combination of an excitation and a resonator. We first present an original technique to model the interaction force for continuous contacts, such as rolling. Visual textures of objects in the environment are reused as a discontinuity map to create audible position-dependent variations during continuous contacts. We then propose a method for a robust and flexible modal analysis to formulate the resonator. Besides allowing to handle a large variety of geometries and proposing a multi-resolution of modal parameters, the technique enables us to solve the problems of coherence between physics simulation and sound synthesis that are frequently encountered in animation. Following a more empirical approach, we propose an innovative method that consists in bridging the gap between direct playback of audio recordings and physically based synthesis by retargetting audio grains extracted from recordings according to the output of a physics engine. In an off-line analysis task, we automatically segment audio recordings into atomic grains and we represent each original recording as a compact series of audio grains. During interactive animations, the grains are triggered individually or in sequence according to parameters reported from the physics engine and/or userdefined procedures. Finally, we address fracture events which commonly appear in virtual environments, especially in video games. Because of their complexity that makes a purely physical-based model prohibitively expensive and an empirical approach impracticable for the large variety of micro-events, this thesis opens the discussion on a hybrid model and the possible strategies to combine a physically based approach and an empirical approach. The model aims at appropriately rendering the sound corresponding to the fracture and to each specific sounding sample when material breaks into piecesL'objectif principal de ce travail est de proposer des outils pour une synthèse en temps-réel, réaliste et expressive, des sons résultant d'interactions physiques entre objets dans une scène virtuelle. De fait, ces effets sonores, à l'exemple des bruits de collisions entre solides ou encore d'interactions continues entre surfaces, ne peuvent être prédéfinis et calculés en phase de pré-production. Dans ce cadre, nous proposons deux approches, la première basée sur une modélisation des phénomènes physiques à l'origine de l'émission sonore, la seconde basée sur le traitement d'enregistrements audio. Selon une approche physique, la source sonore est traitée comme la combinaison d'une excitation et d'un résonateur. Dans un premier temps, nous présentons une technique originale traduisant la force d'interaction entre surfaces dans le cas de contacts continus, tel que le roulement. Cette technique repose sur l'analyse des textures utilisées pour le rendu graphique des surfaces de la scène virtuelle. Dans un second temps, nous proposons une méthode d'analyse modale robuste et flexible traduisant les vibrations sonores du résonateur. Outre la possibilité de traiter une large variété de géométries et d'offrir une multi-résolution des paramètres modaux, la méthode permet de résoudre le problème de cohérence entre simulation physique et synthèse sonore, problème fréquemment rencontré en animation. Selon une approche empirique, nous proposons une technique de type granulaire, exprimant la synthèse sonore par un agencement cohérent de particules ou grains sonores. La méthode consiste tout d'abord en un prétraitement d'enregistrements destiné à constituer un matériel sonore sous forme compacte. Ce matériel est ensuite manipulé en temps réel pour, d'une part, une resynthèse complète des enregistrements originaux, et d'autre part, une utilisation flexible en fonction des données reportées par le moteur de simulation et/ou de procédures prédéfinies. Enfin, l'intérêt est porté sur les sons de fracture, au vu de leur utilisation fréquente dans les environnements virtuels, et en particulier les jeux vidéos. Si la complexité du phénomène rend l'emploi d'un modèle purement physique très coûteux, l'utilisation d'enregistrements est également inadaptée pour la grande variété de micro-événements sonores. Le travail de thèse propose ainsi un modèle hybride et des stratégies possibles afin de combiner une approche physique et une approche empirique. Le modèle ainsi conçu vise à reproduire l'événement sonore de la fracture, de son initiation à la création de micro-débris
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Hällgren, Nina. "Designing with Urban Sound : Exploring methods for qualitative sound analysis of the built environment." Licentiate thesis, KTH, Arkitektur, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-240078.
