Academic literature on the topic 'Flux audio'
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Journal articles on the topic "Flux audio"
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Bryant, M. D. "Bond Graph Models for Linear Motion Magnetostrictive Actuators." Journal of Dynamic Systems, Measurement, and Control 118, no. 1 (March 1, 1996): 161–67. http://dx.doi.org/10.1115/1.2801139.
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Bond graph models for the audio range response of a dynamically continuous, linear motion magnetostrictive actuator are formulated and presented. The actuator involves a continuous rod of magnetostrictive material that extends, contracts, and vibrates in modes when energized by magnetic flux produced by a coil. The left end is fixed, force is extracted from the right end. The bond graph model includes dynamics of the energizing coil, the flux routing circuit, magnetic to mechanical energy conversion, and mechanical elements. Constitutive relations for magnetostriction suggest use of a multipart capacitor with ports for magnetic and mechanical power flow; constraints imposed by modal dynamics require a separate mechanical port for each vibration mode. Values were assigned to bond graph parameters in a non-empirical manner: solely from theory and handbook data. State equations and transfer functions were extracted from the bond graph. For audio range operation, theory (the bond graph model) compared well with experiment (measurements taken on a magnetostrictive actuator designed and built by the author).
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Stupacher, Jan, Michael J. Hove, and Petr Janata. "Audio Features Underlying Perceived Groove and Sensorimotor Synchronization in Music." Music Perception 33, no. 5 (June 1, 2016): 571–89. http://dx.doi.org/10.1525/mp.2016.33.5.571.
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The experience of groove is associated with the urge to move to a musical rhythm. Here we focus on the relevance of audio features, obtained using music information retrieval (MIR) tools, for explaining the perception of groove and music-related movement. In Study 1 we extracted audio features from clips of real music previously rated on perceived groove. Measures of variability, such as the variance of the audio signal’s RMS curve and spectral flux (particularly in low frequencies), predicted groove ratings. Additionally, we dissociated two forms of event density, showing that an algorithm that emphasizes variability between beats predicted groove ratings better. In Study 2 we manipulated RMS levels and groove category (low, mid, and high groove) to confirm that perceived groove is not a function of loudness. In Study 3 we utilized novel music clips that manipulated the frequency of bass and bass drum (low vs. high) and attack time (short vs. long). Groove ratings and tapping velocities tended to be higher and tapping variability tended to be lower when the bass instruments had lower frequencies. The present findings emphasize the multifaceted nature of groove by linking audio and musical qualities to subjective experience and motor behavior.
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Rossetti, Danilo, and Jônatas Manzolli. "Analysis of Granular Acousmatic Music: Representation of sound flux and emergence." Organised Sound 24, no. 02 (August 2019): 205–16. http://dx.doi.org/10.1017/s1355771819000244.
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Analysing electroacoustic music is a challenging task that can be approached by different strategies. In the last few decades, newly emerging computer environments have enabled analysts to examine the sound spectrum content in greater detail. This has resulted in new graphical representation of features extracted from audio recordings. In this article, we propose the use of representations from complex dynamical systems such as phase space graphics in musical analysis to reveal emergent timbre features in granular technique-based acousmatic music. It is known that granular techniques applied to musical composition generate considerable sound flux, regardless of the adopted procedures and available technological equipment. We investigate points of convergence between different aesthetics of the so-called Granular Paradigm in electroacoustic music, and consider compositions employing different methods and techniques. We analyse three works: Concret PH (1958) by Iannis Xenakis, Riverrun (1986) by Barry Truax, and Schall (1996) by Horacio Vaggione. In our analytical methodology, we apply such concepts as volume and emergence, as well as their graphical representation to the pieces. In conclusion we compare our results and discuss how they relate to the three composers’ specific procedures creating sound flux as well as to their compositional epistemologies and ontologies.
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Valiveti, Hima Bindu, Anil Kumar B., Lakshmi Chaitanya Duggineni, Swetha Namburu, and Swaraja Kuraparthi. "Soft computing based audio signal analysis for accident prediction." International Journal of Pervasive Computing and Communications 17, no. 3 (March 26, 2021): 329–48. http://dx.doi.org/10.1108/ijpcc-08-2020-0120.
