Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA

Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 1

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adam_text Volume I TABLE OF CONTENTS Volume 1________________________________________________ SPl Robust Recognition: Signals and Features Feature Parameter Curve Method for High Performance NN-based Speech Recognition ................................................1-1 D. Chen, S. Zhu, T. Huang - Chinese Academy of Sciences, China On the Use of Residual Cepstrum in Speech Recognition .......................................................................................1-5 J. He, L· Liu, G. Palm - University of Ulm, Germany HMM-Based Speech Recognition Using State-Dependent, Linear Transforms on Mel-Warped DFT Features .....................1-9 C. Rathinavelu, L Deng - University of Waterloo, Canada Mixed Malvar- Wavelets for Non-Stationary Signal Representation ...........................................................................1-13 J. Thripuraneni, W. Lou, V. Debrunner - The University of Oklahoma, USA Experiments on a Parametric Nonlinear Spectral Warping for an HMM-based Speech Recognizer .................................1-17 D. Mashao - Brown University, USA Robust Distant Talking Speech Recognition ......................................................................................................1-21 Q. Lin, С Che, D. Yuk L. Jin - CA1P Center, Rutgers University, USA B. Vríes, J. Pearson - David Sarnoff Research Center, USA J. Flanagan - Rutgers University, USA Time-Frequency Representation Based Cepstral Processing For Speech Recognition ...................................................1-25 A. Fineberg, K. Yu - Motorola Lexicus, USA Knowledge-Based Parameters for Speech HMM Recognition .................................................................................1-29 С Espy-Wilson, N. Bitar - Boston University, USA A Phoneme Similarity Based ASR Front-End ......................................................................................................1-33 T. Applebaum, P. Morin, B. Hanson - Speech Technology Laboratory, USA A Model of Dynamic Auditory Perception and its Application to Robust Speech Recognition ..........................................1-37 B. Strope, A. Alwan - University of California at Los Angeles, USA SP2 Robust Recognition: Large Vocabulary Independent Calculation of Power Parameters on PMC Method ..............................................................................1-41 H. Yamamoto, M. Yantada, T. Kosáka, Y. Komori, Y. Ohora ■ Canon Inc., Japan Noisy Speech Recognition Using Variance Adapted Likelihood Measure ..................................................................1-45 J. Chien, L Lee, H. Wang - National Tsing Hua University, ROC An Improved Noise Compensation Algorithm for Speech Recognition in Noise ............................................................1-49 R. Yang, P. Haavisto - Nokia Research Center, Finland Improved Speech Recognition via Speaker Stress Directed Classification ..................................................................1-53 B. Womack, J. Hansen - Duke University, USA High-Accuracy Connected Digit Recognition for Mobile Applications .....................................................................1-57 S. Gupta, F. Soong, R. Haimi-Cohen ■ AT&T Bell Labs, USA Feature Extraction Based on Zero-Crossings with Peak Amplitudes for Robust Speech Recognition in Noisy Environments... 1-61 D. Kim, J. Jeong, J. Kim, S. Lee - Korea Advanced Institute of Science and Technology, ROK Improving Environmental Robustness in Large Vocabulary Speech Recognition .........................................................1-65 P. Woodland, M. Gales, D. Pye - Cambridge University, UK Noise and Room Acoustics Distorted Speech Recognition by HMM Composition .........................................................1-69 S. Nakamura, T. Takiguchi, K. Shikano - Nara Institute of Science and Technology, Japan Developments in Continuous Speech Dictation Using the 1995 ARPA NAB News Task ...................................................1-73 J. Gauvain, L Lamel, G. Adda, D. Matrouf - LIMSI-CNRS, France Evaluation of the Root-Normalized Front-End (RN-LFCC) for Speech Recognition in Wireless GSM Network Environments 1-77 P. Lockwood, S. Dufour, С. Glorien - MATRA Communications, France Volume I SP3 Speaker Recognition Speaker Background Models for Connected Digit Password Speaker Verification ......................................................1-81 Л. Rosenberg, S. Parthasarathy - AT&T Bell Laboratories, USA Cohort Selection and Word Grammar Effects on Speaker Recognition .....................................................................1-85 J. Colombi, D. Ruck - AFIT/ENG, USA, T. Anderson - AL/CFBA, USA, S. Rogers - AFIT/ENG, USA, M, Oxley - AFIT/ENC, USA Discriminative TVaining of GMM for Speaker Identification .................................................................................1-89 С Martin Del Alamo, J. Caminero Gil, С. De La Torre, L Hernandez Gomez - Telefonica I+D, Spain Subword-based Text-dependent Speaker Verification System with User-Selectable Passwords .......................................1-93 M. Sharma, R. Mammone - Rutgers University, USA Robust Methods of Updating Model and A Priori Threshold in Speaker Verification ...................................................1-97 T. Matsui, T. Nishitani, S. Fumi - NTT Human Interface Laboratories, Japan A Further Investigation of AR- Vector Models for Text-Independent Speaker Identification ..........................................1-101 /. Magrin-Chagnolleau, J. Wilke, F. Bimbót - CNRS, France Speaker Identification via Support Vector Classifiers ..........................................................................................1-105 M. Schmidt - BBN, USA Speaker Verification Using Mixture Likelihood Profiles Extracted from Speaker Independent Hidden Markov Models ......1-109 A. Setlur, R. Sukkar, M. Gandhi - AT&T Bell Laboratories, USA The Effects of Handset Variability on Speaker Recognition Performance: Experiments on the Switchboard Corpus .........1-113 D. Reynolds ■ MIT Lincoln Laboratory, USA Speaker Recognition in Reverberant Enclosures ................................................................................................1-117 P. Castellano, S. Sridharan, D. Cole - Signal Processing Research Centre, Australia SP4 Speech Recognition: Noise and Environment Using a Transcription Graph for Large Vocabulary Continuous Speech Recognition ...................................................1-121 Z. Li, D. O Shaughnessy - INRS-Telecommunications, Canada Fast and Accurate Recognition of Very-Large-Vocabulary Continuous Mandarin Speech for Chinese Language with Improved Segmental Probability Modeling .....................................................................................................................1-125 J. Shen, S. Hwang - National Taiwan University, ROC L Lin-shan - Academia Sinica, ROC Decoding Optimal State Sequence with Smooth State Likelihoods ...........................................................................1-129 /. Zeljkovic - AT&T Bell Laboratories, USA Improvements on the Pronunciation Prefix Tree Search Organization .....................................................................1-133 E Alleva, X. Huang, M. Hwang - Microsoft Corporation, USA Minimizing Search Errors Due to Delayed Digrams in Real-Time Speech Recognition Systems ....................................1-137 M. Woszczyna - University of Karlruhe, Germany, M. Finke - University of Karlsruhe, Germany Real-Time Recognition of Broadcast Radio Speech .............................................................................................1-141 G. Cook, J. Christie, P. Clarkson, S. Cooper, M. Hochberg, D. Kershaw, R. Logan, S. Renals, A. Robinson, С. Seymour, S. Waterhouse, P. Zolfaghari - Cambridge University, UK Spontaneous Dialogue Speech Recognition Using Cross-Word Context Constrained Word Graphs .................................1-145 T. Shimizu, H. Yamamoto, H. Masataki, S. Matsunaga, Y. Sagisaka - ATR - ITL Japan Efficient Evaluation of the LVCSR Search Space Using the NOWAY Decoder .........................................................1-149 S. Renais - University of Sheffield, UK, M. Hochberg - University of Cambridge, UK Developments in Large Vocabulary, Continuous Speech Recognition of German ......................................................1-153 M. Adda-Decker, G. Adda, L Lamel, J. Gauvain - UMSI-CNRS, France Speech Recognition on Mandarin Call Home: A Large-Vocabulary, Conversational, and Telephone Speech Corpus .........1-157 F. Liu, M. Picheny, P. Srinivasa, M. Monkowski, J. Chen - IBM T.J. Watson Research Center, USA xii Volume I SP5 Speech-Recognition: Language Modeling II Multilingual Stochastic η -Gram Class Language Models ....................................................................................1-161 M. Jardino - LIMSl-CNRS, France A Variable-Length Category-Based N-Gram Language Model ..............................................................................1-164 T. Niesler, P. Woodland - Cambridge University, UK Improving N-Gram Models by Incorporating Enhanced Distributions .....................................................................1-168 P. O Boyle, J. Ming, J. McMahon, J. Smith - Queen s University of Belfast, UK A Novel Word Clustering Algorithm Based on Latent Semantic Analysis ..................................................................1-172 J. Bellegarda, J. Butzberger, Y. Chow, N. Coccaro, D. Naik - Interactive Media Group, Apple