Advances in digital speech transmission:
Gespeichert in:
| Format: | Buch |
|---|---|
| Sprache: | English |
| Veröffentlicht: |
Chichester, West Sussex, England
John Wiley & Sons
2008
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| Schlagworte: | |
| Online-Zugang: | Inhaltsverzeichnis |
| Beschreibung: | Includes bibliographical references (p. 525-528) and index |
| Beschreibung: | xxv, 543 p. ill. 26 cm |
| ISBN: | 9780470517390 0470517395 |
Internformat
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| 020 | |a 9780470517390 |c alk. paper |9 978-0-470-51739-0 | ||
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| 245 | 1 | 0 | |a Advances in digital speech transmission |c edited by Rainer Martin ... |
| 264 | 1 | |a Chichester, West Sussex, England |b John Wiley & Sons |c 2008 | |
| 300 | |a xxv, 543 p. |b ill. |c 26 cm | ||
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| 500 | |a Includes bibliographical references (p. 525-528) and index | ||
| 650 | 4 | |a Speech processing systems | |
| 650 | 4 | |a Signal processing |x Digital techniques | |
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| 700 | 1 | |a Martin, Rainer |e Sonstige |4 oth | |
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Datensatz im Suchindex
| _version_ | 1804141047627907072 |
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| adam_text | ADVANCES IN DIGITAL SPEECH TRANSMISSION EDITED BY RAINER MARTIN RUHR
UNIVERSITY BOCHUM, BOCHUM, GERMANY ULRICH HEUTE
CHRISTIAN-ALHRECHTS-UNIVERSITY, KIEL, GERMANY CHRISTIANE ANTWEILER RWTH
AACHEN UNIVERSITY, AACHEN, GERMANY JOHN WILEY & SONS, LTD CONTENTS LIST
OF CONTRIBUTORS XXI PREFACE XXVII 1 INTRODUCTION 1 RAINER MARTIN, ULRICH
HEUTE, CHRISTIANE ANTWEILER 1 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 2.3.1 NARROWBAND SPEECH
INTELLIGIBILITY 12 2.3.2 NARROWBAND SPEECH-SOUND QUALITY 12 2.4 WIDEBAND
SPEECH SIGNALS 14 2.4.1 WIDEBAND-SPEECH INTELLIGIBILITY AND SOUND
QUALITY 15 2.4.2 WIDEBAND SPEECH TRANSMISSION AND PROCESSING 18 2.5
SPEECH-QUALITY ASSESSMENT 25 2.5.1 AUDITORY QUALITY DETERMINATION 25
2.5.2 INSTRUMENTAL QUALITY DETERMINATION 27 2.6 WIDEBAND SPEECH-QUALITY
ASSESSMENT 30 2.6.1 INTEGRAL QUALITY DETERMINATION 30 2.6.2
ATTRIBUTE-ORIENTED QUALITY DETERMINATION 34 VIII CONTENTS 2.6.3 COMBINED
DIRECT AND ATTRIBUTE-BASED TOTAL QUALITY DETERMINATION 43 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 3.1.1 SUBJECTIVE LISTENING TESTS AND CLASSES OF OBJECTIVE MEASURES .
