2000 IEEE International Conference On Acoustics, Speech, And Signal Processing

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ITHERM 2000. The Seventh Intersociety Conference on Thermal and Thermomechanical Phenomena in Electronic Systems (Cat. No.00CH37069) (Abstract)

pp. 1165-lxxx

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Author Index (Abstract)

pp. 1241-1251

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Efficient ML training of CDHMM parameters based on prior evolution, posterior intervention and feedback (Abstract)

Qiang Hue, Dept. of Comput. Sci. & Inf. Syst., Hong Kong Univ., China

N. Smith

Bin Ma

pp. II1001-II1004

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Asynchronous-transition HMM (Abstract)

S. Matsuda, Japan Adv. Inst. of Sci. & Technol., Ishikawa, Japan

pp. II1005-II1008

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Factored sparse inverse covariance matrices (Abstract)

J.A. Bilmes, Dept. of Electr. Eng., Washington Univ., Seattle, WA, USA

pp. II1009-II1012

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Sub-state tying in tied mixture hidden Markov models (Abstract)

Liang Gu, Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA, USA

K. Rose

pp. II1013-II1016

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Unified frame and segment based models for automatic speech recognition (Abstract)

Hsiao-Wuen Hon, Speech Technol. Group, Microsoft Res., Redmond, WA, USA

Kuansan Wang

pp. II1017-II1020

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Use of higher level linguistic structure in acoustic modeling for speech recognition (Abstract)

I. Shafran, Dept. of Electr. Eng., Washington Univ., Seattle, WA, USA

M. Ostendorf

pp. II1021-II1024

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Mandarin accent adaptation based on context-independent/context-dependent pronunciation modeling (Abstract)

Mingkuan Liu, Inst. of Autom., Acad. Sinica, Beijing, China

Bo Xu

pp. II1025-II1028

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Towards language independent acoustic modeling (Abstract)

W. Byrne, CLSP, Johns Hopkins Univ., MD, USA

T. Wang

pp. II1029-II1032

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Improving the performance of a small microphone array at low frequencies using critical band and LPC codebooks (Abstract)

Yuchang Cao, Sch. of Electron. & Sci. Eng., Queensland Univ. of Technol., Brisbane, Qld., Australia

S. Sridharan

pp. II1033-II1036

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Speech dereverberation and noise reduction with a combined microphone array approach (Abstract)

J. Gonzalez-Rodriguez, Speech & Signal Process. Group, Univ. Politecnica de Madrid, Spain

J.L. Sanchez-Bote

J. Ortega-Garcia

pp. II1037-II1040

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Exploring permutation inconsistency in blind separation of speech signals in a reverberant environment (Abstract)

M.Z. Ikram, Center for Signal & Image Process., Georgia Inst. of Technol., Atlanta, GA, USA

D.R. Morgan

pp. II1041-II1044

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Intelligibility assessment of a multi-band speech enhancement scheme (Abstract)

A. Hussain, Dept. of Appl. Comput., Dundee Univ., UK

pp. II1045-II1048

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

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

pp. II1049-II1052

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Localization of multiple sound sources based on a CSP analysis with a microphone array (Abstract)

T. Nishiura, Graduate Sch. of Inf. Sci., Nara Inst. of Sci. & Technol., Japan

pp. II1053-II1056

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Automatic enhancement of speech intelligibility (Abstract)

V. Colotte, LORIA, Vandoeuvre-les-Nancy, France

Y. Laprie

pp. II1057-II1060

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Combined acoustic echo and noise reduction using GSVD-based optimal filtering (Abstract)

S. Doclo, Dept. of Electr. Eng., Katholieke Univ., Leuven, Belgium

M. Moonen

E. de Clippel

pp. II1061-II1064

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Speaker-centric score normalisation and time pattern analysis for continuous speaker verification (Abstract)

R. Auckenthaler, Ensigma Ltd., Chepstow, UK

M.J. Carey

J.S.D. Mason

pp. II1065-II1068

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A proposed likelihood transformation for speaker verification (Abstract)

D. Tran, Sch. of Comput., Canberra Univ., ACT, Australia

M. Wagner

pp. II1069-II1072

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The use of sub-band cepstrum in speaker verification (Abstract)

