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Acoustics, Speech, and Signal Processing, IEEE International Conference on (1996)
Atlanta, GA, USA
May 7, 1996 to May 10, 1996
ISBN: 0-7803-3192-3
TABLE OF CONTENTS

Authors Index (Abstract)

pp. A1

Discriminative training of stochastic Markov graphs for speech recognition (Abstract)

F. Wolfertstetter , Inst. for Human-Machine-Commun., Munich Univ. of Technol., Germany
pp. 581-584

Incremental ML estimation of HMM parameters for efficient training (Abstract)

Y. Gotoh , Div. of Eng., Brown Univ., Providence, RI, USA
pp. 585-588

Improved HMM training and scoring strategies with application to accent classification (Abstract)

L.M. Arslan , Robust Speech Processing Lab., Duke Univ., Durham, NC, USA
pp. 589-592

Maximum likelihood successive state splitting (Abstract)

H. Singer , ATR Interpreting Telephony Res. Labs., Kyoto, Japan
pp. 601-604

Training algorithms for a predictive classifier (Abstract)

P.V.S. Rao , Comput. Syst. & Commun. Group, Tata Inst. of Fundamental Res., Bombay, India
pp. 609-612

Stochastic observation hidden Markov models (Abstract)

C.D. Mitchell , AT&T Bell Labs., Murray Hill, NJ, USA
pp. 617-620

Robust noise reduction for speech and audio signals (Abstract)

S.J. Godsill , Dept. of Eng., Cambridge Univ., UK
pp. 625-628

Noise model adaptation in model based speech enhancement (Abstract)

B.L. McKinley , Signal Processing Consultants, Fairfax, VA, USA
pp. 633-636

Selective all-pole modeling of degraded speech using M-band decomposition (Abstract)

C.D. Yoo , Res. Lab. of Electron., MIT, Cambridge, MA, USA
pp. 641-644

Multi-channel signal separation (Abstract)

D.C.B. Chan , Dept. of Eng., Cambridge Univ., UK
pp. 649-652

Signal modeling for speaker identification (Abstract)

Li Liu , Ulm Univ., Germany
pp. 665-668

An HMM approach to text-prompted speaker verification (Abstract)

ChiWei Che , CAIP Center, Rutgers Univ., Piscataway, NJ, USA
pp. 673-676

Text-independent speaker identification (Abstract)

M. Birnbaum , Lockheed Martin Co., Nashua, NH, USA
pp. 677-680

Language independent gender identification (Abstract)

E.S. Parris , Ensigma Ltd., Chepstow, UK
pp. 685-688

Fine structure features for speaker identification (Abstract)

M.J. Carey , Lincoln Lab., MIT, Lexington, MA, USA
pp. 689-692

Speaker verification with low storage requirements (Abstract)

J. Schalkwyk , Center for Spoken Language Understanding, Oregon Graduate Inst. of Sci. & Technol., Portland, OR, USA
pp. 693-696

A Markov random field approach to Bayesian speaker adaptation (Abstract)

B.M. Shahshahani , Nuance Commun., Manlo Park, CA, USA
pp. 697-700

A study of on-line quasi-Bayes adaptation for CDHMM-based speech recognition (Abstract)

Qiang Huo , ATR Interpreting Telephony Res. Labs., Kyoto, Japan
pp. 705-708

Speaker time-drifting adaptation using trajectory mixture hidden Markov models (Abstract)

Jian Su , Dept. of Electron. Eng., City Univ. of Hong Kong, Hong Kong
pp. 709-712

An experimental study of acoustic adaptation algorithms (Abstract)

A. Sankar , Speech Technol. & Res. Lab., SRI Int., Menlo Park, CA, USA
pp. 713-716

Speaker adaptation with autonomous model complexity control by MDL principle (Abstract)

K. Shinoda , Inf. Technol. Res. Labs., NEC Corp., Kawasaki, Japan
pp. 717-720

An approach to speaker adaptation based on analytic functions (Abstract)

