Proceedings On Icassp 93 International Conference On Acoustic Speech And Signal Processing 1993
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| Author | : Saeed V. Vaseghi |
| Publisher | : John Wiley & Sons |
| Total Pages | : 480 |
| Release | : 2006-02-03 |
| Genre | : Science |
| ISBN | : 0470094958 |
Signal processing plays an increasingly central role in the development of modern telecommunication and information processing systems, with a wide range of applications in areas such as multimedia technology, audio-visual signal processing, cellular mobile communication, radar systems and financial data forecasting. The theory and application of signal processing deals with the identification, modelling and utilisation of patterns and structures in a signal process. The observation signals are often distorted, incomplete and noisy and hence, noise reduction and the removal of channel distortion is an important part of a signal processing system. Advanced Digital Signal Processing and Noise Reduction, Third Edition, provides a fully updated and structured presentation of the theory and applications of statistical signal processing and noise reduction methods. Noise is the eternal bane of communications engineers, who are always striving to find new ways to improve the signal-to-noise ratio in communications systems and this resource will help them with this task. * Features two new chapters on Noise, Distortion and Diversity in Mobile Environments and Noise Reduction Methods for Speech Enhancement over Noisy Mobile Devices. * Topics discussed include: probability theory, Bayesian estimation and classification, hidden Markov models, adaptive filters, multi-band linear prediction, spectral estimation, and impulsive and transient noise removal. * Explores practical solutions to interpolation of missing signals, echo cancellation, impulsive and transient noise removal, channel equalisation, HMM-based signal and noise decomposition. This is an invaluable text for senior undergraduates, postgraduates and researchers in the fields of digital signal processing, telecommunications and statistical data analysis. It will also appeal to engineers in telecommunications and audio and signal processing industries.
| Author | : Joseph Mariani |
| Publisher | : John Wiley & Sons |
| Total Pages | : 576 |
| Release | : 2013-03-01 |
| Genre | : Technology & Engineering |
| ISBN | : 1118623754 |
Speech processing addresses various scientific and technological areas. It includes speech analysis and variable rate coding, in order to store or transmit speech. It also covers speech synthesis, especially from text, speech recognition, including speaker and language identification, and spoken language understanding. This book covers the following topics: how to realize speech production and perception systems, how to synthesize and understand speech using state-of-the-art methods in signal processing, pattern recognition, stochastic modelling computational linguistics and human factor studies.
| Author | : Boualem Boashash |
| Publisher | : Academic Press |
| Total Pages | : 1070 |
| Release | : 2015-12-11 |
| Genre | : Technology & Engineering |
| ISBN | : 0123985250 |
Time-Frequency Signal Analysis and Processing (TFSAP) is a collection of theory, techniques and algorithms used for the analysis and processing of non-stationary signals, as found in a wide range of applications including telecommunications, radar, and biomedical engineering. This book gives the university researcher and R&D engineer insights into how to use TFSAP methods to develop and implement the engineering application systems they require. New to this edition: - New sections on Efficient and Fast Algorithms; a "Getting Started" chapter enabling readers to start using the algorithms on simulated and real examples with the TFSAP toolbox, compare the results with the ones presented in the book and then insert the algorithms in their own applications and adapt them as needed. - Two new chapters and twenty three new sections, including updated references. - New topics including: efficient algorithms for optimal TFDs (with source code), the enhanced spectrogram, time-frequency modelling, more mathematical foundations, the relationships between QTFDs and Wavelet Transforms, new advanced applications such as cognitive radio, watermarking, noise reduction in the time-frequency domain, algorithms for Time-Frequency Image Processing, and Time-Frequency applications in neuroscience (new chapter). - A comprehensive tutorial introduction to Time-Frequency Signal Analysis and Processing (TFSAP), accessible to anyone who has taken a first course in signals - Key advances in theory, methodology and algorithms, are concisely presented by some of the leading authorities on the respective topics - Applications written by leading researchers showing how to use TFSAP methods
| Author | : |
| Publisher | : |
| Total Pages | : 664 |
| Release | : 1993 |
| Genre | : Electro-acoustics |
| ISBN | : |
| Author | : Ravi P. Ramachandran |
| Publisher | : Springer Science & Business Media |
| Total Pages | : 471 |
| Release | : 2012-12-06 |
| Genre | : Technology & Engineering |
| ISBN | : 1461522811 |
The term speech processing refers to the scientific discipline concerned with the analysis and processing of speech signals for getting the best benefit in various practical scenarios. These different practical scenarios correspond to a large variety of applications of speech processing research. Examples of some applications include enhancement, coding, synthesis, recognition and speaker recognition. A very rapid growth, particularly during the past ten years, has resulted due to the efforts of many leading scientists. The ideal aim is to develop algorithms for a certain task that maximize performance, are computationally feasible and are robust to a wide class of conditions. The purpose of this book is to provide a cohesive collection of articles that describe recent advances in various branches of speech processing. The main focus is in describing specific research directions through a detailed analysis and review of both the theoretical and practical settings. The intended audience includes graduate students who are embarking on speech research as well as the experienced researcher already working in the field. For graduate students taking a course, this book serves as a supplement to the course material. As the student focuses on a particular topic, the corresponding set of articles in this book will serve as an initiation through exposure to research issues and by providing an extensive reference list to commence a literature survey. Expe rienced researchers can utilize this book as a reference guide and can expand their horizons in this rather broad area.
