Analysis by synthesis speech coding with generalized pitch prediction

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A 16, 24, 32 kbit/s wideband speech codec based on ATCELP (Abstract)

P. Combescure, Aachen Univ. of Technol., Germany

P. Vary

pp. 5-8

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A 6.1 to 13.3-kb/s variable rate CELP codec (VR-CELP) for AMR speech coding (Abstract)

S. Heinen, Inst. of Commun. Syst. & Data Process, Aachen Univ. of Technol., Germany

M. Adratm

O. Steil

P. Vary

W. Xu

pp. 9-12

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CELP speech coding based on an adaptive pulse position codebook (Abstract)

T. Amada, Kansai Res. Lab., Toshiba Corp., Kobe, Japan

K. Miseki

M. Akamine

pp. 13-16

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A multistage search of algebraic CELP codebooks (Abstract)

M.A. Ramirez, Escola Politecnica, Sao Paulo Univ., Brazil

M. Gerken

pp. 17-20

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A fast search method of algebraic codebook by reordering search sequence (Abstract)

M. Gerken, Inf. Process. Sector, Samsung Adv. Inst. of Technol., Kyungki, South Korea

pp. 21-24

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An 8 kbit/s ACELP coder with improved background noise performance (Abstract)

R. Hagen, Audio & Visual Technol. Res., Ericsson Radar Syst. AB, Stockholm, Sweden

E. Ekudden

pp. 25-28

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On phase perception in speech (Abstract)

H. Pobloth, Dept. of Speech Music & Hearing, R. Inst. of Technol., Stockholm, Sweden

W.B. Kleijn

pp. 29-32

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Progress in Broadcast News transcription at Dragon Systems (Abstract)

S. Wegmann, Dragon Syst. Inc., Newton, MA, USA

Puming Zhan

L. Gillick

pp. 33-36

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Recent improvements to IBM's speech recognition system for automatic transcription of broadcast news (Abstract)

S.S. Chen, IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA

pp. 37-40

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Recent experiments in large vocabulary conversational speech recognition (Abstract)

J. Billa, BBN Technol., Cambridge, MA, USA

T. Colhurst

A. El-Jaroudi

R. Iyer

K. Ma

M. Siu

H. Gish

pp. 41-44

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Large vocabulary speech recognition in French (Abstract)

M. Adda-Decker, Lab. d'Informatique pour la Mecanique et les Sci. de l'Ingenieur, CNRS, Orsay, France

G. Adda

J. Gauvain

L. Lamel

pp. 45-48

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The Cambridge University spoken document retrieval system (Abstract)

S.E. Johnson, Dept. of Eng., Cambridge Univ., UK

pp. 49-52

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Improvements in recognition of conversational telephone speech (Abstract)

B. Peskin, Dragon Syst. Inc., Newton, MA, USA

M. Hunt

pp. 53-56

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The 1998 HTK system for transcription of conversational telephone speech (Abstract)

T. Hain, Dept. of Eng., Cambridge Univ., UK

pp. 57-60

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Real-time telephone-based speech recognition in the Jupiter domain (Abstract)

J.R. Glass, Lab. for Comput. Sci., MIT, Cambridge, MA, USA

T.J. Hazen

I.L. Hetherington

pp. 61-64

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Template-driven generation of prosodic information for Chinese concatenative synthesis (Abstract)

Chung-Hsien Hu, Dept. of Comput. Sci. & Inf. Eng., Nat. Cheng Kung Univ., Tainan, Taiwan

Jau-Hung Chen

pp. 65-68

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Speech enhancement using nonlinear microphone array with complementary beamforming (Abstract)

H. Saruwatari, Graduate Sch. of Eng., Nagoya Univ., Japan

pp. 69-72

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A multivariate speech activity detector based on the syllable rate (Abstract)

D.C. Smith, Dept. of Defense, Fort Meade, MD, USA

pp. 73-76

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Discriminating speakers with vocal nodules using aerodynamic and acoustic features (Abstract)

J. Kuo, AT&T Bell Labs., Murray Hill, NJ, USA

E.B. Holmberg

R.E. Hillman

pp. 77-80

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Enhancement of esophageal speech using formant synthesis (Abstract)