J. McDonough , BBN Syst. & Technol. Corp., Cambridge, MA, USA
pp. 721-724

Maximum a posteriori adaptation for large scale HMM recognizers (Abstract)

G. Zavaliagkos , BBN Syst. & Technol. Corp., Cambridge, MA, USA
pp. 725-728

Acoustic adaptation using nonlinear transformations of HMM parameters (Abstract)

V. Abrash , Speech Res. & Technol. Lab., SRI Int., Menlo Park, CA, USA
pp. 729-732

A vector Taylor series approach for environment-independent speech recognition (Abstract)

P.J. Moreno , Dept. of Electr. & Comput. Eng., Carnegie Mellon Univ., Pittsburgh, PA, USA
pp. 733-736

Linked split-vector quantizer of LPC parameters (Abstract)

Jianping Pan , Samsung Adv. Inst. of Technol., KyungKi-Do, South Korea
pp. 741-744

Scalar quantization using vector measure with application to quantization of LSF parameters (Abstract)

H.C. Ng , Dept. of Electron. Eng., City Univ. of Hong Kong, Kowloon Tong, Hong Kong
pp. 745-748

A robust LSP encoding scheme for noisy channel (Abstract)

Junchen Du , Dept. of Electr. Eng., Polytech. Univ., Brooklyn, NY, USA
pp. 749-752

An efficient coding of LSP parameters using multiple type frame segmentation (Abstract)

Y.K. Lee , LG Electron. Res. Centre, Inf. & Technol. Lab., Seoul, South Korea
pp. 753-756

Switched prediction and quantization of LSP frequencies (Abstract)

H. Zarrinkoub , INRS-Telecommun., Quebec Univ., Nuns' Island, Que., Canada
pp. 757-760

Classified nonlinear predictive vector quantization of speech spectral parameters (Abstract)

J.H.Y. Loo , Dept. of Electr. Eng., McGill Univ., Montreal, Que., Canada
pp. 761-764

Exploiting interframe correlation in spectral quantization: a study of different memory VQ schemes (Abstract)

T. Eriksson , Dept. of Inf. Theory, Chalmers Univ. of Technol., Goteborg, Sweden
pp. 765-768

A hybrid input/output spectrum adaptation scheme for LD-CELP coding of speech (Abstract)

Siu-Pun Chui , Dept. of Electron. Eng., City Univ. of Hong Kong, Kowloon, Hong Kong
pp. 773-776

LVCSR-based language identification (Abstract)

T. Schultz , Interactive Syst. Labs., Karlsruhe Univ., Germany
pp. 781-784

Experiments for an approach to language identification with conversational telephone speech (Abstract)

Yonghong Yan , Center for Spoken Language Understanding, Oregon Graduate Inst. of Sci. & Technol., Portland, OR, USA
pp. 789-792

Statistical language identification based on untranscribed training (Abstract)

M.A. Lund , BBN Syst. & Technols., Cambridge, MA, USA
pp. 793-796

A model for efficient formant estimation (Abstract)

L. Welling , Lehrstuhl fur Inf. VI, Tech. Hochschule Aachen, Germany
pp. 797-800

Improved vocal tract model for the analysis of nasal speech sounds (Abstract)

Minsheng Liu , Inst. fur Angewandte Phys., Frankfurt Univ., Germany
pp. 801-804

A scalar homotopy method for parallel and robust tracking of line spectral pairs (Abstract)

U. Pillai , Dept. of Electr. & Comput. Eng., Carnegie Mellon Univ., Pittsburgh, PA, USA
pp. 805-808

A robust method for the estimation of formant frequency modulation in speech signals (Abstract)

P. Rao , Dept. of Electr. Eng., Indian Inst. of Technol., Kanpur, India
pp. 813-816

Visual speech recognition using active shape models and hidden Markov models (Abstract)

