Journal: IEEE Transactions on Audio, Speech & Language Processing

Volume 8, Issue 6

645 -- 0José M. F. Moura. A good read
646 -- 655Paolo Prandoni, Martin Vetterli. R/D optimal linear prediction
656 -- 663Tomas Gänsler, Steven L. Gay, Man Mohan Sondhi, Jacob Benesty. Double-talk robust fast converging algorithms for network echo cancellation
664 -- 675Chin-Teng Lin, Hsi-Wen Nein, Jiing-Yuan Hwu. GA-based noisy speech recognition using two-dimensional cepstrum
676 -- 687Stefan Ortmanns, Hermann Ney. The time-conditioned approach in dynamic programming search for LVCSR
688 -- 694Hui Jiang 0001, Keikichi Hirose, Qiang Hue. A minimax search algorithm for robust continuous speech recognition
695 -- 707Roland Kuhn, Jean-Claude Junqua, Patrick Nguyen, Nancy Niedzielski. Rapid speaker adaptation in eigenvoice space
708 -- 716Tero Tolonen, Matti Karjalainen. A computationally efficient multipitch analysis model
717 -- 727Davide Rocchesso. Fractionally addressed delay lines
728 -- 737Biljana D. Radlovic, Rodney A. Kennedy. Nonminimum-phase equalization and its subjective importance in room acoustics
738 -- 743Michael S. Brandstein, Darren B. Ward. Cell-based beamforming (CE-BABE) for speech acquisition with microphone arrays
744 -- 746Hai Le Vu, László Lois. Efficient distance measure for quantization of LSF and its Karhunen-Loeve transformed parameters
747 -- 751Jun Huang, Yunxin Zhao. A DCT-based fast signal subspace technique for robust speech recognition
751 -- 754William Byrne, Asela Gunawardana. Comments on "Efficient training algorithms for HMMs using incremental estimation"
754 -- 756Boaz Rafaely, Mariano Roccasalva-Firenze. Control of feedback in hearing aids-a robust filter design approach

Volume 8, Issue 5

497 -- 507Futoshi Asano, Satoru Hayamizu, Takeshi Yamada, Satoshi Nakamura. Speech enhancement based on the subspace method
508 -- 518Donald L. Duttweiler. Proportionate normalized least-mean-squares adaptation in echo cancelers
519 -- 532Elmar Nöth, Anton Batliner, Andreas Kießling, Ralf Kompe, Heinrich Niemann. VERBMOBIL: the use of prosody in the linguistic components of a speech understanding system
533 -- 540Ivandro Sanches. Noise-compensated hidden Markov models
541 -- 554Gin-Der Wu, Chin-Teng Lin. Word boundary detection with mel-scale frequency bank in noisy environment
555 -- 566Wolfgang Reichl, Wu Chou. Robust decision tree state tying for continuous speech recognition
567 -- 584Thomas F. Quatieri, Douglas A. Reynolds, Gerald C. O'Leary. Estimation of handset nonlinearity with application to speaker recognition
585 -- 596Qi Li, Biing-Hwang Juang, Chin-Hui Lee. Automatic verbal information verification for user authentication
597 -- 605Gianpaolo Borin, Giovanni De Poli, Davide Rocchesso. Elimination of delay-free loops in discrete-time models of nonlinear acoustic systems
606 -- 618Martin Bouchard, Stephan Quednau. Multichannel recursive-least-square algorithms and fast-transversal-filter algorithms for active noise control and sound reproduction systems
619 -- 625Stan Z. Li. Content-based audio classification and retrieval using the nearest feature line method
626 -- 632Sven C. Martin, Hermann Ney, Christoph Hamacher. Maximum entropy language modeling and the smoothing problem
633 -- 637Miguel Arjona Ramirez, Max Gerken. Joint position and amplitude search of algebraic multipulses
637 -- 641Sangki Kang, SeongJoon Baek, Ki Yong Lee, Koeng-Mo Sung. Mixture IMM for speech enhancement under nonstationary noise