52 3.1.2 OVERVIEW OF OBJECTIVE SPEECH QUALITY MEASURES 55 3.1.3
DEVELOPMENT OF PARAMETRIC MODELS 57 3.2 SIMULATIONS OF GSM AND UMTS
SPEECH TRANSMISSIONS 58 3.2.1 SIMULATION ENVIRONMENT 58 3.2.2 GSM
TRANSMISSION PARAMETERS 62 3.2.3 UMTS TRANSMISSION PARAMETERS 64 3.3
SPEECH QUALITY MEASURES BASED ON TRANSMISSION PARAMETERS 65 3.3.1
CORRELATION ANALYSIS 65 3.3.2 PARAMETRIC SPEECH QUALITY MEASURES 68 3.4
DISCUSSION AND CONCLUSIONS 73 BIBLIOGRAPHY 73 II ADAPTIVE ALGORITHMS IN
ACOUSTIC SIGNAL PROCESSING 77 4 KAIMAN FILTERING IN ACOUSTIC ECHO
CONTROL: A SMOOTH RIDE ON A ROCKY ROAD 79 GERALD ENZNER 4.1 INTRODUCTION
79 4.1.1 ADAPTIVE FILTER STRUCTURES FOR ACOUSTIC ECHO CONTROL 81 4.1.2
CONTROL OF ADAPTIVE FILTERS 82 4.1.3 OPEN PROBLEMS / ORGANIZATION OF
THIS CHAPTER 84 CONTENTS IX 4.2 A COMPREHENSIVE THEORY OF ACOUSTIC ECHO
CONTROL 85 4.2.1 STOCHASTIC MODELING OF THE ECHO PATH 85 4.2.2 MINIMUM
MEAN-SQUARE ERROR (MMSE) SOLUTION 87 4.2.3 MMSE PROCESSOR IN THE
GAUSSIAN CASE 89 4.3 THE KAIMAN FILTER FOR CONDITIONAL MEAN AND
COVARIANCE ESTIMATION . 90 4.3.1 LINEAR ECHO PATH MODEL IN DFT-MATRIX
FORM 91 4.3.2 MARKOV MODEL OF THE TIME-VARYING ECHO PATH 93 4.3.3 EXACT
KAIMAN FILTER FOR THE CONDITIONAL MEAN AND COVARIANCE 94 4.3.4
DIAGONALIZATION OF THE KAIMAN FILTER 96 4.3.5 UNIFICATION OF ADAPTIVE
FILTERING AND ADAPTATION CONTROL ... 98 4.4 AEC PERFORMANCE OF THE
FREQUENCY-DOMAIN ADAPTIVE KAIMAN 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 5.2.1
OPTIMAL SPEECH ESTIMATORS 109 5.3 MEASUREMENT AND ASSESSMENT OF
UNNATURAL FLUCTUATIONS 115 5.3.1 FILTER ANALYSIS VIA APPROXIMATED FILTER
INPUT-OUTPUT CHARACTERISTICS 116 5.3.2 OUTLIER STATISTICS 118 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 5.8.1 MEAN A PRIORI SNR
FOR DIFFERENT FILTER TYPES AND LOW SNR . . . 129 5.8.2 APPROXIMATION OF
THE DECISION-DIRECTED APPROACH FOR LOW SNR . 131 BIBLIOGRAPHY 131 X
CONTENTS 6 ACOUSTIC SOURCE LOCALIZATION WITH MICROPHONE ARRAYS 135
NILESH MADHU, RAINER MARTIN 6.1 INTRODUCTION . 135 6.2 SIGNAL MODEL 136
6.2.1 CONTINUOUS TIME MODEL 136 6.2.2 DISCRETE TIME REPRESENTATION 137
6.2.3 FORMULATION IN THE FREQUENCY DOMAIN 138 6.2.4 SIMPLIFIED MODEL FOR
LOCALIZATION 138 6.3 LOCALIZATION APPROACH TAXONOMY 141 6.4 INDIRECT
LOCALIZATION APPROACHES 141 6.4.1 GENERALIZED CROSS-CORRELATION (GCC)
142 6.4.2 ADAPTIVE EIGENVALUE DECOMPOSITION (AED) 144 6.4.3 INFORMATION
THEORETIC APPROACH TO TDOA ESTIMATION 146 6.4.4 EXTENSION TO MULTIPLE
MICROPHONE PAIRS 147 6.5 DIRECT LOCALIZATION APPROACHES 148 6.5.1
STEERED RESPONSE POWER BEAMFORMING 149 6.5.2 MINIMUM MEAN SQUARE (MMSE)
APPROACH 150 6.5.3 PRACTICAL ASPECTS 151 6.5.4 SUBSPACE BASED APPROACHES
151 6.5.5 MAXIMUM LIKELIHOOD ESTIMATION (MLE) 154 6.6 EVALUATION OF