Повний текст джерелаАнотація:
The licentiate thesis Designing with Urban Sound explores the constitution and qualitative characteristics of urban sonic space from a design-oriented and practice-based perspective. The act of lifting forth and illuminating the interaction between architecture, the creation of sound and a sonic experience aims to examine and develop useful tools and methods for the representation, communication and analysis of the exterior sonic environment in complex architectural spaces. The objective is to generate theoretical and practical knowledge within the field of urban sound planning and design by showing examples of different and complementary ways of communicating and analyzing sound than those which are commonly recognized.Licentiatavhandlingen Designa med stadens ljud undersöker det urbana ljudrummets konstitution och kvalitativa egenskaper utifrån ett designorienterat och praktiknära perspektiv. Avsikten med arbetet är att utveckla verktyg och metoder för representation, kommunikation och analys av stadens exteriöra ljudmiljö genom att synliggöra interaktionen mellan arkitektur, ljudbildning och upplevelse. Genom att visa exempel på andra sätt att kommunicera och analysera ljud i staden än dagens vedertagna metoder, är syftet är att bidra till kunskapsutvecklingen inom fältet för urban ljudplanering- och design.
QC 20181211
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Banwell, Guy. "Impulsive sound quality of tennis rackets." Thesis, Loughborough University, 2013. https://dspace.lboro.ac.uk/2134/13678.
Повний текст джерелаАнотація:
The sound radiated during a tennis impact has been shown to greatly influence the perception of 'feel' but has also been suggested to contribute to the perception of equipment quality, as is the case with many other consumer products. In an industry that is governed by design limitations, tennis racket manufactures are continually aiming to differentiate their products from their competitors' and the sound character of a racket is seen as one such method to do so. In order to control the sound radiated from a tennis racket it was first necessary to identify how a tennis racket radiates sound. Through a number of controlled player tests, involving groundstrokes and serves, the sound was recorded along with measurements of the physical vibrations excited in the frame. Analysis of the data revealed typical characteristics of the sound in the time and frequency domain. The sound was split into two sections to aid the analysis; an initial impulsive component that decayed very quickly and a ringing component that was of much lower amplitude but decayed at a much slower rate. The evolution of the frequency content over time was also investigated, however, the data provided much more information as to where each frequency component originated by analysing the data together with the experimental modal analysis data. The experimental modal analysis of a tennis racket was a vital stage in understanding which components of the racket were responsible for radiating the identified frequency component in the sound spectrum. The investigations identified frame bending modes, out-of-plane and inplane, as well as torsional modes, hoop modes and stringbed modes. To enable direct comparison between the natural frequencies excited during a tennis shot and the frequencies recorded from a freely suspended racket, the effect of the hand on the modal behaviour of the racket was analysed; experimental modal analysis data from a hand-gripped racket was compared with data generated by adding simulated mass to the modal model of the freely suspended racket. The first stringbed mode was identified as a key contributor to the sound of the racket, especially in the ringing component of the sound. Analysis of the physical vibrations in the frame following a tennis shot revealed that the first stringbed mode excited the frame of the racket, which is iv thought to be the reason why stringbed modes contribute to the sound more so than their surface area would suggest that they are capable of. Analysis of the relationship between subjective perceptions of players and calculated sound metrics suggest a negative correlation exists between the duration and loudness of a tennis shot and the appeal of the sound. Further investigations involving a jury to evaluate the appeal of a tennis impact sound, modified in terms of duration, revealed further evidence to support the theory that tennis impact sounds of greater duration are less appealing than those that decay quickly. This is the first study to investigate the sound radiated from a tennis racket and has identified how the sound is radiated from the racket and also suggests which parameters of the sound are considered appealing to tennis players. With this information it is possible to suggest design modifications that would influence the sound of a tennis racket in a predictable manner.
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Lång, Magnus. "Sound and Complete Reachability Analysis under PSO." Thesis, Uppsala universitet, Institutionen för informationsteknologi, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-213286.
Повний текст джерелаАнотація:
Modern multiprocessor systems use weak (relaxed) memory models in order to execute memory sharing multi-threaded code in an efficient manner, but are much harder for programmers to reason about than systems using the sequential consistency memory model. The SB abstraction and its implementation in the Memorax tool allows sound and complete checking of control state reachability under the TSO memory model, used in modern x86 processors. In this paper, I present a formalisation of the PSO memory model using the semantics of the Sun SPARC documentation and an alternate semantic, called Partial Write Serialisation, I conjecture to be equivalent with my formalisation under the control state reachability problem. PWS is proved to be a well-structured system which allows sound and complete reachability analysis. An implementation of PWS is presented as part of the Memorax tool and demonstrated experimentally to be capable of analysing reachability and inferring minimal fence sets on many non-trivial and real world examples in reasonable time and memory usage.
