Author: Erik Larsen Publisher: John Wiley & Sons ISBN: 0470858656 Category : Technology & Engineering Languages : en Pages : 312
Book Description
Bandwidth extension (BWE) refers to various methods that increase either the perceived or real frequency spectrum (bandwidth) of audio signals. Such frequency extension is desirable if at some point the frequency content of the audio signal has been reduced, as can happen for example during recording, transmission or reproduction. This volume, significant in dealing exclusively with BWE, discusses applications to music and speech and places particular emphasis on signal processing techniques. Presents an all-encompassing approach to BWE by covering theory, applications and algorithms Reviews important concepts in psychoacoustics, signal processing and loudspeaker theory Develops the theory and implementation of BWE applied to low-frequency sound reproduction, perceptually coded audio, speech and noise abatement Includes a BWE patent overview Audio Bandwidth Extension pulls together recent developments in to a single volume and presents a coherent framework to the reader. Such an approach will have instant appeal to engineers, specialists, researchers and postgraduate students in the fields of audio, signal processing and speech.
Author: Bernd Iser Publisher: Springer ISBN: 9780387688985 Category : Technology & Engineering Languages : en Pages : 190
Book Description
Bandwidth Extension of Speech Signals describes the theory and methods for quality enhancement of clean speech signals and distorted speech signals such as those that have undergone a band limitation, for instance, in a telephone network. Problems and the respective solutions are discussed for the different approaches. The different approaches are evaluated and a real-time implementation of the most promising approach is presented. The book includes topics related to speech coding, pattern- / speech recognition, speech enhancement, statistics and digital signal processing in general.
Author: Søren Bech Publisher: John Wiley & Sons ISBN: 0470869240 Category : Technology & Engineering Languages : en Pages : 462
Book Description
As audio and telecommunication technologies develop, there is an increasing need to evaluate the technical and perceptual performance of these innovations. A growing number of new technologies (e.g. low bit-rate coding) are based on specific properties of the auditory system, which are often highly non-linear. This means that the auditory quality of such systems cannot be measured by traditional physical measures (such as distortion, frequency response etc.), but only by perceptual evaluations in the form of listening tests. Perceptual Audio Evaluation provides a comprehensive guide to the many variables that need to be considered before, during and after experiments. Including the selection of the content of the programme material to be reproduced, technical aspects of the production of the programme material, the experimental set-up including calibration, and the statistical planning of the experiment and subsequent analysis of the data. Perceptual Audio Evaluation: Provides a complete and accessible guide to the motives, theory and practical application of perceptual evaluation of reproduced sound. Discusses all the variables of perceptual evaluation, their control and their possible influence on the results. Covers in detail all international standards on the topic. Is illustrated throughout with tables, figures and worked solutions. Perceptual Audio Evaluation will appeal to audio and speech engineers as well as researchers in audio and speech laboratories. Postgraduate students in engineering or acoustics and undergraduate students studying psychoacoustics, speech audio processing and signal processing will also find this an essential reference.
Author: Ville Pulkki Publisher: John Wiley & Sons ISBN: 1119252598 Category : Technology & Engineering Languages : en Pages : 410
Book Description
A comprehensive guide that addresses the theory and practice of spatial audio This book provides readers with the principles and best practices in spatial audio signal processing. It describes how sound fields and their perceptual attributes are captured and analyzed within the time-frequency domain, how essential representation parameters are coded, and how such signals are efficiently reproduced for practical applications. The book is split into four parts starting with an overview of the fundamentals. It then goes on to explain the reproduction of spatial sound before offering an examination of signal-dependent spatial filtering. The book finishes with coverage of both current and future applications and the direction that spatial audio research is heading in. Parametric Time-frequency Domain Spatial Audio focuses on applications in entertainment audio, including music, home cinema, and gaming—covering the capturing and reproduction of spatial sound as well as its generation, transduction, representation, transmission, and perception. This book will teach readers the tools needed for such processing, and provides an overview to existing research. It also shows recent up-to-date projects and commercial applications built on top of the systems. Provides an in-depth presentation of the principles, past developments, state-of-the-art methods, and future research directions of spatial audio technologies Includes contributions from leading researchers in the field Offers MATLAB codes with selected chapters An advanced book aimed at readers who are capable of digesting mathematical expressions about digital signal processing and sound field analysis, Parametric Time-frequency Domain Spatial Audio is best suited for researchers in academia and in the audio industry.
Author: Patrick A. Naylor Publisher: Springer Science & Business Media ISBN: 1849960569 Category : Technology & Engineering Languages : en Pages : 388
Book Description
Speech Dereverberation gathers together an overview, a mathematical formulation of the problem and the state-of-the-art solutions for dereverberation. Speech Dereverberation presents current approaches to the problem of reverberation. It provides a review of topics in room acoustics and also describes performance measures for dereverberation. The algorithms are then explained with mathematical analysis and examples that enable the reader to see the strengths and weaknesses of the various techniques, as well as giving an understanding of the questions still to be addressed. Techniques rooted in speech enhancement are included, in addition to a treatment of multichannel blind acoustic system identification and inversion. The TRINICON framework is shown in the context of dereverberation to be a generalization of the signal processing for a range of analysis and enhancement techniques. Speech Dereverberation is suitable for students at masters and doctoral level, as well as established researchers.
Author: Visar Berisha Publisher: Springer Nature ISBN: 3031015215 Category : Technology & Engineering Languages : en Pages : 71
Book Description
Bandwidth extension of speech is used in the International Telecommunication Union G.729.1 standard in which the narrowband bitstream is combined with quantized high-band parameters. Although this system produces high-quality wideband speech, the additional bits used to represent the high band can be further reduced. In addition to the algorithm used in the G.729.1 standard, bandwidth extension methods based on spectrum prediction have also been proposed. Although these algorithms do not require additional bits, they perform poorly when the correlation between the low and the high band is weak. In this book, two wideband speech coding algorithms that rely on bandwidth extension are developed. The algorithms operate as wrappers around existing narrowband compression schemes. More specifically, in these algorithms, the low band is encoded using an existing toll-quality narrowband system, whereas the high band is generated using the proposed extension techniques. The first method relies only on transmitted high-band information to generate the wideband speech. The second algorithm uses a constrained minimum mean square error estimator that combines transmitted high-band envelope information with a predictive scheme driven by narrowband features. Both algorithms make use of novel perceptual models based on loudness that determine optimum quantization strategies for wideband recovery and synthesis. Objective and subjective evaluations reveal that the proposed system performs at a lower average bit rate while improving speech quality when compared to other similar algorithms.
Author: Shoji Makino Publisher: Springer Science & Business Media ISBN: 9783540240396 Category : Hearing Languages : en Pages : 432
Book Description
We live in a noisy world! In all applications (telecommunications, hands-free communications, recording, human-machine interfaces, etc.) that require at least one microphone, the signal of interest is usually contaminated by noise and reverberation. As a result, the microphone signal has to be "cleaned" with digital signal processing tools before it is played out, transmitted, or stored. This book is about speech enhancement. Different well-known and state-of-the-art methods for noise reduction, with one or multiple microphones, are discussed. By speech enhancement, we mean not only noise reduction but also dereverberation and separation of independent signals. These topics are also covered in this book. However, the general emphasis is on noise reduction because of the large number of applications that can benefit from this technology. The goal of this book is to provide a strong reference for researchers, engineers, and graduate students who are interested in the problem of signal and speech enhancement. To do so, we invited well-known experts to contribute chapters covering the state of the art in this focused field. TOC:Introduction.- Study of the Wiener Filter for Noise Reduction.- Statistical Methods for the Enhancement of Noisy Speech.- Single- und Multi-Microphone Spectral Amplitude Estimation Using a Super-Gaussian Speech Model.- From Volatility Modeling of Financial Time-Series to Stochastic Modeling and Enhancement of Speech Signals.- Single-Microphone Noise Suppression for 3G Handsets Based on Weighted Noise Estimation.- Signal Subspace Techniques for Speech Enhancement.- Speech Enhancement: Application of the Kalman Filter in the Estimate-Maximize (EM) Framework.- Speech Distortion Weighted Multichannel Wiener Filtering Techniques for Noise Reduction.- Adpative Microphone Arrays Employing Spatial Quadratic Soft Constraints and Spectral Shaping.- Single-Microphone Blind Dereverberation.- Separation and Dereverberation of Speech Signals with Multiple Microphones.- Frequency-Domain Blind Source Separation.- Subband Based Blind Source Separation.- Real-Time Blind Source Separation for Moving Speech Signals.- Separation of Speech by Computational Auditory Scene Analysis
Author: Marina Bosi Publisher: Springer Science & Business Media ISBN: 1461503272 Category : Science Languages : en Pages : 443
Book Description
Introduction to Digital Audio Coding and Standards provides a detailed introduction to the methods, implementations, and official standards of state-of-the-art audio coding technology. In the book, the theory and implementation of each of the basic coder building blocks is addressed. The building blocks are then fit together into a full coder and the reader is shown how to judge the performance of such a coder. Finally, the authors discuss the features, choices, and performance of the main state-of-the-art coders defined in the ISO/IEC MPEG and HDTV standards and in commercial use today. The ultimate goal of this book is to present the reader with a solid enough understanding of the major issues in the theory and implementation of perceptual audio coders that they are able to build their own simple audio codec. There is no other source available where a non-professional has access to the true secrets of audio coding.
Author: Vijay Madisetti Publisher: CRC Press ISBN: 1420046098 Category : Technology & Engineering Languages : en Pages : 616
Book Description
Now available in a three-volume set, this updated and expanded edition of the bestselling The Digital Signal Processing Handbook continues to provide the engineering community with authoritative coverage of the fundamental and specialized aspects of information-bearing signals in digital form. Encompassing essential background material, technical details, standards, and software, the second edition reflects cutting-edge information on signal processing algorithms and protocols related to speech, audio, multimedia, and video processing technology associated with standards ranging from WiMax to MP3 audio, low-power/high-performance DSPs, color image processing, and chips on video. Drawing on the experience of leading engineers, researchers, and scholars, the three-volume set contains 29 new chapters that address multimedia and Internet technologies, tomography, radar systems, architecture, standards, and future applications in speech, acoustics, video, radar, and telecommunications. This volume, Video, Speech, and Audio Signal Processing and Associated Standards, provides thorough coverage of the basic foundations of speech, audio, image, and video processing and associated applications to broadcast, storage, search and retrieval, and communications.
Author: Erik Larsen
Author: Bernd Iser
Author: Søren Bech
Author: Ville Pulkki
Author: Patrick A. Naylor
Author: Visar Berisha
Author: Shoji Makino
Author: Marina Bosi
Author: Vijay Madisetti
Author: Ian Devlin