LOCALIZATION ALGORITHMS 156 6.6.1 PERFORMANCE OF THE INDIRECT METHODS
158 6.6.2 PERFORMANCE OF THE DIRECT METHODS 160 6.6.3 THE TWO-SOURCE
CASE 165 6.7 CONCLUSIONS 166 BIBLIOGRAPHY 166 CONTENTS XI 7
MULTI-CHANNEL SYSTEM IDENTIFICATION WITH PERFECT SEQUENCES - THEORY AND
APPLICATIONS - 171 CHRISTIANE ANTWEUEER 7.1 INTRODUCTION 171 7.2 SYSTEM
IDENTIFICATION WITH PERFECT SEQUENCES 174 7.2.1 GEOMETRIE INTERPRETATION
OF THE NLMS ALGORITHM 175 7.2.2 OPTIMAL EXCITATION OF THE NLMS ALGORITHM
176 7.2.3 INNUENCE OF ENVIRONMENTAL NOISE, STEPSIZE, AND PERIOD 179
7.2.4 ODD-PERFECT SEQUENCES 181 7.2.5 TRACKING OF TIME-VARIANT SYSTEMS
183 7.2.6 COMPLEXITY 184 7.3 MULTI-CHANNEL SYSTEM IDENTIFICATION 185
7.3.1 THE DUAL-CHANNEL CASE 185 7.3.2 SIMULATION RESULTS 188 7.3.3
GENERALIZATION TO THE MULTI-CHANNEL CASE 190 7.4 APPLICATIONS 191 7.4.1
SIMULATION OF TIME-VARIANT RIRS FOR STEREOPHONIE ECHO CONTROL 191 7.4.2
ACOUSTIC TUBE ENDOSCOPY 194 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, STEPHANE
RAGOT, HERVE TADDEI 8.1 INTRODUCTION 201 8.2 THEORY AND TOOLS OF
EMBEDDED SPEECH CODING 203 8.2.1 BASIC PRINCIPLES 203 8.2.2
APPROXIMATION THEORY 205 8.2.3 HIERARCHICAL VECTOR QUANTIZATION METHODS
208 CONTENTS 8.3 EMBEDDED SPEECH CODING METHODS 212 8.3.1 EMBEDDED DPCM
AND ADPCM 212 8.3.2 EMBEDDED CELP 213 8.3.3 EMBEDDED EXTENSIONS OF CELP
CODERS 216 8.3.4 EMBEDDED PARAMETER QUANTIZATION 218 8.4 STANDARDIZED
EMBEDDED SPEECH CODERS 219 8.4.1 ITU-T G.711 PCM CODEC 219 8.4.2 ITU-T
G.727 AND G.722 ADPCM CODECS 220 8.4.3 MPEG-4 SCALABLE SPEECH CODING 220
8.4.4 EMBEDDED WIDEBAND CODING FOR VOIP: ITU-T G.729.1 . . . . 223 8.5
NETWORK ASPECTS OF EMBEDDED SPEECH CODING 232 8.5.1 IMPLEMENTATION AND
UTILIZATION OF SCALABILITY 232 8.5.2 UNEQUAL ERROR PROTECTION AND
ENCRYPTION 236 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 9.3.1 ESTIMATION OF THE WIDEBAND
SPECTRAL ENVELOPE 255 9.3.2 EXTENSION OF THE EXCITATION SIGNAL 256 9.3.3
PERFORMANCE AND STATE-OF-THE-ART 257 9.4 EMBEDDED WIDEBAND CODING USING
BANDWIDTH EXTENSION TECHNIQUES 257 9.4.1 TRANSMISSION OF BWE INFORMATION
258 9.4.2 EXAMPLES OF EMBEDDED WIDEBAND SPEECH CODECS 260 9.4.3 AUDIO
CODING 262 9.5 COMBINATION OF BANDWIDTH EXTENSION WITH WATERMARKING 262
9.5.1 DIGITAL WATERMARKING OF SPEECH SIGNALS 263 9.5.2 TRANSMISSION OF
BWE INFORMATION VIA WATERMARKING 265 CONTENTS XIII 9.5.3 CHALLENGES AND
STATUS 266 9.6 ADVANCED TRANSMISSION OF HIGHBAND PARAMETERS 267 9.6.1
CODING WITH SIDE INFORMATION 268 9.6.2 ERROR CONCEALMENT WITH SIDE
INFORMATION 270 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 10.2.1 SOURCE ENCODING 283
10.2.2 EQUIVALENT CHANNEL 284 10.2.3 HARD DECISION AND SOFT DECISION