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Mgdob, Hosam Mohamed. "Heart sound acquisition system and signal analysis." Thesis, University of Sussex, 2004. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.400032.
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Hebden, John Edward. "Acquisition and analysis of heart sound data." Thesis, University of Sussex, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.360518.
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Toop, Leslie John. "Cough sound analysis in children with asthma." Thesis, University of Bristol, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.361104.
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Liuni, Marco. "Automatic adaptation of sound analysis and synthesis." Paris 6, 2012. http://www.theses.fr/2012PA066105.
Повний текст джерелаАнотація:
En Analyse Temps-Fréquence, l'adaptativité est la possibilité de concevoir de représentations et opérateurs avec des caractéristiques qui peuvent être modifiées en fonction des objets à analyser. Dans cette thèse, on s'intéresse à des méthodes qui permettent de varier localement la résolution temps-fréquence pour l'analyse et la re-synthèse du son. Le premier objectif fondamental est la définition formelle d'un cadre mathématique qui puisse engendrer des méthodes adaptatives pour l'analyse du son. Le deuxième est de rendre l'adaptation automatique; on établit des critères pour définir localement la meilleure résolution temps-fréquence, en optimisant des mesures de parcimonie appropriées. Afin d'exploiter l'adaptativité dans le traitement spectral du son, on introduit des méthodes de reconstruction efficaces, basées sur des analyses à résolution variable, conçues pour préserver et améliorer les techniques actuelles de manipulation du son. L'idée principale est que les algorithmes adaptatifs puissent contribuer à la simplification de l'utilisation de méthodes de traitement du son qui nécessitent aujourd'hui un haut niveau d'expertise. En particulier, la nécessité d'une configuration manuelle détaillée constitue une limitation majeure dans les applications grand public de traitement du son de haute qualité (par exemple: transposition, compression/dilatation temporelle). Nous montrons des exemples où la gestion automatique de la résolution temps-fréquence permet non seulement de réduire significativement les paramètres à régler, mais aussi d'améliorer la qualité des traitementsIn Time-Frequency Analysis, adaptivity is the possibility to conceive representations and operators whose characteristics can be modeled according to their input. In this work, we look for methods providing for a local variation of the time-frequency resolution for sound analysis and re-synthesis. The first and fundamental objective is thus the formal definition of mathematical models whose interpretation leads to theoretical and algorithmic methods for adaptive sound analysis. The second objective is to make the adaptation automatic; we establish criteria to define the best local time-frequency resolution, with the optimization of appropriate sparsity measures. To be able to exploit adaptivity in spectral sound processing, we then introduce efficient re-synthesis methods based on analyses with varying resolution, designed to preserve and improve the existing sound transformation techniques. Our main assumption is that algorithms based on adaptive representations will help to establish a generalization and simplification for the application of signal processing methods that today still require expert knowledge. In particular, the need to provide manual low level configuration is a major limitation for the use of advanced signal processing methods by large communities. The possibility to dispose of an automatic time-frequency resolution drastically limits the parameters to set, without affecting, and even ameliorating, the treatment quality: the result is an improvement of the user experience, even with high-quality sound processing techniques, like transposition and time-stretch
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Schwarz, Diemo. "Spectral envelopes in sound analysis and synthesis." [S.l.] : Universität Stuttgart , Fakultät Informatik, 1998. http://www.bsz-bw.de/cgi-bin/xvms.cgi?SWB7084238.
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Lee, Siew-Eang. "Transmission of sound through non-homogeneous walls." Thesis, Heriot-Watt University, 1985. http://hdl.handle.net/10399/1656.
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Wang, Shuai School of Electrical Engineering & Telecommunication UNSW. "Soundfield analysis and synthesis: recording, reproduction and compression." Awarded by:University of New South Wales. School of Electrical Engineering and Telecommunication, 2007. http://handle.unsw.edu.au/1959.4/31502.