P. Sivakumaran, Hertfordshire Univ., Hatfield, UK

A.M. Ariyaeeinia

pp. II1073-II1076

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Generation of optimum signature base sequences for speech signals (Abstract)

B.S. Yarman, Dept. of Electron. Eng., Isik Univ., Maslak-Istanbul, Turkey

R. Akdeniz

pp. II1077-II1080

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Effective speaker adaptations for speaker verification (Abstract)

Sungjoo Ahn, Dept. of Electron. Eng., Korea Univ., Seoul, South Korea

Sunmee Kang

Hanseok Ko

pp. II1081-II1084

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GSM speech coding and speaker recognition (Abstract)

L. Besacier, Inst. of Microtechnol., Neuchatel Univ., Switzerland

pp. II1085-II1088

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Speaker recognition using G.729 speech codec parameters (Abstract)

T.F. Quatieri, Lincoln Lab., MIT, Lexington, MA, USA

pp. II1089-II1092

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User validation for mobile telephones (Abstract)

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

R. Auckenthaler

pp. II1093-II1096

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An instantiable speech biometrics module with natural language interface: implementation in the telephony environment (Abstract)

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

J. Kleindienst

S.H. Maes

pp. II1097-II1100

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Speaker verification: minimizing the channel effects using autoassociative neural network models (Abstract)

S.P. Kishore, Dept. of Comput. Sci. & Eng., Indian Inst. of Technol., Madras, India

B. Yegnanarayana

pp. II1101-II1104

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LDA derived cepstral trajectory filters in adverse environmental conditions (Abstract)

M. Lieb, Philips GmbH Forschungslab., Aachen, Germany

R. Haeb-Umbach

pp. II1105-II1108

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Maximum likelihood joint estimation of channel and noise for robust speech recognition (Abstract)

Yunxin Zhao, Dept. of Comput. Eng. & Comput. Sci., Missouri Univ., Columbia, MO, USA

pp. II1109-II1112

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PCA-PMC: a novel use of a priori knowledge for fast parallel model combination (Abstract)

R. Sarikaya, Robust Speech Process. Lab., Colorado Univ., Boulder, CO, USA

J.H.L. Hansen

pp. II1113-II1116

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Feature extraction using non-linear transformation for robust speech recognition on the Aurora database (Abstract)

S. Sharma, Oregon Graduate Inst. of Sci. & Technol., Portland, OR, USA

P. Jain

pp. II1117-II1120

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Asynchrony in multi-band speech recognition (Abstract)

C. Cerisara, LORIA, Henri Poincare Univ., Vandoeuvre-les-Nancy, France

D. Fohr

J.-P. Haton

pp. II1121-II1124

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Residual noise compensation for robust speech recognition in nonstationary noise (Abstract)

Kaisheng Yao, Dept. of Electr. & Electron. Eng., Hong Kong Univ. of Sci. & Technol., Clear Water Bay, China

pp. II1125-II1128

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Maximum likelihood discriminant feature spaces (Abstract)

G. Saon, IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA

M. Padmanabhan

R. Gopinath

S. Chen

pp. II1129-II1132

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Blind speech separation of moving speakers in real reverberant environments (Abstract)

A. Koutvas, Dept. of Electr. & Comput. Eng., Patras Univ., Greece

E. Dermatas

G. Kokkinakis

pp. II1133-II1136

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Mixed excitation linear prediction coding of wideband speech at 8 kbps (Abstract)

Weiran Lin, Sch. of Electr. & Electron. Eng., Nanyang Technol. Univ., Singapore

Weiran Lin

Xiao Lin

pp. II1137-II1140

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An embedded sinusoidal transform codec with measured phases and sampling rate scalability (Abstract)

G. Aguilar, Speech Technol. Group, Lucent InterNetworking Syst., Red Bank, NJ, USA

R. Zopf

pp. II1141-II1144

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High quality embedded wideband speech coding using an inherently layered coding paradigm (Abstract)

S.A. Ramprashad, Lucent Technol. Bell Labs., Murray Hill, NJ, USA

pp. II1145-II1148

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A 16-kbit/s bandwidth scalable audio coder based on the G.729 standard (Abstract)

K. Koishida, Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA, USA

V. Cuperman

A. Gersho

pp. II1149-II1152

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A 14 kb/s wideband speech coder with a parametric highband model (Abstract)