J. Luettin , Dept. of Electron. & Electr. Eng., Sheffield Univ., UK
pp. 817-820

An estimation of speaker sampling in Voice Across Japan database (Abstract)

I. Kudo , Tsukuba Res. & Dev. Center Ltd., Texas Instrum. Inc., Ibaraki, Japan
pp. 825-828

Adaptive bimodal sensor fusion for automatic speechreading (Abstract)

U. Meier , Interactive Syst. Labs., Karlsruhe Univ., Germany
pp. 833-836

Multi channel HMM (Abstract)

Dongkin Xu , Comput. Neuroeng. Lab., Florida Univ., Gainesville, FL, USA
pp. 841-844

The contribution of consonants versus vowels to word recognition in fluent speech (Abstract)

R.A. Cole , Center for Spoken Language Understanding, Oregon Graduate Inst. of Sci. & Technol., Portland, OR, USA
pp. 853-856

A novel word pre-selection method based on phonetic set indexing (Abstract)

R.R. Sarukkai , Dept. of Comput. Sci., Rochester Univ., NY, USA
pp. 857-860

DP-based wordgraph pruning (Abstract)

T. Kuhn , Dept. of Res. & Technol., Daimler-Benz AG, Ulm, Germany
pp. 861-864

A new hybrid system based on MMI-neural networks for the RM speech recognition task (Abstract)

G. Rigoll , Dept. of Comput. Sci., Gerhard-Mercator-Univ. Duisberg, Germany
pp. 865-868

A dependence tree model of phone correlation (Abstract)

O. Ronen , Dept. of Electron. & Comput. Sci., Boston Univ., MA, USA
pp. 873-876

The use of syllable phonotactics for word hypothesization (Abstract)

R. De Mori , McGill Univ., Montreal, Que., Canada
pp. 877-880

Creating speaker-specific phonetic templates with a speaker-independent phonetic recognizer: implications for voice dialing (Abstract)

N. Jain , Center for Spoken Language Understanding, Oregon Graduate Inst. of Sci. & Technol., Portland, OR, USA
pp. 881-884

A time continuous model for speech recognition (Abstract)

S. Euler , Bosch Telecom, Frankfurt, Germany
pp. 889-892

A localization-error-based method for microphone-array design (Abstract)

M.S. Brandstein , Lab. for Eng. Man/Machine Syst., Brown Univ., Providence, RI, USA
pp. 901-904

Nearfield broadband frequency invariant beamforming (Abstract)

R.A. Kennedy , Telecommun. Eng., Australian Nat. Univ., Canberra, ACT, Australia
pp. 905-908

A source subspace tracking array of microphones for double talk situations (Abstract)

S. Affes , INRS-Telecommun., Verdun, Que., Canada
pp. 909-912

Acoustic source location in noisy and reverberant environment using CSP analysis (Abstract)

M. Omologo , Istituto per la Ricerca Sci. e Tecnologica, Trento, Italy
pp. 921-924

Voice-activated AGC for teleconferencing (Abstract)

P.L. Chu , PictureTel Corp., Danvers, MA, USA
pp. 929-932

A two-microphone adaptive broadband array for hearing aids (Abstract)

E.D. McKinney , Sch. of Electr. Eng., Oklahoma Univ., Norman, OK, USA
pp. 933-936

Optimization of a noise reduction preprocessing in an acoustic echo and noise controller (Abstract)

B. Ayad , Lab. Traitement du Signal et de l'Image, Rennes I Univ., France
pp. 953-956

Dynamically regularized fast RLS with application to echo cancellation (Abstract)

S.L. Gay , Dept. of Acoust. Res., AT&T Bell Labs., Murray Hill, NJ, USA
pp. 957-960

Experimental results of subband acoustic echo cancelers under spherically invariant random processes (Abstract)

P.L. De Leon , Dept. of Electr. & Comput. Eng., New Mexico State Univ., Las Cruces, NM, USA
pp. 961-964

The effect of structured uncertainty in multichannel feedforward control systems (Abstract)

