Peter Vary, Rainer Martin

Digital Speech Transmission

Enhancement, Coding and Error Concealment

• Gebundenes Buch

Produktbeschreibung

Robust speech transmission is primarily concerned with issues of spoken language and--due to the progress made in signal processing technology i.e. with the availablilty of powerful, monolithically integrated signal processors--speech signal transmission has developed an important practical significance. This book emphasizes issues of speech coding (data compression), the improvement of distorted speech signals, speech and speaker recognition, and speech synthesis (conversion of text into spoken language). One main area of application which pushes this development ahead is that of digital mobile radio communication.

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The enormous advances in digital signal processing (DSP) technology have contributed to the wide dissemination and success of speech communication devices - be it GSM and UMTS mobile telephones, digital hearing aids, or human-machine interfaces. Digital speech transmission techniques play an important role in these applications, all the more because high quality speech transmission remains essential in all current and next generation communication networks.

Enhancement, coding and error concealment techniques improve the transmitted speech signal at all stages of the transmission chain, from the acoustic front-end to the sound reproduction at the receiver. Advanced speech processing algorithms help to mitigate a number of physical and technological limitations such as background noise, bandwidth restrictions, shortage of radio frequencies, and transmission errors.

Digital Speech Transmission provides a single-source, comprehensive guide to the fundamental issues, algorithms, standards, and trends in speech signal processing and speech communication technology. The authors give a solid, accessible overview of
_ fundamentals of speech signal processing
_ speech coding, including new speech coders for GSM and UMTS
_ error concealment by soft decoding
_ artificial bandwidth extension of speech signals
_ single and multi-channel noise reduction
_ acoustic echo cancellation

This text is an invaluable resource for engineers, researchers, academics, and graduate students in the areas of communications, electrical engineering, and information technology.

Produktdetails

• Verlag: Wiley & Sons
• 1. Auflage
• Seitenzahl: 656
• Erscheinungstermin: 3. März 2006
• Englisch
• Abmessung: 254mm x 176mm x 43mm
• Gewicht: 1330g
• ISBN-13: 9780471560180
• ISBN-10: 0471560189
• Artikelnr.: 12246593

Autorenporträt

Peter Vary is the author of Digital Speech Transmission: Enhancement, Coding and Error Concealment, published by Wiley. Rainer Martin is the author of Digital Speech Transmission: Enhancement, Coding and Error Concealment, published by Wiley.

Inhaltsangabe

1 Introduction. 2 Models of Speech Production and Hearing. 2.1 Organs of
Speech Production. 2.2 Characteristics of Speech Signals. 2.3 Model of
Speech Production. 2.4 Anatomy of Hearing. 2.5 Performance of the Auditory
Organs. Bibliography. 3 Spectral Transformations. 3.1 Fourier Transform of
Continuous Signals. 3.2 Fourier Transform of Discrete Signals. 3.3 Linear
Shift Invariant Systems. 3.4 The z-Transform. 3.5 The Discrete Fourier
Transform. 3.6 Fast Convolution. 3.7 Cepstral Analysis. Bibliography. 4
Filter Banks for Spectral Analysis and Synthesis. 4.1 Spectral Analysis
Using Narrow-Band Filters. 4.2 Polyphase Network Filter Banks. 4.3
QuadratureMirror Filter Banks. Bibliography. 5 Stochastic Signals and
Estimation. 5.1 Basic Concepts. 5.2 Expectations andMoments. 5.3 Bivariate
Statistics. 5.4 Probability and Information. 5.5 Multivariate Statistics.
5.6 Stochastic Processes. 5.7 Estimation of Statistical Quantities by Time
Averages. 5.8 Power Spectral Densities. 5.9 Estimation of the Power
Spectral Density. 5.10 Statistical Properties of Speech Signals. 5.11
Statistical Properties of DFT Coe.cients. 5.12 Optimal Estimation.
Bibliography. 6 Linear Prediction. 6.1 Vocal TractModels and
Short-TermPrediction. 6.2 Optimal Prediction Coe.cients for Stationary
Signals. 6.3 Predictor Adaptation. 6.4 Long-TermPrediction. Bibliography. 7
Quantization. 7.1 Analog Samples and Digital Presentation. 7.2 Uniform
Quantization. 7.3 Non-uniformQuantization. 7.4 OptimalQuantization. 7.5
Adaptive Quantization. 7.6 Vector Quantization. 7.6.1 Principle.
Bibliography. 8 Speech Coding. 8.1 Classi.cation of Speech Coding
Algorithms. 8.2 Model-Based Predictive Coding. 8.3 Di.erentialWaveform
Coding. 8.4 Parametric Coding. 8.5 Hybrid Coding. 8.6 Adaptive
Post.ltering. Bibliography. 9 Error Concealment and Softbit Decoding. 9.1
Hardbit Source Decoding. 9.2 Conventional Error Concealment. 9.3 Softbits
and L-Values. 9.4 Softbit Source Decoding (SD). 9.5 Application toModel
Parameters. 9.6 Further Improvements. Bibliography. 10 Bandwidth Extension
of Speech Signals (BWE). 10.1 Narrowband versusWideband Telephony. 10.2
Speech Coding with Integrated BWE. 10.3 BWE without Auxiliary Transmission.
Bibliography. 11 Single and Dual Channel Noise Reduction. 11.1
Introduction. 11.2 LinearMMSE Estimators. 11.3 Speech Enhancement in the
DFT Domain. 11.4 Optimal Non-Linear Estimators. 11.5 Joint Optimum
Detection and Estimation of Speech. 11.6 Computation of Likelihood Ratios.
11.7 Estimation of the A Priory Probability of Speech Presence. 11.8 VAD
and Noise Estimation Techniques. 11.9 Dual-Channel Noise Reduction.
Bibliography. 12 Multi-Channel Noise Reduction. 12.1 Introduction. 12.2
Spatial Sampling of Sound Fields. 12.3 Beamforming. 12.4
PerformanceMeasures and Spatial Aliasing. 12.5 Design of Fixed Beamformers.
12.6 Adaptive Beamformers. Bibliography. 13 Acoustic Echo Control. 13.1 The
Echo Control Problem. 13.2 Evaluation Criteria. 13.3 TheWiener Solution.
13.4 The LMS and NLMS Algorithm. 13.5 Convergence Analysis and Control of
the LMS Algorithm. 13.6 Geometric Projection Interpretation of the NLMS
Algorithm. 13.7 The A.ne Projection Algorithm. 13.8 Least-Squares and
Recursive Least-Squares Algorithms. 13.9 Block Processing and
Frequency-Domain Adaptive Filters. 13.9.1 Block LMS Algorithm. 13.10
Additional Measures for Echo Control. 13.11 Stereophonic Acoustic Echo
Control. Bibliography. A Codec Standards. B Speech Quality Assessment.
Bibliography.