Computer USA Statistical Natural Language Understanding ......................................................................................................1-176 M. Epstein, K. Papineri, S. Roukos, T. Ward - IBM T.J. Watson Research Center, USA, S. Della Pietra · Renaissance Technologies, USA Clustering Words for Statistical Language Models Based on Contextual Word Similarity .............................................1-180 A. Farhat, J. Isabelle, D. O Shaughnessy - INRS-Telecommunications, Canada Domain Word Translation By Space-Frequency Analysis of Context Length Histograms .............................................1-184 P. Fung - Columbia University, USA Variable-Order N-Gram Generation by Word-class Splitting and Consecutive Word Sequence Grouping ........................1-188 H. Masataki, Y. Sagisaka - ATR Interpreting Telecommunications Research Laboratories, Japan Back-off Method for N-Gram Smoothing Based on Binomial Posteriori Distribution ...................................................1-192 T. Kawabata, M. Tamoto - NTT Basic Research Labs, Japan Ergodic Multigram HMM Integrating Word Segmentation and Class Tagging for Chinese Language Modeling ...............1-196 H. Law, С Chan - The University of Hong Kong, Hong Kong SP6 Low-Rate Speech Coding A 2.4 kbit/s MELP Coder Candidate for the New U.S. Federal Standard ..................................................................1-200 A. McCree - Texas Instruments, USA, K. Truong - Atlanta Signal Processors, Inc., USA, E. George - Texas Instruments, USA T. Barnwell - Atlanta Signal Processors, Inc., USA, V. Viswanathan - Texas Instruments, USA Harmonic - Stochastic eXcitation (HSX) Speech Coding Below 4Kbits .....................................................................1-204 C. Laflamme, R. Salami, R. Matmti, J. Adoul - University of Sherbrooke, Canada A High Quality MBE-LPC-FE Speech Coder at 2.4kbps and 1.2kbps .....................................................................1-208 T. Wang, K. Tang, C. Feng - Tsinghua University, China A Low-Complexity Waveform Interpolation Coder .............................................................................................1-212 W. Kleijn, Y. Shoham, D. Sen, R. Hagen - AT&T Bell Laboratories, USA Mixed-Domain Coding of Speech at 3 Kbps ......................................................................................................1-216 J. De Martin - Politecnico di Torino, Italy A. Gersho - University of California at Santa Barbara, USA Source Driven/Variable Bit Rate Protoype Interpolation Coding ...........................................................................1-220 C. Xydeas, B. Cao - University of Manchester, UK A New Approach to Very Low-Rate Speech Coding Using Temporal Decomposition ...................................................1-224 S. Ghaemmaghami, M. Deriche - Queensland University of Technology, Australia A Variable Frame Pitch Estimator and Test Results .............................................................................................1-228 X. Qian, R. Kumaresan - University of Rhode Island, USA Robust Method of Measurement of Fundamental Frequency by ACLOS - Autocorrelation of Log Spectrum - ..................1-232 N. Kunieda, T. Shimamura, J. Suzuki - Satiama University, Japan Lag-Indexed VQ for Pitch Filter Coding .........................................................................................................1-236 S. McClellan - University of Alabama-Birmingham, USA J.Gibson-Texas A&M University, USA Volume I SP7 Wideband Coding and Emerging Techniques Embedded Algebraic Vector Quantizer (EAVQ) with Application to Wideband Speech Coding ....................................1-240 M. Xie, J. Adotti - University of Sherbrooke, Canada The Two-Dimensional Discrete Cosine Transform Applied to Speech Data ...............................................................1-244 L Baghai-Ravary, S. Beet, M. Tokhi - University of Sheffield, UK Real-Time High Accurate Cell Loss Recovery Technique For Speech Over ATM Networks ..........................................1-248 K. Matsumoto ■ NTT LSI Laboratories, Japan Predictive Fractal Interpolation Mapping: Differential Speech Coding at Low Bit Rates .............................................1-251 Z. Wang - University of Waterloo, Canada lékbit/s Wideband Speech Coding Based on Unequal Subbands ...........................................................................1-255 J. Paulus, J. Schmitzler - IND, Aachen University of Technology, Germany Low Delay IIR QMF Banks with High Perceptive Quality for Speech Processing ......................................................1-259 T. Kleinmann, A. Lacroix - University of Frankfurt, Germany Demodulators for AM-FM Models of Speech Signals: A Comparison .....................................................................1-263 