A. McCree, DSP Solutions R&D Center, Texas Instrum. Inc., Dallas, TX, USA

pp. II1153-II1156

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Hi-BIN: an alternative approach to wideband speech coding (Abstract)

R. Taori, Digital Signal Process. Group, Philips Res. Lab., Eindhoven, Netherlands

R.J. Sluijter

A.J. Gerrits

pp. II1157-II1160

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A high-fidelity speech and audio codec with low delay and low complexity (Abstract)

Juin-Hwey Chen, Speech Technol. Group, Lucent InterNetworking Syst., Red Bank, NJ, USA

pp. II1161-II1164

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Stochastic-algebraic wideband LSF quantization (Abstract)

S. Ragot, Dept. of Electr. & Comput. Eng., Sherbrooke Univ., Que., Canada

pp. II1169-II1172

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A silence compression algorithm for multi-rate/dual-bandwidth MPEG-4 CELP standard (Abstract)

M. Serizawa, C&C Media Res. Labs., NEC Corp., Kawasaki, Japan

H. Ito

T. Nomura

pp. II1173-II1176

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A speaker tracking system based on speaker turn detection for NIST evaluation (Abstract)

J.-F. Bonastre, LIA/CERI, Avignon Univ., France

pp. II1177-II1180

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Speaker identification in mismatch training and testing conditions (Abstract)

C. Alonso-Martinez, Univ. Politecnica de Catalunya, Barcelona, Spain

M. Faundez-Zanuy

pp. II1181-II1184

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An iterative technique for training speaker verification systems (Abstract)

W.M. Campbell, Human Interface Lab., Motorola Inc., Tempe, AZ, USA

pp. II1185-II1188

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Search-space reduction for fast, optimal HMM decoding in speaker verification (Abstract)

Qi Li, Multimedia Commun. Res. Lab., Lucent Technol. Bell Labs., Murray Hill, NJ, USA

pp. II1189-II1192

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A two-stage scoring method combining world and cohort models for speaker verification (Abstract)

W.D. Zhang, Dept. of Electron. & Inf. Eng., Hong Kong Polytech. Univ., China

M.W. Mak

X. He

pp. II1193-II1196

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Behavior of a Bayesian adaptation method for incremental enrollment in speaker verification (Abstract)

C. Fredouille, UBS-Ubilab, Zurich, Switzerland

pp. II1197-II1200

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Evolutive HMM for multi-speaker tracking system (Abstract)

S. Meignier, LIA/CERI, Avignon Univ., France

pp. II1201-II1204

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Speaker recognition in two-speaker data: recent results from Dragon Systems (Abstract)

F. Weber, Dragon Syst. Inc., Newton, MA, USA

pp. II1205-II1208

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A novel rank-based classifier combination scheme for speaker identification (Abstract)

H. Altincay, Dept. of Electr. & Electron. Eng., Middle East Tech. Univ., Ankara, Turkey

H. Demirekler

pp. II1209-II1212

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Improved normalization without recourse to an impostor database for speaker verification (Abstract)

M. Hebert, Nuance Commun., Menlo Park, CA, USA

S.D. Peters

pp. II1213-II1216

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Probabilistic simulation of human-machine dialogues (Abstract)

K. Scheffler, Dept. of Eng., Cambridge Univ., UK

S. Young

pp. II1217-II1220

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Fundamental performance analysis for spoken dialogue systems based on a quantitative simulation approach (Abstract)

Bor-Shen Lin, Dept. of Electr. Eng., Nat. Taiwan Univ., Taipei, Taiwan

Lin-Shan Lee

pp. II1221-II1224

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Parser adaptation via Householder transform (Abstract)

Xiaoqiang Luo, IBM Thomas J. Watson Res. Center, Yorktown Heights, NY, USA

pp. II1225-II1228

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CU FOREX: a bilingual spoken dialog system for foreign exchange enquiries (Abstract)

H.M. Meng, Dept. of Syst. Eng. & Eng. Manage., Chinese Univ. of Hong Kong, Shatin, China

S. Lee

C. Wai

pp. II1229-II1232

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Fast reinforcement learning of dialog strategies (Abstract)