A. Omoto , Dept. of Acoust. Design, Kyushu Inst. of Design, Fukuoka, Japan
pp. 965-968

Optimisation of controlled acoustic shadows (Abstract)

S.E. Wright , Sch. of Eng., Huddersfield Polytech., UK
pp. 973-976

AM-FM separation using auditory-motivated filters (Abstract)

T.F. Quatieri , Lincoln Lab., MIT, Lexington, MA, USA
pp. 977-980

A 'gammachirp' function as an optimal auditory filter with the Mellin transform (Abstract)

I. Toshio , NTT Basic Res. Labs., Kanagawa, Japan
pp. 981-984

A binaural auditory model for sound quality measurements and spatial hearing studies (Abstract)

M. Karajalainen , Lab. of Acoustics & Audio Signal Process., Helsinki Univ. of Technol., Espoo, Finland
pp. 985-988

A study on auditory resolution using Bark-FAMlet clicks (Abstract)

U.K. Laine , Acoust. Lab., Helsinki Univ. of Technol., Espoo, Finland
pp. 989-992

Real-time discrimination of broadcast speech/music (Abstract)

J. Saunders , Lockheed Martin Co., Nashua, NH, USA
pp. 993-996

Analysis and resynthesis of musical instrument sounds using energy separation (Abstract)

R.B. Sussman , CAIP Center, Rutgers Univ., Piscataway, NJ, USA
pp. 997-1000

Automatic audio morphing (Abstract)

M. Slaney , Interval Res. Corp., Palo Alto, CA, USA
pp. 1001-1004

Residual modeling in music analysis-synthesis (Abstract)

M. Goodwin , CNMAT, California Univ., Berkeley, CA, USA
pp. 1005-1008

Computationally efficient algorithm for time scale modification (GLS-TSM) (Abstract)

S. Yim , Tsukuba Res. & Dev., Texas Inst., Japan
pp. 1009-1012

Multichannel dynamic range compression for music signals (Abstract)

J.C. Schmidt , Dept. of Electr. Eng. & Comput. Sci., Northwestern Univ., Evanston, IL, USA
pp. 1013-1016

A bi-dimensional coding scheme applied to audio bitrate reduction (Abstract)

L. Mainard , CCETT, Cesson Sevigne, France
pp. 1017-1020

A test of MPEG using time-inverted spoken audio (Abstract)

T. McLaughlin , Nat. Libr. Service for the Blind & Physically Handicapped, Washington, DC, USA
pp. 1025-1028

Extension and complexity reduction of TwinVQ audio coder (Abstract)

T. Moriya , NTT Human Interface Labs., Tokyo, Japan
pp. 1029-1032

Minimising the effects of subband quantisation of the time domain aliasing cancellation filter bank (Abstract)

C. Jakob , Dept. of Commun. & Electr. Eng., R. Melbourne Inst. of Technol., Vic., Australia
pp. 1033-1036

Speech analysis and coding using a multi-resolution sinusoidal transform (Abstract)

D.V. Anderson , Sch. of Electr. & Compu. Eng., Georgia Inst. of Technol., Atlanta, GA, USA
pp. 1037-1040

Audio coding using the wavelet packet transform and a combined scalar-vector quantization (Abstract)

S. Boland , Signal Process. Res. Centre, Queensland Univ., Brisbane, Qld., Australia
pp. 1041-1044

Low bit rate high quality audio coding with combined harmonic and wavelet representations (Abstract)

K.N. Hamdy , Dept. of Electr. Eng., Minnesota Univ., Minneapolis, MN, USA
pp. 1045-1048

A high performance software implementation of MPEG audio encoder (Abstract)

M. Kumar , IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA
pp. 1049-1052

Signal processing techniques for efficient use of transmit diversity in wireless communications (Abstract)

G.W. Wornell , Dept. of Electr. Eng. & Comput. Sci., MIT, Cambridge, MA, USA
pp. 1057-1060

