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Biblioteca

MARC Lista (tabellare)
Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Pubbl/distr/stampa Chichester, West Sussex, England ; , : John Wiley & Sons, , c2008
Descrizione fisica 1 online resource (573 p.)
Disciplina 621.39/9
Altri autori (Persone) MartinRainer
HeuteUlrich
AntweilerChristiane
Soggetto topico Speech processing systems
Signal processing - Digital techniques
ISBN 1-281-32198-2
9786611321987
0-470-72718-7
0-470-72717-9
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto -- List of Contributors xxi -- Preface xxvii -- 1 Introduction 1 /Rainer Martin, Ulrich Heute, Christiane Antweiler -- I Speech Quality Assessment 7 -- 2 Speech-Transmission Quality: Aspects and Assessment for Wideband vs. Narrowband Signals 9 /Ulrich Heute -- 2.1 Introduction 9 -- 2.2 Speech Signals . 10 -- 2.3 Telephone-Band Speech Signals 11 -- 2.4 Wideband Speech Signals 14 -- 2.5 Speech-Quality Assessment 25 -- 2.6 Wideband Speech-Quality Assessment 30 -- 2.7 Concluding Remarks 43 -- Bibliography 44 -- 3 Parametric Quality Assessment of Narrowband Speech in Mobile Communication Systems 51 /Marc Werner -- 3.1 Introduction 51 -- 3.2 Simulations of GSM and UMTS Speech Transmissions 58 -- 3.3 Speech Quality Measures based on Transmission Parameters 65 -- 3.4 Discussion and Conclusions 73 -- Bibliography 73 -- II Adaptive Algorithms in Acoustic Signal Processing 77 -- 4 Kalman Filtering in Acoustic Echo Control: A Smooth Ride on a Rocky Road 79 /Gerald Enzner -- 4.1 Introduction 79 -- 4.2 A Comprehensive Theory of Acoustic Echo Control 85 -- 4.3 The Kalman Filter for Conditional Mean and Covariance Estimation 90 -- 4.4 AEC Performance of the Frequency-Domain Adaptive Kalman Filter 100 -- 4.5 Discussion and Conclusions 102 -- Bibliography 103 -- 5 Noise Reduction - Statistical Analysis and Control of Musical Noise 107 /Colin Breithaupt, Rainer Martin -- 5.1 Introduction 107 -- 5.2 Speech Enhancement in the DFT Domain 109 -- 5.3 Measurement and Assessment of Unnatural Fluctuations 115 -- 5.4 Avoidance of Processing Artifacts 120 -- 5.5 Control of Spectral Fluctuations in the Cepstral Domain 123 -- 5.6 Discussion and Conclusions 128 -- 5.7 Acknowledgements 129 -- 5.8 Appendix 129 -- Bibliography 131 -- 6 Acoustic Source Localization with Microphone Arrays 135 /Nilesh Madhu, Rainer Martin -- 6.1 Introduction 135 -- 6.2 SignalModel 136 -- 6.3 Localization Approach Taxonomy 141 -- 6.4 Indirect Localization Approaches 141 -- 6.5 Direct Localization Approaches 148 -- 6.6 Evaluation of Localization Algorithms 156.
6.7 Conclusions 166 -- Bibliography 166 / /7 Multi-Channel System Identification with Perfect Sequences / Theory and Applications / 171 /Christiane Antweiler -- 7.1 Introduction 171 -- 7.2 System Identification with Perfect Sequences 174 -- 7.3 Multi-Channel System Identification 185 -- 7.4 Applications 191 -- 7.5 Discussion and Conclusions 195 -- Bibliography 195 -- III Speech Coding for Heterogeneous Networks 199 -- 8 Embedded Speech Coding: From G.711 to G.729.1 201 /Bernd Geiser, Stéphane Ragot, Hervé Taddei -- 8.1 Introduction 201 -- 8.2 Theory and Tools of Embedded Speech Coding 203 -- 8.3 Embedded Speech Coding Methods 212 -- 8.4 Standardized Embedded Speech Coders 219 -- 8.5 Network Aspects of Embedded Speech Coding 232 -- 8.6 Conclusions and Perspectives 237 -- Bibliography 238 -- 9 Backwards Compatible Wideband Telephony 249 /Peter Jax -- 9.1 Introduction 249 -- 9.2 From Narrowband Telephony to Wideband Telephony 250 -- 9.3 Stand-Alone Bandwidth Extension 254 -- 9.4 Embedded Wideband Coding Using Bandwidth Extension Techniques 257 -- 9.5 Combination of Bandwidth Extension with Watermarking 262 -- 9.6 Advanced Transmission of Highband Parameters 267 -- 9.7 Conclusions 274 -- Bibliography 274 -- IV Joint Source-Channel Coding 279 -- 10 Parameter Models and Estimators in Soft Decision Source Decoding 281 /Tim Fingscheidt -- 10.1 Introduction 281 -- 10.2 Overview to Soft Decision Source Decoding 283 -- 10.3 The Markovian Parameter Model 287 -- 10.4 Basic Extrapolative Estimators 290 -- 10.5 Joint Extrapolative Estimation of Two Different Parameters 298 -- 10.6 Extrapolative Estimation with Repeated Parameter Transmission 301 -- 10.7 Interpolative Estimation of a Parameter 304 -- 10.8 Discussion and Conclusions 307 -- Bibliography 307 -- 11 Optimal MMSE Estimation for Vector Sources with Spatially and Temporally Correlated Elements 311 /Stefan Heinen, Marc Adrat -- 11.1 Introduction 311 -- 11.2 Source Model 312 -- 11.3 Transmission Channel 316 -- 11.4 Optimal MMSE Parameter Estimator 316.
