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| Titre : | Digital Signal Processing in Audio and Acoustical Engineering |
| Titre original: | Traitement numérique du signal en ingénierie audio et acoustique |
| Auteurs : | Francis F Li, Auteur ; Cox,Trevor J., Auteur |
| Type de document : | texte imprimé |
| Editeur : | New york [U.S.A.] : CRC press, 2019 |
| ISBN/ISSN/EAN : | 978-1-4665-9388-6 |
| Format : | XIII-228 p. / couv. ill. en coul.; ill. / 24 cm. |
| Langues: | Anglais |
| Langues originales: | Anglais |
| Index. décimale : | 621.382 2 (Traitement du signal (Théorie de l'information dans les techniques de télécommunications)) |
| Catégories : | |
| Mots-clés: | Digital Signal processing ; Sound |
| Résumé : |
Starting with essential maths, fundamentals of signals and systems, and classical concepts of DSP, this book presents, from an application-oriented perspective, modern concepts and methods of DSP including machine learning for audio acoustics and engineering. Content highlights include but are not limited to room acoustic parameter measurements, filter design, codecs, machine learning for audio pattern recognition and machine audition, spatial audio, array technologies and hearing aids. Some research outcomes are fed into book as worked examples. As a research informed text, the book attempts to present DSP and machine learning from a new and more relevant angle to acousticians and audio engineers.
Some MATLAB® codes or frameworks of algorithms are given as downloads available on the CRC Press website. Suggested exploration and mini project ideas are given for "proof of concept" type of exercises and directions for further study and investigation. The book is intended for researchers, professionals, and senior year students in the field of audio acoustics. |
| Note de contenu : |
Sommaire :
Chapitre 1: Acoustic signals and audio systems Chapitre 2: Sampling quantization and discrete fourier Chapitre 3: DSP in acoustical transfer function measurements Chapitre 4: Digital filter and z-transform Chapitre 5: Audio codecs Chapitre 6: DSP in binaural hearing and microphone arrays Chapitre 7: Adaptive filters Chapitre 8: Machine learning in acoustic DSP Chapitre 9: Unsupervised learning and blind source separation Chapitre 10: DSP in hearing aids. |
Exemplaires (2)
| Cote | Support | Localisation | Section | Disponibilité |
|---|---|---|---|---|
| F8/12007 | Livre | Bibliothèque de la Faculté de Technologie | Salle des livres | Disponible |
| F8/12008 | Livre | Bibliothèque de la Faculté de Technologie | Salle des livres | Disponible |
