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Artykuły w czasopismach na temat "Traitement audio numérique"
Malti, Rachid, i Martial Leyney. "Do Ré Mi Fa Sol La S T M 32". J3eA 22 (2023): 1037. http://dx.doi.org/10.1051/j3ea/20231037.
Pełny tekst źródłaLópez Charles, Carlos. "La transduction dans la création de la musique visuelle en temps différé". Revue Francophone Informatique et Musique 1, nr 1 (2022). http://dx.doi.org/10.56698/rfim.202.
Pełny tekst źródłaTangedal, Kirsten, Jennifer Bolt, Suzanne Len i Ali Bell. "Baseline Competency Assessment of Pharmacists Prescribing and Managing Vancomycin Therapy in the Regina Qu’Appelle Health Region". Canadian Journal of Hospital Pharmacy 70, nr 5 (30.10.2017). http://dx.doi.org/10.4212/cjhp.v70i5.1694.
Pełny tekst źródłaRozprawy doktorskie na temat "Traitement audio numérique"
Fillon, Thomas. "Traitement numérique du signal acoustique pour une aide aux malentendants". Phd thesis, Télécom ParisTech, 2004. http://pastel.archives-ouvertes.fr/pastel-00001201.
Pełny tekst źródłaNesvadba, Jan. "Segmentation sémantique des contenus audio-visuels". Bordeaux 1, 2007. http://www.theses.fr/2007BOR13456.
Pełny tekst źródłaLapierre, Jimmy. "Approches paramétriques pour le codage audio multicanal". Mémoire, Université de Sherbrooke, 2007. http://savoirs.usherbrooke.ca/handle/11143/1355.
Pełny tekst źródłaGonzález, Santos Ángel de Dios. "Circuits de traitement de signal numérique en temps continu ultra-faible consommation en technologie 28nm FDSOI pour applications audio". Thesis, Lille 1, 2020. http://www.theses.fr/2020LIL1I047.
Pełny tekst źródłaThe focus of this work is the study and development of a feature extraction system using Continuous-Time Digital Signal Processing (CT DSP) techniques, to mitigate the drawbacks of existing implementations based on traditional analog and digital solutions of always-on monitoring sensors for the Internet of Things (IoT). The target is to extract the spectral content of an audio signal using a novel architecture based on a cascade of configurable CT DSP Finite Impulse Response (FIR) filters. An efficient cascade scheme is enabled by the proposed glitch elimination and delta encoding techniques. Additionally, this work introduces a CT function to estimate the instantaneous power within selected frequency bands to build an output spectrogram. The proposed 12-band system has been validated using behavioral simulations. The key element for the implementation of this system is the digital delay element. A new delay element has been designed and fabricated in 28nm FDSOI technology and achieves a record tuning range from 30 ns to 97 µs with a power consumption of 15 fJ/event. By extrapolating this result, the system would have an overall peak power consumption of 2.85 µW when processing typical female speech, while consuming approximately 100 nW when no events are generated. Thus, the average system power consumption outperforms state-of-the-art feature extraction circuits
Hassaïne, Abdelâali. "Restauration des pistes sonores optiques cinématographiques : approche par traitement d'images". Phd thesis, École Nationale Supérieure des Mines de Paris, 2009. http://pastel.archives-ouvertes.fr/pastel-00005981.
Pełny tekst źródłaLapierre, Jimmy. "Amélioration de codecs audio standardisés avec maintien de l'interopérabilité". Thèse, Université de Sherbrooke, 2016. http://hdl.handle.net/11143/8816.
Pełny tekst źródłaAbstract : Digital audio applications have grown exponentially during the last decades, in good part because of the establishment of international standards. However, imposing such norms necessarily introduces hurdles that can impede the improvement of technologies that have already been deployed, potentially leading to a proliferation of new standards. This thesis shows that existent coders can be better exploited by improving their quality or their bitrate, even within the rigid constraints posed by established standards. Three aspects are studied, being the enhancement of the encoder, the decoder and the bit stream. In every case, the compatibility with the other elements of the existent coder is maintained. Thus, it is shown that the audio signal can be improved at the decoder without transmitting new information, that an encoder can produce an improved signal without modifying its decoder, and that a bit stream can be optimized for a new application. In particular, this thesis shows that even a standard like G.711, which has been deployed for decades, has the potential to be significantly improved after the fact. This contribution has even served as the core for a new standard embedded coder that had to maintain that compatibility. It is also shown that the subjective and objective audio quality of the AAC (Advanced Audio Coding) decoder can be improved, without adding any extra information from the encoder, by better exploiting the knowledge of the coder model’s limitations. Finally, it is shown that the fixed rate bit stream of the AMR-WB+ (Extended Adaptive Multi-Rate Wideband) can be compressed more efficiently when considering a variable bit rate scenario, showing the need to adapt a coder to its use case.
Parekh, Sanjeel. "Learning representations for robust audio-visual scene analysis". Thesis, Université Paris-Saclay (ComUE), 2019. http://www.theses.fr/2019SACLT015/document.