Adaptive suppression of narrowband digital interferers from spread spectrum signals (Abstract)

H.V. Poor , Dept. of Electr. Eng., Princeton Univ., NJ, USA
pp. 1061-1064

Signal processing for interference suppression in direct-sequence CDMA systems (Abstract)

U. Madhow , Coordinated Sci. Lab., Illinois Univ., Urbana, IL, USA
pp. 1065-1068

Digital video in a fading interference wireless environment (Abstract)

L.C. Yun , Dept. of Electr. Eng. & Comput. Sci., California Univ., Berkeley, CA, USA
pp. 1069-1072

Singular value analysis of space-time equalization in the GSM mobile system (Abstract)

A.L. van der Veen , Dept. of Electr. Eng., Delft Univ. of Technol., Netherlands
pp. 1073-1076

Geometric properties of the blind digital co-channel communications problem (Abstract)

L.K. Hansen , Dept. of Electr. & Comput. Eng., Texas Univ., Austin, TX, USA
pp. 1085-1088

Estimation and equalization of fading channels with random coefficients (Abstract)

M.K. Tsatsanis , Dept. of Electr. Eng. & Comput. Sci., Stevens Inst. of Technol., Hoboken, NJ, USA
pp. 1093-1096

Real-time DSP for sophomores (Abstract)

K.H. Chiang , Dept. of Electr. Eng. & Comput. Sci., California Univ., Berkeley, CA, USA
pp. 1097-1100

Multi-media and World Wide Web resources for teaching DSP (Abstract)

J.H. McClellan , Sch. of Electr. & Comput. Eng., Georgia Inst. of Technol., Atlanta, GA, USA
pp. 1101-1104

Developing Internet-based VHDL course material (Abstract)

J. Calhoun , Microsyst. Prototyping Lab., Eng. Res. Center, Mississippi State, MS, USA
pp. 1105-1108

Distance teaming experiments in undergraduate DSP (Abstract)

D.M. Etter , Dept. of Electr. & Comput. Eng., Colorado Univ., Boulder, CO, USA
pp. 1109-1112

A system characterization/identification laboratory teaching tool for Internet (Abstract)

S. Chatfield , Telecommun. Res. Center, Arizona State Univ., Tempe, AZ, USA
pp. 1113-1116

An integrated environment for modeling, simulation, digital signal processing, and control (Abstract)

C.C. Crane , Dept. of Electr. Eng., Bucknell Univ., Lewisburg, PA, USA
pp. 1121-1124

Spectra-a hands-on DSP learning experience (Abstract)

F. Taylor , Florida Univ., Gainesville, FL, USA
pp. 1125-1128

A laboratory course for designing and testing spoken dialogue systems (Abstract)

D. Colton , Center for Spoken Language, Oregon Graduate Inst. of Sci. & Technol., Portland, OR, USA
pp. 1129-1132

A MATLAB software tool for the introduction of speech coding fundamentals in a DSP course (Abstract)

E. Painter , Dept. of Electr. Eng., Arizona State Univ., Tempe, AZ, USA
pp. 1133-1136

Vocoder intelligibility and quality test methods (Abstract)

J.D. Tardelli , Arcon Corp., Waltham, MA, USA
pp. 1145-1148

Speaker recognizability testing for voice coders (Abstract)

A. Schmidt-Nielsen , Naval Res. Lab., Washington, DC, USA
pp. 1149-1156

Communicability Testing For Voice Coders (Abstract)

E.W. Kreamer , Naval Res. Lab., Washington, DC, USA
pp. 1153-1156

Criteria for the DoD 2400 bps vocoder selection (Abstract)

M.A. Kohler , US Dept. of Defense, Ft. Meade, MD, USA
pp. 1161-1164

Evaluation of partially adaptive STAP algorithms on the Mountain Top data set (Abstract)

Y. Seliktar , Sch. of Electr. & Comput. Eng., Georgia Inst. of Technol., Atlanta, GA, USA
pp. 1169-1172

Reduced rank space-time adaptive radar processing (Abstract)

