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9780470062012

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

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  • ISBN13:

    9780470062012

  • ISBN10:

    0470062010

  • Edition: 1st
  • Format: Hardcover
  • Copyright: 2007-11-12
  • Publisher: WILEY
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Summary

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.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.

Author Biography

Saeed Vaseghi is Professor of Communications and Signal Processing at Brunel University’s 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.

Table of Contents

Prefacep. xiii
Acknowledgementp. xvii
Symbolsp. xix
Abbreviationsp. xxiii
Basic Digital Signal Processingp. 1
Introductionp. 3
Signals and Informationp. 3
Signal Processing Methodsp. 5
Applications of Digital Signal Processingp. 8
Summaryp. 23
Fourier Analysis and Synthesisp. 25
Introductionp. 25
Fourier Series: Representation of Periodic Signalsp. 27
Fourier Transform: Representation of Nonperiodic Signalsp. 33
Discrete Fourier Transformp. 48
Short-Time Fourier Transformp. 57
Fast Fourier Transform (FFT)p. 59
2-D Discrete Fourier Transform (2-D DFT)p. 65
Discrete Cosine Transform (DCT)p. 66
Some Applications of the Fourier Transformp. 68
Summaryp. 74
z-Transformp. 79
Introductionp. 79
Derivation of the z-Transformp. 81
The z-Plane and the Unit Circlep. 83
Properties of z-Transformp. 88
z-Transfer Function, Poles (Resonance) and Zeros (Anti-resonance)p. 91
z-Transform of Analysis of Exponential Transient Signalsp. 100
Inverse z-Transformp. 104
Summaryp. 106
Digital Filtersp. 111
Introductionp. 111
Linear Time-Invariant Digital Filtersp. 113
Recursive and Non-Recursive Filtersp. 115
Filtering Operation: Sum of Vector Products, A Comparison of Convolution and Correlationp. 117
Filter Structures: Direct, Cascade and Parallel Formsp. 119
Linear Phase FIR Filtersp. 122
Design of Digital FIR Filter-banksp. 136
Quadrature Mirror Sub-band Filtersp. 139
Design of Infinite Impulse Response (IIR) Filters by Pole-zero Placementsp. 145
Issues in the Design and Implementation of a Digital Filterp. 148
Summaryp. 148
Sampling and Quantisationp. 155
Introductionp. 155
Sampling a Continuous-Time Signalp. 158
Quantisationp. 162
Sampling Rate Conversion: Interpolation and Decimationp. 166
Summaryp. 171
Model-Based Signal Processingp. 173
Information Theory and Probability Modelsp. 175
Introduction: Probability and Information Modelsp. 176
Random Processesp. 177
Probability Models of Random Signalsp. 182
Information Modelsp. 189
Stationary and Non-Stationary Random Processesp. 199
Statistics (Expected Values) of a Random Processp. 202
Some Useful Practical Classes of Random Processesp. 212
Transformation of a Random Processp. 225
Search Engines: Citation Rankingp. 230
Summaryp. 231
Bayesian Inferencep. 233
Bayesian Estimation Theory: Basic Definitionsp. 233
Bayesian Estimationp. 242
Expectation Maximisation Methodp. 255
Cramer-Rao Bound on the Minimum Estimator Variancep. 257
Design of Gaussian Mixture Models (GMM)p. 260
Bayesian Classificationp. 263
Modelling the Space of a Random Processp. 270
Summaryp. 273
Least Square Error, Wiener-Kolmogorov Filtersp. 275
Least Square Error Estimation: Wiener-Kolmogorov Filterp. 275
Block-Data Formulation of the Wiener Filterp. 280
Interpretation of Wiener Filter as Projection in Vector Spacep. 282
Analysis of the Least Mean Square Error Signalp. 284
Formulation of Wiener Filters in the Frequency Domainp. 285