K. Matsui, Central Res. Lab., Matsushita Electr. Ind. Co. Ltd., Seika, Japan

N. Hara

pp. 81-84

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Development of rules for controlling the HLsyn speech synthesizer (Abstract)

H.M. Hanson, Sensimetrics Corp., Somerville, MA, USA

pp. 85-88

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On the characteristics and effects of loudness during utterance production in continuous speech recognition (Abstract)

D. Tapias, Telefonica Investigacion y Desarrollo, Madrid, Spain

C. Cazassus

pp. 89-92

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A multichannel speech/silence detector based on time delay estimation and fuzzy classification (Abstract)

F. Bentelli, Ist. di Inf. e Telecommun., Catania Univ., Italy

S. Casale

A. Cavallaro

pp. 93-96

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Noise suppression using a time-varying, analysis/synthesis gamma chirp filterbank (Abstract)

T. Irino, ATR Human Inf. Process. Res. Lab., Kyoto, Japan

pp. 97-100

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Author Index (Abstract)

pp. a1-a13

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Experimental comparison of signal subspace based noise reduction methods (Abstract)

P.S.K. Hansen, Dept. of Math. Modelling, Tech. Univ., Lyngby, Denmark

pp. 101-104

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Using a large vocabulary continuous speech recognizer for a constrained domain with limited training (Abstract)

M. Siu, BBN Technol./GTE Internetworking, Cambridge, UK

M. Jonas

H. Gish

pp. 105-108

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Initial evaluation of hidden dynamic models on conversational speech (Abstract)

J. Picone, Center for Language & Speech Process., Johns Hopkins Univ., Baltimore, MD, USA

S. Pike

R. Regan

T. Kamm

J. Bridle

L. Deng

Z. Ma

H. Richards

M. Schuster

pp. 109-112

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Convolutional density estimation in hidden Markov models for speech recognition (Abstract)

S. Matsoukas, BBN Technol./GTE Internetworking, Cambridge, UK

G. Zavaliagkos

pp. 113-116

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Automatic clustering and generation of contextual questions for tied states in hidden Markov models (Abstract)

R. Singh, Sch. of Electr. & Comput. Eng., Carnegie Mellon Univ., Pittsburgh, PA, USA

B. Raj

R.M. Stern

pp. 117-120

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Partly hidden Markov model and its application to speech recognition (Abstract)

T. Iobayashi, Waseda Univ., Tokyo, Japan

J. Furuyama

K. Masumitsu

pp. 121-124

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Hidden Markov models with divergence based vector quantized variances (Abstract)

J. Kim, Philips Consumer Commun., Piscataway, NJ, USA

R. Haimi-Cohen

F. Soonge

pp. 125-128

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HMM training based on quality measurement (Abstract)

Yuqing Gao, IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA

pp. 129-132

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Prosodic word boundary detection using statistical modeling of moraic fundamental frequency contours and its use for continuous speech recognition (Abstract)

K. Iwano, Dept. of Inf. & Commun. Eng., Tokyo Univ., Japan

K. Hirose

pp. 133-136

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Voice recognition focusing on vowel strings on a fixed-point 20-MIPS DSP board (Abstract)

Y. Nishida, Speech & Acoust. Lab., NTT Human Interface Labs., Kanagawa, Japan

H. Sato

pp. 137-140

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Speech interface VLSI for car applications (Abstract)

M. Shozakai, LSI Labs., Asahi Chem. Ind. Co. Ltd., Kanagawa, Japan

pp. 141-144

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Recognition of elderly speech and voice-driven document retrieval (Abstract)

S. Anderson, Dragon Syst. Inc., Newton, MA, USA

E. Cate

pp. 145-148

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A comparison of features for speech, music discrimination (Abstract)

M.J. Carey, Ensigma Ltd., Chepstow, UK

E.S. Parris

H. Lloyd-Thomas

pp. 149-152

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Recognizing connected digits in a natural spoken dialog (Abstract)

M. Rahim, AT&T Labs.-Res., Florham Park, NJ, USA

pp. 153-156

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Telephone speech recognition using neural networks and hidden Markov models (Abstract)

DongSuk Yuk, CAIP Center, Rutgers Univ., Piscataway, NJ, USA

J. Flanagan

pp. 157-160

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Improving speech recognition performance by using multi-model approaches (Abstract)