SOURCE DECODING 285 10.3 THE MARKOVIAN PARAMETER MODEL 287 10.3.1
DESCRIPTION OF A PRIORI KNOWLEDGE 287 10.3.2 QUANTIFICATION OF
UTILIZABLE RESIDUAL REDUNDANCY 288 10.3.3 CHOICE OF THE MODEL ORDER 289
10.4 BASIC EXTRAPOLATIVE ESTIMATORS 290 10.4.1 INTRODUCTION AND
SIMULATION SETTINGS 290 10.4.2 ESTIMATORS 291 10.4.3 SIMULATION RESULTS
294 10.5 JOINT EXTRAPOLATIVE ESTIMATION OF TWO DIFFERENT PARAMETERS 298
10.5.1 ESTIMATORS 298 10.5.2 SIMULATION RESULTS 299 10.6 EXTRAPOLATIVE
ESTIMATION WITH REPEATED PARAMETER TRANSMISSION . . . 301 10.6.1
ESTIMATORS 301 10.6.2 SIMULATION RESULTS 303 10.7 INTERPOLATIVE
ESTIMATION OF A PARAMETER 304 XIV CONTENTS 10.7.1 ESTIMATORS 304 10.7.2
SIMULATION RESULTS 306 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 12.3.1
DEFINITION 333 12.3.2 DECODING OF SOURCE OPTIMIZED CHANNEL CODES 334
12.3.3 DESIGN OF SOURCE OPTIMIZED CHANNEL CODES 335 12.3.4 NUMERICAL
ASPECTS OF SOCC DESIGN 336 12.3.5 BIT ALLOCATION BETWEEN SOURCE AND
CHANNEL CODING 336 12.3.6 RELATION TO CHANNEL OPTIMIZED VECTOR
QUANTIZATION 338 12.4 SOURCE CONTROLLED CHANNEL DECODING (SCCD) 341
12.4.1 CHANNEL CODING AND DECODING IN SCCD 341 CONTENTS XV 12.4.2 A
PRIORI KNOWLEDGE IN CHANNEL DECODING 345 12.4.3 CHANNEL DECODING USING
INTRA-PARAMETER CORRELATION 347 12.4.4 CHANNEL DECODING USING
INTER-FRAME CORRELATION 349 12.4.5 CHANNEL DECODING USING
INTRA-PARAMETER AND INTER-FRAME CORRELATION 350 12.4.6 SIMULATION
RESULTS 352 12.4.7 EXPLOITING A PRIORI KNOWLEDGE IN SOURCE AND/OR
CHANNEL DECODING 354 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 13.2.1 TERMS OF RELIABILITY INFORMATION 367 13.2.2
EXTRINSIC INFORMATION OF CHANNEL DECODING 368 13.2.3 EXTRINSIC
INFORMATION OF SOURCE DECODING 371 13.2.4 EXTRINSIC INFORMATION OF
DEMODULATION 374 13.2.5 EXIT CHARTS 376 13.3 ITERATIVE SOURCE-CHANNEL
DECODING (ISCD) 379 13.3.1 TRANSMISSION SYSTEM AND ALGORITHM 379 13.3.2
SIMULATION EXAMPLES 382 13.3.3 ADVANCEMENTS AND OPTIMIZATIONS 385 13.4
TURBO DECODULATION (TDEC) 387 13.4.1 TRANSMISSION SYSTEM AND ALGORITHM
388 13.4.2 SIMULATION EXAMPLES 390 13.4.3 ADVANCEMENTS AND OPTIMIZATIONS
393 13.5 CONCLUSIONS 394 BIBLIOGRAPHY 395 XVI CONTENTS 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 14.1.1 MONAURAL HEARING AIDS - STATE
OF THE ART 402 14.1.2 BINAURAL HEARING AIDS 404 14.1.3 ORGANIZATION OF
THIS CHAPTER 405 14.2 WIRELESS SYSTEM FOR HEARING AIDS 405 14.2.1
COMPARISON OF WIRELESS SYSTEMS 405 14.2.2 FUNCTIONAL DESCRIPTION OF THE
WIRELESS SYSTEM FOR HEARING AIDS 406 14.2.3 APPLICATIONS OF THE WIRELESS
SYSTEM FOR HEARING AIDS 409 14.3 BINAURAL CLASSIFICATION SYSTEMS 410
14.3.1 MOTIVATION AND BASIC PRINCIPLE 410 14.3.2 BINAURAL CLASSIFICATION