Some Applications of Wiener Filtersp. 286
Implementation of Wiener Filtersp. 292
Summaryp. 294
Adaptive Filters: Kalman, RLS, LMSp. 297
Introductionp. 297
State-Space Kalman Filtersp. 299
Sample Adaptive Filtersp. 307
Recursive Least Square (RLS) Adaptive Filtersp. 309
The Steepest-Descent Methodp. 313
LMS Filterp. 317
Summaryp. 321
Linear Prediction Modelsp. 323
Linear Prediction Codingp. 323
Forward, Backward and Lattice Predictorsp. 332
Short-Term and Long-Term Predictorsp. 339
MAP Estimation of Predictor Coefficientsp. 341
Formant-Tracking LP Modelsp. 343
Sub-Band Linear Prediction Modelp. 344
Signal Restoration Using Linear Prediction Modelsp. 345
Summaryp. 350
Hidden Markov Modelsp. 353
Statistical Models for Non-Stationary Processesp. 353
Hidden Markov Modelsp. 355
Training Hidden Markov Modelsp. 361
Decoding Signals Using Hidden Markov Modelsp. 367
HMM in DNA and Protein Sequencesp. 371
HMMs for Modelling Speech and Noisep. 372
Summaryp. 378
Eigenvector Analysis, Principal Component Analysis and Independent Component Analysisp. 381
Introduction - Linear Systems and Eigenanalysisp. 382
Eigenvectors and Eigenvaluesp. 386
Principal Component Analysis (PCA)p. 389
Independent Component Analysisp. 393
Summaryp. 412
Applications of Digital Signal Processing to Speech, Music and Telecommunicationsp. 415
Music Signal Processing and Auditory Perceptionp. 417
Introductionp. 418
Musical Notes, Intervals and Scalesp. 418
Musical Instrumentsp. 426
Review of Basic Physics of Soundsp. 439
Music Signal Features and Modelsp. 447
Anatomy of the Ear and the Hearing Processp. 451
Psychoacoustics of Hearingp. 462
Music Coding (Compression)p. 471
High Quality Audio Coding: MPEG Audio Layer-3 (MP3)p. 475
Stereo Music Codingp. 478
Summaryp. 480
Speech Processingp. 483
Speech Communicationp. 483
Acoustic Theory of Speech: The Source-filter Modelp. 484
Speech Models and Featuresp. 490
Linear Prediction Models of Speechp. 491
Harmonic Plus Noise Model of Speechp. 492
Fundamental Frequency (Pitch) Informationp. 496
Speech Codingp. 500
Speech Recognitionp. 510
Summaryp. 525
Speech Enhancementp. 527
Introductionp. 528
Single-Input Speech Enhancement Methodsp. 528
Speech Bandwidth Extension - Spectral Extrapolationp. 547
Interpolation of Lost Speech Segments - Packet Loss Concealmentp. 553
Multi-Input Speech Enhancement Methodsp. 562
Speech Distortion Measurementsp. 565
Summaryp. 569
Echo Cancellationp. 573
Introduction: Acoustic and Hybrid Echop. 573
Telephone Line Hybrid Echop. 575
Hybrid (Telephone Line) Echo Suppressionp. 577
Adaptive Echo Cancellationp. 578
Acoustic Echop. 581
Sub-Band Acoustic Echo Cancellationp. 584
Echo Cancellation with Linear Prediction Pre-whiteningp. 585
Multi-Input Multi-Output Echo Cancellationp. 586
Summaryp. 589
Channel Equalisation and Blind Deconvolutionp. 591
Introductionp. 591
Blind Equalisation Using Channel Input Power Spectrump. 598
Equalisation Based on Linear Prediction Modelsp. 601
Bayesian Blind Deconvolution and Equalisationp. 603
Blind Equalisation for Digital Communication Channelsp. 611
Equalisation Based on Higher-Order Statisticsp. 616
Summaryp. 623
Signal Processing in Mobile Communicationp. 625
Introduction to Cellular Communicationp. 625
Communication Signal Processing in Mobile Systemsp. 631
Capacity, Noise, and Spectral Efficiencyp. 632
Multi-path and Fading in Mobile Communicationp. 634
Smart Antennas - Space-Time Signal Processingp. 639
Summaryp. 642
Indexp. 643
Table of Contents provided by Ingram. All Rights Reserved.

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