Multimedia Signal Processing: Theory and Applications in Speech ...

Brochure More information from http://www.researchandmarkets.com/reports/569388/

Multimedia Signal Processing: Theory and Applications in Speech, Music and Communications Description:

Multimedia Signal Processing is a comprehensive and accessible text to the theory and applications of digital signal processing (DSP). The applications of DSP are pervasive and include multimedia systems, cellular communication, adaptive network management, radar, pattern recognition, medical signal processing, financial data forecasting, artificial intelligence, decision making, control systems and search engines. This book is organised in to three major parts making it a coherent and structured presentation of the theory and applications of digital signal processing. A range of important topics are covered in basic signal processing, model-based statistical signal processing and their applications. The aim of this book is to provide a coherent and structured presentation of the theory and applications of statistical signal processing in three sections: Part 1: Basic Digital Signal Processing gives an introduction to the topic, discussing sampling and quantization, Fourier analysis and synthesis, Z-transform, and digital filters. Part 2: Model-based Signal Processing covers probability and information models, Bayesian inference, Wiener filter, adaptive filters, linear prediction hidden Markov models and independent component analysis. Part 3: Applications of Signal Processing in Speech, Music and Telecommunications explains the topics of speech and music processing, echo cancellation, deconvolution and channel equalization, and mobile communication signal processing. -Covers music signal processing, explains the anatomy and psychoacoustics of hearing and the design of MP3 music coder -Examines speech processing technology including speech models, speech coding for mobile phones and speech recognition -Covers single-input and multiple-inputs denoising methods, bandwidth extension and the recovery of lost speech packets in applications such as voice over IP (VoIP) -Illustrated throughout, including numerous solved problems, Matlab experiments and demonstrations -Companion website features Matlab and C++ programs with electronic copies of all figures. This book is ideal for researchers, postgraduates and senior undergraduates in the fields of digital signal processing, telecommunications and statistical data analysis. It will also be a valuable text to professional engineers in telecommunications and audio and signal processing industries. About the author: Saeed Vaseghi is Professor of Communications and Signal Processing at Brunel Universitys Department of Electronics and Computer Engineering and is Group Leader for the Communications & Multimedia Signal Processing Group. Previously, Saeed obtained a first in Electrical and Electronics Engineering from Newcastle University, and a PhD in Digital Signal Processing from Cambridge University. His PhD in noisy signal restoration led to establishment of CEDAR, the world's leading system for restoration of audio signals. Saeed also held a British Telecom lectureship at UEA Norwich, and a readership at Queen's University of Belfast before his move to Brunel

Contents:

Preface Acknowledgement

Symbols Abbreviations Part I Basic Digital Signal Processing 1 Introduction 1.1 1.2 1.3 1.4

Signals and Information Signal Processing Methods Applications of Digital Signal Processing Summary

2 Fourier Analysis and Synthesis 2.1 Introduction 2.2 Fourier Series: Representation of Periodic Signals 2.3 Fourier Transform: Representation of Nonperiodic Signals 2.4 Discrete Fourier Transform 2.5 Short-Time Fourier Transform 2.6 Fast Fourier Transform (FFT) 2.7 2-D Discrete Fourier Transform (2-D DFT) 2.8 Discrete Cosine Transform (DCT) 2.9 Some Applications of the Fourier Transform 2.10 Summary 3 z-Transform 3.1 3.2 3.3 3.4 3.5 3.6 3.7 3.8

Introduction Derivation of the z-Transform The z-Plane and the Unit Circle Properties of z-Transform z-Transfer Function, Poles (Resonance) and Zeros (Anti-resonance) z-Transform of Analysis of Exponential Transient Signals Inverse z-Transform Summary