11.5 Near-Optimal MMSE Parameter Estimator 320 -- 11.6 Illustrative Comparison 323 -- 11.7 Simulation Results 325 -- 11.8 Conclusions 327 -- Bibliography 327 -- 12 Source Optimized Channel Codes & Source Controlled Channel Decoding 329 /Stefan Heinen, Thomas Hindelang -- 12.1 Introduction 329 -- 12.2 The Transmission System Used as Reference 330 -- 12.3 Source Optimized Channel Coding (SOCC) 332 -- 12.4 Source Controlled Channel Decoding (SCCD) 341 -- 12.5 Comparison of SOCC versus SCCD 357 -- 12.6 Conclusions 362 -- Bibliography 363 -- 13 Iterative Source-Channel Decoding & Turbo DeCodulation 365 /Marc Adrat, Thorsten Clevorn, Laurent Schmalen -- 13.1 Introduction 365 -- 13.2 The Key of the Turbo Principle: Extrinsic Information 366 -- 13.3 Iterative Source-Channel Decoding (ISCD) 379 -- 13.4 Turbo DeCodulation (TDeC) 387 -- 13.5 Conclusions 394 -- Bibliography 395 -- V Speech Processing in Hearing Instruments 399 -- 14 Binaural Signal Processing in Hearing Aids 401 /Volkmar Hamacher, Ulrich Kornagel, Thomas Lotter, Henning Puder -- 14.1 Introduction 401 -- 14.2 Wireless System for Hearing Aids 405 -- 14.3 Binaural Classification Systems 410 -- 14.4 Binaural Beamformer 415 -- 14.5 Blind Source Separation (BSS): An Application for a Binaural Directional Microphone Array in Hearing Aids 422 -- 14.6 Conclusions 427 -- Bibliography 428 -- 15 Auditory-profile-based Physical Evaluation of Multi-microphone Noise Reduction Techniques in Hearing Instruments 431 /Koen Eneman, Arne Leijon, Simon Doclo, Ann Spriet, Marc Moonen, Jan Wouters -- 15.1 Introduction 431 -- 15.2 Multi-microphone Noise Reduction in Hearing Instruments 434 -- 15.3 Auditory-profile-based Physical Evaluation 441 -- 15.4 Test Conditions 449 -- 15.5 Simulation Results 450 -- 15.6 Discussion 452 -- 15.7 Conclusions 455 -- Bibliography 456 -- VI Speech Processing for Human / Machine Interfaces 459 -- 16 Automatic Speech Recognition in Adverse Acoustic Conditions 461 /Hans-GŠunter Hirsch -- 16.1 Introduction 461.
16.2 Structure of Speech Recognition Systems 462 -- 16.3 Acoustic Scenarios during Speech Input 468 -- 16.4 Improving the Recognition Performance in Adverse Conditions 476 -- 16.5 Conclusions 493 -- Bibliography 494 -- 17 Speaker Classification for Next-Generation Voice-Dialog Systems 497 /Felix Burkhardt, Florian Metze, Joachim Stegmann -- 17.1 Introduction 497 -- 17.2 Speaker Classification 498 -- 17.3 Detection of Age and Gender 505 -- 17.4 Detection of Anger 510 -- 17.5 Applications in IVR Systems 517 -- 17.6 Discussion and Conclusion 523 -- Bibliography 525 -- Index 529 -- Permissions List 541.
Record Nr. UNINA-9910144580303321
Chichester, West Sussex, England ; , : John Wiley & Sons, , c2008
Materiale a stampa
Lo trovi qui: Univ. Federico II
Opac: Controlla la disponibilità qui
Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Edizione [1st ed.]