Ji Ming, Sch. of Comput. Sci., Queen's Univ., Belfast, UK

pp. 161-164

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Speaker-dependent name dialing in a car environment with out-of-vocabulary rejection (Abstract)

C.S. Ramalingam, Media Technol. Lab., Texas Instrum. Inc., Dallas, TX, USA

pp. 165-168

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An new method used in HMM for modeling frame correlation (Abstract)

Guo Qing, Dept. of Comput. Sci. & Technol., Tsinghua Univ., Beijing, China

Zheng Fang

Wu Jian

Wu Wenhu

pp. 169-172

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N-best based supervised and unsupervised adaptation for native and non-native speakers in cars (Abstract)

P. Nguyen, Speech Technol. Lab., Panasonic Technol. Inc., Santa Barbara, CA, USA

pp. 173-176

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Performance assessment of tandem connection of enhanced cellular coders (Abstract)

S.F. Campos Neto, COMSAT Lab., Clarksburg, MD, USA

F.L. Corcoran

A. Karahisar

pp. 177-180

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TTS based very low bit rate speech coder (Abstract)

Ki-Seung Lee, AT&T Labs.-Res., Florham Park, NJ, USA

R.V. Cox

pp. 181-184

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Wideband speech coding with toll quality based on IA-model (Abstract)

R.V. Cox, Sch. of Electr. & Electron. Eng., Nanyang Technol. Univ., Singapore

Gang Li

Xiao Lin

Guoan Bi

pp. 185-188

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4 kb/s multi-pulse based CELP speech coding using excitation switching (Abstract)

K. Ozawa, C&C Media Res. Labs., NEC Corp., Kanagawa, Japan

pp. 189-192

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An adaptive multi-rate speech coder for digital cellular telephony (Abstract)

E. Paksoy, DSP Solutions R&D Center, Texas Instrum. Inc., Dallas, TX, USA

pp. 193-196

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An adaptive post-filtering technique based on the modified Yule-Walker filter (Abstract)

A. Mustapha, COMSAT Lab., Clarksburg, MD, USA

S. Yeldener

pp. 197-200

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A modular approach to speech enhancement with an application to speech coding (Abstract)

A.J. Accardi, AT&T Labs.-Res., Florham Park, NJ, USA

R.V. Cox

pp. 201-204

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On speech coding in a perceptual domain (Abstract)

G. Kubin, Tech. Univ. Wien, Austria

W. Bastiaan Kleijn

pp. 205-208

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Speech analysis/synthesis/conversion by using sequential processing (Abstract)

B. Panuthat, Fac. of Eng., Kanazawa Univ., Japan

T. Funada

N. Kanedera

pp. 209-212

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Modelling energy flow in the vocal tract with applications to glottal closure and opening detection (Abstract)

D.M. Brookes, Dept. of Electr. & Electron. Eng., Imperial Coll. of Sci., Technol. & Med., London, UK

H.P. Loke

pp. 213-216

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Fitting the Mel scale (Abstract)

S. Umesh, Dept. of Electr. Eng., Indian Inst. of Technol., Kanpur, India

L. Cohen

D. Nelson

pp. 217-220

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Fast accent identification and accented speech recognition (Abstract)

D. Nelson, Dept. of Electr. & Electron. Eng., Univ. of Sci. & Technol. of China, Hefei, China

P. Fung

pp. 221-224

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Relevancy of time-frequency features for phonetic classification measured by mutual information (Abstract)

H. Yang, Dept. of Comput. Sci. & Eng., Oregon Graduate Inst. of Sci. & Technol., Beaverton, OR, USA

S. van Vuuren

H. Hermansky

pp. 225-228

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Hidden Markov models based on multi-space probability distribution for pitch pattern modeling (Abstract)

K. Tokuda, Dept. of Comput. Eng., Nagoya Inst. of Technol., Japan

pp. 229-232

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An algorithm for glottal volume velocity estimation (Abstract)

A. Alkhairy, MIT, Cambridge, MA, USA

pp. 233-236

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Frame level noise classification in mobile environments (Abstract)

K. El-Maleh, Dept. of Electr. & Comput. Eng., McGill Univ., Montreal, Que., Canada

A. Samouelian

P. Kabal

pp. 237-240

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Low delay multi-level decomposition and quantisation techniques for WI coding (Abstract)