D. Goddeau, Cambridge Res. Lab., Compaq Comput. Corp., MA, USA

J. Pineau

pp. II1233-II1236

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Confidence measures for dialogue management in the CU Communicator system (Abstract)

R. San-Segundo, Center for Spoken Language Understanding, Colorado Univ., Boulder, CO, USA

B. Pellom

W. Ward

J.M. Pardo

pp. II1237-II1240

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Time-frequency signal decomposition using energy mixture models (Abstract)

M. Coates, Dept. of Electr. & Comput. Sci., Rice Univ., Houston, TX, USA

W. Fitzgerald

pp. II633-II636

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Optimal estimation of non-stationary phase and amplitude processes (Abstract)

C. Andrieu, Dept. of Eng., Cambridge Univ., UK

pp. II637-II640

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Improved frequency marginal estimates for time-frequency distributions (Abstract)

S. Aviyente, Dept. of Electr. Eng. & Comput. Sci., Michigan Univ., Ann Arbor, MI, USA

W.J. Williams

pp. II641-II644

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SNR analysis of time-frequency distributions (Abstract)

Weifeng Mu, Dept. of Electr. & Comput. Eng., Villanova Univ., PA, USA

M.G. Amin

pp. II645-II648

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Data-driven time-frequency classification techniques applied to tool-wear monitoring (Abstract)

B.W. Gillespie, Dept. of Electr. Eng., Washington Univ., Seattle, WA, USA

L.E. Atlas

pp. II649-II652

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A fast Gabor spectrogram (Abstract)

Shie Qian, DSP Group, Nat. Instrum. Corp., Austin, TX, USA

Yong Rao

Dapang Chen

pp. II653-II656

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Adaptive instantaneous frequency estimation of multi-component FM signals (Abstract)

Z.M. Hussain, Signal Processing Res. Centre, Queensland Univ. of Technol., Brisbane, Qld., Australia

B. Boashash

pp. II657-II660

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IF estimation of linear FM signals corrupted by multiplicative and additive noise: a time-frequency approach (Abstract)

B. Barkat, Signal Processing Res. Centre, Queensland Univ. of Technol., Brisbane, Qld., Australia

B. Boashash

pp. II661-II664

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Characterization of transient wandering tones by dynamic modeling of fractional-Fourier features (Abstract)

P.L. Ainsleigh, Naval Underwater Warfare Center, Newport, RI, USA

N. Kehtarnavaz

pp. II665-II668

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Nontrivial analytic signals with positive instantaneous frequency and band-limited amplitude (Abstract)

M.I. Doroslovacki, Dept. of Electr. & Comput. Eng., George Washington Univ., Washington, DC, USA

pp. II669-II672

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Enhancing weak signal components in time-frequency distributions by wavelet pre-processing (Abstract)

Mingui Sun, Dept. of Neurosurgery & Electr. Eng., Pittsburgh Univ., PA, USA

R.J. Sclabassi

pp. II673-II676

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A new method for estimating the instantaneous frequency based on maximum likelihood (Abstract)

F.A.C. de Bastos, Inst. Militar de Engenharia, Rio Janeiro, Brazil

M.L.R. de Campos

pp. II677-II680

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Hybrid linear/quadratic time-frequency attributes (Abstract)

R. Baraniuk, Dept. of Electr. & Comput. Eng., Rice Univ., Houston, TX, USA

M. Coates

P. Steeghs

pp. II681-II684

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A Kalman filter based approach for estimating nonstationary VAR models via pole tracking (Abstract)

M. Elling, Iowa State Univ., Ames, IA, USA

P. Sherman

pp. II685-II688

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Wavelet packets based features selection for voiceless plosives classification (Abstract)

E. Lukasik, Inst. of Comput. Sci., Poznan Univ. of Technol., Poland

pp. II689-II692

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A high resolution time frequency representation with significantly reduced cross-terms (Abstract)

A.K. Ozdemir, Dept. of Electr. & Electron. Eng., Bilkent Univ., Ankara, Turkey

O. Arikan

pp. II693-II696

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An efficient algorithm to extract components of a composite signal (Abstract)

O. Arikan, Dept. of Electr. & Electron. Eng., Bilkent Univ., Ankara, Turkey

A. Kemal Ozdemir

pp. II697-II700

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Monte Carlo filtering and smoothing with application to time-varying spectral estimation (Abstract)