Повний текст джерелаАнотація:
Globally, the ever increasing consumer interest in multichannel audio is a major factor driving the research intent in soundfield reconstruction and compression. The popularity of the well commercialized 5.1 surround sound system and its 6-Channel audio has been strongly supported by the advent of powerful storage medium, DVD, as well as the use of efficient telecommunication techniques. However, this popularity has also revealed potential problems in the development of soundfield systems. Firstly, currently available soundfield systems have rather poor compatibility with irregular speaker arrangements. Secondly, bandwidth requirement is dramatically increased for multichannel audio representation with good temporal and spatial fidelity. This master???s thesis addresses these two major issues in soundfield systems. It introduces a new approach to analyze and sysnthesize soundfield, and compares this approach with currently popular systems. To facilitate this comparison, the behavior of soundfield has been reviewed from both physical and psychoacoustic perspectives, along with an extensive study of past and present soundfield systems and multichannel audio compression algorithms. The 1th order High Spatial Resolution (HSR) soundfield recording and reproduction has been implemented in this project, and subjectively evaluated using a series of MUSHRA tests to finalize the comparison.
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Bastos, Maíza de Lavenère. "Sound and language in discourse." Florianópolis, SC, 2007. http://repositorio.ufsc.br/xmlui/handle/123456789/90508.
Повний текст джерелаАнотація:
Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro de Comunicação e Expressão. Programa de Pós-Graduação em Inglês e Literatura Correspondente.Made available in DSpace on 2012-10-23T11:41:47Z (GMT). No. of bitstreams: 1 244416.pdf: 495366 bytes, checksum: ece5b31c790c4a28c9f7071f79b9f72a (MD5)
Tendo em vista o crescente interesse por análises multimodais de diferentes tipos de mídia, meu objetivo neste trabalho é investigar como as relações e representações entre alguns dos personagens do filme "Cidade dos Sonhos", de David Lynch, são estabelecidos através de elementos não-verbais e verbais. Os elementos analisados são aspectos da trilha sonora e as escolhas lingüísticas dos personagens em quatro cenas do filme. Para isso, base teórica para a análise de trilha sonora desenvolvida neste trabalho se situa na área de multimodalidade, mais especificamente na área de semiótica do som proposta por van Leeuwen (1999). Somados à análise de elementos da trilha sonora, a perspectiva teórica aplicada à analise dos elementos verbais tem como base aspectos da Gramática Sistêmica Funcional proposta por Halliday (1984; Halliday & Matthiessen, 2004). Os resultados possibilitam o entendimento da construção de relações entre os personagens e classificação de detalhes importantes no filme, que são evidenciados pelo uso de qualidade de voz, juntamente com as escolhas lingüísticas dos personagens. Tais conclusões podem contribuir para o entendimento de como uma comunicação não-verbal e verbal pode construir relações e representações. Bearing in mind the increasing interest for multimodal analysis of different types of media, my objective in this study is to investigate how relationships and representations among some of the characters of the film Mulholland Drive, by David Lynch, are established through non-verbal and verbal modes. The analysis focuses on elements of the soundtrack and the linguistic choices of specific characters in four scenes of the film. The theoretical basis for the analysis of the elements of the soundtrack derives from the area of multimodality according to the semiotic of sound as proposed by van Leeuwen (1999). And the theoretical perspective for the analysis of the verbal elements draws on aspects of Systemic Functional Linguistics as proposed by Halliday (1984; Halliday & Matthiessen, 2004). Results enable the understanding of the establishment of relationships among characters and classification of important details in the film, which are evidenced by the use of voice quality together with their linguistic choices. Such findings may contribute to the understanding of how representations and relationships may be built through non-verbal and verbal communication.
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Brgulja, Amir. "A Real-Time Electronic Sound Analysis System with Graphical User Interface." Thesis, University of North Texas, 2011. https://digital.library.unt.edu/ark:/67531/metadc84180/.