N.R. Chong, Whisper Labs., Wollongong Univ., NSW, Australia

I.S. Burnett

J.F. Chicharo

pp. 241-244

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An improved mixed excitation linear prediction (MELP) coder (Abstract)

T. Unno, Center for Signal & Image Process., Georgia Inst. of Technol., Atlanta, GA, USA

T. Unno

Kwan Truong

pp. 245-248

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Split band LPC based adaptive multi-rate GSM candidate (Abstract)

S. Villette, Centre for Commun. Syst. Res., Surrey Univ., Guildford, UK

M. Stefanovic

A. Kondoz

pp. 249-252

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Frequency-domain spectral envelope estimation for low rate coding of speech (Abstract)

M. Jelinek, Sherbrooke Univ., Que., Canada

J.-P. Adoul

pp. 253-256

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Robust closed-loop pitch estimation for harmonic coders by time scale modification (Abstract)

Chunyan Li, Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA, USA

V. Cuperman

A. Gersho

pp. 257-260

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Phase adjustment in waveform interpolation (Abstract)

Hong-Goo Kang, AT&T Labs.-Res., Florham Park, NJ, USA

D. Sen

pp. 261-264

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A low resolution pulse position coding method for improved excitation modeling of speech transition (Abstract)

Jongseo Sohn, Sch. of Electr. Eng., Seoul Nat. Univ., South Korea

Wonyong Sung

pp. 265-268

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Dispersion phase vector quantization for enhancement of waveform interpolative coder (Abstract)

O. Gottesmann, Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA, USA

pp. 269-272

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The Teager energy based feature parameters for robust speech recognition in car noise (Abstract)

F. Jabloun, Dept. of Electr. Eng., Bilkent Univ., Ankara, Turkey

A. Enis Cetin

pp. 273-276

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Avoiding distortions due to speech coding and transmission errors in GSM ASR tasks (Abstract)

A. Gallardo-Antolin, Dept. de Tecnologias de las Commun., Univ. Carlos III de Madrid, Spain

F. Diaz-de-Maria

F. Valverde-Albacete

pp. 277-280

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Binaural bark subband pre-processing of nonstationary signals for noise robust speech feature extraction (Abstract)

M. Peters, Res. & Dev. Centre, BMW AG, Munich, Germany

pp. 281-284

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Speaker normalized spectral subband parameters for noise robust speech recognition (Abstract)

S. Tsuge, ATR Interpreting Telephony Res. Labs., Kyoto, Japan

T. Fukuda

H. Singer

pp. 285-288

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Temporal patterns (TRAPs) in ASR of noisy speech (Abstract)

H. Hermansky, Graduate Inst. of Sci. & Technol., Portland, OR, USA

S. Sharma

pp. 289-292

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Signal modeling for isolated word recognition (Abstract)

M. Karnjanadecha, Dept. of Electr. & Comput. Eng., Old Dominion Univ., Norfolk, VA, USA

S.A. Zahorian

pp. 293-296

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Transforming HMMs for speaker-independent hands-free speech recognition in the car (Abstract)

Y. Gong, Speech Res., Media Technols. Lab., Texas Instrum. Inc., Dallas, TX, USA

J.J. Godfrey

pp. 297-300

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Channel and noise adaptation via HMM mixture mean transform and stochastic matching (Abstract)

J.J. Godfrey, Dept. of Electr. & Electron. Eng., Hong Kong Univ. of Sci. & Technol., Clearwater Bay, Hong Kong

Bertram Shi

pp. 301-304

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Speaker verification performance and the length of test sentence (Abstract)

Jialong He, Dept. of Spech & Hearing Sci., Arizona State Univ., Tempe, AZ, USA

Li Liu

pp. 305-308

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On the use of some divergence measures in speaker recognition (Abstract)

R. Vergin, Moncton Univ., NB, Canada

D. O'Shaughnessy

pp. 309-312

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Improving a GMM speaker verification system by phonetic weighting (Abstract)

R. Auckenthaler, Ensigma Ltd., Chepstow, UK

E.S. Parris

M.J. Carey

pp. 313-316

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A hybrid score measurement for HMM-based speaker verification (Abstract)