4 Digital Filters 4.1 Introduction 4.2 Linear Time-Invariant Digital Filters 4.3 Recursive and Non-Recursive Filters 4.4 Filtering Operation: Sum of Vector Products, A Comparison of Convolution and Correlation 4.5 Filter Structures: Direct, Cascade and Parallel Forms 4.6 Linear Phase FIR Filters 4.7 Design of Digital FIR Filter-banks 4.8 Quadrature Mirror Sub-band Filters 4.9 Design of Infinite Impulse Response (IIR) Filters by Pole–zero Placements 4.10 Issues in the Design and Implementation of a Digital Filter 4.11 Summary 5 Sampling and Quantisation 5.1 Introduction 5.2 Sampling a Continuous-Time Signal 5.3 Quantisation 5.4 Sampling Rate Conversion: Interpolation and Decimation 5.5 Summary Part II Model-Based Signal Processing 6 Information Theory and Probability Models 6.1 Introduction: Probability and Information Models

6.2 Random Processes 6.3 Probability Models of Random Signals 6.4 Information Models 6.5 Stationary and Non-Stationary Random Processes 6.6 Statistics (Expected Values) of a Random Process 6.7 Some Useful Practical Classes of Random Processes 6.8 Transformation of a Random Process 6.9 Search Engines: Citation Ranking 6.10 Summary 7 Bayesian Inference 7.1 7.2 7.3 7.4 7.5 7.6 7.7 7.8

Bayesian Estimation Theory: Basic Definitions Bayesian Estimation Expectation Maximisation Method Cramer–Rao Bound on the Minimum Estimator Variance Design of Gaussian Mixture Models (GMM) Bayesian Classification Modelling the Space of a Random Process Summary

8 Least Square Error, Wiener–Kolmogorov Filters 8.1 8.2 8.3 8.4 8.5 8.6 8.7 8.8

Least Square Error Estimation: Wiener–Kolmogorov Filter Block-Data Formulation of the Wiener Filter Interpretation of Wiener Filter as Projection in Vector Space Analysis of the Least Mean Square Error Signal Formulation of Wiener Filters in the Frequency Domain Some Applications of Wiener Filters Implementation of Wiener Filters Summary

9 Adaptive Filters: Kalman, RLS, LMS 9.1 9.2 9.3 9.4 9.5 9.6 9.7

Introduction State-Space Kalman Filters Sample Adaptive Filters Recursive Least Square (RLS) Adaptive Filters The Steepest-Descent Method LMS Filter Summary

10 Linear Prediction Models 10.1 10.2 10.3 10.4 10.5 10.6 10.7 10.8

Linear Prediction Coding Forward, Backward and Lattice Predictors Short-Term and Long-Term Predictors MAP Estimation of Predictor Coefficients Formant-Tracking LP Models Sub-Band Linear Prediction Model Signal Restoration Using Linear Prediction Models Summary

11 Hidden Markov Models 11.1 11.2 11.3 11.4 11.5 11.6 11.7

Statistical Models for Non-Stationary Processes Hidden Markov Models Training Hidden Markov Models Decoding Signals Using Hidden Markov Models HMM in DNA and Protein Sequences HMMs for Modelling Speech and Noise Summary

12 Eigenvector Analysis, Principal Component Analysis and Independent Component Analysis 12.1 12.2 12.3 12.4 12.5

Introduction – Linear Systems and Eigenanalysis Eigenvectors and Eigenvalues Principal Component Analysis (PCA) Independent Component Analysis Summary

Part III Applications of Digital Signal Processing to Speech, Music and Telecommunications 13 Music Signal Processing and Auditory Perception 13.1 Introduction 13.2 Musical Notes, Intervals and Scales 13.3 Musical Instruments 13.4 Review of Basic Physics of Sounds 13.5 Music Signal Features and Models 13.6 Anatomy of the Ear and the Hearing Process 13.7 Psychoacoustics of Hearing 13.8 Music Coding (Compression) 13.9 High Quality Audio Coding: MPEG Audio Layer-3 (MP3) 13.10 Stereo Music Coding 13.11 Summary 14 Speech Processing 14.1 14.2 14.3 14.4 14.5 14.6 14.7 14.8 14.9