S. Lu, P. Doerschuk - Purdue University, USA Synthesis and Coding of Continuous Speech with the Nonlinear Oscillator Model ......................................................1-267 G. Kubin - Vienna University of Technology, Austria Variable Frame Rate Parameter Encoding via Adaptive Frame Selection Using Dynamic Programming ........................1-271 E. George, A. McCree, V. Viswanathan ■ Texas Instruments, Inc., USA Transform Predictive Coding of Wideband Speech Signals ....................................................................................1-275 J. Chen ■ AT&T Bell Labs, USA D. Wang - Georgia Institute of Technology, USA SP8 Topic Identification and Spoken Information Retrieval A System for Unrestricted Topic Retrieval From Radio News Broadcasts ...............................................................1-279 D. James - Union Bank of Switzerland, Switzerland Automated Generation of N-Best Pronunciations of Proper Nouns ........................................................................1-283 N. Deshmukh, M. Weber, J. Picone ■ Mississippi State University, USA An Efficient Voice Retrieval System for Very-Large-Vocabulary Chinese Textual Databases with a Clustered Language Model .. .1-287 5. Lin - National Taiwan University, ROC L Chien, К. Chen, L Lee - Academia Sinica, ROC Concept-based Phrase Spotting Approach for Spontaneous Speech Understanding ...................................................1-291 T. Kawahara, N. Kitaoka, S. Doshita - Kyoto University, Japan A Dictionary-Based Method for Determining Topics in Text and Transcribed Speech 1-295 P. Schone, D. Nelson ■ Department of Defense, USA Keyword Spotting for Video Soundtrack Indexing .............................................................................................1-299 P. Gelin, C. Wellekens - Institut Eurecom, France Improvements in Switchboard Recognition and Topic Identification ........................................................................1-303 B. Peskin, S. Connolly, L Gillick, S. Lowe, D. McAllaster, V. Nagesha, P. van Mulbregt, S. Wegmann - Dragon Systems, Inc., USA Statistical Models for Topic Identification Using Phoneme Substrings .....................................................................1-307 J. Wright - University of Bristol, UK M. Carey, E. Partis - ENSIGMA Limited, UK Robust Talker-Independent Audio Document Retrieval .......................................................................................1-311 G. Jones, J. Foote, K. Spark Jones, S. Young - Cambridge University, UK Unsupervised Topic Clustering of Switchboard Speech Messages ...........................................................................1-315 B. Carlson ■ MIT Lincoln Laboratory, USA Volume I SP9 Robust Recognition: Compensation and Normalization Speaker Recognition and Speaker Normalization by Projection to Speaker Subspace ................................................1-319 К Ariki, S. Tagashira, M. Nishijima - Ryukoku University, Japan Compensated Mel Frequency Cepstrum Coefficients ..........................................................................................1-323 R. Vergin, D. O Shaughnessy, V. Gupta - /WRS-Telecommunications, Canada Adaptation Method Based on HMM Composition and EM Algorithm .....................................................................1-327 Y. Minami, S. Fumi - NTT Human Interface Laboratories, Japan SNR-Normalisation for Robust Speech Recognition .............................................................................................1-331 T. Claes, D. Van Compernolle - KU Leuven, Belgium Towards Robustness to Fast Speech in ASR ...................................................................................................... I-33S N. Mirghafori, E. Fosler, N. Morgan - International Computer Science Institute, USA Speaker Normalization on Conversational Telephone Speech .................................................................................1-339 S. Wegmann, D. Mc Allas ter, J. Orloff, B. Peskin ■ Dragon Systems, USA Speaker and Gender Normalization for Continuous-Density Hidden Markov Models ................................................1-342 A. Acero, X. Huang - Microsoft Corporation, USA A Parametric Approach to Vocal Tract Length Normalization ..............................................................................1-346 E. Eide, H. Gish - BBN Systems and Technologies, USA A Study on Speech Recognition for Children and the Elderly .................................................................................1-349 J.Wilpon- AT&T Bell Labs, USA C. Jacobsen - TeleDanmark/Jydsk Telefon, Denmark Speaker Normalization Using Efficient Frequency Warping Procedures ..................................................................1-353 L. Lee - Massachusetts Institute of Technology, USA R. Rose - AT&T Bell Laboratories, USA SPIO Speech Synthesis A Fast Stochastic Parser for Determining Phrase Boundaries for Text-to-Speech Synthesis ..........................................1-357 R. Sharman - IBM Laboratories, U.K. J. Wright - University of Bristol, U.K. Speech Concatenation and Synthesis Using an Overlap-add Sinusoidal Model .........................................................1-361 M. Macon, M. Clements - Georgia Institute of Technology, USA Voice Conversion Using Partitions of Spectral Feature Space .................................................................................1-365 W. Verhelst, J. Mertens - Vrije Universiteit Brussel, Belgium Determination of Vocal-Tract Shapes from Formant Frequencies Based on Perturbation Theory and Interpolation Method 1-369 Z Yu, P. Ching - Chinese University of Hong Kong, Hong Kong Unit Selection in a Concatenative Speech Synthesis System Using a Large Speech Database ..........................................1-373 A. Hunt, A. Black - ATR Interpreting Telecommunications Research Laboratories, Japan Parametric Hybrid Source Models for Voiced and Voiceless Fricative Consonants ......................................................1-377 S. Narayanan - AT&T Bell Laboratories, USA A. Alwan - University of California at Los Angeles, USA High Quality Speech Synthesis Using Context-Dependent Syllabic Units ..................................................................1-381 T. Saito, Y. Hashimoto, M. Sakamoto - IBM, Japan Articulatory Copy Synthesis Using a Nine Parameter Vocal Tract Model ..................................................................1-385 С Goodyear, D. Wei - University of Liverpool, UK Speech Synthesis Using HMMs with Dynamic Features .......................................................................................1-389 T. Masuko, K. Tokuda, T. Kobayashi, S. lmai - Tokyo Institute of Technology, Japan Interpolating V/UV Mixture Functions of a Harmonic Model for Concatenative Speech Synthesis .................................1-393 K. Lam, C Chan - City University of Hong Kong, Hong Kong Volume I SPII Speech Recognition: Language Modelling I An Efficient Top-Down Parsing Algorithm for Understanding Speech by Using Stochastic Syntactic and Semantic Models ...1-397 H. Stahl, J. Muller, M. Lang - Munich University of Technology, Germany Data-Driven Discourse Modeling for Semantic Interpretation ..............................................................................1-401 F. Caminero-Gil, J. Alvarez-Cercadillo, С Crespo-Casas, D. Tapias-Merino - Telefonica I+D, Spain Statistical Language Modeling for Speech Disfluencies .......................................................................................1-405 A. Stoicke, E. Shriberg - SRI International, USA JANUS II-Translation of Spontaneous Conversational Speech ..............................................................................1-409 A. Weibel, M. Finke, D. Gates, M. Wosczyna, M. Gavalda, T. Kemp, A. Lavie, L Levin, M. Maier - University ofKarkruhe, Germany Language Model Acquisition From a Text Corpus for Speech Understanding ............................................................1-413 T. Matsuoka - NTT Human Interface Laboratories, Japan R. Hasson - Eurecom Institute, France M. Barlow, S. Furui - NTT Human Interface Laboratories, Japan A Class Based Language Model for Speech Recognition .......................................................................................1-416 W. Ward, S. Issar - Carnegie Mellon University, USA An Integrated Model of Acoustics and Language Using Semantic Classification Trees ................................................1-419 E. Noth, R. DeMori, J. Fischer, A. Gebhard, S. Harbeck, R. Котре - Universität Erlangen-Numberg, Germany R. Kuhn, H. Niemann, M. Mast - Centre de Recherche Informatique de Montreal, Canada Combining Stochastic and Linguistic Language Models for Recognition of Spontaneous Speech ....................................1-423 W. Eckert, F. Gallwitz, H. Niemann - Universität Erlangen-Nurnberg, Germany Error Correction via a Post-Processor for Continuous Speech Recognition ...............................................................1-427 E. Ringger, J. Allen - University of Rochester, USA Integration of Concept-Driven Semantic Interpretation with Speech Recognition ......................................................1-431 A. Nogai, Y. Ishikawa, K. Nakajima - Mitsubishi Electric Corporation, Japan SP12 Speech Recognition Acoustic Modeling A Second-Order HMM for High Performance Word and Phoneme-Based Continuous Speech Recognition .....................1-435 J. -F. Mart, D. Fohr, J.