Yong Gu, Vocalis Ltd., Cambridge, UK

T. Thomas

pp. 317-320

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Polynomial classifier techniques for speaker verification (Abstract)

W.M. Campbell, SSG, Motorola Inc., Scottsdale, AZ, USA

K.T. Assaleh

pp. 321-324

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Channel-robust speaker identification using modified-mean cepstral mean normalization with frequency warping (Abstract)

A.A. Garcia, SpeakEZT-NETIX Inc., Piscataway, NJ, USA

R.J. Mammone

pp. 325-328

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Feature selection using genetics-based algorithm and its application to speaker identification (Abstract)

M. Demirekler, EE Eng. Dept., METU, Ankaraa, Turkey

A. Haydar

pp. 329-332

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Frame discrimination training for HMMs for large vocabulary speech recognition (Abstract)

D. Povey, Dept. of Eng., Cambridge Univ., UK

P.C. Woodland

pp. 333-336

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Discriminative mixture weight estimation for large Gaussian mixture models (Abstract)

F. Beaufays, Speech Technol. & Res. Lab., SRI Int., Menlo Park, CA, USA

M. Weintraub

Yochai Konig

pp. 337-340

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Modeling disfluency and background events in ASR for a natural language understanding task (Abstract)

R.C. Rose, Res. Dept., AT&T Bell Labs., Florham Park, NJ, USA

G. Riccardi

pp. 341-344

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Decision tree state tying based on penalized Bayesian information criterion (Abstract)

Wu Chou, Lucent Technols., Bell Labs., Murray Hill, NJ, USA

W. Reichl

pp. 345-348

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A 2D extended HMM for speech recognition (Abstract)

Jiayu Li, Dept. of Stat., Chicago Univ., IL, USA

A. Murua

pp. 349-352

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Probabilistic classification of HMM states for large vocabulary continuous speech recognition (Abstract)

Xiaoqiang Luo, Dept. of Electr. & Comput. Eng., Johns Hopkins Univ., Baltimore, MD, USA

F. Jelinek

pp. 353-356

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The HDM: a segmental hidden dynamic model of coarticulation (Abstract)

H.B. Richards, Dragon Syst. UK, Cheltenham, UK

J.S. Bridle

pp. 357-360

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Maximum likelihood estimates for exponential type density families (Abstract)

S. Basu, IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA

C.A. Micchelli

P.A. Olsen

pp. 361-364

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On the limits of speech recognition in noise (Abstract)

S.D. Peters, Nortel Technol., Montreal, Que., Canada

P. Stubley

J.-M. Valin

pp. 365-368

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Recognition of spectrally degraded speech in noise with nonlinear amplitude mapping (Abstract)

Qian-Jie Fu, Dept. of Auditory Implants & Perception, House Ear Inst., Los Angeles, CA, USA

R.V. Shannon

pp. 369-372

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Phrase splicing and variable substitution using the IBM trainable speech synthesis system (Abstract)

R.E. Donovan, IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA

pp. 373-376

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Assessment and correction of voice quality variabilities in large speech databases for concatenative speech synthesis (Abstract)

Y. Stylianou, SIPS, AT&T Bell Labs., Florham Park, NJ, USA

pp. 377-380

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Shape invariant time-scale modification of speech using a harmonic model (Abstract)

D. O'Brien, Sch. of Comput. Applications, Dublin City Univ., Ireland

A. Monaghan

pp. 381-384

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Using a sigmoid transformation for improved modeling of phoneme duration (Abstract)

K.E.A. Silverman, Spoken Language Group, Apple Comput. Inc., Cupertino, CA, USA

J.R. Bellegarda

pp. 385-388

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Nonlinear dynamic modeling of the voiced excitation for improved speech synthesis (Abstract)

K. Narasimhan, Comput. Neuroeng. Lab., Florida Univ., Gainesville, FL, USA

J.C. Principe

D.G. Childers

pp. 389-392

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Results on perceptual invariants to transformations on speech (Abstract)

A. Robert, Dept. of Electr. Eng., Fed. Inst. of Technol., Lausanne, Switzerland

pp. 393-396

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Investigations on inter-speaker variability in the feature space (Abstract)