Повний текст джерелаАнотація:
Noise-induced hearing loss is a serious problem common to musical environments. Current dosimetry technology is primarily designed for industrial environments and not suited for musical settings. At present, there are no government regulations that apply to the educational music environment as it relates to monitoring and prevention of hearing loss. Also, no system exists than can serve as a proactive tool in observation and reporting of sound exposure levels with the goal of hearing conservation. Newly proposed system takes a software based approach in designing a proactive dosimetry system that can assess the risk of sound noise exposure. It provides real-time feedback trough a graphical user interface that is capable of database storage for further study.
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Barlow, G. N. D. "Sound speed measurement and variabilities in the sea." Thesis, Bangor University, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.332805.
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Wong, Lawrence Yik-Lung. "Adaptive system modelling for active attenuation of sound." Thesis, Open University, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.316967.
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Silcox, Richard J. "Active control of multi-modal sound in waveguides." Thesis, University of Southampton, 1994. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.339481.
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Orduna-Bustamante, Felipe. "Digital signal processing for multi-channel sound reproduction." Thesis, University of Southampton, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.261565.
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Joseph, P. F. "Active control of high frequency enclosed sound fields." Thesis, University of Southampton, 1990. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.280927.
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Hughes, Ian John. "The scattering of sound from some practical surfaces." Thesis, University of Cambridge, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.292984.
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Roemer, Jake. "Practical High-Coverage Sound Predictive Race Detection." The Ohio State University, 2019. http://rave.ohiolink.edu/etdc/view?acc_num=osu1563505463237874.
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Lundin, Magnus. "Sound transmission through double walls usingStatistical Energy Analysis." Thesis, KTH, MWL Strömningsakustik, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-30642.
Повний текст джерелаАнотація:
A new Statistical Energy Analysis (SEA) formulation for double walls isderived. The new formulation uses three elements to describe the double wall,one for each plate and the third for the cavity including the mass-controlledmodes of the plates. This means that the influence of the double wall resonance(mass-air-mass resonance) can be predicted by SEA. Calculations aremade and compared to measurements, showing fair agreement.
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Mattes, Symeon. "Perceptual models for sound field analysis and synthesis." Thesis, University of Southampton, 2016. https://eprints.soton.ac.uk/397216/.
Повний текст джерелаАнотація:
This thesis describes the methodology that has been followed for the implementation of a biologically inspired auditory signal processing model that predicts human sound localization of stationary acoustic sound sources in 3D space. The intended use of the the model is for the evaluation of audio systems. An attempt is made to develop both a theoretical and mathematical framework that can be adopted as a generalized theory for the development of biologically inspired models of human sound localization. The model makes use of a combination of monaural and binaural cues and within a psychoacoustical framework makes predictions of the location of a sound source given the sound presence signals delivered to the ears. Finally, the effectiveness of the model is evaluated through comparison with the experimental results of a listening test in which a number of human subjects made judgements of the location of real sound sources in 3D space under anechoic conditions.
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Pietila, Glenn M. "Intelligent Systems Approaches to Product Sound Quality Analysis." University of Cincinnati / OhioLINK, 2013. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1367936769.
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40
Akil, Husein Avicenna. "The scattering of sound by fittings in industrial buildings." Thesis, University of Liverpool, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.318263.
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41
Tussing, Timothy Mark. "Analysis of Effects on Sound Using the Discrete Fourier Transform." The Ohio State University, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=osu1338371732.
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42
Picard-Limpens, Cécile. "Expressive Sound Synthesis for Animation." Phd thesis, Université de Nice Sophia-Antipolis, 2009. http://tel.archives-ouvertes.fr/tel-00440417.