Pełny tekst źródłaThe goal of this thesis is to design algorithms that enable robust detection of objectsand events in videos through joint audio-visual analysis. This is motivated by humans’remarkable ability to meaningfully integrate auditory and visual characteristics forperception in noisy scenarios. To this end, we identify two kinds of natural associationsbetween the modalities in recordings made using a single microphone and camera,namely motion-audio correlation and appearance-audio co-occurrence.For the former, we use audio source separation as the primary application andpropose two novel methods within the popular non-negative matrix factorizationframework. The central idea is to utilize the temporal correlation between audio andmotion for objects/actions where the sound-producing motion is visible. The firstproposed method focuses on soft coupling between audio and motion representationscapturing temporal variations, while the second is based on cross-modal regression.We segregate several challenging audio mixtures of string instruments into theirconstituent sources using these approaches.To identify and extract many commonly encountered objects, we leverageappearance–audio co-occurrence in large datasets. This complementary associationmechanism is particularly useful for objects where motion-based correlations are notvisible or available. The problem is dealt with in a weakly-supervised setting whereinwe design a representation learning framework for robust AV event classification,visual object localization, audio event detection and source separation.We extensively test the proposed ideas on publicly available datasets. The experimentsdemonstrate several intuitive multimodal phenomena that humans utilize on aregular basis for robust scene understanding
Mbaye, Amadou. "Linéarisation des amplificateurs de puissance large-bande pour des applications de communications tactiques et de diffusion audio ou vidéo numérique". Thesis, Paris Est, 2015. http://www.theses.fr/2015PEST1021/document.
Pełny tekst źródłaPower amplifier is one of the most critical element within radiocommunications systems. The PA is their main source of nonlinearities and it has a great contribution on the emitter's power consumption. Running the PA with highest power efficiency is thus as crucial as having it linear for a good communication quality. However these two specifications of the PA are antagonistic and PA manifacturers need to find a compromise between linearity and power efficiency. Digital Predistortion (DPD) and Crest factor Reduction techniques are intended to improve power efficiency while preserving linearity or inversely. Linearization of wideband RF power amplifiers using Digital Predistortion is the focus of this thesis. Three DPD issues are investigated in these works. The first issue deals with multiband linearization where signals with various waveforms located at different frequency bands are amplified. The second objective of this thesis is to study a concurrent DPD/CFR systems based on an automatic estimation of the necessary CFR gain. The last part of this dissertation deals with PA linearization under antenna load variations. Indeed, the impedance of antenna may vary because of electromagnetic objects that are present in its vicinity. Those impedance variations may instigate signal reflections toward the PA, that modify some of its main specifications (linearity, delivered power and efficiency). Our goal in this field is to preserve DPD linearization performances under antenna load mismatch
Gillet, Olivier. "Transcription des signaux percussifs : application à l'analyse de scènes musicales audiovisuelles". Phd thesis, Télécom ParisTech, 2007. http://pastel.archives-ouvertes.fr/pastel-00002805.
Pełny tekst źródłaBayle, Yann. "Apprentissage automatique de caractéristiques audio : application à la génération de listes de lecture thématiques". Thesis, Bordeaux, 2018. http://www.theses.fr/2018BORD0087/document.
Pełny tekst źródłaThis doctoral dissertation presents, discusses and proposes tools for the automatic information retrieval in big musical databases.The main application is the supervised classification of musical themes to generate thematic playlists.The first chapter introduces the different contexts and concepts around big musical databases and their consumption.The second chapter focuses on the description of existing music databases as part of academic experiments in audio analysis.This chapter notably introduces issues concerning the variety and unequal proportions of the themes contained in a database, which remain complex to take into account in supervised classification.The third chapter explains the importance of extracting and developing relevant audio features in order to better describe the content of music tracks in these databases.This chapter explains several psychoacoustic phenomena and uses sound signal processing techniques to compute audio features.New methods of aggregating local audio features are proposed to improve song classification.The fourth chapter describes the use of the extracted audio features in order to sort the songs by themes and thus to allow the musical recommendations and the automatic generation of homogeneous thematic playlists.This part involves the use of machine learning algorithms to perform music classification tasks.The contributions of this dissertation are summarized in the fifth chapter which also proposes research perspectives in machine learning and extraction of multi-scale audio features
Książki na temat "Traitement audio numérique"
V, Madisetti, red. Video, speech, and audio signal processing and associated standards. Boca Raton, FL: CRC Press, 2009.
Znajdź pełny tekst źródłaV, Madisetti, red. Video, speech, and audio signal processing and associated standards. Boca Raton, FL: CRC Press, 2009.
Znajdź pełny tekst źródłaBlair, Benson K., red. Compression technologies for video and audio. New York: McGraw-Hill, 2000.
Znajdź pełny tekst źródłaGold, Bernard. Speech and audio signal processing: Processing and perception of speech and music. Wyd. 2. Hoboken, N.J: Wiley, 2011.
Znajdź pełny tekst źródłaNelson, Morgan, red. Speech and audio signal processing: Processing and perception of speech and music. New York: John Wiley, 2000.
Znajdź pełny tekst źródłaInc, NetLibrary, red. Applications of digital signal processing to audio and acoustics. New York: Kluwer Academic, 2002.
Znajdź pełny tekst źródłaFrédéric, Bath M'Wom, red. MP3. Paris: Micro Application, 2000.
Znajdź pełny tekst źródłaBennett, Christopher L. Digital Audio Theory. Taylor & Francis Group, 2020.
Znajdź pełny tekst źródłaBennett, Christopher L. Digital Audio Theory: A Practical Guide. Taylor & Francis Group, 2020.
Znajdź pełny tekst źródłaBennett, Christopher L. Digital Audio Theory: A Practical Guide. Taylor & Francis Group, 2020.
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