Pubbl/distr/stampa Chichester, West Sussex, England ; ; Hoboken, NJ, : John Wiley & Sons, c2008
Descrizione fisica 1 online resource (573 p.)
Disciplina 621.39/9
Altri autori (Persone) MartinRainer
HeuteUlrich
AntweilerChristiane
Soggetto topico Speech processing systems
Signal processing - Digital techniques
ISBN 1-281-32198-2
9786611321987
0-470-72718-7
0-470-72717-9
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto -- List of Contributors xxi -- Preface xxvii -- 1 Introduction 1 /Rainer Martin, Ulrich Heute, Christiane Antweiler -- I Speech Quality Assessment 7 -- 2 Speech-Transmission Quality: Aspects and Assessment for Wideband vs. Narrowband Signals 9 /Ulrich Heute -- 2.1 Introduction 9 -- 2.2 Speech Signals . 10 -- 2.3 Telephone-Band Speech Signals 11 -- 2.4 Wideband Speech Signals 14 -- 2.5 Speech-Quality Assessment 25 -- 2.6 Wideband Speech-Quality Assessment 30 -- 2.7 Concluding Remarks 43 -- Bibliography 44 -- 3 Parametric Quality Assessment of Narrowband Speech in Mobile Communication Systems 51 /Marc Werner -- 3.1 Introduction 51 -- 3.2 Simulations of GSM and UMTS Speech Transmissions 58 -- 3.3 Speech Quality Measures based on Transmission Parameters 65 -- 3.4 Discussion and Conclusions 73 -- Bibliography 73 -- II Adaptive Algorithms in Acoustic Signal Processing 77 -- 4 Kalman Filtering in Acoustic Echo Control: A Smooth Ride on a Rocky Road 79 /Gerald Enzner -- 4.1 Introduction 79 -- 4.2 A Comprehensive Theory of Acoustic Echo Control 85 -- 4.3 The Kalman Filter for Conditional Mean and Covariance Estimation 90 -- 4.4 AEC Performance of the Frequency-Domain Adaptive Kalman Filter 100 -- 4.5 Discussion and Conclusions 102 -- Bibliography 103 -- 5 Noise Reduction - Statistical Analysis and Control of Musical Noise 107 /Colin Breithaupt, Rainer Martin -- 5.1 Introduction 107 -- 5.2 Speech Enhancement in the DFT Domain 109 -- 5.3 Measurement and Assessment of Unnatural Fluctuations 115 -- 5.4 Avoidance of Processing Artifacts 120 -- 5.5 Control of Spectral Fluctuations in the Cepstral Domain 123 -- 5.6 Discussion and Conclusions 128 -- 5.7 Acknowledgements 129 -- 5.8 Appendix 129 -- Bibliography 131 -- 6 Acoustic Source Localization with Microphone Arrays 135 /Nilesh Madhu, Rainer Martin -- 6.1 Introduction 135 -- 6.2 SignalModel 136 -- 6.3 Localization Approach Taxonomy 141 -- 6.4 Indirect Localization Approaches 141 -- 6.5 Direct Localization Approaches 148 -- 6.6 Evaluation of Localization Algorithms 156.
6.7 Conclusions 166 -- Bibliography 166 / /7 Multi-Channel System Identification with Perfect Sequences / Theory and Applications / 171 /Christiane Antweiler -- 7.1 Introduction 171 -- 7.2 System Identification with Perfect Sequences 174 -- 7.3 Multi-Channel System Identification 185 -- 7.4 Applications 191 -- 7.5 Discussion and Conclusions 195 -- Bibliography 195 -- III Speech Coding for Heterogeneous Networks 199 -- 8 Embedded Speech Coding: From G.711 to G.729.1 201 /Bernd Geiser, Stéphane Ragot, Hervé Taddei -- 8.1 Introduction 201 -- 8.2 Theory and Tools of Embedded Speech Coding 203 -- 8.3 Embedded Speech Coding Methods 212 -- 8.4 Standardized Embedded Speech Coders 219 -- 8.5 Network Aspects of Embedded Speech Coding 232 -- 8.6 Conclusions and Perspectives 237 -- Bibliography 238 -- 9 Backwards Compatible Wideband Telephony 249 /Peter Jax -- 9.1 Introduction 249 -- 9.2 From Narrowband Telephony to Wideband Telephony 250 -- 9.3 Stand-Alone Bandwidth Extension 254 -- 9.4 Embedded Wideband Coding Using Bandwidth Extension Techniques 257 -- 9.5 Combination of Bandwidth Extension with Watermarking 262 -- 9.6 Advanced Transmission of Highband Parameters 267 -- 9.7 Conclusions 274 -- Bibliography 274 -- IV Joint Source-Channel Coding 279 -- 10 Parameter Models and Estimators in Soft Decision Source Decoding 281 /Tim Fingscheidt -- 10.1 Introduction 281 -- 10.2 Overview to Soft Decision Source Decoding 283 -- 10.3 The Markovian Parameter Model 287 -- 10.4 Basic Extrapolative Estimators 290 -- 10.5 Joint Extrapolative Estimation of Two Different Parameters 298 -- 10.6 Extrapolative Estimation with Repeated Parameter Transmission 301 -- 10.7 Interpolative Estimation of a Parameter 304 -- 10.8 Discussion and Conclusions 307 -- Bibliography 307 -- 11 Optimal MMSE Estimation for Vector Sources with Spatially and Temporally Correlated Elements 311 /Stefan Heinen, Marc Adrat -- 11.1 Introduction 311 -- 11.2 Source Model 312 -- 11.3 Transmission Channel 316 -- 11.4 Optimal MMSE Parameter Estimator 316.