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Purpose Road accidents, an inadvertent mishap can be detected automatically and alerts sent instantly with the collaboration of image processing techniques and on-road video surveillance systems. However, to rely exclusively on visual information especially under adverse conditions like night times, dark areas and unfavourable weather conditions such as snowfall, rain, and fog which result in faint visibility lead to incertitude. The main goal of the proposed work is certainty of accident occurrence. Design/methodology/approach The authors of this work propose a method for detecting road accidents by analyzing audio signals to identify hazardous situations such as tire skidding and car crashes. The motive of this project is to build a simple and complete audio event detection system using signal feature extraction methods to improve its detection accuracy. The experimental analysis is carried out on a publicly available real time data-set consisting of audio samples like car crashes and tire skidding. The Temporal features of the recorded audio signal like Energy Volume Zero Crossing Rate 28ZCR2529 and the Spectral features like Spectral Centroid Spectral Spread Spectral Roll of factor Spectral Flux the Psychoacoustic features Energy Sub Bands ratio and Gammatonegram are computed. The extracted features are pre-processed and trained and tested using Support Vector Machine (SVM) and K-nearest neighborhood (KNN) classification algorithms for exact prediction of the accident occurrence for various SNR ranges. The combination of Gammatonegram with Temporal and Spectral features of the validates to be superior compared to the existing detection techniques. Findings Temporal, Spectral, Psychoacoustic features, gammetonegram of the recorded audio signal are extracted. A High level vector is generated based on centroid and the extracted features are classified with the help of machine learning algorithms like SVM, KNN and DT. The audio samples collected have varied SNR ranges and the accuracy of the classification algorithms is thoroughly tested. Practical implications Denoising of the audio samples for perfect feature extraction was a tedious chore. Originality/value The existing literature cites extraction of Temporal and Spectral features and then the application of classification algorithms. For perfect classification, the authors have chosen to construct a high level vector from all the four extracted Temporal, Spectral, Psycho acoustic and Gammetonegram features. The classification algorithms are employed on samples collected at varied SNR ranges.
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Stanton, Polly. "Sound, listening and the moving image." Qualitative Research Journal 19, no. 1 (February 4, 2019): 65–71. http://dx.doi.org/10.1108/qrj-12-2018-0019.
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Purpose As an artist working with sound and the moving image, an in-between space is revealed, a flux between two distinct mediums that intersect as temporal experience and sensory synchronisation. The audio–visual relationship is a pattern of constantly shifting moments of connection and discordance, an ephemeral dance of timing and rhythm that binds together to create a cinematic expression of time and event. The paper aims to discuss this issue. Design/methodology/approach In this paper, the author will consider the audio-visual event and the space that exists between the visual and the sonic via the frame of my own art practice. Through this context, the author will examine audio–visual relations from practice through to presentation, challenging the belief that sound is merely a support for the moving image and propose that it is an equal if not driving force in the audio-visual contract. The author will also investigate sound-based disciplines that the author utilize in my own work, all of which highlight the materiality of sound and how it can be engaged to directly affect the production and installation of moving image works in a gallery context. Findings Utilizing listening in this way has revealed surprising or overlooked connections that visually the author would otherwise have not acknowledged. It has helped link together interests across geography and cartography by expanding on what is not seen and can only be heard, and therefore revealing a new space of information. And it has emboldened the author to investigate the geographies of sound by supplying a way to follow associative connections across a range of environments. Originality/value This paper is an original work that is related to the author’s current doctoral research that considers how listening expands visual comprehension.
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Istvanek, Matej, Zdenek Smekal, Lubomir Spurny, and Jiri Mekyska. "Enhancement of Conventional Beat Tracking System Using Teager–Kaiser Energy Operator." Applied Sciences 10, no. 1 (January 4, 2020): 379. http://dx.doi.org/10.3390/app10010379.
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Beat detection systems are widely used in the music information retrieval (MIR) research field for the computation of tempo and beat time positions in audio signals. One of the most important parts of these systems is usually onset detection. There is an understandable tendency to employ the most accurate onset detector. However, there are options to increase the global tempo (GT) accuracy and also the detection accuracy of beat positions at the expense of less accurate onset detection. The aim of this study is to introduce an enhancement of a conventional beat detector. The enhancement is based on the Teager–Kaiser energy operator (TKEO), which pre-processes the input audio signal before the spectral flux calculation. The proposed approach is first evaluated in terms of the ability to estimate the GT and beat positions accuracy of given audio tracks compared to the same conventional system without the proposed enhancement. The accuracy of the GT and average beat differences (ABD) estimation is tested on the manually labelled reference database. Finally, this system is used for analysis of a string quartet music database. Results suggest that the presence of the TKEO lowers onset detection accuracy but also increases the GT and ABD estimation. The average deviation from the reference GT in the reference database is 9.99 BPM (11.28%), which improves the conventional methodology, where the average deviation is 18.19 BPM (17.74%). This study has a pilot character and provides some suggestions for improving the beat tracking system for music analysis.
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Hao, Yiya, Yaobin Chen, Weiwei Zhang, Gong Chen, and Liang Ruan. "A real-time music detection method based on convolutional neural network using Mel-spectrogram and spectral flux." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 263, no. 1 (August 1, 2021): 5910–18. http://dx.doi.org/10.3397/in-2021-11599.