Повний текст джерелаАнотація:
L'objectif principal de ce travail est de proposer des outils pour une synthèse en temps-réel, réaliste et expressive, des sons résultant d'interactions physiques entre objets dans une scène virtuelle. De fait, ces effets sonores, à l'exemple des bruits de collisions entre solides ou encore d'interactions continues entre surfaces, ne peuvent être prédéfinis et calculés en phase de pré-production. Dans ce cadre, nous proposons deux approches, la première basée sur une modélisation des phénomènes physiques à l'origine de l'émission sonore, la seconde basée sur le traitement d'enregistrements audio. Selon une approche physique, la source sonore est traitée comme la combinaison d'une excitation et d'un résonateur. Dans un premier temps, nous présentons une technique originale traduisant la force d'interaction entre surfaces dans le cas de contacts continus, tel que le roulement. Cette technique repose sur l'analyse des textures utilisées pour le rendu graphique des surfaces de la scène virtuelle. Dans un second temps, nous proposons une méthode d'analyse modale robuste et flexible traduisant les vibrations sonores du résonateur. Outre la possibilité de traiter une large variété de géométries et d'offrir une multi-résolution des paramètres modaux, la méthode permet de résoudre le problème de cohérence entre simulation physique et synthèse sonore, problème fréquemment rencontré en animation. Selon une approche empirique, nous proposons une technique de type granulaire, exprimant la synthèse sonore par un agencement cohérent de particules ou grains sonores. La méthode consiste tout d'abord en un prétraitement d'enregistrements destiné à constituer un matériel sonore sous forme compacte. Ce matériel est ensuite manipulé en temps réel pour, d'une part, une resynthèse complète des enregistrements originaux, et d'autre part, une utilisation flexible en fonction des données reportées par le moteur de simulation et/ou de procédures prédéfinies. Enfin, l'intérêt est porté sur les sons de fracture, au vu de leur utilisation fréquente dans les environnements virtuels, et en particulier les jeux vidéos. Si la complexité du phénomène rend l'emploi d'un modèle purement physique très coûteux, l'utilisation d'enregistrements est également inadaptée pour la grande variété de micro-événements sonores. Le travail de thèse propose ainsi un modèle hybride et des stratégies possibles afin de combiner une approche physique et une approche empirique. Le modèle ainsi conçu vise à reproduire l'événement sonore de la fracture, de son initiation à la création de micro-débris.
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Miao, Hsu-Chiang. "Analysis of nonsymmetric effects in finite amplitude sound beams." Diss., Georgia Institute of Technology, 1985. http://hdl.handle.net/1853/15959.
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Pyž, Gražina. "Analysis and synthesis of Lithuanian phoneme dynamic sound models." Doctoral thesis, Lithuanian Academic Libraries Network (LABT), 2013. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2013~D_20131125_134056-50408.
Повний текст джерелаАнотація:
Speech is the most natural way of human communication. Text-to-speech (TTS) problem arises in various applications: reading email aloud, reading text from e-book aloud, services for the people with speech disorders. Construction of speech synthesizer is a very complex task. Researchers are trying to automate speech synthesis. In order to solve the problem of Lithuanian speech synthesis, it is necessary to develop mathematical models for Lithuanian speech sounds. The research object of the dissertation is Lithuanian vowel and semivowel phoneme models. The proposed vowel and semivowel phoneme models can be used for developing a TTS formant synthesizer. Lithuanian vowel and semivowel phoneme modelling framework based on a vowel and semivowel phoneme mathematical model and an automatic procedure of estimation of the vowel phoneme fundamental frequency and input determining is proposed. Using this framework, the phoneme signal is described as the output of a linear multiple-input and single-output (MISO) system. The MISO system is a parallel connection of single-input and single-output (SISO) systems whose input impulse amplitudes vary in time. Within this framework two synthesis methods are proposed: harmonic and formant. Simulation has revealed that that the proposed framework gives sufficiently good vowel and semivowel synthesis quality.Kalba yra natūralus žmonių bendravimo būdas. Teksto-į-šneką (TTS) problemos atsiranda įvairiose srityse: elektroninių laiškų skaitymas balsu, teksto iš elektroninių knygų skaitymas balsu, paslaugos kalbos sutrikimų turintiems žmonėms. Kalbos sintezatoriaus kūrimas yra be galo sudėtingas uždavinys. Įvairių šalių mokslininkai bando automatizuoti kalbos sintezę. Siekiant išspręsti lietuvių kalbos sintezės problemą, būtina kurti naujus lietuvių kalbos garsų matematinius modelius. Disertacijos tyrimo objektas yra dinaminiai lietuviškos šnekos balsių ir pusbalsių fonemų modeliai. Pasiūlyti balsių ir pusbalsių fonemų dinaminiai modeliai gali būti panaudoti kuriant formantinį kalbos sintezatorių. Garsams aprašyti pasiūlyta modeliavimo sistema pagrįsta balsių ir pusbalsių fonemų matematiniu modeliu bei pagrindinio tono ir įėjimų nustatymo automatine procedūra. Fonemos signalas yra gaunamas kai daugelio-įėjimų ir vieno-išėjimo (MISO) sistemos išėjimas. MISO sistema susideda iš lygiagrečiai sujungtų vieno-įėjimo ir vieno-išėjimo (SISO) sistemų, kurių įėjimų amplitudes kinta laike. Disertacijoje du sintezės metodai sukurti: harmoninis ir formantinis. Eksperimentiniai rezultatai parodė, kad balsiai ir pusbalsiai sintezuoti minėta sistema skamba pakankamai natūraliai.
