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Digital Speech Transmission Enhancement, Coding and Error Concealment [Other digital carrier]

(Ruhr-Universitat Bochum, Germany), (RWTH Aachen University, Germany)
  • Formaat: Other digital carrier, 644 pages, kõrgus x laius x paksus: 251x179x40 mm, kaal: 1262 g
  • Ilmumisaeg: 30-Jun-2006
  • Kirjastus: John Wiley & Sons Ltd
  • ISBN-10: 0470031743
  • ISBN-13: 9780470031742
Digital Speech Transmission  Enhancement, Coding and Error Concealment
  • Formaat: Other digital carrier, 644 pages, kõrgus x laius x paksus: 251x179x40 mm, kaal: 1262 g
  • Ilmumisaeg: 30-Jun-2006
  • Kirjastus: John Wiley & Sons Ltd
  • ISBN-10: 0470031743
  • ISBN-13: 9780470031742
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.
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.