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Audio processing, including speech enhancement system, improves speech intelligibility and quality in real-time communication (RTC) such as online meetings and online education. However, such processing, primarily noise suppression and automatic gain control, is harmful to music quality when the captured signal is music instead of speech. A music detector can solve the issue above by switching off the speech processing when the music is detected. In RTC scenarios, the music detector should be low-complexity and cover various situations, including different types of music, background noises, and other acoustical environments. In this paper, a real-time music detection method with low-computation complexity is proposed, based on a convolutional neural network (CNN) using Mel-spectrogram and spectral flux as input features. The proposed method achieves overall 90.63% accuracy under different music types (classical music, instruments solos, singing-songs, etc.), speech languages (English and Mandarin), and noise types. The proposed method is constructed on a lightweight CNN model with a small feature size, which guarantees real-time processing.
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Luck, Geoff, and Petri Toiviainen. "Ensemble Musicians’ Synchronization With Conductors’ Gestures: An Automated Feature-Extraction Analysis." Music Perception 24, no. 2 (December 1, 2006): 189–200. http://dx.doi.org/10.1525/mp.2006.24.2.189.
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Previous work suggests that the perception of a visual beat in conductors’ gestures is related to certain physical characteristics of the movements they produce, most notably to periods of negative acceleration, and low position in the vertical axis. These findings are based on studies that have presented participants with somewhat simple gestures, and in which participants have been required to simply tap in time with the beat. Thus, it is not clear how generalizable these findings are to real-world conducting situations, in which a conductor uses considerably more complex gestures to direct an ensemble of musicians playing actual instruments. The aims of the present study were to examine the features of conductors’ gestures with which ensemble musicians synchronize their performance in an ecologically valid setting and to develop automatic feature extraction methods for the analysis of audio and movement data. An optical motion capture system was used to record the gestures of an expert conductor directing an ensemble of expert musicians over a 20-minute period. A simultaneous audio recording of the performance of the ensemble was also made and synchronized with the motion capture data. Four short excerpts were selected for analysis, two in which the conductor communicated the beat with high clarity, and two in which the beat was communicated with low clarity. Twelve movement variables were computationally extracted from the movement data and cross-correlated with the pulse of the ensemble’s performance, the latter based on the spectral flux of the audio signal. Results of the analysis indicated that the ensemble’s performance tended to be most highly synchronized with periods of maximal deceleration along the trajectory, followed by periods of high vertical velocity (a higher correlation than deceleration but a longer delay).
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Purnomo, Endra Dwi, Ubaidillah Ubaidillah, Fitrian Imaduddin, Iwan Yahya, and Saiful Amri Mazlan. "Preliminary experimental evaluation of a novel loudspeaker featuring magnetorheological fluid surround absorber." Indonesian Journal of Electrical Engineering and Computer Science 17, no. 2 (February 1, 2020): 922. http://dx.doi.org/10.11591/ijeecs.v17.i2.pp922-928.
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<span>A novel design of magnetorheological fluids (MRF) based surround device in a loudspeaker system was studied in this article. The main objective of this research is to design a new surround device of the loudspeaker that can be easily controlled its damping. Therefore, it was predicted that the audio pressure level on the loudspeaker could be easily manipulated at a different sound source by applying a certain magnetic field. This function could not be reached using one conventional speaker system. Firstly, a set of an electromagnetic device containing MRF was designed to replace the conventional rubber surround. The magnetic circuit was then evaluated using the finite element method magnetics to study the flux distribution in the MRF area. The current was varied from 0.25 to 0.75 A by an interval of 0.25 A. The magnetic flux resulted from the simulation was then logged and used as the based value for predicting the change of shear yield stress. The base properties of the shear yield stress of the MRF against the magnetic flux was obtained from previous experimental result. Therefore, it was hopefully the prediction could be closed to the real system. Based on the simulation result, the shear yield stress varied from 43 to 49 Mpa or about 15 % increment. </span><span lang="IN">A simple experimental work was carried out. By applying particular direct current into the coil, the sound quality generated by the loudspeaker shows different values</span><span>.</span><span lang="IN"> Based on the preliminary experiment, the level of decibel decreased about 3 dB as the application of magnetic fields. The idea has been proven in this preliminary experimental evaluation.</span>
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Mauch, Matthias, Robert M. MacCallum, Mark Levy, and Armand M. Leroi. "The evolution of popular music: USA 1960–2010." Royal Society Open Science 2, no. 5 (May 2015): 150081. http://dx.doi.org/10.1098/rsos.150081.
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In modern societies, cultural change seems ceaseless. The flux of fashion is especially obvious for popular music. While much has been written about the origin and evolution of pop, most claims about its history are anecdotal rather than scientific in nature. To rectify this, we investigate the US Billboard Hot 100 between 1960 and 2010. Using music information retrieval and text-mining tools, we analyse the musical properties of approximately 17 000 recordings that appeared in the charts and demonstrate quantitative trends in their harmonic and timbral properties. We then use these properties to produce an audio-based classification of musical styles and study the evolution of musical diversity and disparity, testing, and rejecting, several classical theories of cultural change. Finally, we investigate whether pop musical evolution has been gradual or punctuated. We show that, although pop music has evolved continuously, it did so with particular rapidity during three stylistic ‘revolutions’ around 1964, 1983 and 1991. We conclude by discussing how our study points the way to a quantitative science of cultural change.
Dissertations / Theses on the topic "Flux audio"
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Nesvadba, Jan. "Segmentation sémantique des contenus audio-visuels." Bordeaux 1, 2007. http://www.theses.fr/2007BOR13456.
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Dans ce travail, nous avons mis au point une méthode de segmentation des contenus audiovisuels applicable aux appareils de stockage domestiques pour cela nous avons expérimenté un système distribué pour l'analyse du contenu composé de modules individuels d'analyse : les service unit. L'un entre eux a été dédié à la caractérisation des éléments hors contenu, i. E. Les publicités, et offre de bonnes perfermances. Parallélement, nous avons testé différents détecteurs de changement de plans afin de retenir le meilleur d'ente eux pour la suite. Puis, nous avons proposé une étude des règles de production des films, i. E. Grammaire de films, qui a permis de définir les séquences de parallel shot. Nous avons, ainsi, testé quatre méthodes de regroupement basées similarité afin de retenir la meilleure d'entre elles pour la suite. Finalement, nous avons recherché différentes méthodes de détection des frontières de scènes et avons obtenu les meilleurs résultats en combinant une méthode basée couleur avec un critère de longueur de plan. Ce dernier offre des performances justifiant son intégration dans les appareils de stockage grand public.
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Ramona, Mathieu. "Classification automatique de flux radiophoniques par Machines à Vecteurs de Support." Phd thesis, Télécom ParisTech, 2010. http://pastel.archives-ouvertes.fr/pastel-00529331.
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Nous présentons ici un système de classification audio parole/musique tirant parti des excellentes propriétés statistiques des Machines à Vecteurs de Support. Ce problème pose les trois questions suivantes : comment exploiter efficacement les SVM, méthode d'essence discriminatoire, sur un problème à plus de deux classes, comment caractériser un signal audio de manière pertinente, et enfin comment traiter l'aspect temporel du problème ? Nous proposons un système hybride de classification multi-classes tirant parti des approches un-contre-un et par dendogramme, et permettant l'estimation de probabilités a posteriori. Ces dernières sont exploitées pour l'application de méthodes de post-traitement prenant en compte les interdépendances entre trames voisines. Nous proposons ainsi une méthode de classification par l'application de Modèles de Markov Cachés (HMM) sur les probabilités a posteriori, ainsi qu'une approche basée sur la détection de rupture entre segments au contenu acoustique "homogène". Par ailleurs, la caractérisation du signal audio étant opérée par une grande collection des descripteurs audio, nous proposons de nouveaux algorithmes de sélection de descripteurs basés sur le récent critère d'Alignement du noyau ; critère que nous avons également exploité pour la sélection de noyau dans le processus de classification. Les algorithmes proposés sont comparés aux méthodes les plus efficaces de l'état de l'art auxquelles elles constituent une alternative pertinente en termes de coût de calcul et de stockage. Le système construit sur ces contributions a fait l'objet d'une participation à la campagne d'évaluation ESTER 2, que nous présentons, accompagnée de nos résultats.
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Soldi, Giovanni. "Diarisation du locuteur en temps réel pour les objets intelligents." Electronic Thesis or Diss., Paris, ENST, 2016. http://www.theses.fr/2016ENST0061.
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La diarisation du locuteur en temps réel vise à détecter "qui parle maintenant" dans un flux audio donné. La majorité des systèmes de diarisation en ligne proposés a mis l'accent sur des domaines moins difficiles, tels que l’émission des nouvelles et discours en plénière, caractérisé par une faible spontanéité. La première contribution de cette thèse est le développement d'un système de diarisation du locuteur complètement un-supervisé et adaptatif en ligne pour les données de réunions qui sont plus difficiles et spontanées. En raison des hauts taux d’erreur de diarisation, une approche semi-supervisé pour la diarisation en ligne, ou les modèles des interlocuteurs sont initialisés avec une quantité modeste de données étiquetées manuellement et adaptées par une incrémentale maximum a-posteriori adaptation (MAP) procédure, est proposée. Les erreurs obtenues peuvent être suffisamment bas pour supporter des applications pratiques. La deuxième partie de la thèse aborde le problème de la normalisation phonétique pendant la modélisation des interlocuteurs avec petites quantités des données. Tout d'abord, Phone Adaptive Training (PAT), une technique récemment proposé, est évalué et optimisé au niveau de la modélisation des interlocuteurs et dans le cadre de la vérification automatique du locuteur (ASV) et est ensuite développée vers un système entièrement un-supervise en utilisant des transcriptions de classe acoustiques générées automatiquement, dont le nombre est contrôlé par analyse de l'arbre de régression. PAT offre des améliorations significatives dans la performance d'un système ASV iVector, même lorsque des transcriptions phonétiques précises ne sont pas disponiblesOn-line speaker diarization aims to detect “who is speaking now" in a given audio stream. The majority of proposed on-line speaker diarization systems has focused on less challenging domains, such as broadcast news and plenary speeches, characterised by long speaker turns and low spontaneity. The first contribution of this thesis is the development of a completely unsupervised adaptive on-line diarization system for challenging and highly spontaneous meeting data. Due to the obtained high diarization error rates, a semi-supervised approach to on-line diarization, whereby speaker models are seeded with a modest amount of manually labelled data and adapted by an efficient incremental maximum a-posteriori adaptation (MAP) procedure, is proposed. Obtained error rates may be low enough to support practical applications. The second part of the thesis addresses instead the problem of phone normalisation when dealing with short-duration speaker modelling. First, Phone Adaptive Training (PAT), a recently proposed technique, is assessed and optimised at the speaker modelling level and in the context of automatic speaker verification (ASV) and then is further developed towards a completely unsupervised system using automatically generated acoustic class transcriptions, whose number is controlled by regression tree analysis. PAT delivers significant improvements in the performance of a state-of-the-art iVector ASV system even when accurate phonetic transcriptions are not available
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Poignant, Johann. "Identification non-supervisée de personnes dans les flux télévisés." Phd thesis, Université de Grenoble, 2013. http://tel.archives-ouvertes.fr/tel-00958774.
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Ce travail de thèse a pour objectif de proposer plusieurs méthodes d'identi- fication non-supervisées des personnes présentes dans les flux télévisés à l'aide des noms écrits à l'écran. Comme l'utilisation de modèles biométriques pour reconnaître les personnes présentes dans de larges collections de vidéos est une solution peu viable sans connaissance a priori des personnes à identifier, plusieurs méthodes de l'état de l'art proposent d'employer d'autres sources d'informations pour obtenir le nom des personnes présentes. Ces méthodes utilisent principalement les noms prononcés comme source de noms. Cependant, on ne peut avoir qu'une faible confiance dans cette source en raison des erreurs de transcription ou de détection des noms et aussi à cause de la difficulté de savoir à qui fait référence un nom prononcé. Les noms écrits à l'écran dans les émissions de télévision ont été peu utilisés en raison de la difficulté à extraire ces noms dans des vidéos de mauvaise qualité. Toutefois, ces dernières années ont vu l'amélioration de la qualité des vidéos et de l'incrustation des textes à l'écran. Nous avons donc ré-évalué, dans cette thèse, l'utilisation de cette source de noms. Nous avons d'abord développé LOOV (pour Lig Overlaid OCR in Vidéo), un outil d'extraction des textes sur-imprimés à l'image dans les vidéos. Nous obtenons avec cet outil un taux d'erreur en caractères très faible. Ce qui nous permet d'avoir une confiance importante dans cette source de noms. Nous avons ensuite comparé les noms écrits et les noms prononcés dans leurs capacités à fournir le nom des personnes présentes dans les émissions de télévisions. Il en est ressorti que deux fois plus de personnes sont nommables par les noms écrits que par les noms prononcés extraits automatiquement. Un autre point important à noter est que l'association entre un nom et une personne est intrinsèquement plus simple pour les noms écrits que pour les noms prononcés. Cette très bonne source de noms nous a donc permis de développer plusieurs méthodes de nommage non-supervisé des personnes présentes dans les émissions de télévision. Nous avons commencé par des méthodes de nommage tardives où les noms sont propagés sur des clusters de locuteurs. Ces méthodes remettent plus ou moins en cause les choix fait lors du processus de regroupement des tours de parole en clusters de locuteurs. Nous avons ensuite proposé deux méthodes (le nommage intégré et le nommage précoce) qui intègrent de plus en plus l'information issue des noms écrits pendant le processus de regroupement. Pour identifier les personnes visibles, nous avons adapté la méthode de nommage précoce pour des clusters de visages. Enfin, nous avons aussi montré que cette méthode fonctionne aussi pour nommer des clusters multi-modaux voix-visage. Avec cette dernière méthode, qui nomme au cours d'un unique processus les tours de paroles et les visages, nous obtenons des résultats comparables aux meilleurs systèmes ayant concouru durant la première campagne d'évaluation REPERE.
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Trad, Abdelbasset. "Déploiement à grande échelle de la voix sur IP dans des environnements hétérogènes." Phd thesis, Nice, 2006. http://tel.archives-ouvertes.fr/tel-00406513.
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Dans cette thèse, nous nous intéressons au déploiement à grande échelle de la Voix sur IP (VoIP) dans des environnements Internet hétérogènes. Après une description des mécanismes de codage et de transmission de la voix sur l'Internet, nous étudions dans une première partie de la thèse, les limites de performance dans le cas d'une transmission d'un grand nombre de flux de voix sur IP entre deux passerelles téléphoniques. Nous discutons le besoin d'utilisation de mécanismes de contrôle de congestion pour le trafic de voix sur IP qui est en croissance continue sur l'Internet. Nous proposons un nouveau schéma de contrôle de congestion de la voix sur IP. Ce schéma combine le multiplexage de flux RTP et le mécanisme de contrôle TCP-amical (TCP-friendly) afin d'améliorer l'efficacité et la performance de la transmission des flux de voix sur IP et de garantir l'équité avec les autres types de trafic coexistant sur l'Internet. La deuxième partie de la thèse est consacrée à l'étude de la transmission de la voix dans des environnements de réseaux locaux sans fil IEEE 802.11e. Nous développons un modèle analytique permettant d'évaluer la capacité d'un réseau 802.11e en nombre de communications de voix sur IP en fonction des paramètres de l'application (codage audio utilisé) ainsi que des paramètres relatifs aux canal de transmission sans fil. Ce modèle peut être utilisé pour ajuster ces paramètres afin d'augmenter la capacité du réseau sans fil tout en considérant les contraintes strictes des communications interactives de la voix sur IP. Dans la dernière partie de la thèse, nous étudions le cas de la transmission de la voix sur IP dans des environnements Internet hétérogènes (filaires/sans fil) constitués en partie par des liens d'accès sans fil. Nous proposons une architecture basée sur une passerelle de voix sur IP placée au bord du réseau sans fil. Cette passerelle est utilisée pour adapter les flux de voix aux caractéristiques du réseau sans fil. Le mécanisme d'adaptation proposé estime dynamiquement l'état de congestion du canal sans fil et permet la différentiation entre les pertes de paquets causées par la congestion et celles dûes aux erreurs de transmission sur le canal sans fil. L'adaptation appropriée est alors appliquée. Le mécanisme d'adaptation proposé, ne nécessite pas de modifications du protocole de contrôle d'accès au canal sans fil (MAC), ce qui facilite son déploiement sur l'infrastructure réseau existante.
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Michaud, Jérôme. "Re-conceptualiser notre expérience de l’environnement audio-visuel qui nous entoure : l’individuation, entre attention et mémoire." Thèse, 2016. http://hdl.handle.net/1866/16151.
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Notre mémoire prend en charge de re-conceptualiser notre nouvel environnement audio-visuel et l’expérience que nous en faisons. À l’ère du numérique et de la dissémination généralisée des images animées, nous circonscrivons une catégorie d’images que nous concevons comme la plus à même d’avoir un impact sur le développement humain. Nous les appelons des images-sons synchrono-photo-temporalisées. Plus spécifiquement, nous cherchons à mettre en lumière leur puissance d’affection et de contrôle en démontrant qu’elles ont une influence certaine sur le processus d’individuation, influence qui est grandement facilitée par l’isotopie structurelle qui existe entre le flux de conscience et leur flux d’écoulement. Par le biais des recherches de Bernard Stiegler, nous remarquons également l’important rôle que jouent l’attention et la mémoire dans le processus d’individuation. L’ensemble de notre réflexion nous fait réaliser à quel point le système d’éducation actuel québécois manque à sa tâche de formation citoyenne en ne dispensant pas un enseignement adéquat des images animées.This thesis re-conceptualizes our new audio-visual environment and analyses the experience we make of it. In the digital age marked by the dissemination of moving images, we circumscribe a category of images which we see as the most likely to have an impact on human development. We call it synchrono-photo-temporalized images-sounds. Specifically, we seek to highlight their power of affection and control by showing that they have some influence on the process of individuation, an influence which is greatly facilitated by the structural isotopy between the stream of consciousness and the flow of motion images. By examining the research of Bernard Stiegler, we also note the important roles attention and memory play in the process of individuation. This thinking makes us realize how the current education system in Quebec fails in its mission to give a good civic education by not providing an adequate teaching of moving images.
Books on the topic "Flux audio"
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Colbert, Don. Bible Cure for Colds, Flu & Sinus Infections (Bible Cure (Oasis Audio)). Oasis Audio, 2004.
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(Narrator), Greg Wheatley, ed. The Bible Cure for Colds, Flu and Sinus Infections (Bible Cure (Oasis Audio)). Oasis Audio, 2004.
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Shatzkin, Mike, and Robert Paris Riger. The Book Business. Oxford University Press, 2019. http://dx.doi.org/10.1093/wentk/9780190628031.001.0001.
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Many of us read books every day, either electronically or in print. We remember the books that shaped our ideas about the world as children, go back to favorite books year after year, give or lend books to loved ones and friends to share the stories we've loved especially, and discuss important books with fellow readers in book clubs and online communities. But for all the ways books influence us, teach us, challenge us, and connect us, many of us remain in the dark as to where they come from and how the mysterious world of publishing truly works. How are books created and how do they get to readers? The Book Business: What Everyone Needs to Know® introduces those outside the industry to the world of book publishing. Covering everything from the beginnings of modern book publishing early in the 20th century to the current concerns over the alleged death of print, digital reading, and the rise of Amazon, Mike Shatzkin and Robert Paris Riger provide a succinct and insightful survey of the industry in an easy-to-read question-and-answer format. The authors, veterans of "trade publishing," or the branch of the business that puts books in our hands through libraries or bookstores, answer questions from the basic to the cutting-edge, providing a guide for curious beginners and outsiders. How does book publishing actually work? What challenges is it facing today? How have social media changed the game of book marketing? What does the life cycle of a book look like in 2019? They focus on how practices are changing at a time of great flux in the industry, as digital creation and delivery are altering the commercial realities of the book business. This book will interest not only those with no experience in publishing looking to gain a foothold on the business, but also those working on the inside who crave a bird's eye view of publishing's evolving landscape. This is a moment of dizzyingly rapid change wrought by the emergence of digital publishing, data collection, e-books, audio books, and the rise of self-publishing; these forces make the inherently interesting business of publishing books all the more fascinating.
Book chapters on the topic "Flux audio"
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Elrom, Elad. "Facilitating Audio and Video." In AdvancED Flex 4, 461–503. Berkeley, CA: Apress, 2010. http://dx.doi.org/10.1007/978-1-4302-2484-6_14.
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Richardson, Darren, Paul Milbourne, Steve Webster, Todd Yard, and Sean McSharry. "Using Audio." In Foundation ActionScript 3.0 for Flash and Flex, 301–53. Berkeley, CA: Apress, 2009. http://dx.doi.org/10.1007/978-1-4302-1919-4_8.
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McSharry, Sean. "Using Audio." In Foundation ActionScript 3.0 with Flash CS3 and Flex, 293–343. Berkeley, CA: Apress, 2008. http://dx.doi.org/10.1007/978-1-4302-0196-0_8.
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I. Al-Shoshan, Abdullah. "Classification and Separation of Audio and Music Signals." In Multimedia Information Retrieval [Working Title]. IntechOpen, 2020. http://dx.doi.org/10.5772/intechopen.94940.
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This chapter addresses the topic of classification and separation of audio and music signals. It is a very important and a challenging research area. The importance of classification process of a stream of sounds come up for the sake of building two different libraries: speech library and music library. However, the separation process is needed sometimes in a cocktail-party problem to separate speech from music and remove the undesired one. In this chapter, some existed algorithms for the classification process and the separation process are presented and discussed thoroughly. The classification algorithms will be divided into three categories. The first category includes most of the real time approaches. The second category includes most of the frequency domain approaches. However, the third category introduces some of the approaches in the time-frequency distribution. The approaches of time domain discussed in this chapter are the short-time energy (STE), the zero-crossing rate (ZCR), modified version of the ZCR and the STE with positive derivative, the neural networks, and the roll-off variance. The approaches of the frequency spectrum are specifically the roll-off of the spectrum, the spectral centroid and the variance of the spectral centroid, the spectral flux and the variance of the spectral flux, the cepstral residual, and the delta pitch. The time-frequency domain approaches have not been yet tested thoroughly in the process of classification and separation of audio and music signals. Therefore, the spectrogram and the evolutionary spectrum will be introduced and discussed. In addition, some algorithms for separation and segregation of music and audio signals, like the independent Component Analysis, the pitch cancelation and the artificial neural networks will be introduced.
Conference papers on the topic "Flux audio"
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Wang, Wengen, Xiaoqing Yu, Yun Hui Wang, and Ram Swaminathan. "Audio fingerprint based on Spectral Flux for audio retrieval." In 2012 International Conference on Audio, Language and Image Processing (ICALIP). IEEE, 2012. http://dx.doi.org/10.1109/icalip.2012.6376781.
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Lee, Sangkil, Jieun Kim, and Insung Lee. "Speech/Audio Signal Classification Using Spectral Flux Pattern Recognition." In 2012 IEEE Workshop on Signal Processing Systems (SiPS). IEEE, 2012. http://dx.doi.org/10.1109/sips.2012.36.
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Xu, Y., P. Smagacz, J. Lapinskas, J. Webster, P. Shaw, and R. P. Taleyarkhan. "Neutron Detection with Centrifugally-Tensioned Metastable Fluid Detectors (CTMFD)." In 14th International Conference on Nuclear Engineering. ASMEDC, 2006. http://dx.doi.org/10.1115/icone14-89199.
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Tensioned metastable liquid states at room temperature were utilized to display sensitivity to impinging nuclear radiation, that manifests itself via audio-visual signals that one can see and hear. A centrifugally-tensioned metastable fluid detector (CTMFD), a diamond shaped spinning device rotating about its axis, was used to induce tension states, i.e. negative (sub-vacuum) pressures in liquids. In this device, radiation induced cavitation is audible due to liquid fracture and is visible from formed bubbles, so called hearing and seeing radiation. This type of detectors is selectively insensitive to Gamma rays and associated indication devices could be extremely simple, reliable and inexpensive. Furthermore, any liquids with large neutron interaction cross sections could be good candidates. Two liquids, isopentane and methanol, were tested with three neutron sources of Cf-252, PuBe and Pulsed Neutron Generator (PNG) under various configurations of neutron spectra and fluxes. The neutron count rates were measured using a liquid scintillation detector. The CTMFD was operated at preset values of rotating frequency and a response time was recorded when a cavitation occurred. Other parameters, including ambient temperature, ramp rate, delay time between two consecutive cavitations, were kept constant. The distance between the menisci of the liquid in the CTMFD was measured before and after each experiment. In general, the response of liquid molecules in a CTMFD varies with the neutron spectrum and flux. The response time follows an exponential trend with negative pressures for a given neutron count rate and spectra conditions. Isopentane was found to exhibit lower tension thresholds than methanol. On the other hand, methanol offered a larger tension metastability state variation for the various types of neutron sources, indicating the potential for offering significantly better energy resolution abilities for spectroscopic applications.
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Zheng, Haiming, and Tieqiao Guo. "Relative Accuracy Test Audit Evaluation for Flue Gas Continuous Emission Monitoring Systems in Power Plant." In 2008 Pacific-Asia Workshop on Computational Intelligence and Industrial Application (PACIIA). IEEE, 2008. http://dx.doi.org/10.1109/paciia.2008.123.
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Meng, Liu, Chen Yang, Zhong Zhuhai, Zhang Xiaodan, Deng Guoliang, Mingyan Yin, Jun Li, and Qi Sun. "Numerical Tests on the Effect Factors of the Last Stage Blade for Low Pressure Exhaust Hood Simulation." In ASME Turbo Expo 2017: Turbomachinery Technical Conference and Exposition. American Society of Mechanical Engineers, 2017. http://dx.doi.org/10.1115/gt2017-63964.
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Kinetic energy recovery is a key objective for low pressure exhaust hood design and optimization. Numerical simulation of the exhaust hood helps the engineers to explore and confirm the causes of the loss in the hood. Many studies have suggested that it is necessary for the simulation to include the last stage blade to get a realistic assessment. For the sole exhaust hood study, the inlet boundary condition is hard to set precisely like the downstream flow of the last stage blade. And the studies have also shown that the performances generated from the simulations may vary evidently between the sole exhaust hood and exhaust hood with last stage blade. It is obvious that the blade influences the exhaust hood, but the exact effect factors of the blade and the way they work are not thoroughly discussed. This paper has conducted many numerical tests to audit the influence of the common effect factors of the last stage blade. The internal flow field of the exhaust hood was numerically investigated using three-dimensional Reynolds-Averaged Navier-Stokes (RANS) solutions based on the ANSYS-CFX. In the first part of the paper, the tests are conducted by changing each effect factor of the inlet boundary condition for sole exhaust hood studies. These factors include the mass flow flux, the angle of the exit flow of the last stage, both the circumferential and the radial ones, and the speed and position of the jet-flow downstream of the seal over the shroud of the bucket. The tests show that each factor has its own distinctive style and extent for influence. Some of them may maximize the performance at some certain point, and some may deteriorate the performance rapidly beyond a threshold. And some factors may change the performance insignificantly within a wide range. However, these influences are not good enough to be consistent with the difference between the sole exhaust hood and the hood with blade simulations. In the second part of this paper, the focus locates on the direction of the jet-flow of the bucket seal. The tests prove that this direction is the prominent factor to influence the exhaust hood performance. Some extra tests for the seal have also been conducted to analyze this factor. The static pressure recovery for the simulation with labyrinth seal is about only half of the sole exhaust hood simulation. The discussion of these tests show that the seal jet is the main cause for this performance dive, and explain how the seal jet direction changes the flow field of the exhaust hood. It also suggests that the procedure to optimize the seal design is not mature yet, for some nature of the jet-flow remains unclear. It may need more detailed study in the future.