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Powell, Kenneth John. "Multivariate analysis of underwater sounds." Thesis, University of Exeter, 1997. http://create.canterbury.ac.uk/10470/.
Повний текст джерелаАнотація:
This thesis considers the use of multivariate statistical methods in relation to a common signal processing problem, that of detecting features in sound recordings which contain interference, distortion and background noise. Two separate but related areas of study are undertaken, first, the compression and noise reduction of sounds; second, the detection of intermittent departures ('signals') from the background sound environment ('noise'), where the latter may be evolving and changing over time. Compression and noise reduction are two closely related areas that have been studied for a wide range of signals, both one dimensional (such as sound) and two dimensional (such as images). Many well known techniques used in this field are based on the Fourier transform. In this work, we show how the comparative recent wavelet transform is superior for sound data involving short duration signals (such as shrimp clicks) whilst being at least as good as the Fourier transform for longer duration signals (such as dolphin whistles). Various noise reduction techniques involving thresholding wavelet transforms are examined and compared. We show how none of the standard threshold methods cope well with underwater sounds to any reasonable degree and propose a new technique, known as RunsThresh to overcome the perceived problems. The performance of this new method is contrasted with that of various standard thresholds. Signal identification for underwater sounds is an area that has been examined in detail in much previous work. Here, we build upon the results gleaned from noise reduction to develop a methodology for detecting signals. The underwater noise environment is dynamically modelled using recursive density estimation of certain summary features of its wavelet decomposition. Observations which are considered to be outliers from this distribution are flagged as 'signal'. The performance of our signal detection method is illustrated on artificial data, containing known signals, and on real data. This performance is compared with standard Fourier based methods for both cases. Finally in this thesis, several ideas are presented and discussed which consider how the noise reduction and signal detection techniques examined in earlier chapters could be developed further, for example, in order to classify detected signals into different classes. These ideas are presented in outline only and are not followed up in detail, since they represent interesting directions for future study, rather than a primary focus of this thesis.
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Jamieson, Gary. "Spectral analysis of pulmonary sounds." Thesis, University of Liverpool, 1993. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.240595.
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47
Hosseini-Hashemi, S. "The sound and vibration resulting from the impact of spheres." Thesis, City University London, 1985. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.370937.
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48
Khirnykh, Konstantin. "The development of non fibre-filled resonant sound absorbing systems." Thesis, London South Bank University, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.303943.
Повний текст джерелаАнотація:
The development of a novel type of resonant sound absorber is described, designed specifically for the absorption of sound at low frequencies and at high intensities. A review of previous work on resonant sound absorbers shows that existing theoretical models for describing the phenomena are incomplete and that there remains a need for further extensions to the relevant models to cater for non-linear effects which become particularly important at high intensities. It is also shown that there are limitations to the current methods for testing absorbers which make them less suitable at low frequency and when the signals are nonharmonic. In the present work a theoretical model of a Helmholtz type of absorption resonator working in the nonlinear regime is developed using nonlinear hydrodynamic equations for viscous incompressible fluids. The model is able to predict the input impedance, the resonant frequency and the absorption coefficient of the device under nonlinear conditions. The model is also used to account for "difference frequency" generation, the reflection of signals of any shape from the surface, and the isolation characteristics of this type of absorber when a porous layer is fixed to the back of the resonant cavity. A new method for the development of acoustic characteristic~ of resonant sound absorbers is described, which was developed as part of the present work and which overcomes the limitations of existing methods. Measurements carried out using this and other techniques show that the acoustic characteristics of the absorber described are very close to those predicted from the model. A new type of packless absorber (one without fibrous material) based on the above theoretical model is described. Such an absorber has been constructed and tested in a reverberant sound field and is shown to provide effective sound absorbtion under conditions typical of a working industrial environment.
