default search action
IEEE Transactions on Speech and Audio Processing, Volume 8
Volume 8, Number 1, January 2000
- James R. Glass, Ronald Rosenfeld
:
Guest editorial introduction to the special issue on language modeling and dialogue systems. 1-2 - Giuseppe Riccardi, Allen L. Gorin:
Stochastic language adaptation over time and state in natural spoken dialog systems. 3-10 - Esther Levin, Roberto Pieraccini, Wieland Eckert:
A stochastic model of human-machine interaction for learning dialog strategies. 11-23 - Hermann Ney, Sonja Nießen, Franz Josef Och, Hassan Sawaf, Christoph Tillmann, Stephan Vogel:
Algorithms for statistical translation of spoken language. 24-36 - Stanley F. Chen, Ronald Rosenfeld
A survey of smoothing techniques for ME models. 37-50 - Bernd Souvignier, Andreas Kellner, Bernhard Rueber, Hauke Schramm, Frank Seide:
The thoughtful elephant: strategies for spoken dialog systems. 51-62 - Man-Hung Siu, Mari Ostendorf:
Variable n-grams and extensions for conversational speech language modeling. 63-75 - Jerome R. Bellegarda:
Large vocabulary speech recognition with multispan statistical language models. 76-84 - Victor Zue, Stephanie Seneff, James R. Glass, Joseph Polifroni, Christine Pao, Timothy J. Hazen, I. Lee Hetherington:
JUPlTER: a telephone-based conversational interface for weather information. 85-96
Volume 8, Number 2, March 2000
- Linkai Bu, Tzi-Dar Church:
Perceptual speech processing and phonetic feature mapping for robust vowel recognition. 105-114 - Lawrence K. Saul, Mazin G. Rahim:
Maximum likelihood and minimum classification error factor analysis for automatic speech recognition. 115-125 - Eduardo Lleida
Utterance verification in continuous speech recognition: decoding and training procedures. 126-139 - Ramon Arean, Jelena Kovacevic, Vivek K. Goyal
Multiple description perceptual audio coding with correlating transforms. 140-145 - Daniel Graupe, Dusan Veselinovic:
Blind adaptive filtering of speech from noise of unknown spectrum using a virtual feedback configuration. 146-158 - Udar Mittal, Nam Phamdo:
Signal/noise KLT based approach for enhancing speech degraded by colored noise. 159-167 - Jacob Benesty
A new class of doubletalk detectors based on cross-correlation. 168-172 - James G. Ryan, Rafik A. Goubran:
Array optimization applied in the near field of a microphone array. 173-176 - James DeLucia, Fred Kochman:
A new noniterative algorithm for computing acoustically constrained vocal tract area functions. 177-183 - Lauri Savioja, Vesa Välimäki
Reducing the dispersion error in the digital waveguide mesh using interpolation and frequency-warping techniques. 184-194 - Hong Kook Kim, Seung Ho Choi, Hwang Soo Lee:
On approximating line spectral frequencies to LPC cepstral coefficients. 195-199 - Qiang Huo, Chin-Hui Lee:
A Bayesian predictive classification approach to robust speech recognition. 200-204 - Mukund Padmanabhan, L. R. Ban:
Model complexity adaptation using a discriminant measure. 205-208 - Lúcio Martins da Silva, Abraham Alcaim:
Differential coding of speech LSF parameters using hybrid vector quantization and bidirectional prediction. 208-211 - Darren B. Ward:
Joint least squares optimization for robust acoustic crosstalk cancellation. 211-215
Volume 8, Number 3, May 2000
- Manohar N. Murthi, Bhaskar D. Rao:
All-pole modeling of speech based on the minimum variance distortionless response spectrum. 221-239 - Ashwin Rao, Ramdas Kumaresan:
On decomposing speech into modulated components. 240-254 - Yunxin Zhao:
Frequency-domain maximum likelihood estimation for automatic speech recognition in additive and convolutive noises. 255-266 - Bayya Yegnanarayana, P. Satyanarayana Murthy:
Enhancement of reverberant speech using LP residual signal. 267-281 - Ki Yong Lee, Souhwan Jung:
Time-domain approach using multiple Kalman filters and EM algorithm to speech enhancement with nonstationary noise. 282-291 - Rafid A. Sukkar, Malan B. Gandhi, Anand R. Setlur:
Speaker verification using mixture decomposition discrimination. 292-299 - Tero Tolonen, Vesa Välimäki
Modeling of tension modulation nonlinearity in plucked strings. 300-310 - Biljana D. Radlovic, Robert C. Williamson, Rodney A. Kennedy
Equalization in an acoustic reverberant environment: robustness results. 311-319 - Lucas C. Parra, Clay Spence:
Convolutive blind separation of non-stationary sources. 320-327 - Shrikanth S. Narayanan, Abeer Alwan:
Noise source models for fricative consonants. 328-344 - Nicola R. Chong
A new waveform interpolation coding scheme based on pitch synchronous wavelet transform decomposition. 345-348 - Jae Bum Kim, K. Y. Lee, C. W. Lee:
On the applications of the interacting multiple model algorithm for enhancing noisy speech. 349-352 - Gang Li, Lunji Qiu, Ling Kok Ng:
Signal representation based on instantaneous amplitude models with application to speech synthesis. 353-357
Volume 8, Number 4, July 2000
- Jan Skoglund
On time-frequency masking in voiced speech. 361-369 - Jan Linden:
Channel optimized predictive vector quantization. 370-384 - Per Hedelin, Jan Skoglund