412 14.4 BINAURAL BEAMFORMER 415 14.4.1 DUAL CHANNEL INPUT-OUTPUT
BEAMFORMER DESIGN 416 14.4.2 MULTICHANNEL POSTFILTER 419 14.4.3
PERFORMANCE EVALUATION 420 14.5 BLIND SOURCE SEPARATION (BSS): AN
APPLICATION FOR A BINAURAL DIRECTIONAL MICROPHONE ARRAY IN HEARING AIDS
422 14.5.1 APPLICATION SCENARIO 422 14.5.2 SPECIFIC HEARING AID
CHALLENGES AND SOLUTIONS 423 14.5.3 SIGNAL SEPARATION WITH HEARING AID
CONSTRAINTS 424 14.5.4 OUTPUT SIGNAL SELECTION 425 14.5.5 BINAURAL
OUTPUT GENERATION 426 14.5.6 CONCLUDING REMARKS 427 14.6 CONCLUSIONS 427
BIBLIOGRAPHY 428 CONTENTS XVUE 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 15.2.1 CLASSICAL SOLUTIONS 434
15.2.2 GENERALIZED SIDELOBE CANCELER 436 15.2.3 ADAPTIVE TWO-STAGE
BEAMFORMING APPROACH 438 15.2.4 SPATIALLY PREPROCESSED
SPEECH-DISTORTION-WEIGHTED MULTICHANNEL WIENER FILTERING 439 15.3
AUDITORY-PROFILE-BASED PHYSICAL EVALUATION 441 15.3.1 SIMULATION OF
HEARING-IMPAIRED PERCEPTION 442 15.3.2 PHYSICAL EVALUATION MEASURES 444
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-GUENTER HIRSCH 16.1 INTRODUCTION 461 16.2
STRUCTURE OF SPEECH RECOGNITION SYSTEMS 462 16.2.1 MEL FREQUENCY
CEPSTRAL ANALYSIS 463 16.2.2 MODELING SPEECH UNITS AS HMMS 466 16.3
ACOUSTIC SCENARIOS DURING SPEECH INPUT 468 16.3.1 SIMULATION OF THE
ACOUSTIC ENVIRONMENT 469 16.3.2 RECOGNITION RESULTS FOR DIFFERENT
DISTORTION EFFECTS 473 16.4 IMPROVING THE RECOGNITION PERFORMANCE IN
ADVERSE CONDITIONS . . . . 476 XVIII CONTENTS 16.4.1 ADAPTING HMMS TO
REVERBERATION 477 16.4.2 ADAPTATION OF DELTA PARAMETERS 482 16.4.3
RECOGNITION EXPERIMENTS ON HANDS-FREE SPEECH INPUT 485 16.4.4 COMBINED
ADAPTATION TO ALL DISTORTION EFFECTS 487 16.4.5 RECOGNITION EXPERIMENTS
ON HANDS-FREE SPEECH INPUT IN NOISY ENVIRONMENTS 490 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 17.2.1
OVERVIEW 498 17.2.2 FEATURE EXTRACTION 499 17.2.3 CLASSIFICATION
ALGORITHMS 501 17.2.4 EVALUATION OF CLASSIFIERS 503 17.3 DETECTION OF
AGE AND GENDER 505 17.3.1 BACKGROUND 505 17.3.2 ALGORITHMS 506 17.3.3
RESULTS 509 17.4 DETECTION OF ANGER 510 17.4.1 BACKGROUND 510 17.4.2
ALGORITHM 513 17.4.3 RESULTS 514 17.5 APPLICATIONS IN IVR SYSTEMS 517
17.5.1 ADAPTIVE VOICE-DIALOGS 518 17.5.2 A VOICE PORTAL BASED ON
AGE/GENDER DETECTION 519 17.5.3 CUSTOMER SELF-SERVICE BASED ON ANGER
DETECTION 521 17.6 DISCUSSION AND CONCLUSION 523 BIBLIOGRAPHY 525
CONTENTS X ; X INDEX 529 PERMISSIONS LIST 541
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| discipline | Elektrotechnik / Elektronik / Nachrichtentechnik |
| format | Book |
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| id | DE-604.BV036019834 |
| illustrated | Illustrated |
| indexdate | 2024-07-09T22:09:40Z |
| institution | BVB |
| isbn | 9780470517390 0470517395 |
