Record:   Prev Next
作者 Martin, Rainer
書名 Advances in Digital Speech Transmission
出版項 New York : John Wiley & Sons, Incorporated, 2008
©2008
國際標準書號 9780470727171 (electronic bk.)
9780470517390
book jacket
版本 1st ed
說明 1 online resource (573 pages)
text txt rdacontent
computer c rdamedia
online resource cr rdacarrier
附註 Intro -- Advances in Digital Speech Transmission -- Contents -- List of Contributors -- Preface -- 1 Introduction -- I Speech Quality Assessment -- 2 Speech-Transmission Quality: Aspects and Assessment for Wideband vs. Narrowband Signals -- 2.1 Introduction -- 2.2 Speech Signals -- 2.3 Telephone-Band Speech Signals -- 2.3.1 Narrowband Speech Intelligibility -- 2.3.2 Narrowband Speech-Sound Quality -- 2.4 Wideband Speech Signals -- 2.4.1 Wideband-Speech Intelligibility and Sound Quality -- 2.4.2 Wideband Speech Transmission and Processing -- 2.5 Speech-Quality Assessment -- 2.5.1 Auditory Quality Determination -- 2.5.2 Instrumental Quality Determination -- 2.6 Wideband Speech-Quality Assessment -- 2.6.1 Integral Quality Determination -- 2.6.2 Attribute-Oriented Quality Determination -- 2.6.3 Combined Direct and Attribute-Based Total Quality Determination -- 2.7 Concluding Remarks -- Bibliography -- 3 Parametric Quality Assessment of Narrowband Speech in Mobile Communication Systems -- 3.1 Introduction -- 3.1.1 Subjective Listening Tests and Classes of Objective Measures -- 3.1.2 Overview of Objective Speech Quality Measures -- 3.1.3 Development of Parametric Models -- 3.2 Simulations of GSM and UMTS Speech Transmissions -- 3.2.1 Simulation Environment -- 3.2.2 GSM Transmission Parameters -- 3.2.3 UMTS Transmission Parameters -- 3.3 Speech Quality Measures based on Transmission Parameters -- 3.3.1 Correlation Analysis -- 3.3.2 Parametric Speech Quality Measures -- 3.4 Discussion and Conclusions -- Bibliography -- II Adaptive Algorithms in Acoustic Signal Processing -- 4 Kalman Filtering in Acoustic Echo Control: A Smooth Ride on a Rocky Road -- 4.1 Introduction -- 4.1.1 Adaptive Filter Structures for Acoustic Echo Control -- 4.1.2 Control of Adaptive Filters -- 4.1.3 Open Problems / Organization of this Chapter
4.2 A Comprehensive Theory of Acoustic Echo Control -- 4.2.1 Stochastic Modeling of the Echo Path -- 4.2.2 Minimum Mean-Square Error (MMSE) Solution -- 4.2.3 MMSE Processor in the Gaussian Case -- 4.3 The Kalman Filter for Conditional Mean and Covariance Estimation -- 4.3.1 Linear Echo Path Model in DFT-Matrix Form -- 4.3.2 Markov Model of the Time-Varying Echo Path -- 4.3.3 Exact Kalman Filter for the Conditional Mean and Covariance -- 4.3.4 Diagonalization of the Kalman Filter -- 4.3.5 Unification of Adaptive Filtering and Adaptation Control -- 4.4 AEC Performance of the Frequency-Domain Adaptive Kalman Filter -- 4.5 Discussion and Conclusions -- Bibliography -- 5 Noise Reduction - Statistical Analysis and Control of Musical Noise -- 5.1 Introduction -- 5.2 Speech Enhancement in the DFT Domain -- 5.2.1 Optimal Speech Estimators -- 5.3 Measurement and Assessment of Unnatural Fluctuations -- 5.3.1 Filter Analysis via Approximated Filter Input-Output Characteristics -- 5.3.2 Outlier Statistics -- 5.4 Avoidance of Processing Artifacts -- 5.5 Control of Spectral Fluctuations in the Cepstral Domain -- 5.6 Discussion and Conclusions -- 5.7 Acknowledgements -- 5.8 Appendix -- 5.8.1 Mean a priori SNR for different filter types and low SNR -- 5.8.2 Approximation of the decision-directed approach for low SNR -- Bibliography -- 6 Acoustic Source Localization with Microphone Arrays -- 6.1 Introduction -- 6.2 Signal Model -- 6.2.1 Continuous Time Model -- 6.2.2 Discrete Time Representation -- 6.2.3 Formulation in the Frequency Domain -- 6.2.4 Simplified Model for Localization -- 6.3 Localization Approach Taxonomy -- 6.4 Indirect Localization Approaches -- 6.4.1 Generalized Cross-Correlation (GCC) -- 6.4.2 Adaptive Eigenvalue Decomposition (AED) -- 6.4.3 Information Theoretic Approach to TDOA Estimation -- 6.4.4 Extension to Multiple Microphone Pairs