A. Doucet, Signal Processing Lab., Cambridge Univ., UK

S.J. Godsill

M. West

pp. II701-II704

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Markov chain Monte Carlo data association for target tracking (Abstract)

N. Bergman, Dept. of Electr. Eng., Linkoping Univ., Sweden

A. Doucet

pp. II705-II708

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A new approach to period estimation (Abstract)

D.D. Muresan, Dept. of Electr. Eng., Cornell Univ., Ithaca, NY, USA

T.W. Parks

pp. II709-II712

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H/sub /spl infin// smoothing (Abstract)

E. Blanco, Univ. Claude Bernard, Villeurbanne, France

P. Neveux

G. Thomas

pp. II713-II716

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A modified Baum-Welch algorithm for hidden Markov models with multiple observation spaces (Abstract)

P.M. Baggenstoss, Naval Underwater Warfare Center, Newport, RI, USA

pp. II717-II720

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Least squares deconvolution in wavelet domain for 1/f driven LTI systems (Abstract)

M.I. Izzetoglu, Dept. of Electr. & Comput. Eng., Drexel Univ., Philadelphia, PA, USA

pp. II721-II724

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Bayesian detection of transient signals in colored noise (Abstract)

Yufei Huang, Dept. of Electr. & Comput. Eng., State Univ. of New York, Stony Brook, NY, USA

P.M. Djuric

pp. II725-II728

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Detection of frequency modulated signals in Rayleigh fading channels based on time-frequency distributions (Abstract)

T.L. Nguyen, Signal Processing Res. Centre, Queensland Univ. of Technol., Brisbane, Qld., Australia

B. Senadji

pp. II729-II732

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Detection of signal number based on statistics of maximum likelihood (Abstract)

M. Suzuki, Res. Inst. for Electron. Sci., Hokkaido Univ., Sapporo, Japan

H. Sanada

N. Naga

pp. II733-II736

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An efficient square-root algorithm for BLAST (Abstract)

B. Hassibi, Lucent Technol. Bell Labs., Murray Hill, NJ, USA

pp. II737-II740

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Retrieving sinusoids in colored Rayleigh noise by a cumulant-based FBLP approach (Abstract)

R.R. Ghauieb, Fac. of Eng., Toyama Univ., Japan

Y. Horita

T. Murai

pp. II741-II744

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Synthesis of vibrato singing (Abstract)

Y. Meron, Dept. of Inf. & Commun. Eng., Tokyo Univ., Japan

K. Hirose

pp. II745-II748

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Music summarization using key phrases (Abstract)

B. Logan, Res. Labs., Compaq Comput. Corp., Cambridge, MA, USA

S. Chu

pp. II749-II752

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Musical instrument recognition using cepstral coefficients and temporal features (Abstract)

A. Eronen, Signal Process. Lab., Tampere Univ. of Technol., Finland

A. Klapuri

pp. II753-II756

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A robust predominant-F0 estimation method for real-time detection of melody and bass lines in CD recordings (Abstract)

M. Goto, Electrotech. Lab., Ibaraki, Japan

pp. II757-II760

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Sound analysis using MPEG compressed audio (Abstract)

G. Tzanetakis, Dept. of Comput. Sci., Princeton Univ., NJ, USA

F. Cook

pp. II761-II764

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Separation of harmonic sound sources using sinusoidal modeling (Abstract)

T. Virtanen, Signal Process. Lab., Tampere Univ. of Technol., Finland

A. Klapuri

pp. II765-II768

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Model-based sound synthesis of tanbur, a Turkish long-necked lute (Abstract)

C. Erkut, Lab. of Acoust. & Audio Signal Processing, Helsinki Univ. of Technol., Espoo, Finland

V. Valimaki

pp. II769-II772

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Acoustic sound from the electric guitar using DSP techniques (Abstract)

M. Karjalainen, Lab. of Acoust. & Audio Signal Processing, Helsinki Univ. of Technol., Espoo, Finland

H. Penttinen

V. Valimaki

pp. II773-II776

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Practical and efficient implementation of the Levinson algorithm for multichannel sound reproduction (Abstract)