11.5 Near-Optimal MMSE Parameter Estimator 320 -- 11.6 Illustrative Comparison 323 -- 11.7 Simulation Results 325 -- 11.8 Conclusions 327 -- Bibliography 327 -- 12 Source Optimized Channel Codes & Source Controlled Channel Decoding 329 /Stefan Heinen, Thomas Hindelang -- 12.1 Introduction 329 -- 12.2 The Transmission System Used as Reference 330 -- 12.3 Source Optimized Channel Coding (SOCC) 332 -- 12.4 Source Controlled Channel Decoding (SCCD) 341 -- 12.5 Comparison of SOCC versus SCCD 357 -- 12.6 Conclusions 362 -- Bibliography 363 -- 13 Iterative Source-Channel Decoding & Turbo DeCodulation 365 /Marc Adrat, Thorsten Clevorn, Laurent Schmalen -- 13.1 Introduction 365 -- 13.2 The Key of the Turbo Principle: Extrinsic Information 366 -- 13.3 Iterative Source-Channel Decoding (ISCD) 379 -- 13.4 Turbo DeCodulation (TDeC) 387 -- 13.5 Conclusions 394 -- Bibliography 395 -- V Speech Processing in Hearing Instruments 399 -- 14 Binaural Signal Processing in Hearing Aids 401 /Volkmar Hamacher, Ulrich Kornagel, Thomas Lotter, Henning Puder -- 14.1 Introduction 401 -- 14.2 Wireless System for Hearing Aids 405 -- 14.3 Binaural Classification Systems 410 -- 14.4 Binaural Beamformer 415 -- 14.5 Blind Source Separation (BSS): An Application for a Binaural Directional Microphone Array in Hearing Aids 422 -- 14.6 Conclusions 427 -- Bibliography 428 -- 15 Auditory-profile-based Physical Evaluation of Multi-microphone Noise Reduction Techniques in Hearing Instruments 431 /Koen Eneman, Arne Leijon, Simon Doclo, Ann Spriet, Marc Moonen, Jan Wouters -- 15.1 Introduction 431 -- 15.2 Multi-microphone Noise Reduction in Hearing Instruments 434 -- 15.3 Auditory-profile-based Physical Evaluation 441 -- 15.4 Test Conditions 449 -- 15.5 Simulation Results 450 -- 15.6 Discussion 452 -- 15.7 Conclusions 455 -- Bibliography 456 -- VI Speech Processing for Human / Machine Interfaces 459 -- 16 Automatic Speech Recognition in Adverse Acoustic Conditions 461 /Hans-GŠunter Hirsch -- 16.1 Introduction 461.
16.2 Structure of Speech Recognition Systems 462 -- 16.3 Acoustic Scenarios during Speech Input 468 -- 16.4 Improving the Recognition Performance in Adverse Conditions 476 -- 16.5 Conclusions 493 -- Bibliography 494 -- 17 Speaker Classification for Next-Generation Voice-Dialog Systems 497 /Felix Burkhardt, Florian Metze, Joachim Stegmann -- 17.1 Introduction 497 -- 17.2 Speaker Classification 498 -- 17.3 Detection of Age and Gender 505 -- 17.4 Detection of Anger 510 -- 17.5 Applications in IVR Systems 517 -- 17.6 Discussion and Conclusion 523 -- Bibliography 525 -- Index 529 -- Permissions List 541.
Record Nr. UNINA-9910811945903321
Chichester, West Sussex, England ; ; Hoboken, NJ, : John Wiley & Sons, c2008
Materiale a stampa
Lo trovi qui: Univ. Federico II
Opac: Controlla la disponibilità qui