6.5 Direct Localization Approaches -- 6.5.1 Steered Response Power Beamforming -- 6.5.2 Minimum Mean Square (MMSE) Approach -- 6.5.3 Practical aspects -- 6.5.4 Subspace Based Approaches -- 6.5.5 Maximum Likelihood Estimation (MLE) -- 6.6 Evaluation of Localization Algorithms -- 6.6.1 Performance of the Indirect Methods -- 6.6.2 Performance of the Direct Methods -- 6.6.3 The Two-Source Case -- 6.7 Conclusions -- Bibliography -- 7 Multi-Channel System Identification with Perfect Sequences - Theory and Applications - -- 7.1 Introduction -- 7.2 System Identification with Perfect Sequences -- 7.2.1 Geometric Interpretation of the NLMS Algorithm -- 7.2.2 Optimal Excitation of the NLMS Algorithm -- 7.2.3 Influence of Environmental Noise, Stepsize, and Period -- 7.2.4 Odd-Perfect Sequences -- 7.2.5 Tracking of Time-Variant Systems -- 7.2.6 Complexity -- 7.3 Multi-Channel System Identification -- 7.3.1 The Dual-Channel Case -- 7.3.2 Simulation Results -- 7.3.3 Generalization to the Multi-Channel Case -- 7.4 Applications -- 7.4.1 Simulation of Time-Variant RIRs for Stereophonic Echo Control -- 7.4.2 Acoustic Tube Endoscopy -- 7.5 Discussion and Conclusions -- Bibliography -- III Speech Coding for Heterogeneous Networks -- 8 Embedded Speech Coding: From G.711 to G.729.1 -- 8.1 Introduction -- 8.2 Theory and Tools of Embedded Speech Coding -- 8.2.1 Basic Principles -- 8.2.2 Approximation Theory -- 8.2.3 Hierarchical Vector Quantization Methods -- 8.3 Embedded Speech Coding Methods -- 8.3.1 Embedded DPCM and ADPCM -- 8.3.2 Embedded CELP -- 8.3.3 Embedded Extensions of CELP Coders -- 8.3.4 Embedded Parameter Quantization -- 8.4 Standardized Embedded Speech Coders -- 8.4.1 ITU-T G.711 PCMCodec -- 8.4.2 ITU-T G.727 and G.722 ADPCM Codecs -- 8.4.3 MPEG-4 Scalable Speech Coding -- 8.4.4 Embedded Wideband Coding for VoIP: ITU-T G.729.1
8.5 Network Aspects of Embedded Speech Coding -- 8.5.1 Implementation and Utilization of Scalability -- 8.5.2 Unequal Error Protection and Encryption -- 8.6 Conclusions and Perspectives -- Bibliography -- 9 Backwards Compatible Wideband Telephony -- 9.1 Introduction -- 9.2 From Narrowband Telephony toWideband Telephony -- 9.3 Stand-Alone Bandwidth Extension -- 9.3.1 Estimation of the Wideband Spectral Envelope -- 9.3.2 Extension of the Excitation Signal -- 9.3.3 Performance and State-of-the-Art -- 9.4 Embedded Wideband Coding Using Bandwidth Extension Techniques -- 9.4.1 Transmission of BWE Information -- 9.4.2 Examples of Embedded Wideband Speech Codecs -- 9.4.3 Audio Coding -- 9.5 Combination of Bandwidth Extension with Watermarking -- 9.5.1 Digital Watermarking of Speech Signals -- 9.5.2 Transmission of BWE Information via Watermarking -- 9.5.3 Challenges and Status -- 9.6 Advanced Transmission of Highband Parameters -- 9.6.1 Coding with Side Information -- 9.6.2 Error Concealment with Side Information -- 9.7 Conclusions -- Bibliography -- IV Joint Source-Channel Coding -- 10 Parameter Models and Estimators in Soft Decision Source Decoding -- 10.1 Introduction -- 10.2 Overview to Soft Decision Source Decoding -- 10.2.1 Source Encoding -- 10.2.2 Equivalent Channel -- 10.2.3 Hard Decision and Soft Decision Source Decoding -- 10.3 The Markovian Parameter Model -- 10.3.1 Description of A Priori Knowledge -- 10.3.2 Quantification of Utilizable Residual Redundancy -- 10.3.3 Choice of the Model Order -- 10.4 Basic Extrapolative Estimators -- 10.4.1 Introduction and Simulation Settings -- 10.4.2 Estimators -- 10.4.3 Simulation Results -- 10.5 Joint Extrapolative Estimation of Two Different Parameters -- 10.5.1 Estimators -- 10.5.2 Simulation Results -- 10.6 Extrapolative Estimation with Repeated Parameter Transmission -- 10.6.1 Estimators