R. Haeb-Umbach, Philips Res. Lab., Aachen, Germany

pp. 397-400

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LSP weighting functions based on spectral sensitivity and mel-frequency warping for speech recognition in digital communication (Abstract)

R. Haeb-Umbach, Dept. of Electr. Eng., Korea Adv. Inst. of Sci. & Technol., Seoul, South Korea

R. Haeb-Umbach

pp. 401-404

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Two-dimensional multi-resolution analysis of speech signals and its application to speech recognition (Abstract)

C.P. Chan, Dept. of Electron. Eng., Chinese Univ. of Hong Kong, Shatin, Hong Kong

Y.W. Wong

Tan Lee

P.C. Ching

pp. 405-408

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Hierarchical subband linear predictive cepstral (HSLPC) features for HMM-based speech recognition (Abstract)

R. Chengalvarayan, Speech Process. Group, Lucent Technol., Naperville, IL, USA

pp. 409-412

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Towards a robust/fast continuous speech recognition system using a voiced-unvoiced decision (Abstract)

D. O'Shaughnessy, INRS-Telecommun., Quebec Univ., Verdun, Que., Canada

H. Tolba

pp. 413-416

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A C/V segmentation algorithm for Mandarin speech signal based on wavelet transforms (Abstract)

H. Tolba, Dept. of Electr. Eng., Nat. Cheng Kung Univ., Tainan, Taiwan

Shi-Huang Chen

pp. 417-420

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Feature extraction for speech recognition based on orthogonal acoustic-feature planes and LDA (Abstract)

T. Nitta, Multimeda Eng. Lab., Toshiba Corp., Kawasaki, Japan

pp. 421-424

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Distinctive feature detection using support vector machines (Abstract)

P. Niyogi, Bell Labs., Lucent Technol., USA

C. Burges

P. Ramesh

pp. 425-428

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Time-varying noise compensation using multiple Kalman filters (Abstract)

P. Ramesh, Dept. of Electr. Eng., Seoul Nat. Univ., South Korea

pp. 429-432

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A segment-based C/sub 0/ adaptation scheme for PMC-based noisy Mandarin speech recognition (Abstract)

Wei-Tyng Hong, Dept. of Commun. Eng., Nat. Chiao Tung Univ., Hsinchu, Taiwan

Sin-Horng Chen

pp. 433-436

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Improved parallel model combination techniques with split Gaussian mixtures for speech recognition under noisy conditions (Abstract)

Jeih-weih Hung, Inst. of Inf. Sci., Acad. Sinica, Taipei, Taiwan

Jia-lin Shen

Lin-shan Lee

pp. 437-440

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Speech recognition and enhancement by a nonstationary AR HMM with gain adaptation under unknown noise (Abstract)

G. Ruske, Inst. for Human-Machine-Commun., Munich Univ. of Technol., Germany

G. Ruske

pp. 441-444

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Database and online adaptation for improved speech recognition in car environments (Abstract)

A. Fischer, Philips Res. Lab., Aachen, Germany

V. Stahl

pp. 445-448

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Training of HMM with filtered speech material for hands-free recognition (Abstract)

D. Giuliani, ITC-IRST, Centro per la Ricerca Sci. e Technol., Trento, Italy

pp. 449-452

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Incremental enrolment of speech recognizers (Abstract)

C. Mokbel, CNET, Lannion, France

O. Collin

pp. 453-456

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Automatic speech recognition: a communication perspective (Abstract)

B.S. Atal, AT&T Labs., Florham Park, NJ, USA

pp. 457-460

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Split band CELP (SB-CELP) speech coder (Abstract)

M.R. Nakhai, Dept. of Electron. & Electr. Eng., King's Coll., London, UK

F.A. Marvasti

pp. 461-464

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Log amplitude modeling of sinusoids in voiced speech (Abstract)

N. Malik, Sch. of Electr. & Telecommun. Eng., New South Wales Univ., Sydney, NSW, Australia

W.H. Holmes

pp. 465-468

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1.2 kbit/s harmonic coder using auditory filters (Abstract)

M. Kohata, Chiba Inst. of Technol., Narashino, Japan

pp. 469-472

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Exponential sinusoidal modeling of transitional speech segments (Abstract)