J.J. Lopez, Dept. de Comunicaciones, Univ. Politecnica de Valencia, Spain

A. Gonzalez

pp. II777-II780

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A new robust system for 3D audio using loudspeakers (Abstract)

D.B. Ward, Sch. of Electr. Eng., New South Wales Univ., Kensington, NSW, Australia

G.W. Elko

pp. II781-II784

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Common acoustical-poles/zeros modeling for 3D sound processing (Abstract)

M.C. Chen, Dept. of Commun. Eng., Nat. Chiao Tung Univ., Hsinchu, Taiwan

S.F. Hsieh

pp. II785-II788

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Frequency-domain adaptive filtering revisited, generalization to the multi-channel case, and application to acoustic echo cancellation (Abstract)

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

D.R. Morgan

pp. II789-II792

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A robust proportionate affine projection algorithm for network echo cancellation (Abstract)

T. Gansler, AT&T Bell Labs., Murray Hill, NJ, USA

pp. II793-II796

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Study on combining multi-channel echo cancellers with beamformers (Abstract)

M. Kallinger, Dept. of Telecommun., Bremen Univ., Germany

J. Bitzer

K.-D. Kammeyer

pp. II797-II800

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A new adaptive algorithm for stereophonic acoustic echo canceller (Abstract)

Yang-Won Jung, Dept. of Electr. & Comput. Eng., Yonsei Univ., Seoul, South Korea

pp. II801-II804

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Nonlinear acoustic echo cancellation with fast converging memoryless preprocessor (Abstract)

A. Stenger, Telecommun. Lab., Erlangen-Nurnberg Univ., Germany

W. Kellermann

pp. II805-II808

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A new sampling of echo paths in North American networks (Abstract)

K.C. Ho, Dept. of Electr. Eng., Missouri Univ., Columbia, MO, USA

pp. II809-II812

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Channel-number-compressed multi-channel acoustic echo canceller for high-presence teleconferencing system with large display (Abstract)

A. Nakagawa, NTT Cyber Space Labs., Tokyo, Japan

S. Aoki

pp. II813-II816

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Synthetic stereo acoustic echo cancellation structure with microphone array beamforming for VoIP conferences (Abstract)

T. Yensen, Dept. of Syst. & Comput. Eng., Carleton Univ., Ottawa, Ont., Canada

J. Ryan

R. Goubran

I. Lambadaris

pp. II817-II820

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Towards a perceptually optimal spectral amplitude estimator for audio signal enhancement (Abstract)

P.J. Wolfe, Dept. of Eng., Cambridge Univ., UK

S.J. Godsill

pp. II821-II824

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Multichannel active noise control algorithms using inverse filters (Abstract)

F.Yu.M. Bouchard, Sch. of Inf. Technol. & Eng., Ottawa Univ., Ont., Canada

pp. II825-II828

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Active control of synchronous machine vibrations: an adaptive optimal MIMO controller (Abstract)

P. Granjon, CNRS, Domaine Univ., France

C. Serviere

A. Foggia

pp. II829-II832

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Time domain recursive deconvolution in sound reproduction (Abstract)

A. Gonzalez, Dept. de Commun., Univ. Politecnica de Valencia, Spain

J.J. Lopez

pp. II833-II836

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Some practical insights in multichannel active noise control equalization (Abstract)

M. De Diego, Dept. of Comunicaciones, Univ. Politecnica de Valencia, Spain

pp. II837-II840

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The performance surface in nonlinear mean square estimation: application to the active noise control problem (Abstract)

M.H. Costa, Nucl. de Engenharia Biomed., Univ. Catolica de Pelotas, Brazil

J.C.M. Bermudez

N.J. Bershad

pp. II841-II844

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A preconditioned LMS algorithm for rapid adaptation of feedforward controllers (Abstract)

S.J. Elliott, Inst. of Sound & Vibration Res., Southampton Univ., UK

J.G. Cook

pp. II845-II848

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An architecture for wavelet-packet based speech enhancement for hearing aids (Abstract)

M.A. Trenas, Dept. Comput. Architecture, Malaga Univ., Spain

pp. II849-II852

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Modulation enhancement of speech as a preprocessing for reverberant chambers with the hearing-impaired (Abstract)

