Vary / Martin Digital Speech Transmission

Enhancement, Coding and Error Concealment
1. Auflage 2006
ISBN: 978-0-470-03175-9
Verlag: John Wiley & Sons
Format: PDF
Kopierschutz: Adobe DRM (»Systemvoraussetzungen)

E-Book, Englisch, 644 Seiten, E-Book

The enormous advances in digital signal processing (DSP) technologyhave contributed to the wide dissemination and success of speechcommunication devices - be it GSM and UMTS mobile telephones,digital hearing aids, or human-machine interfaces. Digital speechtransmission techniques play an important role in theseapplications, all the more because high quality speech transmissionremains essential in all current and next generation communicationnetworks.
Enhancement, coding and error concealment techniques improve thetransmitted speech signal at all stages of the transmission chain,from the acoustic front-end to the sound reproduction at thereceiver. Advanced speech processing algorithms help tomitigate a number of physical and technological limitations such asbackground noise, bandwidth restrictions, shortage of radiofrequencies, 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 speechcommunication technology. The authors give a solid, accessibleoverview of
* fundamentals of speech signal processing
* speech coding, including new speech coders for GSM andUMTS
* 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.

Autoren/Hrsg.

Vary, Peter

Martin, Rainer

Inhaltsverzeichnis

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.
A Codec Standards.
B Speech Quality Assessment.
Bibliography.