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Caracalla, Hugo. "Sound texture synthesis from summary statistics." Electronic Thesis or Diss., Sorbonne université, 2019. http://www.theses.fr/2019SORUS676.
Повний текст джерелаАнотація:
Les textures sonores sont une catégorie de sons incluant le bruit de la pluie, le brouhaha d’une foule ou les pépiements d’un groupe d’oiseaux. Tous ces sons contiennent une part d’imprévisibilité qui n’est habituellement pas recherchée en synthèse sonore, et rend ainsi indispensable l’utilisation d’algorithmes dédiés. Cependant, la grande diversité de leurs propriétés complique la création d’un algorithme capable de synthétiser un large panel de textures. Cette thèse s’intéresse à la synthèse paramétrique de textures sonores. Dans ce paradigme, un ensemble de statistiques sont extraites d’une texture cible et progressivement imposées sur un bruit blanc. Si l’ensemble de statistiques est pertinent, le bruit blanc est alors modifié jusqu’à ressembler à la cible, donnant l’illusion d’avoir été enregistré quelques instants après. Dans un premier temps, nous proposons l’amélioration d’une méthode paramétrique basée sur des statistiques perceptuelles. Cette amélioration vise à améliorer la synthèse d’évènements à forte attaque ou singuliers en modifiant et simplifiant le processus d’imposition. Dans un second temps, nous adaptons une méthode paramétrique de synthèse de textures visuelles basée sur des statistiques extraites par un réseau de neurones convolutifs (CNN) afin de l’utiliser sur des textures sonores. Nous modifions l’ensemble de statistiques utilisées afin de mieux correspondre aux propriétés des signaux sonores, changeons l’architecture du CNN pour l’adapter aux événements présents dans les textures sonores et utilisons une représentation temps-fréquence prenant en compte à la fois amplitude et phaseSound textures are a wide class of sounds that includes the sound of the rain falling, the hubbub of a crowd and the chirping of flocks of birds. All these sounds present an element of unpredictability which is not commonly sought after in sound synthesis, requiring the use of dedicated algorithms. However, the diverse audio properties of sound textures make the designing of an algorithm able to convincingly recreate varied textures a complex task. This thesis focuses on parametric sound texture synthesis. In this paradigm, a set of summary statistics are extracted from a target texture and iteratively imposed onto a white noise. If the set of statistics is appropriate, the white noise is modified until it resemble the target, sounding as if it had been recorded moments later. In a first part, we propose improvements to perceptual-based parametric method. These improvements aim at making its synthesis of sharp and salient events by mainly altering and simplifying its imposition process. In a second, we adapt a parametric visual texture synthesis method based statistics extracted by a Convolutional Neural Networks (CNN) to work on sound textures. We modify the computation of its statistics to fit the properties of sound signals, alter the architecture of the CNN to best fit audio elements present in sound textures and use a time-frequency representation taking both magnitude and phase into account
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Metcalfe, Paul N. "Learning and teaching the art of sound design: An analysis into determining best practices." Thesis, Queensland University of Technology, 2017. https://eprints.qut.edu.au/102704/8/Paul_Metcalfe_Thesis.pdf.
Повний текст джерелаАнотація:
The aim of this study was to investigate the learning and teaching of sound design in higher education contexts. Two main aspects were examined: firstly, an investigation into published resources, addressing sound design as creative practice, and, secondly, an examination of the relationship between technical process and creative skills development within curriculum design. The perceived educational benefits were also investigated, of collaborative projects and other practical exercises, contrasting them with technical, skills-based learning. Finally, the study suggests avenues for future development that may provide better access to reference materials that support sound design students' development as creative, industry ready practitioners.