A. Kusumoto, Dept. of Electr. & Electron. Eng., Sophia Univ., Tokyo, Japan

T. Arai

pp. II853-II856

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Simulation of hearing impairment based on the Fourier time transformation (Abstract)

J. Chalupper, Inst. for Human-Machine Commun., Tech. Univ. of Munich, Germany

H. Fastl

pp. II857-II860

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A learning algorithm for adaptive microphone arrays based on mathematical programming methods (Abstract)

K. Suyama, Sci. Univ. of Tokyo, Japan

R. Hirabayashi

pp. II861-II864

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Binaural sound localization in an artificial neural network (Abstract)

C. Schauer, Dept. of Neuroinf., Ilmenau Tech. Univ., Germany

T. Zahn

P. Paschke

H.-M. Gross

pp. II865-II868

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Bias of feedback cancellation algorithms based on direct closed loop identification (Abstract)

J. Hellgren, Dept. of Neurosci. & Locomotion, Linkoping Univ., Sweden

U. Forssell

pp. II869-II872

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Multiple frequency harmonics analysis and synthesis of audio signals (Abstract)

A. Jbira, Centre for Commun. Syst. Res., Surrey Univ., Guildford, UK

A. Kondoz

pp. II873-II876

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A 6Kbps to 85Kbps scalable audio coder (Abstract)

T.S. Verma, Lab. of Acoustics & Audio Signal Processing, Helsinki Univ. of Technol., Espoo, Finland

T.H.Y. Meng

pp. II877-II880

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Audio coding using a psychoacoustic pre- and post-filter (Abstract)

B. Edler, Multimedia Commun. Res. Lab., AT&T Bell Labs., Murray Hill, NJ, USA

G. Schuller

pp. II881-II884

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Lossless coding of MPEG-1 Layer III encoded audio streams (Abstract)

F. Golchin, Sch. of Microelectron. Eng., Griffith Univ., Brisbane, Qld., Australia

K.K. Paliwal

pp. II885-II888

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A design of lossy and lossless scalable audio coding (Abstract)

T. Moriya, NTT LSI Labs., Tokyo, Japan

N. Iwakami

A. Jin

T. Mori

pp. II889-II892

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Perceptual bit allocation for low rate coding of narrowband audio (Abstract)

H. Najafzadeh, Dept. of Electr. & Comput. Eng., McGill Univ., Montreal, Que., Canada

P. Kabal

pp. II893-II896

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Evaluation of a warped linear predictive coding scheme (Abstract)

A. Harma, Lab. of Acoustics & Audio Signal Processing, Helsinki Univ. of Technol., Espoo, Finland

pp. II897-II900

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Exploiting time and frequency masking in consistent sinusoidal analysis-synthesis (Abstract)

R. Vafin, Dept. of Speech Commun. & Music Acoust., R. Inst. of Technol., Stockholm, Sweden

S.V. Andersen

W.B. Kleijn

pp. II901-II904

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Frame erasure concealment using sinusoidal analysis-synthesis and its application to MDCT-based codecs (Abstract)

V.N. Parikh, Speech Technol. Group, Lucent InterNetworking Syst., Red Bank, NJ, USA

Juin-Hwey Chen

G. Aguilar

pp. II905-II908

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Passive acoustic source localization for video camera steering (Abstract)

Yiteng Huang, Center for Image & Signal Processing, Georgia Inst. of Technol., Atlanta, GA, USA

J. Benesty

G.W. Elko

pp. II909-II912

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Multiband warped filter equalizer design for loudspeaker systems (Abstract)

Wang Peng, Sch. of Electr. & Electron. Eng., Nanyang Technol. Inst., Singapore

Wee Ser

Ming Zhang

pp. II913-II916

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The effect of polarity inversion of speech on human perception and data hiding as an application (Abstract)

S. Sakaguchi, Dept. of Electr. & Electron. Eng., Sophia Univ., Tokyo, Japan

T. Arai

Y. Murahara

pp. II917-II920

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Multi-source localization in reverberant environments by ROOT-MUSIC and clustering (Abstract)

E.D. Di Claudio, INFOCOM Dept., Rome Univ., Italy

R. Parisi

G. Orlandi

pp. II921-II924

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A fast subband room response simulator (Abstract)