| language | English |
| lccn | 2007039292 |
| oai_aleph_id | oai:aleph.bib-bvb.de:BVB01-018911982 |
| oclc_num | 173480524 |
| open_access_boolean | |
| owner | DE-706 |
| owner_facet | DE-706 |
| physical | xxv, 543 p. ill. 26 cm |
| publishDate | 2008 |
| publishDateSearch | 2008 |
| publishDateSort | 2008 |
| publisher | John Wiley & Sons |
| record_format | marc |
| spelling | Advances in digital speech transmission edited by Rainer Martin ... Chichester, West Sussex, England John Wiley & Sons 2008 xxv, 543 p. ill. 26 cm txt rdacontent n rdamedia nc rdacarrier Includes bibliographical references (p. 525-528) and index Speech processing systems Signal processing Digital techniques Sprachübertragung (DE-588)4129249-2 gnd rswk-swf Digitale Sprachverarbeitung (DE-588)4233857-8 gnd rswk-swf Codierung (DE-588)4070059-8 gnd rswk-swf Digitale Sprachverarbeitung (DE-588)4233857-8 s Sprachübertragung (DE-588)4129249-2 s Codierung (DE-588)4070059-8 s DE-604 Martin, Rainer Sonstige oth GBV Datenaustausch application/pdf http://bvbr.bib-bvb.de:8991/F?func=service&doc_library=BVB01&local_base=BVB01&doc_number=018911982&sequence=000001&line_number=0001&func_code=DB_RECORDS&service_type=MEDIA Inhaltsverzeichnis |
| spellingShingle | Advances in digital speech transmission Speech processing systems Signal processing Digital techniques Sprachübertragung (DE-588)4129249-2 gnd Digitale Sprachverarbeitung (DE-588)4233857-8 gnd Codierung (DE-588)4070059-8 gnd |
| subject_GND | (DE-588)4129249-2 (DE-588)4233857-8 (DE-588)4070059-8 |
| title | Advances in digital speech transmission |
| title_auth | Advances in digital speech transmission |
| title_exact_search | Advances in digital speech transmission |
| title_full | Advances in digital speech transmission edited by Rainer Martin ... |
| title_fullStr | Advances in digital speech transmission edited by Rainer Martin ... |
| title_full_unstemmed | Advances in digital speech transmission edited by Rainer Martin ... |
| title_short | Advances in digital speech transmission |
| title_sort | advances in digital speech transmission |
| topic | Speech processing systems Signal processing Digital techniques Sprachübertragung (DE-588)4129249-2 gnd Digitale Sprachverarbeitung (DE-588)4233857-8 gnd Codierung (DE-588)4070059-8 gnd |
| topic_facet | Speech processing systems Signal processing Digital techniques Sprachübertragung Digitale Sprachverarbeitung Codierung |
| url | http://bvbr.bib-bvb.de:8991/F?func=service&doc_library=BVB01&local_base=BVB01&doc_number=018911982&sequence=000001&line_number=0001&func_code=DB_RECORDS&service_type=MEDIA |
| work_keys_str_mv | AT martinrainer advancesindigitalspeechtransmission |