Speech Coding (eBook)
XXI, 240 Seiten
Springer-Verlag
978-3-319-50204-5 (ISBN)
This book provides scientific understanding of the most central techniques used in speech coding both for advanced students as well as professionals with a background in speech audio and or digital signal processing. It provides a clear connection between the Why's?, How's?, and What's, such that the necessity, purpose and solutions provided by tools should be always within sight, as well as their strengths and weaknesses in each respect. Equivalently, this book sheds light on the following perspectives for each technology presented:
Objective: What do we want to achieve and especially why is this goal important?
Resource / Information: What information is available and how can it be useful?
Resource / Platform: What kind of platforms are we working with and what are the capabilities/restrictions of those platforms? This includes properties such as computational, memory, acoustic and transmission capacity of devices used.
Solutions: Which solutions have been proposed and how can they be used to reach the stated goals?
Strengths and weaknesses: In which ways do the solutions fulfill the objectives and where are they insufficient? Are resources used efficiently?
This book concentrates solely on code excited linear prediction and its derivatives since mainstream speech codecs are based on linear prediction It also concentrates exclusively on time domain techniques because frequency domain tools are to a large extent common with audio codecs.
Tom Bäckström is Professor for Speech Coding at University of Erlangen-Nuremberg and Member of the International Audio Labs Erlangen, funded by Fraunhofer IIS. He is active as a researcher in mathematical methods in the modeling of the voice and audio. His interests are in developing the mathematical side even more in the intersection of digital signal processing, matrix and polynomial algebra and functional analysis.
Tom Bäckström is Professor for Speech Coding at University of Erlangen-Nuremberg and Member of the International Audio Labs Erlangen, funded by Fraunhofer IIS. He is active as a researcher in mathematical methods in the modeling of the voice and audio. His interests are in developing the mathematical side even more in the intersection of digital signal processing, matrix and polynomial algebra and functional analysis.
Introduction.- Usage.- Known Problems.- Implementation.
Erscheint lt. Verlag | 29.3.2017 |
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Reihe/Serie | Signals and Communication Technology |
Zusatzinfo | XXI, 240 p. 76 illus. |
Verlagsort | Cham |
Sprache | englisch |
Themenwelt | Mathematik / Informatik ► Informatik ► Web / Internet |
Technik ► Elektrotechnik / Energietechnik | |
Technik ► Nachrichtentechnik | |
Schlagworte | Algebraic Codebooks • Filter Windowing • Formant Enhancement • Jitter Buffer • Linear Predictive Coding • packet loss • perceptual modelling • Perceptual Windowing • Source Modelling |
ISBN-10 | 3-319-50204-2 / 3319502042 |
ISBN-13 | 978-3-319-50204-5 / 9783319502045 |
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