10.6.2 Simulation Results -- 10.7 Interpolative Estimation of a Parameter -- 10.7.1 Estimators -- 10.7.2 Simulation Results -- 10.8 Discussion and Conclusions -- Bibliography -- 11 Optimal MMSE Estimation for Vector Sources with Spatially and Temporally Correlated Elements -- 11.1 Introduction -- 11.2 Source Model -- 11.3 Transmission Channel -- 11.4 Optimal MMSE Parameter Estimator -- 11.5 Near-Optimal MMSE Parameter Estimator -- 11.6 Illustrative Comparison -- 11.7 Simulation Results -- 11.8 Conclusions -- Bibliography -- 12 Source Optimized Channel Codes & Source Controlled Channel Decoding -- 12.1 Introduction -- 12.2 The Transmission System Used as Reference -- 12.3 Source Optimized Channel Coding (SOCC) -- 12.3.1 Definition -- 12.3.2 Decoding of Source Optimized Channel Codes -- 12.3.3 Design of Source Optimized Channel Codes -- 12.3.4 Numerical Aspects of SOCC Design -- 12.3.5 Bit Allocation between Source and Channel Coding . -- 12.3.6 Relation to Channel Optimized Vector Quantization -- 12.4 Source Controlled Channel Decoding (SCCD) -- 12.4.1 Channel Coding and Decoding in SCCD -- 12.4.2 A Priori Knowledge in Channel Decoding -- 12.4.3 Channel Decoding Using Intra-Parameter Correlation -- 12.4.4 Channel Decoding Using Inter-Frame Correlation -- 12.4.5 Channel Decoding Using Intra-Parameter and Inter-Frame Correlation -- 12.4.6 Simulation Results -- 12.4.7 Exploiting A Priori Knowledge in Source and/or Channel Decoding -- 12.5 Comparison of SOCC versus SCCD -- 12.6 Conclusions -- Bibliography -- 13 Iterative Source-Channel Decoding & Turbo DeCodulation -- 13.1 Introduction -- 13.2 The Key of the Turbo Principle: Extrinsic Information -- 13.2.1 Terms of Reliability Information -- 13.2.2 Extrinsic Information of Channel Decoding -- 13.2.3 Extrinsic Information of Source Decoding -- 13.2.4 Extrinsic Information of Demodulation
13.2.5 EXIT Charts
Dr. Ing. Rainer Martin is a Professor of Information Technology at Ruhr University and Head of the Institute of Communication Acoustics, Bochum, Germany. His research interests are signal processing for voice communication systems, acoustics, and human-machine interfaces. He has worked on algorithms for noise reduction, acoustic echo cancellation, microphone arrays, and speech recognition. He is coauthor of the book Vary/Martin "Digital Speech Transmission", John Wiley, 2006. Ulrich Heute is a Professor for circuit and system theory at Christian-Albrechts University, Kiel. His research interests include digital signal processing, filters and filter banks, and spectral analysis, with applications in medical, audio, and, especially, speech-signal processing (combined source and channel coding, enhancement, modeling, speaker characterization, and instrumental quality assessment). Christiane Antweiler is a senior scientist at the Institute of Communication Systems and Data Processing of the RWTH Aachen University. Her interests are the design and implementation of digital signal and speech processing algorithms for real-time applications, and her special focus lies on speech coding for cellular mobile radio and the enhancement of digital speech signals. Furthermore she is interested in algorithms for system identification, in adaptive filter theory and in digital signal processing algorithms for medical diagnostics
Description based on publisher supplied metadata and other sources
Electronic reproduction. Ann Arbor, Michigan : ProQuest Ebook Central, 2020. Available via World Wide Web. Access may be limited to ProQuest Ebook Central affiliated libraries
鏈接 Print version: Martin, Rainer Advances in Digital Speech Transmission New York : John Wiley & Sons, Incorporated,c2008 9780470517390
主題 Speech processing systems.;Signal processing -- Digital techniques
Electronic books
Alt Author Heute, Ulrich
Antweiler, Christiane
Heute, Prof Ulrich
Record:   Prev Next