Speech Communication Acoustic Theory of Speech: The Source–filter Model Speech Models and Features Linear Prediction Models of Speech Harmonic Plus Noise Model of Speech Fundamental Frequency (Pitch) Information Speech Coding Speech Recognition Summary

15 Speech Enhancement 15.1 15.2 15.3 15.4 15.5 15.6 15.7

Introduction Single-Input Speech Enhancement Methods Speech Bandwidth Extension – Spectral Extrapolation Interpolation of Lost Speech Segments – Packet Loss Concealment Multi-Input Speech Enhancement Methods Speech Distortion Measurements Summary

16 Echo Cancellation 16.1 16.2 16.3 16.4 16.5 16.6 16.7 16.8 16.9

Introduction: Acoustic and Hybrid Echo Telephone Line Hybrid Echo Hybrid (Telephone Line) Echo Suppression Adaptive Echo Cancellation Acoustic Echo Sub-Band Acoustic Echo Cancellation Echo Cancellation with Linear Prediction Pre-whitening Multi-Input Multi-Output Echo Cancellation Summary

17 Channel Equalisation and Blind Deconvolution 17.1 Introduction 17.2 Blind Equalisation Using Channel Input Power Spectrum 17.3 Equalisation Based on Linear Prediction Models

17.4 17.5 17.6 17.7

Bayesian Blind Deconvolution and Equalisation Blind Equalisation for Digital Communication Channels Equalisation Based on Higher-Order Statistics Summary

18 Signal Processing in Mobile Communication 18.1 18.2 18.3 18.4 18.5 18.6

Introduction to Cellular Communication Communication Signal Processing in Mobile Systems Capacity, Noise, and Spectral Efficiency Multi-path and Fading in Mobile Communication Smart Antennas – Space–Time Signal Processing Summary

Index

Ordering:

Order Online - http://www.researchandmarkets.com/reports/569388/ Order by Fax - using the form below Order by Post - print the order form below and sent to Research and Markets, Guinness Centre, Taylors Lane, Dublin 8, Ireland.

Page 1 of 2 Fax Order Form To place an order via fax simply print this form, fill in the information below and fax the completed form to 646-6071907 (from USA) or +353-1-481-1716 (from Rest of World). If you have any questions please visit http://www.researchandmarkets.com/contact/

Order Information Please verify that the product information is correct. Product Name:

Multimedia Signal Processing: Theory and Applications in Speech, Music and Communications

Web Address:

http://www.researchandmarkets.com/reports/569388/

Office Code:

OC8HJNQTTPPOY

Product Format Please select the product format and quantity you require: Quantity Hard Copy:

EURO €98.00 + Euro €25.00 Shipping/Handling

Contact Information Please enter all the information below in BLOCK CAPITALS Title: First Name:

Mr

Mrs

Dr

Miss

Ms

Last Name:

Email Address: * Job Title: Organisation: Address: City: Postal / Zip Code: Country: Phone Number: Fax Number: * Please refrain from using free email accounts when ordering (e.g. Yahoo, Hotmail, AOL)

Prof

Page 2 of 2 Payment Information Please indicate the payment method you would like to use by selecting the appropriate box. Pay by credit card:

American Express Diners Club Master Card Visa Cardholder's Name Cardholder's Signature Expiry Date Card Number CVV Number Issue Date (for Diners Club only)

Pay by check:

Please post the check, accompanied by this form, to: Research and Markets, Guinness Center, Taylors Lane, Dublin 8, Ireland.

Pay by wire transfer:

Please transfer funds to: Account number

833 130 83

Sort code

98-53-30

Swift code

ULSBIE2D

IBAN number

IE78ULSB98533083313083

Bank Address

Ulster Bank, 27-35 Main Street, Blackrock, Co. Dublin, Ireland.

If you have a Marketing Code please enter it below: Marketing Code: Please note that by ordering from Research and Markets you are agreeing to our Terms and Conditions at http://www.researchandmarkets.com/info/terms.asp

Please fax this form to: (646) 607-1907 or (646) 964-6609 - From USA +353 1 481 1716 or +353 1 653 1571 - From Rest of World