-C. Junqua - CRIN-CNRS & INRIA, France Evaluation of Segmental Unit Input HMM ......................................................................................................1-439 5. Nakagawa, K. Yamamoto - Toyohashi University of Technology, Japan Design of a Speech Recognition System Based on Non-Uniform Segmental Units ......................................................1-443 M. Bacchiani - ATR Interpreting Telecommunications Research Laboratories, Japan M. Ostendorf- Boston University, USA Y. Sagisaka - ATR Interpreting Telecommunications Research Laboratories, Japan K. Paliwal - Griffith University, Australia Modeling Speech Variability with Segmental HMMs ............................................... .............1-447 W. Holmes, M. Russell - DRA Malvem, UK Context-Dependent Units for Vocabulary-Independent Spanish Speech Recognition ...................................................1-451 L VUlarrubia, L Gomez, J- Elvira, J. Torrecilla - Telefonica I+D, Spain Context-Dependent Acoustic Models For Chinese Speech Recognition .....................................................................1-455 B. Ma, T. Huang, B. Xu, X. Zhang, F. Qu - Chinese Academy of Sciences, China Automatic Recognition of Danish Natural Numbers for Telephone Applications .........................................................1-459 C. Jacobsen - TeleDanmark/Jydsk Telefon, Denmark J. Wilpon - AT&T Bell Laboratories, USA Explicit Modeling of Coarticulation in a Statistical Speech Recognizer .....................................................................1-463 R. Chen, L Jamieson - Purdue University, USA Tied-Structure HMM Based on Parameter Correlation for Efficient Model Training ...................................................1-467 5. Takahashi, S. Sagayama - NTT Human Interface Laboratories, Japan A Semi-Continuous Stochastic Trajectory Model for Phoneme-Based Continuous Speech Recognition ..................1-471 O. Siohan, Y. Gong - CRIN-CNRS & INRIA Lorraine, France xvi Volume I SP13 Speech Coding Quality Assessment Automatic Evaluation of Speaker Recognizability of Coded Speech ........................................................................1-475 K. Assaleh ■ Motorola, USA A Perceptually-Based Objective Measure for Speech Coders Using Abductive Network .............................................1-479 M. Meky, T. Saadawi - City University of New York, USA Objectively Measured Descriptors Applied to Speaker Characterization ..................................................................1-483 B. Necioglu, M. Clements, T. Barnwell - Georgia Institute of Technology, USA Objective Speech Quality Measure for Cellular Phone .......................................................................................1-487 K. Lam, О. Аи, С. Chan, К. Hui, S. Lau - Hong Kong University of Science and Technology, Hong Kong Vector Quantization Techniques for Output-Based Objective Speech Quality ............................................................1-491 C. Jin, R. Kubichek - University of Wyoming, USA Objective Measures for Speech Quality Assessment in Wireless Communication s ......................................................1-495 A. Bayya, M. Vis - US West Advanced Technologies, USA Performance Assessment of 4.8 kbit/s AMBE Coding Under Aeronautical Environmental Conditions ..............................1-499 5. Campos Neto, F. Corcoran, J. Phipps, S. Dimolitsas - COMSAT, USA Normalization of Cellular Telephone Speech for Recognition by Adaptive Vector Quantization ....................................1-503 R. Rajasekaran, M. Sonmez - Texas Instruments, Inc., USA J. Baras - University of Maryland at College Park, USA SP14 Speech Recognition Out-of-Vocabulary Modeling and Rejection Efficient Decoding and Training Procedures for Utterance Verification in Continuous Speech Recognition .....................1-507 E. Ueida, R. Rose - ATT Bell Laboratories, USA Confidence Measures for the SWITCHBOARD Database ....................................................................................1-511 5. Cox, R. Rose - AT&T Bell Labs, USA A Phone-Dependent Confidence Measure for Utterance Rejection ...........................................................................1-515 Z Rivlin, M. Cohen, V. Abrash, T. Chung - SRI International, USA Utterance Verification of Keyword Strings Using Word-Based Minimum Verification Error (WB-MVE) Training ............1-516 R. Sukkar, A. Setlur, M. Rahim, С Lee - AT&T Bell Laboratories, USA Discriminative Utterance Verification Using Minimum String Verification Error (MSVE) Training ........................... VI-3585 M. Rahim, С Lee, B. Juang, W. Chou - AT&T Bell Laboratories, USA (at time of printing this paper was placed in Volume 6) Murray Hill, NJ, USA Fast Implementation Methods for Viterbi-based Word-Spotting ...........................................................................1-522 K. Knill, S. Young - Cambridge University, UK Improving