J. Jensen, Center for PeronKommunikaion, Aalborg Univ., Denmark

S.H. Jensen

E. Hansen

pp. 473-476

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Harmonic+noise coding using improved V/UV mixing and efficient spectral quantization (Abstract)

E.W.M. Yu, Dept. of Electron. Eng., City Univ. of Hong Kong, Kowloon, Hong Kong

Cheung-Fat Chan

pp. 477-480

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A 4 kb/s toll quality harmonic excitation linear predictive speech coder (Abstract)

S. Yeldener, COMSAT Lab., Clarksburg, MD, USA

pp. 481-484

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High quality MELP coding at bit-rates around 4 kb/s (Abstract)

J. Stachurski, Texas Instrum. Inc., Dallas, TX, USA

A. McCree

V. Viswanathan

pp. 485-488

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Pitch quantization in low bit-rate speech coding (Abstract)

T. Eriksson, AT&T Labs. Res., Florham Park, NJ, USA

Hong-Goo Kang

pp. 489-492

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Incorporating confidence measures in the Dutch train timetable information system developed in the ARISE project (Abstract)

G. Bouwman, Nijmegen Univ., Netherlands

J. Sturm

L. Boves

pp. 493-496

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HMM and neural network based speech act detection (Abstract)

K. Ries, Language Tech. Inst., Carnegie Mellon Univ., Pittsburgh, PA, USA

pp. 497-500

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The LIMSI ARISE system for train travel information (Abstract)

L. Lamel, CNRS, Orsay, France

pp. 501-504

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Improving the suitability of imperfect transcriptions for information retrieval from spoken documents (Abstract)

M. Siegler, Dept. of Electr. & Comput. Eng., Carnegie Mellon Univ., Pittsburgh, PA, USA

M. Withrock

pp. 505-508

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Automatic topic identification for two-level call routing (Abstract)

J. Golden, BBN Syst. & Technol. Corp., Cambridge, MA, USA

O. Kimball

Man-Hung Siu

H. Gish

pp. 509-512

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Named entity tagged language models (Abstract)

Y. Gotoh, Dept. of Comput. Sci., Sheffield Univ., UK

S. Renals

G. Williams

pp. 513-516

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Speech translation: coupling of recognition and translation (Abstract)

H. Ney, Lehrstuhl fur Inf. VI, Tech. Hochschule Aachen, Germany

pp. 517-520

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Probabilistic models for topic detection and tracking (Abstract)

F. Walls, BBN Syst. & Technol. Corp., Cambridge, MA, USA

H. Jin

S. Sista

R. Schwartz

pp. 521-524

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Discriminative estimation of interpolation parameters for language model classifiers (Abstract)

V. Warnke, Erlangen-Nurnberg Univ., Germany

E. Noth

pp. 525-528

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Combination of words and word categories in varigram histories (Abstract)

R. Blasig, Philips Res. Lab., Aachen, Germany

pp. 529-532

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Multi-class composite N-gram based on connection direction (Abstract)

H. Yamamoto, ATR Interpreting Telephony Res. Labs., Kyoto, Japan

Y. Sagisaka

pp. 533-536

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A class-based language model for large-vocabulary speech recognition extracted from part-of-speech statistics (Abstract)

C. Samuelsson, AT&T Bell Labs., Murray Hill, NJ, USA

W. Reichl

pp. 537-540

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Improved topic-dependent language modeling using information retrieval techniques (Abstract)

M. Mahajan, Microsoft Corp., Redmond, WA, USA

D. Beeferman

X.D. Huang

pp. 541-544

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Smoothing methods in maximum entropy language modeling (Abstract)

S.C. Martin, Lehrstuhl fur Inf., Tech. Hochschule Aachen, Germany

H. Ney

J. Zaplo

pp. 545-548

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Efficient sampling and feature selection in whole sentence maximum entropy language models (Abstract)

S.F. Chen, Sch. of Comput. Sci., Carnegie Mellon Univ., Pittsburgh, PA, USA

R. Rosenfeld

pp. 549-552

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A maximum entropy language model integrating N-grams and topic dependencies for conversational speech recognition (Abstract)

S. Khudanpur, Center for Language & Speech Process., Johns Hopkins Univ., Baltimore, MD, USA

Jun Wu

pp. 553-556

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1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258) (Abstract)

pp. i-556

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