Vector quantization based on Gaussian mixture models. 385-401 - Jonathan Huang, Kuan-Chieh Yen, Yunxin Zhao:
Subband-based adaptive decorrelation filtering for co-channel speech separation. 402-406 - Ho-Young Jung, Soo-Young Lee:
On the temporal decorrelation of feature parameters for noise-robust speech recognition. 407-416 - Mark J. F. Gales:
Cluster adaptive training of hidden Markov models. 417-428 - Sahar E. Bou-Ghazale, John H. L. Hansen:
A comparative study of traditional and newly proposed features for recognition of speech under stress. 429-442 - Marcio G. Siqueira, Abeer Alwan:
Steady-state analysis of continuous adaptation in acoustic feedback reduction systems for hearing-aids. 443-453 - Akira Watanabe, Shingo Tomishige, Masahiro Nakatake:
Speech visualization by integrating features for the hearing impaired. 454-466 - Andrew Horner:
Low peak amplitudes for wavetable synthesis. 467-470 - Jean Laroche:
Synthesis of sinusoids via non-overlapping inverse Fourier transform. 471-477 - S. Gökhun Tanyer
Voice activity detection in nonstationary noise. 478-482 - Chi-Min Liu, Chin-Chih Chiu, Hung-Yuan Chang:
Design of vocabulary-independent Mandarin keyword spotters. 483-487 - João Manuel Rodrigues
On the use of backward adaptation in a perceptual audio coder. 488-490
Volume 8, Number 5, September 2000
- Futoshi Asano, Satoru Hayamizu, Takeshi Yamada, Satoshi Nakamura:
Speech enhancement based on the subspace method. 497-507 - Donald L. Duttweiler:
Proportionate normalized least-mean-squares adaptation in echo cancelers. 508-518 - Elmar Nöth
VERBMOBIL: the use of prosody in the linguistic components of a speech understanding system. 519-532 - Ivandro Sanches
Noise-compensated hidden Markov models. 533-540 - Gin-Der Wu, Chin-Teng Lin
Word boundary detection with mel-scale frequency bank in noisy environment. 541-554 - Wolfgang Reichl, Wu Chou:
Robust decision tree state tying for continuous speech recognition. 555-566 - Thomas F. Quatieri, Douglas A. Reynolds, Gerald C. O'Leary:
Estimation of handset nonlinearity with application to speaker recognition. 567-584 - Qi Li, Biing-Hwang Juang, Chin-Hui Lee:
Automatic verbal information verification for user authentication. 585-596 - Gianpaolo Borin, Giovanni De Poli
Elimination of delay-free loops in discrete-time models of nonlinear acoustic systems. 597-605 - Martin Bouchard
Multichannel recursive-least-square algorithms and fast-transversal-filter algorithms for active noise control and sound reproduction systems. 606-618 - Stan Z. Li:
Content-based audio classification and retrieval using the nearest feature line method. 619-625 - Sven C. Martin, Hermann Ney, Christoph Hamacher:
Maximum entropy language modeling and the smoothing problem. 626-632 - Miguel Arjona Ramírez
Joint position and amplitude search of algebraic multipulses. 633-637 - SangKi Kang, SeongJoon Baek, Ki Yong Lee, Koeng-Mo Sung:
Mixture IMM for speech enhancement under nonstationary noise. 637-641
Volume 8, Number 6, November 2000
- José M. F. Moura:
A good read. 645 - Paolo Prandoni, Martin Vetterli
R/D optimal linear prediction. 646-655 - Tomas Gänsler, Steven L. Gay, Man Mohan Sondhi, Jacob Benesty
Double-talk robust fast converging algorithms for network echo cancellation. 656-663 - Chin-Teng Lin
GA-based noisy speech recognition using two-dimensional cepstrum. 664-675 - Stefan Ortmanns, Hermann Ney:
The time-conditioned approach in dynamic programming search for LVCSR. 676-687 - Hui Jiang, Keikichi Hirose, Qiang Hue:
A minimax search algorithm for robust continuous speech recognition. 688-694 - Roland Kuhn, Jean-Claude Junqua, Patrick Nguyen, Nancy Niedzielski:
Rapid speaker adaptation in eigenvoice space. 695-707 - Tero Tolonen, Matti Karjalainen:
A computationally efficient multipitch analysis model. 708-716 - Davide Rocchesso
Fractionally addressed delay lines. 717-727 - Biljana D. Radlovic, Rodney A. Kennedy
Nonminimum-phase equalization and its subjective importance in room acoustics. 728-737 - Michael S. Brandstein, Darren B. Ward:
Cell-based beamforming (CE-BABE) for speech acquisition with microphone arrays. 738-743 - Hai Le Vu
Efficient distance measure for quantization of LSF and its Karhunen-Loeve transformed parameters. 744-746 - Jun Huang, Yunxin Zhao:
A DCT-based fast signal subspace technique for robust speech recognition. 747-751 - William Byrne, Asela Gunawardana:
Comments on "Efficient training algorithms for HMMs using incremental estimation". 751-754 - Boaz Rafaely
Control of feedback in hearing aids-a robust filter design approach. 754-756
manage site settings
To protect your privacy, all features that rely on external API calls from your browser are turned off by default. You need to opt-in for them to become active. All settings here will be stored as cookies with your web browser. For more information see our F.A.Q.
last updated on 2024-10-31 20:03 CET by the dblp team
all metadata released as open data under CC0 1.0 license
see also: Terms of Use | Privacy Policy | Imprint
dblp was originally created in 1993 at:
since 2018, dblp has been operated and maintained by:
the dblp computer science bibliography is funded and supported by: