Academic literature on the topic 'Audio analysi'

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Journal articles on the topic "Audio analysi"

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Pečiulytė, Indrė, and Juozas Ruževičius. "Kokybės auditas: koncepcija ir metodologijos tobulinimas." Informacijos mokslai 68 (January 1, 2014): 23–43. http://dx.doi.org/10.15388/im.2014..3922.

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Audito metodologija plačiai nagrinėjama tiek užsienio, tiek Lietuvos autorių mokslinėje literatūroje, tačiau išsamių studijų, kuriose būtų sistemingai analizuojamos visos šiuo metu naudojamos audito rūšys, nėra daug. Kokybės audito metodologija taip pat nėra iki galo išnagrinėta ir aiškiai susisteminta. Nagrinėjant audito koncepcijas sunku vienareikšmiškai nustatyti, kuriam audito tipui – veiklos ar valdymo – yra priskiriamas kokybės auditas. Atlikta mokslinės literatūros analizė atskleidė, kad visoje audito sistemoje kokybės auditą galima identifikuoti kaip veiklos audito sudedamąją dalį. Kokybės auditas yra svarbus visoje audito sistemoje, nes jis yra dalis vertinimo proceso, kuriuo auditoriai atskleidžia tobulintinas organizacijos veiklos kokybės sritis. Straipsnyje, remiantis teorinės literatūros sistemine analize ir kokybės auditorių bei ekspertų anketinės apklausos rezultatais, suformuotas audito sistemos pagal funkcinį požymį modelis, identifikuoti kokybės audito atlikimo būdų pranašumai ir trūkumai, pateiktos kokybės audito metodologijos tobulinimo rekomendacijos.Pagrindiniai žodžiai: kokybės auditas, audito procesas, audito sistema, audito metodologija, tobulinimas, modelis, apklausa. Quality audit: concept and methodology improvementIndrė Pečiulytė, Juozas Ruževičius SummaryThe methodology of audit is extensively researched by both foreign and Lithuanian authors, but there isn’t a complete theory of audit, in which all the currently existing types of audit are mentioned. This is due to the fact that the quality audit methodology is not fully explored and clearly systematized. When analyzing audit concepts, it is a challenge to determine which type of audit – performance or management – is attributable to the quality audit. The theoretical analysis has shown that in the audit system the quality audit can be assigned as part of the performance audit. Quality audit is important throughout the audit system as it is part of the evaluation process, which helps auditors to reveal areas for improvement within their organizations. According to the theoretical literature systematic analysis and the quality auditors’ questionnaire survey results, in this study the audit system model by the functional character has been formed, the advantages and disadvantages of quality audit techniques have been identified, and recommendations for the quality audit methodology improvement are presented.Keywords: quality audit, auditing process, audit system, audit methodology, improvement, survey, model.
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Braga, Marcelo, and Otavio Lube dos Santos. "MÉTODODE ANÁLISE DE ÁUDIO PARA DETECÇÃO DE FALHAS DE COMUNICAÇÃO." Revista Científica Faesa 15, no. 2 (July 22, 2019): 81–98. http://dx.doi.org/10.5008/1809.7367.159.

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Aleliūnas, Irmantas, and Zenona Atkočiūnienė. "Informacijos auditas kitų audito rūšių kontekste." Informacijos mokslai 54 (January 1, 2010): 7–16. http://dx.doi.org/10.15388/im.2010.0.3178.

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Mokslininkai, analizuodami informacijos auditą, pažymi šio audito svarbą ir įvairius jo aspektus, tačiau praktinėje veikloje informacijos auditas neįsitvirtinęs. Informacijos auditas dar neturi parengtų standartų ir metodikų bei labiau panašus į mokslinę veiklą. Siekiant sukurti pagrindą legitimiai bazei, straipsnyje yra atliekamas informacijos audito pozicionavimas bendroje audito aplinkoje. Tyrimo metu įvertinamos audito rūšys, jos struktūrizuojamos ir atliekama informacijos audito bei kitų audito rūšių lyginamoji analizė pagal funkcinį ir kitus požymius.Pagrindiniai žodžiai: informacijos auditas, duomenų (informacinių sistemų) auditas, veiklos auditas, atitikimo (teisėtumo) auditas, informacijos valdymas, auditorių nepriklausomumasInformation Audit in the Context of Other Audit TypesIrmantas Aleliūnas, Zenona Atkočiūnienė Summary Information auditing requires globally applicable standards that pertain specifically to information auditing. Globally applicable standards are known in financial, performance, information systems (electronic data processing) and other types of audits. However, information audit is rather scientific than practical activity – information audit standards are not agreed yet, therefore:• minimum level of acceptable performance required to meet the professional responsibilities is not yet set out • Stakeholders and shareholders have no information concerning the work of information auditing practitionersThese problems incited the purpose of the research – to analyse relationships between information audit and other audit types and to find a basis for the concepts and standards of information audit.;
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Jolly, Jasmine, and Mehbooba P. Shareef. "Audio Watermarking Schemes: A Comparative Analysis." International Journal of Engineering Technology and Management Sciences 4, no. 4 (July 28, 2020): 57–61. http://dx.doi.org/10.46647/ijetms.2020.v04i04.009.

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The growth of internet and its reachability to all sectors of people have never been greater. Internet has become the best marketplace, the best library and may be the best guide for everything. But this revolution comes with some bigger problems. One of the most challenging problems among them would be copyright protection of digital data being transferred over internet. Digital images, videos and audios undergo illegal re-production and re-distributions, tampering and other acts of copyright violation. This is proved to have leading the film and other prominent industries to loss of millions of dollars per year. Encrypting the data provides security to it. In this case only people who pay to buy the secret key that should be used for decryption can use the data. But the problem is that once decrypted, the data can be re-produced into any number of copies and can be re-distributed without any permission from the author. Watermarking is an intelligent solution for this problem where the presence of watermark can be checked to distinguish pirated copies from the actual ones. A lot of methods have been developed for image and video watermarking, but the research on audio watermarking started a little bit later. The reason might be the fact that audio watermarking is tedious compared to image and video as Human Auditory System (HAS) is more sensitive compared to Human Visionary System. So ensuring the imperceptibility of audio watermarks is a tougher task. In this thesis various audio watermarking schemes introduced so far in the literature and their merits and demerits are studied.
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Barrio Fraile, Estrella, Ana María Enrique Jiménez, María Luz Barbeito Veloso, and Anna Fajula Payet. "Sonic identity and audio branding elements in Spanish radio advertising." Anàlisi 65 (December 22, 2021): 103–19. http://dx.doi.org/10.5565/rev/analisi.3330.

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The emotional potential of sound is an excellent resource for companies and institutions seeking to test new ways of communicating with their stakeholders through the senses. However, there are still few organizations that include sound as a conveyer of their corporate identity. Audio branding aims to expand the contact points with audiences by incorporating sound elements that facilitate the recognition of the brand's values. This research attempts to understand the use that corporations make of sound as a communication tool and to evaluate its presence, but above all it wants to find out if this use is due to a strategic approach or if it is a specific decision marked by the temporality of advertising campaigns. For this reason, in this paper we have taken the radio as an advertising media. On the one hand, because it is the sound media par excellence. On the other hand, because it is where we can find the most well-known modes of audio branding such as the brand song, the jingle or the sonotype. The sample, composed of 239 inserts from the 3 generalist radio channels with the highest audiences in Spain, reveals that only 21% of items contain an element of audio branding, indicating that this is a field yet to be explored by organizations.
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Greenspun, Philip, and Leigh Klotz. "Audio Analysis VI: Testing Audio Cables." Computer Music Journal 12, no. 1 (1988): 58. http://dx.doi.org/10.2307/3679837.

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Kischinhevsky, Marcelo, Itala Maduell Vieira, João Guilherme Bastos dos Santos, Viktor Chagas, Miguel de Andrade Freitas, and Alessandra Aldé. "WhatsApp audios and the remediation of radio: Disinformation in Brazilian 2018 presidential election." Radio Journal: International Studies in Broadcast & Audio Media 18, no. 2 (October 1, 2020): 139–58. http://dx.doi.org/10.1386/rjao_00021_1.

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This article brings the results of an investigation into the role of WhatsApp audio messages in the 2018 Brazilian presidential elections, proposing that instant voice messaging borrows elements from radio language. We started from a broader research, conducted by the Brazilian National Institute of Science and Technology in Digital Democracy (INCT.DD, in its Portuguese acronym), which identified a network composed of 220 WhatsApp groups – all of them with open-entry links – supporting six different candidates. Those groups put together thousands of anonymized profiles linked through connections to similar groups, configuring an extensive network. More than 1 million messages, including 98,000 audios, were gathered and downloaded during 2018 Brazilian electoral period (from June to October). We focused on eighteen audios with major circulation (totalling 3622 appearances) among the ones shared at least 100 times, which were extracted and analysed. The use of radio content analysis techniques pointed out strong evidence that audio messaging remediate radiophonic elements such as intimacy and colloquial language to accelerate disinformation campaigns.
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Sierra García, Laura, Emiliano Ruiz Barbadillo, and Manuel Orta Pérez. "Análisis de la Influencia de la Función de Auditoría Interna sobre las Cuotas de Auditoría." Revista de Contabilidad 22, no. 1 (January 1, 2019): 100–111. http://dx.doi.org/10.6018/rc-sar.22.1.354351.

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The aim of this paper is to analyse the relationship between the internal audit function and auditor's fees using a sample of Spanish listed companies between 2003 and 2011. We use the audit fees model that was proposed by Simunic (1980). Our results indicate that fees are greater in those companies that have an internal audit function. In addition, if there are meetings between the audit committee and the internal audit function, this is statistically significant with audit fees. This paper contributes to literature in several ways. Firstly, there are no previous papers in Spain, unlike other contexts, on the relationship between internal audit and auditor fees. Secondly, the empirical results detected in previous studies are conflicting; therefore, there is a need to complement them. Finally, the findings have practical implications for companies, external auditors and regulatory agencies themselves. El objetivo de este trabajo es analizar la relación existente entre la función de auditoría interna y los honorarios del auditor sobre una muestra de empresas que cotizan en el mercado continuo español entre 2003 y 2011. Utilizando el modelo de honorarios de auditoría propuesto por Simunic (1980), los resultados indican que los honorarios de auditoría son mayores en aquellas compañías que cuentan con la existencia de un departamento de auditoría interna. Asimismo, la existencia de reuniones entre el comité de auditoría y la función de auditoría interna es estadísticamente significativa con los honorarios. Este estudio contribuye a la literatura de diversas formas. En primer lugar, no existen trabajos anteriores en España, a diferencia de otros contextos, sobre la relación entre auditoría interna y honorarios. En segundo lugar, los resultados empíricos detectados en estudios previos resultan contradictorios por lo tanto lleva a la necesidad de complementarlos. Finalmente, los resultados tiene implicaciones prácticas para la empresa, los auditores externos y los propios organismos reguladores.
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Suhandinata, Sebastian, Reyhan Achmad Rizal, Dedy Ongky Wijaya, Prabhu Warren, and Srinjiwi Srinjiwi. "ANALISIS PERFORMA KRIPTOGRAFI HYBRID ALGORITMA BLOWFISH DAN ALGORITMA RSA." JURTEKSI (Jurnal Teknologi dan Sistem Informasi) 6, no. 1 (December 10, 2019): 1–10. http://dx.doi.org/10.33330/jurteksi.v6i1.395.

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Abstract:Computer data security relies on preventing data theft from irresponsible parties by using cryptography method. Some cryptography algorithms have good and poor performance in encrypting and decrypting data depending on the key types. Therefore the purpose of this research is to measure the performance of the hybrid algorithm, consisting a symmetric keyBlowfish algorithm and an asymmetric key RSA algorithm, in encrypting and decrypting multiple types of data such as documents, photos, audios, and videos. The result is the performance of the hybrid algorithm is almost on par with Blowfish and provides a more secure data encryption and decryption by taking advantage of RSA algorithm. The average encryption performance of hybrid algorithm is 0.85s on document, 1.06s on photo, 3.38s on audio, and 15.56s on video. While the average decryption performance of hybrid algorithm is 1.01s on document, 1.38s on photo, 4.3s on audio, and 27.56s on video. Keywords:Hybrid cryptography, Data security, Performance, Blowfish, RSA Abstrak:Keamanan data komputer berhubungan dengan pencegahan dari pencurian data oleh pihak yang tidak bertanggung jawab, salah satu cara pengamanan data komputer yaitu dengan teknik kriptografi. Beberapa metode kriptografi memiliki performa yang baik dan buruk tergantung dengan tipe kuncinya. Maka dari itu, tujuan dari penelitian ini adalah mengukur tingkat kecepatan kriptografi hybrid, terdiri dari algoritma simetris Blowfish dan algoritma asimetris RSA, dengan beberapa tipe data seperti dokumen, foto, audio dan video. Hasil dari penelitian ini adalah algoritma hybrid memiliki performa yang tidak jauh berbeda dari algoritma Blowfish dan membuat proses enkripsi dan dekripsi data lebih aman dengan keunggulan dari algoritma RSA. Rata-rata kecepatan enkripsi algoritma hybrid untuk dokumen 0,85 detik, gambar 1,06 detik, audio 3,38 detik, dan video 15,56 detik. Sedangkan rata-rata kecepatan dekripsi algoritma hybrid untuk dokumen 1,01 detik, gambar 1,38 detik, audio 4,3 detik, dan video 27,56 detik. Kata kunci:Kriptografihybrid, Keamanan data, Performa, Blowfish, RSA
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Auken, V. M. "Анализ взаимодействия государственных доходов и аудита." INTERNATIONAL JOURNAL OF INFORMATION AND COMMUNICATION TECHNOLOGIES, no. 8(8) (March 4, 2022): 43–46. http://dx.doi.org/10.54309/ijict.2021.8.8.008.

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The article examines the relationship between unforeseen government revenue and the actual audit conduct. It has been found that overprofits can worsen the public organizations’ work, while government audits increase their effectiveness. The findings were obtained through macroeconomic management modeling that was structurally evaluated using econometric formulas and quasi government data. Based on the estimated model, the correlation between over income and the number of audits on the public administration’s effectiveness has been determined. В статье исследуется взаимосвязь непредвиденных государственных доходов и фактического проведения аудита. Выяснилось, что сверхприбыль может ухудшить работу общественных организаций, а государственный аудит повысит их эффективность. Результаты были получены с помощью моделирования макроэкономического управления, которое было структурно оценено с использованием эконометрических формул с использованием квазигосударственных данных. На основе оценочной модели была определена корреляция между избыточным доходом и количеством проверок эффективности государственного управления.
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Dissertations / Theses on the topic "Audio analysi"

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CHEMLA, ROMEU SANTOS AXEL CLAUDE ANDRE'. "MANIFOLD REPRESENTATIONS OF MUSICAL SIGNALS AND GENERATIVE SPACES." Doctoral thesis, Università degli Studi di Milano, 2020. http://hdl.handle.net/2434/700444.

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Tra i diversi campi di ricerca nell’ambito dell’informatica musicale, la sintesi e la generazione di segnali audio incarna la pluridisciplinalità di questo settore, nutrendo insieme le pratiche scientifiche e musicale dalla sua creazione. Inerente all’informatica dalla sua creazione, la generazione audio ha ispirato numerosi approcci, evolvendo colle pratiche musicale e gli progressi tecnologici e scientifici. Inoltre, alcuni processi di sintesi permettono anche il processo inverso, denominato analisi, in modo che i parametri di sintesi possono anche essere parzialmente o totalmente estratti dai suoni, dando una rappresentazione alternativa ai segnali analizzati. Per di più, la recente ascesa dei algoritmi di l’apprendimento automatico ha vivamente interrogato il settore della ricerca scientifica, fornendo potenti data-centered metodi che sollevavano diversi epistemologici interrogativi, nonostante i sui efficacia. Particolarmente, un tipo di metodi di apprendimento automatico, denominati modelli generativi, si concentrano sulla generazione di contenuto originale usando le caratteristiche che hanno estratti dei dati analizzati. In tal caso, questi modelli non hanno soltanto interrogato i precedenti metodi di generazione, ma anche sul modo di integrare questi algoritmi nelle pratiche artistiche. Mentre questi metodi sono progressivamente introdotti nel settore del trattamento delle immagini, la loro applicazione per la sintesi di segnali audio e ancora molto marginale. In questo lavoro, il nostro obiettivo e di proporre un nuovo metodo di audio sintesi basato su questi nuovi tipi di generativi modelli, rafforazti dalle nuove avanzati dell’apprendimento automatico. Al primo posto, facciamo una revisione dei approcci esistenti nei settori dei sistemi generativi e di sintesi sonore, focalizzando sul posto di nostro lavoro rispetto a questi disciplini e che cosa possiamo aspettare di questa collazione. In seguito, studiamo in maniera più precisa i modelli generativi, e come possiamo utilizzare questi recenti avanzati per l’apprendimento di complesse distribuzione di suoni, in un modo che sia flessibile e nel flusso creativo del utente. Quindi proponiamo un processo di inferenza / generazione, il quale rifletta i processi di analisi/sintesi che sono molto usati nel settore del trattamento del segnale audio, usando modelli latenti, che sono basati sull’utilizzazione di un spazio continuato di alto livello, che usiamo per controllare la generazione. Studiamo dapprima i risultati preliminari ottenuti con informazione spettrale estratte da diversi tipi di dati, che valutiamo qualitativamente e quantitativamente. Successiva- mente, studiamo come fare per rendere questi metodi più adattati ai segnali audio, fronteggiando tre diversi aspetti. Primo, proponiamo due diversi metodi di regolarizzazione di questo generativo spazio che sono specificamente sviluppati per l’audio : una strategia basata sulla traduzione segnali / simboli, e una basata su vincoli percettivi. Poi, proponiamo diversi metodi per fronteggiare il aspetto temporale dei segnali audio, basati sull’estrazione di rappresentazioni multiscala e sulla predizione, che permettono ai generativi spazi ottenuti di anche modellare l’aspetto dinamico di questi segnali. Per finire, cambiamo il nostro approccio scientifico per un punto di visto piú ispirato dall’idea di ricerca e creazione. Primo, descriviamo l’architettura e il design della nostra libreria open-source, vsacids, sviluppata per permettere a esperti o non-esperti musicisti di provare questi nuovi metodi di sintesi. Poi, proponiamo una prima utilizzazione del nostro modello con la creazione di una performance in real- time, chiamata ægo, basata insieme sulla nostra libreria vsacids e sull’uso di une agente di esplorazione, imparando con rinforzo nel corso della composizione. Finalmente, tramo dal lavoro presentato alcuni conclusioni sui diversi modi di migliorare e rinforzare il metodo di sintesi proposto, nonché eventuale applicazione artistiche.
Among the diverse research fields within computer music, synthesis and generation of audio signals epitomize the cross-disciplinarity of this domain, jointly nourishing both scientific and artistic practices since its creation. Inherent in computer music since its genesis, audio generation has inspired numerous approaches, evolving both with musical practices and scientific/technical advances. Moreover, some syn- thesis processes also naturally handle the reverse process, named analysis, such that synthesis parameters can also be partially or totally extracted from actual sounds, and providing an alternative representation of the analyzed audio signals. On top of that, the recent rise of machine learning algorithms earnestly questioned the field of scientific research, bringing powerful data-centred methods that raised several epistemological questions amongst researchers, in spite of their efficiency. Especially, a family of machine learning methods, called generative models, are focused on the generation of original content using features extracted from an existing dataset. In that case, such methods not only questioned previous approaches in generation, but also the way of integrating this methods into existing creative processes. While these new generative frameworks are progressively introduced in the domain of image generation, the application of such generative techniques in audio synthesis is still marginal. In this work, we aim to propose a new audio analysis-synthesis framework based on these modern generative models, enhanced by recent advances in machine learning. We first review existing approaches, both in sound synthesis and in generative machine learning, and focus on how our work inserts itself in both practices and what can be expected from their collation. Subsequently, we focus a little more on generative models, and how modern advances in the domain can be exploited to allow us learning complex sound distributions, while being sufficiently flexible to be integrated in the creative flow of the user. We then propose an inference / generation process, mirroring analysis/synthesis paradigms that are natural in the audio processing domain, using latent models that are based on a continuous higher-level space, that we use to control the generation. We first provide preliminary results of our method applied on spectral information, extracted from several datasets, and evaluate both qualitatively and quantitatively the obtained results. Subsequently, we study how to make these methods more suitable for learning audio data, tackling successively three different aspects. First, we propose two different latent regularization strategies specifically designed for audio, based on and signal / symbol translation and perceptual constraints. Then, we propose different methods to address the inner temporality of musical signals, based on the extraction of multi-scale representations and on prediction, that allow the obtained generative spaces that also model the dynamics of the signal. As a last chapter, we swap our scientific approach to a more research & creation-oriented point of view: first, we describe the architecture and the design of our open-source library, vsacids, aiming to be used by expert and non-expert music makers as an integrated creation tool. Then, we propose an first musical use of our system by the creation of a real-time performance, called aego, based jointly on our framework vsacids and an explorative agent using reinforcement learning to be trained during the performance. Finally, we draw some conclusions on the different manners to improve and reinforce the proposed generation method, as well as possible further creative applications.
À travers les différents domaines de recherche de la musique computationnelle, l’analysie et la génération de signaux audio sont l’exemple parfait de la trans-disciplinarité de ce domaine, nourrissant simultanément les pratiques scientifiques et artistiques depuis leur création. Intégrée à la musique computationnelle depuis sa création, la synthèse sonore a inspiré de nombreuses approches musicales et scientifiques, évoluant de pair avec les pratiques musicales et les avancées technologiques et scientifiques de son temps. De plus, certaines méthodes de synthèse sonore permettent aussi le processus inverse, appelé analyse, de sorte que les paramètres de synthèse d’un certain générateur peuvent être en partie ou entièrement obtenus à partir de sons donnés, pouvant ainsi être considérés comme une représentation alternative des signaux analysés. Parallèlement, l’intérêt croissant soulevé par les algorithmes d’apprentissage automatique a vivement questionné le monde scientifique, apportant de puissantes méthodes d’analyse de données suscitant de nombreux questionnements épistémologiques chez les chercheurs, en dépit de leur effectivité pratique. En particulier, une famille de méthodes d’apprentissage automatique, nommée modèles génératifs, s’intéressent à la génération de contenus originaux à partir de caractéristiques extraites directement des données analysées. Ces méthodes n’interrogent pas seulement les approches précédentes, mais aussi sur l’intégration de ces nouvelles méthodes dans les processus créatifs existants. Pourtant, alors que ces nouveaux processus génératifs sont progressivement intégrés dans le domaine la génération d’image, l’application de ces techniques en synthèse audio reste marginale. Dans cette thèse, nous proposons une nouvelle méthode d’analyse-synthèse basés sur ces derniers modèles génératifs, depuis renforcés par les avancées modernes dans le domaine de l’apprentissage automatique. Dans un premier temps, nous examinerons les approches existantes dans le domaine des systèmes génératifs, sur comment notre travail peut s’insérer dans les pratiques de synthèse sonore existantes, et que peut-on espérer de l’hybridation de ces deux approches. Ensuite, nous nous focaliserons plus précisément sur comment les récentes avancées accomplies dans ce domaine dans ce domaine peuvent être exploitées pour l’apprentissage de distributions sonores complexes, tout en étant suffisamment flexibles pour être intégrées dans le processus créatif de l’utilisateur. Nous proposons donc un processus d’inférence / génération, reflétant les paradigmes d’analyse-synthèse existant dans le domaine de génération audio, basé sur l’usage de modèles latents continus que l’on peut utiliser pour contrôler la génération. Pour ce faire, nous étudierons déjà les résultats préliminaires obtenus par cette méthode sur l’apprentissage de distributions spectrales, prises d’ensembles de données diversifiés, en adoptant une approche à la fois quantitative et qualitative. Ensuite, nous proposerons d’améliorer ces méthodes de manière spécifique à l’audio sur trois aspects distincts. D’abord, nous proposons deux stratégies de régularisation différentes pour l’analyse de signaux audio : une basée sur la traduction signal/ symbole, ainsi qu’une autre basée sur des contraintes perceptives. Nous passerons par la suite à la dimension temporelle de ces signaux audio, proposant de nouvelles méthodes basées sur l’extraction de représentations temporelles multi-échelle et sur une tâche supplémentaire de prédiction, permettant la modélisation de caractéristiques dynamiques par les espaces génératifs obtenus. En dernier lieu, nous passerons d’une approche scientifique à une approche plus orientée vers un point de vue recherche & création. Premièrement, nous présenterons notre librairie open-source, vsacids, visant à être employée par des créateurs experts et non-experts comme un outil intégré. Ensuite, nous proposons une première utilisation musicale de notre système par la création d’une performance temps réel, nommée ægo, basée à la fois sur notre librarie et sur un agent d’exploration appris dynamiquement par renforcement au cours de la performance. Enfin, nous tirons les conclusions du travail accompli jusqu’à maintenant, concernant les possibles améliorations et développements de la méthode de synthèse proposée, ainsi que sur de possibles applications créatives.
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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.

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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.
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3

Djebbar, Fatiha. "Contributions to Audio Steganography : Algorithms and Robustness Analysis." Thesis, Brest, 2012. http://www.theses.fr/2012BRES0005.

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La stéganographie numérique est une technique récente qui a émergé comme une source importante pour la sécurité des données. Elle consiste à envoyer secrètement et de manière fiable des informations dissimulées et non pas seulement à masquer leur présence. Elle exploite les caractéristiques des fichiers médias numériques anodins, tels que l’image, le son et la vidéo,et les utilise comme support pour véhiculer des informations secrète d’une façon inaperçue. Les techniques de cryptage et de tatouage sont déjà utilisées pour résoudre les problèmes liés à la sécurité des données. Toutefois,l’évolution des tentatives d’interception et de déchiffrement des données numériques nécessitent de nouvelles techniques pour enrayer les tentatives malveillantes et d’élargir le champ des applications y associées. L’objectif principal des systèmes stéganographiques consiste à fournir de nouveaux moyens sécurisés, indétectables et imperceptibles pour dissimuler des données.La stéganographie est utilisée sous l’hypothèse qu’elle ne sera pas détectée si personne n’essaye de la découvrir. Les techniques récentes destéganographie ont déjà été employées dans diverses applications. La majorité de ces applications ont pour objectif d’assurer la confidentialité des données.D’autres par contre sont utilisées malicieusement. L’utilisation de la stéganographie à des fins criminelles, de terrorisme, d’espionnage ou de piraterie constitue une menace réelle. Ces tentatives malveillantes de communiquer secrètement ont récemment conduit les chercheurs à inclure une nouvelle branche de recherche: la stéganalyse, pour contrer les techniques stéganographique. L’objectif principal de la stéganalyse est de détecter la résence potentielle d’un message dissimulé dans un support numérique et ne considère pas nécessairement son extraction. La parole numérique constitue un terrain de prédilection pour dissimuler des données numériques. En effet, elle est présente en abondance grâce aux technologies de télécommunications fixes ou mobiles et aussi à travers divers moyens de stockage de l’audio numérique. Cette thèse étudie la stéganographie et la stéganalyse utilisant la voix numérisée comme support et vise à (1) présenter un algorithme qui répond aux exigences des systèmes stéganographiques reliées à la capacité élevée, l’indétectabilité et l’imperceptibilité des données dissimulées, (2) le contrôle de la distorsion induite par le processus de dissimulation des données (3) définir un nouveau concept de zones spectrales dans le domaine de Fourier utilisant et l’amplitude et la phase (4) introduire un nouveau algorithme de stéganalyse basé sur les techniques de compression sans pertes d’information à la fois simple et efficace. La performance de l’algorithme stéganographique est mesurée par des méthodes d’évaluation perceptuelles et statistiques. D’autre part, la performance de l’algorithme de stéganalyse est mesurée par la capacité du système à distinguer entre un signal audio pur et un signal audio stéganographié. Les résultats sont très prometteurs et montrent des compromis de performance intéressants par rapport aux méthodes connexes. Les travaux futurs incluent essentiellement le renforcement de l’algorithme de stéganalyse pour qu’il soit en mesure de détecter une faible quantité de données dissimulées. Nous planifions également l’intégration de notre algorithme de stéganographie dans certaines plateformes émergentes telles que l’iPhone. D’autres perspectives consistent à améliorer l’algorithme stéganographique pour que les données dissimulées soit résistantes au codage de la parole, au bruit et à la distorsion induits parles canaux de transmission
Digital steganography is a young flourishing science emerged as a prominent source of data security. The primary goal of steganography is to reliably send hidden information secretly, not merely to obscure its presence. It exploits the characteristics of digital media files such as: image, audio, video, text by utilizing them as carriers to secretly communicate data. Encryption and watermarking techniques are already used to address concerns related to datasecurity. However, constantly-changing attacks on the integrity of digital data require new techniques to break the cycle of malicious attempts and expand the scope of involved applications. The main objective of steganographic systems is to provide secure, undetectable and imperceptible ways to conceal high-rate of data into digital medium. Steganography is used under the assumption that it will not be detected if no one is attempting to uncover it. Steganography techniques have found their way into various and versatile applications. Some of these applications are used for the benefit of people others are used maliciously. The threat posed by criminals, hackers, terrorists and spies using steganography is indeed real. To defeat malicious attempts when communicating secretly, researchers’ work has been lately extended toinclude a new and parallel research branch to countermeasure steganagraphy techniques called steganalysis. The main purpose of steganalysis technique is to detect the presence or not of hidden message and does not consider necessarily its successful extraction. Digital speech, in particular, constitutes a prominent source of data-hiding across novel telecommunication technologies such as covered voice-over-IP, audio conferencing, etc. This thesis investigatesdigital speech steganography and steganalysis and aims at: (1) presenting an algorithm that meets high data capacity, undetectability and imperceptibility requirements of steganographic systems, (2) controlling the distortion induced by the embedding process (3) presenting new concepts of spectral embedding areas in the Fourier domain which is applicable to magnitude and phase spectrums and (4) introducing a simple yet effective speech steganalysis algorithm based on lossless data compression techniques. The steganographic algorithm’s performance is measured by perceptual and statistical evaluation methods. On the other hand, the steganalysis algorithm’s performance is measured by how well the system can distinguish between stego- and cover-audio signals. The results are very promising and show interesting performance tradeoffs compared to related methods. Future work is based mainly on strengthening the proposed steganalysis algorithm to be able to detect small hiding capacity. As for our steganographic algorithm, we aim at integrating our steganographic in some emerging devices such as iPhone and further enhancing the capabilities of our steganographic algorithm to ensure hidden-data integrity under severe compression, noise and channel distortion
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Kafentzis, George. "Adaptive Sinusoidal Models for Speech with Applications in Speech Modifications and Audio Analysis." Thesis, Rennes 1, 2014. http://www.theses.fr/2014REN1S085/document.

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La modélisation sinusoïdale est une des méthodes les plus largement utilisés paramétriques pour la parole et le traitement des signaux audio. Inspiré par le récemment introduit Modèle aQHM et Modèle aHM, nous la vue d’ensemble de la théorie de l’ adaptation sinusoïdale modélisation et nous proposons un modèle nommé la Modèle eaQHM, qui est un non modèle paramétrique de mesure d’ajuster les amplitudes et les phases instantanées de ses fonctions de base aux caractéristiques variant dans le temps de sous-jacents du signal de parole, ainsi atténuer significativement la dite hypothèse de stationnarité locale. Le eaQHM est montré à surperformer aQHM dans l’analyse et la resynthèse de la parole voisée. Sur la base de la eaQHM , un système hybride d’analyse / synthèse de la parole est présenté (eaQHNM), et aussi d’ une version hybride de l’ aHM (aHNM). En outre, nous présentons la motivation pour une représentation pleine bande de la parole en utilisant le eaQHM, c’est, représentant toutes les parties du discours comme haute résolution des sinusoıdes AM-FM. Les expériences montrent que l’adaptation et la quasi-harmonicité est suffisante pour fournir une qualité de transparence dans la parole non voisée resynthèse. La pleine bande analyse eaQHM et système de synthèse est présenté à côté, ce qui surpasse l’état de l’art des systèmes, hybride ou pleine bande, dans la reconstruction de la parole, offrant une qualité transparente confirmé par des évaluations objectives et subjectives. En ce qui concerne les applications, le eaQHM et l’ aHM sont appliquées sur les modifications de la parole (de temps et pas mise à l’échelle). Les modifications qui en résultent sont de haute qualité, et suivent des règles très simples, par rapport à d’autres systèmes de modification état de l’art. Les résultats montrent que harmonicité est préféré au quasi- harmonicité de modifications de la parole du fait de la simplicité de la représentation intégrée. En outre, la pleine bande eaQHM est appliquée sur le problème de la modélisation des signaux audio, et en particulier d’instrument de musique retentit. Le eaQHM est évaluée et comparée à des systèmes à la pointe de la technologie, et leur est montré surpasser en termes de qualité de resynthèse, représentant avec succès l’attaque , transitoire, et une partie stationnaire d’un son d’instruments de musique. Enfin, une autre application est suggéré, à savoir l’analyse et la classification des discours émouvant. Le eaQHM est appliqué sur l’analyse des discours émouvant, offrant à ses paramètres instantanés comme des caractéristiques qui peuvent être utilisés dans la reconnaissance et la quantification vectorielle à base classification du contenu émotionnel de la parole. Bien que les modèles sinusoidaux sont pas couramment utilisés dans ces tâches, les résultats sont prometteurs
Sinusoidal Modeling is one of the most widely used parametric methods for speech and audio signal processing. The accurate estimation of sinusoidal parameters (amplitudes, frequencies, and phases) is a critical task for close representation of the analyzed signal. In this thesis, based on recent advances in sinusoidal analysis, we propose high resolution adaptive sinusoidal models for analysis, synthesis, and modifications systems of speech. Our goal is to provide systems that represent speech in a highly accurate and compact way. Inspired by the recently introduced adaptive Quasi-Harmonic Model (aQHM) and adaptive Harmonic Model (aHM), we overview the theory of adaptive Sinusoidal Modeling and we propose a model named the extended adaptive Quasi-Harmonic Model (eaQHM), which is a non-parametric model able to adjust the instantaneous amplitudes and phases of its basis functions to the underlying time-varying characteristics of the speech signal, thus significantly alleviating the so-called local stationarity hypothesis. The eaQHM is shown to outperform aQHM in analysis and resynthesis of voiced speech. Based on the eaQHM, a hybrid analysis/synthesis system of speech is presented (eaQHNM), along with a hybrid version of the aHM (aHNM). Moreover, we present motivation for a full-band representation of speech using the eaQHM, that is, representing all parts of speech as high resolution AM-FM sinusoids. Experiments show that adaptation and quasi-harmonicity is sufficient to provide transparent quality in unvoiced speech resynthesis. The full-band eaQHM analysis and synthesis system is presented next, which outperforms state-of-the-art systems, hybrid or full-band, in speech reconstruction, providing transparent quality confirmed by objective and subjective evaluations. Regarding applications, the eaQHM and the aHM are applied on speech modifications (time and pitch scaling). The resulting modifications are of high quality, and follow very simple rules, compared to other state-of-the-art modification systems. Results show that harmonicity is preferred over quasi-harmonicity in speech modifications due to the embedded simplicity of representation. Moreover, the full-band eaQHM is applied on the problem of modeling audio signals, and specifically of musical instrument sounds. The eaQHM is evaluated and compared to state-of-the-art systems, and is shown to outperform them in terms of resynthesis quality, successfully representing the attack, transient, and stationary part of a musical instrument sound. Finally, another application is suggested, namely the analysis and classification of emotional speech. The eaQHM is applied on the analysis of emotional speech, providing its instantaneous parameters as features that can be used in recognition and Vector-Quantization-based classification of the emotional content of speech. Although the sinusoidal models are not commonly used in such tasks, results are promising
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SIMONETTA, FEDERICO. "MUSIC INTERPRETATION ANALYSIS. A MULTIMODAL APPROACH TO SCORE-INFORMED RESYNTHESIS OF PIANO RECORDINGS." Doctoral thesis, Università degli Studi di Milano, 2022. http://hdl.handle.net/2434/918909.

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This Thesis discusses the development of technologies for the automatic resynthesis of music recordings using digital synthesizers. First, the main issue is identified in the understanding of how Music Information Processing (MIP) methods can take into consideration the influence of the acoustic context on the music performance. For this, a novel conceptual and mathematical framework named “Music Interpretation Analysis” (MIA) is presented. In the proposed framework, a distinction is made between the “performance” – the physical action of playing – and the “interpretation” – the action that the performer wishes to achieve. Second, the Thesis describes further works aiming at the democratization of music production tools via automatic resynthesis: 1) it elaborates software and file formats for historical music archiving and multimodal machine-learning datasets; 2) it explores and extends MIP technologies; 3) it presents the mathematical foundations of the MIA framework and shows preliminary evaluations to demonstrate the effectiveness of the approach
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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.

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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 AV
Humans 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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Elfitri, I. "Analysis by synthesis spatial audio coding." Thesis, University of Surrey, 2013. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.590657.

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Spatial Audio Coding (SAC) is a technique used to encode multichannel audio signals by extract ing the spatial parameters and downmixing the audio signals to a mono or stereo audio signal. Recently, various SAC techniques have been proposed to efficiently encode multichannel audio signals. However, all of them operate in open-loop, where the encoder and decoder operate sequentially and independently, and, thus, lack a mechanism for minimising the decoded audio reconstruction error. This thesis proposes a novel SAC technique that utilises the closed-loop system configuration, termed Analysis by Synthesis (AbS), in order to optimise the downmix: signal and the spatial parameters, so as to minimise the decoded signal error. In order to show the effect of the AbS optimisations, the Reverse One-To-Two (R-OTT) module, used in the MPEG Surround (MPS) , must first be applied in the frequency domain to recalculate the downmix and residual signals based on the quantised spatial parameters. These parameters show that the AbS scheme can minimise the quantisation errors of the spatial parameters. As the full AbS is far too complicated to be applied in practice, a simplified AbS algorithm for finding sub-optimal parameters, based on the adapted R-OTT module, is also proposed. Subjective tests show that the proposed Analysis by Synthesis Spatial Audio Coding (AbS-SAC), encoding 5-channel audio signals at a bitrate of 51.2 kb/s per audio channel, achieves higher Subjective Difference Grade (SDG) scores than the tested Advanced Audio Coding (AAC) technique. Furthermore, the objective test also shows that the proposed AbS-SAC method, operating at bitrates of 40 to 96 kb/s per audio channel, significantly outperforms (in terms of Objective Difference Grade (ODG) scores) the tested AAC multichannel technique.
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Fazekas, György. "Semantic audio analysis utilities and applications." Thesis, Queen Mary, University of London, 2012. http://qmro.qmul.ac.uk/xmlui/handle/123456789/8443.

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Extraction, representation, organisation and application of metadata about audio recordings are in the concern of semantic audio analysis. Our broad interpretation, aligned with recent developments in the field, includes methodological aspects of semantic audio, such as those related to information management, knowledge representation and applications of the extracted information. In particular, we look at how Semantic Web technologies may be used to enhance information management practices in two audio related areas: music informatics and music production. In the first area, we are concerned with music information retrieval (MIR) and related research. We examine how structured data may be used to support reproducibility and provenance of extracted information, and aim to support multi-modality and context adaptation in the analysis. In creative music production, our goals can be summarised as follows: O↵-the-shelf sound editors do not hold appropriately structured information about the edited material, thus human-computer interaction is inefficient. We believe that recent developments in sound analysis and music understanding are capable of bringing about significant improvements in the music production workflow. Providing visual cues related to music structure can serve as an example of intelligent, context-dependent functionality. The central contributions of this work are a Semantic Web ontology for describing recording studios, including a model of technological artefacts used in music production, methodologies for collecting data about music production workflows and describing the work of audio engineers which facilitates capturing their contribution to music production, and finally a framework for creating Web-based applications for automated audio analysis. This has applications demonstrating how Semantic Web technologies and ontologies can facilitate interoperability between music research tools, and the creation of semantic audio software, for instance, for music recommendation, temperament estimation or multi-modal music tutoring.
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Steinhour, Jacob B. "The Social and Pedagogical Advantages of Audio Forensics and Restoration Education." Ohio University Honors Tutorial College / OhioLINK, 2010. http://rave.ohiolink.edu/etdc/view?acc_num=ouhonors1276014966.

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Xiao, Zhongzhe Chen Liming. "Recognition of emotions in audio signals." Ecully : Ecole Centrale de Lyon, 2008. http://bibli.ec-lyon.fr/exl-doc/zxiao.pdf.

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Books on the topic "Audio analysi"

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Schuller, Björn W. Intelligent Audio Analysis. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-36806-6.

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Schuller, Björn W. Intelligent Audio Analysis. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013.

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Maher, Robert C. Principles of Forensic Audio Analysis. Cham: Springer International Publishing, 2018. http://dx.doi.org/10.1007/978-3-319-99453-6.

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Lerch, Alexander. Audio content analysis: An introduction. Hoboken, N.J: Wiley, 2012.

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Lerch, Alexander. An Introduction to Audio Content Analysis. Hoboken, NJ, USA: John Wiley & Sons, Inc., 2012. http://dx.doi.org/10.1002/9781118393550.

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Pollock, Alan. Surgical audit. London: Butterworths, 1989.

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Cotton, Courtenay. Characterizing Audio Events for Video Soundtrack Analysis. [New York, N.Y.?]: [publisher not identified], 2013.

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author, Pikrakis Aggelos, ed. Introduction to audio analysis: A MATLAB approach. Kidlington, Oxford: Academic Press is an imprint of Elsevier, 2014.

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Willborn, Walter W. O. Audit standards: A comparative analysis. 2nd ed. Milwaukee, Wis: ASQC Quality Press, 1993.

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Willborn, Walter W. O. Audit standards: A comparative analysis. Milwaukee, Wis: American Society for Quality Control, 1987.

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Book chapters on the topic "Audio analysi"

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Dawson, Catherine. "Audio analysis." In A–Z of Digital Research Methods, 10–16. Abingdon, Oxon ; New York, NY : Routledge, 2019.: Routledge, 2019. http://dx.doi.org/10.4324/9781351044677-3.

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Schuller, Björn. "Audio Data." In Intelligent Audio Analysis, 23–40. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-36806-6_5.

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Schuller, Björn. "Audio Features." In Intelligent Audio Analysis, 41–97. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-36806-6_6.

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Schuller, Björn. "Audio Recognition." In Intelligent Audio Analysis, 99–138. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-36806-6_7.

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Schuller, Björn. "Audio Source Separation." In Intelligent Audio Analysis, 139–47. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-36806-6_8.

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Zhang, Tong, and C. C. Jay Kuo. "Audio Feature Analysis." In Content-Based Audio Classification and Retrieval for Audiovisual Data Parsing, 35–54. Boston, MA: Springer US, 2001. http://dx.doi.org/10.1007/978-1-4757-3339-6_3.

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Lu, Lie, and Alan Hanjalic. "Audio Content Analysis." In Encyclopedia of Database Systems, 1–3. New York, NY: Springer New York, 2016. http://dx.doi.org/10.1007/978-1-4899-7993-3_1528-2.

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Lu, Lie, and Alan Hanjalic. "Audio Content Analysis." In Encyclopedia of Database Systems, 154–56. Boston, MA: Springer US, 2009. http://dx.doi.org/10.1007/978-0-387-39940-9_1528.

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Braddock, John Paul. "Mastering audio analysis." In Mastering in Music, 104–18. [1.] | New York : Taylor and Francis, 2021. | Series: Perspectives on music production: Focal Press, 2020. http://dx.doi.org/10.4324/9780429276590-7.

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Lu, Lie, and Alan Hanjalic. "Audio Content Analysis." In Encyclopedia of Database Systems, 198–201. New York, NY: Springer New York, 2018. http://dx.doi.org/10.1007/978-1-4614-8265-9_1528.

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Conference papers on the topic "Audio analysi"

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Barreto, Cephas A. S., Victor V. Targino, Tales V. de M. Alves, Lucas V. Bazante, Rafael V. R. de Oliveira, Ricardo A. R. do A. Junior, João C. Xavier-Júnior, and Anne Magály de P. Canuto. "Applying Feature Selection Combination in Audios of Whale for Improving Classification." In Encontro Nacional de Inteligência Artificial e Computacional. Sociedade Brasileira de Computação - SBC, 2022. http://dx.doi.org/10.5753/eniac.2022.227616.

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Audio signal processing has been under investigation for the last decades. The majority of the works found in literature focus on signal analysis and classification. Most of them integrate Machine Learning (ML) algorithms with the audio signal processing techniques. As the performance of any ML algorithm depends on the features of a dataset used for training and testing purposes, using a dataset derived from the extraction of features from an audio is not trivial due to the fact that the correct combination of extraction techniques with the selection of the most relevant attributes needs to take place. In this sense, this paper proposes an empirical analysis on different audio extraction techniques combined with feature selection for improving Whale audio classification. Usually, the application of audio extraction techniques results in poor classification performance. However, the combination of feature selection can achieve better results. The experimental results have been promising, indicating that the idea of combining different audio extraction techniques with feature selection can improve the performance of ML classification algorithms over whales’ audios by 22 percentage points.
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Zhan, Yunzhen, and Xiaochen Yuan. "Audio post-processing detection and identification based on audio features." In 2017 International Conference on Wavelet Analysis and Pattern Recognition (ICWAPR). IEEE, 2017. http://dx.doi.org/10.1109/icwapr.2017.8076681.

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Couvreur, L., F. Bettens, J. Hancq, and M. Mancas. "Normalized auditory attention levels for automatic audio surveillance." In RISK ANALYSIS 2008. Southampton, UK: WIT Press, 2008. http://dx.doi.org/10.2495/risk080441.

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Delgado, Alejandro, SKoT McDonald, Ning Xu, and Mark Sandler. "A New Dataset for Amateur Vocal Percussion Analysis." In AM'19: Audio Mostly. New York, NY, USA: ACM, 2019. http://dx.doi.org/10.1145/3356590.3356844.

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Pfeiffer, Silvia, Stephan Fischer, and Wolfgang Effelsberg. "Automatic audio content analysis." In the fourth ACM international conference. New York, New York, USA: ACM Press, 1996. http://dx.doi.org/10.1145/244130.244139.

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Ramirez, Rafael. "Session details: Audio analysis." In MM '10: ACM Multimedia Conference. New York, NY, USA: ACM, 2010. http://dx.doi.org/10.1145/3258351.

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Perez, Mauricio, Rodolfo Coelho De Souza, and Regis Rossi Alves Faria. "Digital Design of Audio Signal Processing Using Time Delay." In Simpósio Brasileiro de Computação Musical. Sociedade Brasileira de Computação - SBC, 2019. http://dx.doi.org/10.5753/sbcm.2019.10449.

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This poster describes the design in PureData of some audio signals processes in real time like delay, echo, reverb, chorus, flanger e phaser. We analyze the technical characteristics of each process and the psychoacoustic effects produced by them in human perception and audio applications. A deeper comprehension of the consequences of sound processes based on delay lines helps the decision-making in professional audio applications such as the audio recording, mixing, besides music composition that employs sound effects in preprocessed or real-time.
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Antsiferova, V., T. Pesetskaya, I. Yuldoshev, Syanyan Lu, Cin Van, and V. Lavlinskiy. "ANALYSIS OF RELEVANT CHARACTERISTICS OF INTERJECTION AUDIO SIGNALS ON THE EXAMPLE OF THE RUSSIAN LANGUAGE." In Modern aspects of modeling systems and processes. FSBE Institution of Higher Education Voronezh State University of Forestry and Technologies named after G.F. Morozov, 2021. http://dx.doi.org/10.34220/mamsp_9-14.

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This article describes a method for analyzing the relevant characteristics of audio signals of in-terjections on the example of the Russian language. Audio signals interjections WOW! recorded on a dictaphone and pronounced with different intonation. Interjections are recorded as WAV files for the Russian language using the Praat software product.
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Belloch, Jose A., Christian Antoñanzas, Pablo Gutierrez-Parera, and Mª Angeles Simarro. "Audiovisual Tool for understanding Audio concepts for being used in bachelor’s degree programmes." In HEAd'16 - International Conference on Higher Education Advances. Valencia: Universitat Politècnica València, 2016. http://dx.doi.org/10.4995/head16.2016.2923.

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In the Audio Signal Processing field, there exists difficulties in order to explain different concepts such as, compression, masking, quantization, sampling, among others. Further, most of these concepts require the use of audio laboratories and multiple practical session that must carry out students. Another issue is that there are students that are not able to internalize these concepts straightforwardly and require more practical sessions. In order to address these problems, we have developed an audiovisual tool, designed with Matlab, that can be used for professors and students. This tool allows to analyze, test and apply the audio concepts to real audio signals. The developed tool has been successfully experienced by professors of the audio signal processing field that recommend its use in upcoming academic courses.
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Engeln, Lars, Nhat Long Le, Matthew McGinity, and Rainer Groh. "Similarity Analysis of Visual Sketch-based Search for Sounds." In AM '21: Audio Mostly 2021. New York, NY, USA: ACM, 2021. http://dx.doi.org/10.1145/3478384.3478423.

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Reports on the topic "Audio analysi"

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Wisher, Robert A., Annette N. Priest, and Edward C. Glover. Audio Teletraining for Unit Clerks: A Cost-Effectiveness Analysis. Fort Belvoir, VA: Defense Technical Information Center, June 1997. http://dx.doi.org/10.21236/ada337689.

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Tololiu, Kevin Efrain, Arie Kurnianto, and krisztina Csokasi. Audio Intervention for Acute Pain Management - Protocol of Systematic Review and Meta-Analysis. INPLASY - International Platform of Registered Systematic Review and Meta-analysis Protocols, January 2023. http://dx.doi.org/10.37766/inplasy2023.1.0002.

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Review question / Objective: This study aims to investigate, review, and asses existing literature concerning audio intervention to reduce acute pain. Condition being studied: The study will focus on acute pain experienced by adults in any diseases or surgical procedures. Eligibility criteria: This study will exclude papers published older than ten years ago to collect updated data, non RCTs, non-English literature, paper with combined interventions, and papers with an incomplete essential statical value of pain for meta-analysis.
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Brady-Herbst, Brenene. An Analysis of Spondee Recognition Thresholds in Auditory-only and Audio-visual Conditions. Portland State University Library, January 2000. http://dx.doi.org/10.15760/etd.7094.

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Paranjothi, Gokul, Jonathan Morgenstein, and Hallie Lucas. MFVI Energy Efficiency Audit Training - Module 1.2: Lighting Analysis. Office of Scientific and Technical Information (OSTI), September 2022. http://dx.doi.org/10.2172/1886871.

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Sapiro, Guillermo. Structured and Collaborative Signal Models: Theory and Applications in Image, Video, and Audio Analysis. Fort Belvoir, VA: Defense Technical Information Center, January 2013. http://dx.doi.org/10.21236/ada586672.

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Jajodia, Sushi. Integration of Audit Data Analysis and Mining Techniques into Aide. Fort Belvoir, VA: Defense Technical Information Center, July 2006. http://dx.doi.org/10.21236/ada456840.

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Paranjothi, Gokul, Jonathan Morgenstein, and Hallie Lucas. MFVI Energy Efficiency Audit Training Module 2.2: Plug Loads Analysis. Office of Scientific and Technical Information (OSTI), September 2022. http://dx.doi.org/10.2172/1889268.

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Khan, Mahreen. Evaluating External Government Audit. Institute of Development Studies, September 2022. http://dx.doi.org/10.19088/k4d.2022.140.

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This rapid literature review of primary and grey sources found substantial evidence of the merits of donor support to Public Financial Management (PFM) initiatives but no specific evidence assessing donor support for external government audit, such as Supreme Audit Institutions (SAIs). PFM reforms are established as being generally beneficial, assist in reducing or preventing corruption, increasing transparency and accountability, as well as improving service delivery quality, although the exact impacts are difficult to measure. Performance auditing has recently attracted more attention than traditional financial or compliance auditing and is seen by many sources to be conducive to improving accountability, although compliance and financial auditing are still viewed as the core of external audit. There is a substantial body of literature on donor-assisted PFM reforms but a paucity of focused study or discussion of donor support to external audit specifically. This evidence gap may be due to the cost of examining the narrow focus required on donor-assisted external audit specifically. This is compounded by the complexity of gathering a sufficiently large database through surveys combined with the lack of access (for individual academics) to official datasets across countries. Furthermore, measuring the impact of SAIs, for example, is difficult due to the variety of regulatory structures that exist, inhibiting comparative cross-country studies, which has resulted in a preference for in-depth analyses. Only multilateral institutions have conducted comprehensive cross-country surveys. However, the evidence does show that strengthened PFM systems and SAIs,1 if they are independent and fully resourced, increase transparency and accountability, helping to combat corruption, when governments are made answerable to their audit findings. The evidence on the effectiveness of SAIs (against corruption) is mixed and not as strong as for PFM reforms in general. The impact of PFM interventions in preventing or reducing corruption increases when reforms are sector-specific and complemented by societal awareness initiatives, citizen participation, and infomediary advocacy. This finding seems applicable to SAIs as the discourse is increasingly on improving comprehension of audit reports and wider dissemination to relevant stakeholders.
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Qi, Yuan. Learning Algorithms for Audio and Video Processing: Independent Component Analysis and Support Vector Machine Based Approaches. Fort Belvoir, VA: Defense Technical Information Center, August 2000. http://dx.doi.org/10.21236/ada458739.

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Kiianovska, N. M. The development of theory and methods of using cloud-based information and communication technologies in teaching mathematics of engineering students in the United States. Видавничий центр ДВНЗ «Криворізький національний університет», December 2014. http://dx.doi.org/10.31812/0564/1094.

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The purpose of the study is the analysis of the development of the theory and methods of ICT usage while teaching higher mathematics engineering students in the United States. It was determined following tasks: to analyze the problem source, to identify the state of its elaboration, to identify key trends in the development of theory and methods of ICT usage while teaching higher mathematics engineering students in the United States, the object of study – the use of ICT in teaching engineering students, the research methods are: analysis of scientific, educational, technical, historical sources; systematization and classification of scientific statements on the study; specification, comparison, analysis and synthesis, historical and pedagogical analysis of the sources to establish the chronological limits and implementation of ICT usage in educational practice of U.S. technical colleges. In article was reviewed a modern ICT tools used in learning of fundamental subjects for future engineers in the United States, shown the evolution and convergence of ICT learning tools. Discussed experience of the «best practices» using online ICT in higher engineering education at United States. Some of these are static, while others are interactive or dynamic, giving mathematics learners opportunities to develop visualization skills, explore mathematical concepts, and obtain solutions to self-selected problems. Among ICT tools are the following: tools to transmit audio and video data, tools to collaborate on projects, tools to support object-oriented practice. The analysis leads to the following conclusion: using cloud-based tools of learning mathematic has become the leading trend today. Therefore, university professors are widely considered to implement tools to assist the process of learning mathematics such properties as mobility, continuity and adaptability.
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