Wordspotting Performance with Artificially Generated Data ..................................................................1-526 E. Chang, R. Lippmann - Corona Corporation, USA Modelling Unknown Words in Spontaneous Speech .............................................................................................1-530 T. Kemp - University of Karlsruhe, Germany A. J usek - University of Bielefeld, Germany Improved Modeling of OOV Words in Spontaneous Speech .................................................................................1-534 P. Fetter, A. Kaltenmeier, T. Kuhn, P. Regel-Brietzmann - Research Center Daimler-Benz, Germany Two-Pass Strategy for Continuous Speech Recognition with Detection and Transcription of Unknown Words ..................1-538 S. Matsunaga, H. Sakamoto - ATR Interpreting Telecommunications Research Laboratories, USA SP15 Topics in Speech Coding A Modified Generalised Lloyd Algorithm for VQ Codebook Design ........................................................................1-542 C. Chen, S. Koh, P. Sivaprakasapillai - Nanyang Technological University, Singapore Robust Classification of Speech Based on the Dyadic Wavelet Transform with Application to CELP Coding .....................1-546 J. Stegmann, G. Shroeder, К. Fischer - Deutsche Telekom, Germany xvu Volume I Optimal Wavelet Packets for Low-Delay Audio Coding .......................................................................................1-550 P. Philippe, F. Moreau de St-Martin, M. Lever - CCETT, France J. Soumagne - Supelec, France A Fast VSELP Speech Coder Based on Mutually Orthonormal Regular Pulse Vectors ................................................1-554 Y. Choi, H. Kang, D. Youn - Yonsei University, Korea Dual-Pulse CS-CELP: A Toll Quality Low-Complexity Speech Coder at 7.8 kbit/s ......................................................1-558 H. Ohmuro, J. Ikedo, T. Moriya, A. Kataoka, S. Hayashi, K. Mano - NTT Human Interface Laboratories, Japan Low-Delay CELP With Multi-Pulse VQ and Fast Search for GSM EFR ..................................................................1-562 S. Taumi, K. Ozawa, T. Nomura, M. Serizawa - NEC Corporation, Japan Speech Compression with Cosine and Wavelet Packet Near-Best Bases ..................................................................1-566 C. Taswell - Stanford University, USA An Enhanced Full Rate Speech Coder for Digital Cellular Applications ..................................................................1-569 W. Leblanc, С. Liu, V. Viswanathan - Texas Instruments, USA Optimum Harmonics Tracking Filter for Auditory Scene Analysis ........................................................................1-573 K. Nishi ■ University of Electro-Communications, Japan S. Ando, S. Aida - The University of Tokyo, Japan Selective Error Protection of ITU-T G. 729 Codec for Digital Cellular Channels ......................................................1-577 K. Swaminathan, A. Hammons - Hughes Network Systems, USA M. Austin - BellSouth Cellular Corporation, USA XVIII
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illustrated Illustrated
indexdate 2024-11-25T17:14:19Z
institution BVB
institution_GND (DE-588)5200316-4
language Undetermined
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physical LVII, 580, 6 S. Ill., graph. Darst.
publishDate 1996
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publishDateSort 1996
publisher IEEE Service Center
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spellingShingle Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
subject_GND (DE-588)1071861417
title Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
title_auth Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
title_exact_search Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
title_full Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 1 The 1996 International Conference on Acoustics, Speech, and Signal Processing
title_fullStr Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 1 The 1996 International Conference on Acoustics, Speech, and Signal Processing
title_full_unstemmed Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 1 The 1996 International Conference on Acoustics, Speech, and Signal Processing
title_short Conference proceedings
title_sort conference proceedings may 7 10 1996 marriott marquis hotel atlanta georgia usa
title_sub May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
topic_facet Konferenzschrift
url http://bvbr.bib-bvb.de:8991/F?func=service&doc_library=BVB01&local_base=BVB01&doc_number=007241627&sequence=000002&line_number=0001&func_code=DB_RECORDS&service_type=MEDIA
volume_link (DE-604)BV010834406
work_keys_str_mv AT icasspatlantaga conferenceproceedingsmay7101996marriottmarquishotelatlantageorgiausa1

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