| Author | : Raghunath S. Holambe |
| Publisher | : Springer Science & Business Media |
| Total Pages | : 109 |
| Release | : 2012-02-21 |
| Genre | : Technology & Engineering |
| ISBN | : 1461415055 |
Advances in Non-Linear Modeling for Speech Processing includes advanced topics in non-linear estimation and modeling techniques along with their applications to speaker recognition. Non-linear aeroacoustic modeling approach is used to estimate the important fine-structure speech events, which are not revealed by the short time Fourier transform (STFT). This aeroacostic modeling approach provides the impetus for the high resolution Teager energy operator (TEO). This operator is characterized by a time resolution that can track rapid signal energy changes within a glottal cycle. The cepstral features like linear prediction cepstral coefficients (LPCC) and mel frequency cepstral coefficients (MFCC) are computed from the magnitude spectrum of the speech frame and the phase spectra is neglected. To overcome the problem of neglecting the phase spectra, the speech production system can be represented as an amplitude modulation-frequency modulation (AM-FM) model. To demodulate the speech signal, to estimation the amplitude envelope and instantaneous frequency components, the energy separation algorithm (ESA) and the Hilbert transform demodulation (HTD) algorithm are discussed. Different features derived using above non-linear modeling techniques are used to develop a speaker identification system. Finally, it is shown that, the fusion of speech production and speech perception mechanisms can lead to a robust feature set.
| Author | : Lajos Hanzo |
| Publisher | : John Wiley & Sons |
| Total Pages | : 880 |
| Release | : 2008-06-05 |
| Genre | : Technology & Engineering |
| ISBN | : 9780470516027 |
Voice communications remains the most important facet of mobile radio services, which may be delivered over conventional fixed links, the Internet or wireless channels. This all-encompassing volume reports on the entire 50-year history of voice compression, on recent audio compression techniques and the protection as well as transmission of these signals in hostile wireless propagation environments. Audio and Voice Compression for Wireless and Wireline Communications, Second Edition is divided into four parts with Part I covering the basics, while Part II outlines the design of analysis-by-synthesis coding, including a 100-page chapter on virtually all existing standardised speech codecs. The focus of Part III is on wideband and audio coding as well as transmission. Finally, Part IV concludes the book with a range of very low rate encoding techniques, scanning a range of research-oriented topics. Fully updated and revised second edition of “Voice Compression and Communications”, expanded to cover Audio features Includes two new chapters, on narrowband and wideband AMR coding, and MPEG audio coding Addresses the new developments in the field of wideband speech and audio compression Covers compression, error resilience and error correction coding, as well as transmission aspects, including cutting-edge turbo transceivers Presents both the historic and current view of speech compression and communications. Covering fundamental concepts in a non-mathematical way before moving to detailed discussions of theoretical principles, future concepts and solutions to various specific wireless voice communication problems, this book will appeal to both advanced readers and those with a background knowledge of signal processing and communications.
| Author | : Curtis Roads |
| Publisher | : Routledge |
| Total Pages | : 493 |
| Release | : 2013-12-19 |
| Genre | : Music |
| ISBN | : 1134379706 |
Compiled by an international array of musical and technical specialists, this book deals with some of the most important topics in modern musical signal processing. Beginning with basic concepts, and leading to advanced applications, it covers such essential areas as sound synthesis (including detailed studies of physical modelling and granular synthesis) ,control signal synthesis, sound transformation (including convolution), analysis/resynthesis (phase vocodor, wavelets, analysis by chaotic functions), object-oriented and artificial intelligence representations, musical interfaces and the integration of signal processing techniques in concert performance.
| Author | : Jean-Claude Junqua |
| Publisher | : Springer Science & Business Media |
| Total Pages | : 277 |
| Release | : 2013-03-09 |
| Genre | : Language Arts & Disciplines |
| ISBN | : 9401597197 |
In this book we address robustness issues at the speech recognition and natural language parsing levels, with a focus on feature extraction and noise robust recognition, adaptive systems, language modeling, parsing, and natural language understanding. This book attempts to give a clear overview of the main technologies used in language and speech processing, along with an extensive bibliography to enable topics of interest to be pursued further. It also brings together speech and language technologies often considered separately. Robustness in Language and Speech Technology serves as a valuable reference and although not intended as a formal university textbook, contains some material that can be used for a course at the graduate or undergraduate level.
| Author | : |
| Publisher | : |
| Total Pages | : 456 |
| Release | : 1999 |
| Genre | : Prosthesis |
| ISBN | : |