J.S. Goldstein , USAF Rome Lab., Griffiss AFB, NY, USA
pp. 1173-1176

Beamspace techniques for hot clutter cancellation (Abstract)

S.M. Kogon , Sch. of Electr. & Comput. Eng., Georgia Inst. of Technol., Atlanta, GA, USA
pp. 1177-1180

Linear constraints in hot clutter cancellation (Abstract)

L.J. Griffiths , Dept. of Electr. & Comput. Eng., Colorado Univ., Boulder, CO, USA
pp. 1181-1184

Robust matched-field beamforming with benchmark shallow-water acoustic array data (Abstract)

J.L. Krolik , Dept. of Electr. & Comput. Eng., Duke Univ., Durham, NC, USA
pp. 1185-1188

A coherent approach to broadband matched-field processing: application in the Hudson Canyon (Abstract)

M.B. Porter , Dept. of Math., New Jersey Inst. of Technol., Newark, NJ, USA
pp. 1189-1192

A new 3D segmentation method for region-based video coding (Abstract)

Hang-Bong Kang , DSP Lab., Samsung Adv. Inst. of Technol., Suwon, South Korea
pp. 1201-1204

A hierarchical video coder with cache motion information (Abstract)

K.K. Truong , Atlanta Signal Process. Inc., GA, USA
pp. 1209-1212

Evaluation of a mosaic based approach to video compression (Abstract)

B. Tannenbaum , David Sarnoff Res. Center, Princeton, NJ, USA
pp. 1213-1215

JACOB: just a content-based query system for video databases (Abstract)

M. La Cascia , Dipartimento di Ingegneria Elettrica, Palermo Univ., Italy
pp. 1216-1219

Nonlinear editing by Generative Video (Abstract)

R.S. Jasinschi , Dept. of Electr. & Comput. Eng., Carnegie Mellon Univ., Pittsburgh, PA, USA
pp. 1220-1223

Key frame selection by motion analysis (Abstract)

W. Wolf , Dept. of Electr. Eng., Princeton Univ., NJ, USA
pp. 1228-1231

Executable requirements: opportunities and impediments (Abstract)

G.A. Shaw , Lincoln Lab., MIT, Lexington, MA, USA
pp. 1232-1235

Rapid prototyping of DSP chip-sets via functional reuse (Abstract)

M.S.B. Romdhane , Sch. of Electr. & Comput. Eng., Georgia Inst. of Technol., Atlanta, GA, USA
pp. 1236-1239

Heuristic techniques for synthesis of hard real-time DSP application specific systems (Abstract)

M. Potkonjak , Dept. of Comput. Sci., California Univ., Los Angeles, CA, USA
pp. 1240-1243

Hardware/software co-design for DSP applications via the HMS framework (Abstract)

M. Sheliga , Dept. of Comput. Sci. & Eng., Notre Dame Univ., IN, USA
pp. 1248-1251

An architectural trade capability using the Ptolemy kernel (Abstract)

E.K. Pauer , Lockheed Sanders Avionics, Nashua, NH, USA
pp. 1252-1255

Rapid prototyping of DSP systems via system interface module generation (Abstract)

S. Famorzadeh , Sch. of Electr. & Comput. Eng., Georgia Inst. of Technol., Atlanta, GA, USA
pp. 1256-1259

Autocoding: an enabling technology for rapid prototyping (Abstract)

C.B. Robbins , Manage. Commun. & Control Inc., Arlington, VA, USA
pp. 1260-1263

Middleware for realtime multicomputer tool development (Abstract)

B. Isenstein , Mercury Comput. Syst. Inc., Chelmsford, MA, USA
pp. 1264-1267

Interface synthesis in heterogeneous system-level DSP design tools (Abstract)

J.L. Pino , Dept. of Electr. Eng. & Comput. Sci., California Univ., Berkeley, CA, USA
pp. 1268-1271
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