F. Jabloun, Electr. & Comput. Eng. Dept., McGill Univ., Montreal, Que., Canada

B. Champagne

pp. II925-II928

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Construction of the acoustic inventory for a Greek text-to-speech concatenative synthesis system (Abstract)

C. Christogiannis, Dept. of Electr. & Comput. Eng., Nat. Tech. Univ. of Athens, Greece

C. Shih

pp. II929-II932

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Concatenating syllables for response generation in spoken language applications (Abstract)

A. Arvaniti, Dept. of Syst. Eng. & Eng. Manage., Chinese Univ. of Hong Kong, Shatin, China

H.M. Meng

pp. II933-II936

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Segment pre-selection in decision-tree based speech synthesis systems (Abstract)

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

pp. II937-II940

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Spectral modification for concatenative speech synthesis (Abstract)

J. Wouters, Center for Spoken Language Understanding, Oregon Graduate Inst. of Sci. & Technol., Beaverton, OR, USA

M.W. Macon

pp. II941-II944

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Transition-based speech synthesis using neural networks (Abstract)

G. Corrigan, Motorola Labs., Schaumburg, IL, USA

N. Massey

O. Schnurr

pp. II945-II948

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Stochastic modeling of spectral adjustment for high quality pitch modification (Abstract)

A. Kain, Center for Spoken Language Understanding, Oregon Graduate Inst. of Sci. & Technol., Beaverton, OR, USA

Y. Stylianou

pp. II949-II952

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Voice quality conversion in TD-PSOLA speech synthesis (Abstract)

Xuejing Sun, Dept. of Commun. Sci. & Disorders, Northwestern Univ., Evanston, IL, USA

pp. II953-II956

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On the implementation of the harmonic plus noise model for concatenative speech synthesis (Abstract)

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

pp. II957-II960

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On-line incremental speaker adaptation with automatic speaker change detection (Abstract)

Zhi-Peng Zhang, Dept. of Comput. Sci., Tokyo Inst. of Technol., Japan

S. Furui

K. Ohtsuki

pp. II961-II964

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Joint maximum a posteriori estimation of transformation and hidden Markov model parameters (Abstract)

O. Siohan, Lucent Technol. Bell Labs., Murray Hill, NJ, USA

C. Chesta

Chin-Hui Lee

pp. II965-II968

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Full covariance modelling and adaptation in sub-bands (Abstract)

B. Doherty, Sch. of Electr. & Electron. Eng., Queen's Univ., Belfast, UK

S. Vaseghi

P. McCourt

pp. II969-II972

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Hierarchical Bayes approach to adapting delta- and delta-delta cepstra (Abstract)

A.C. Surendran, Lucent Technol. Bell Labs., Murray Hill, NJ, USA

pp. II973-II976

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On-line Bayesian speaker adaptation using tree-structured transformation and robust priors (Abstract)

Shaojun Wang, Beckman Inst. for Adv. Sci. & Technol., Illinois Univ., Urbana, IL, USA

Yunxin Zhao

pp. II977-II980

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Speaker adaptation based on combination of MAP estimation and weighted neighbor regression (Abstract)

Lei He, Dept. of Comput. Sci. & Technol., Tsinghua Univ., Beijing, China

Jian Wu

Ditang Fang

Wenhu Wu

pp. II981-II984

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Robust estimation for rapid speaker adaptation using discounted likelihood techniques (Abstract)

A. Gunawardana, Center for Language & Speech Procesing, Johns Hopkins Univ., Baltimore, MD, USA

W. Byrne

pp. II985-II988

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Fast speaker adaptation of large vocabulary continuous density HMM speech recognizer using a basis transform approach (Abstract)

C. Boulis, Dept. of Electron. & Comput. Eng., Tech. Univ. of Crete, Chania, Greece

V. Digalakis

pp. II989-II992

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A generalization of the maximum a posteriori training algorithm for mixture priors (Abstract)

E.R. Buhrke, Human Interface lab., Motorola Inc., Naperville, IL, USA

Chen Liu

pp. II993-II996

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Linear regression under maximum a posteriori criterion with Markov random field prior (Abstract)

Xintian Wu, Oregon Graduate Inst. of Sci. & Technol., Beaverton, OR, USA

Yonghong Yan

pp. II997-I1000

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