Volume 8, Issue 4

361 -- 369Jan Skoglund, W. Bastiaan Kleijn. On time-frequency masking in voiced speech
370 -- 384Jan Linden. Channel optimized predictive vector quantization
385 -- 401Per Hedelin, Jan Skoglund. Vector quantization based on Gaussian mixture models
402 -- 406Jonathan Huang, Kuan-Chieh Yen, Yunxin Zhao. Subband-based adaptive decorrelation filtering for co-channel speech separation
407 -- 416Ho-Young Jung, Soo-Young Lee. On the temporal decorrelation of feature parameters for noise-robust speech recognition
417 -- 428Mark J. F. Gales. Cluster adaptive training of hidden Markov models
429 -- 442Sahar E. Bou-Ghazale, John H. L. Hansen. A comparative study of traditional and newly proposed features for recognition of speech under stress
443 -- 453Marcio G. Siqueira, Abeer Alwan. Steady-state analysis of continuous adaptation in acoustic feedback reduction systems for hearing-aids
454 -- 466Akira Watanabe, Shingo Tomishige, Masahiro Nakatake. Speech visualization by integrating features for the hearing impaired
467 -- 470Andrew Horner. Low peak amplitudes for wavetable synthesis
471 -- 477Jean Laroche. Synthesis of sinusoids via non-overlapping inverse Fourier transform
478 -- 482S. Gökhun Tanyer, Hamza Özer. Voice activity detection in nonstationary noise
483 -- 487Chi-Min Liu, Chin-Chih Chiu, Hung-Yuan Chang. Design of vocabulary-independent Mandarin keyword spotters
488 -- 490João M. Rodrigues, Ana Maria Tomé. On the use of backward adaptation in a perceptual audio coder

Volume 8, Issue 3

221 -- 239Manohar N. Murthi, Bhaskar D. Rao. All-pole modeling of speech based on the minimum variance distortionless response spectrum
240 -- 254Ashwin Rao, Ramdas Kumaresan. On decomposing speech into modulated components
255 -- 266Yunxin Zhao. Frequency-domain maximum likelihood estimation for automatic speech recognition in additive and convolutive noises
267 -- 281Bayya Yegnanarayana, P. Satyanarayana Murthy. Enhancement of reverberant speech using LP residual signal
282 -- 291Ki Yong Lee, Souhwan Jung. Time-domain approach using multiple Kalman filters and EM algorithm to speech enhancement with nonstationary noise
292 -- 299Rafid A. Sukkar, Malan B. Gandhi, Anand R. Setlur. Speaker verification using mixture decomposition discrimination
300 -- 310Tero Tolonen, Vesa Välimäki, Matti Karjalainen. Modeling of tension modulation nonlinearity in plucked strings
311 -- 319Biljana D. Radlovic, Robert C. Williamson, Rodney A. Kennedy. Equalization in an acoustic reverberant environment: robustness results
320 -- 327Lucas C. Parra, Clay Spence. Convolutive blind separation of non-stationary sources
328 -- 344Shrikanth Narayanan, Abeer Alwan. Noise source models for fricative consonants
345 -- 348Nicola R. Chong, Ian S. Burnett, Joe F. Chicharo. A new waveform interpolation coding scheme based on pitch synchronous wavelet transform decomposition
349 -- 352Jae Bum Kim, K. Y. Lee, C. W. Lee. On the applications of the interacting multiple model algorithm for enhancing noisy speech
353 -- 357Gang Li, Lunji Qiu, Ling Kok Ng. Signal representation based on instantaneous amplitude models with application to speech synthesis

Volume 8, Issue 2

105 -- 114Linkai Bu, Tzi-Dar Church. Perceptual speech processing and phonetic feature mapping for robust vowel recognition
115 -- 125Lawrence K. Saul, Mazin G. Rahim. Maximum likelihood and minimum classification error factor analysis for automatic speech recognition
126 -- 139Eduardo Lleida, Richard C. Rose. Utterance verification in continuous speech recognition: decoding and training procedures
140 -- 145Ramon Arean, Jelena Kovacevic, Vivek K. Goyal. Multiple description perceptual audio coding with correlating transforms
146 -- 158Daniel Graupe, Dusan Veselinovic. Blind adaptive filtering of speech from noise of unknown spectrum using a virtual feedback configuration
159 -- 167Udar Mittal, Nam Phamdo. Signal/noise KLT based approach for enhancing speech degraded by colored noise
168 -- 172Jacob Benesty, Dennis R. Morgan, Jun H. Cho. A new class of doubletalk detectors based on cross-correlation
173 -- 176James G. Ryan, Rafik A. Goubran. Array optimization applied in the near field of a microphone array
177 -- 183James DeLucia, Fred Kochman. A new noniterative algorithm for computing acoustically constrained vocal tract area functions
184 -- 194Lauri Savioja, Vesa Välimäki. Reducing the dispersion error in the digital waveguide mesh using interpolation and frequency-warping techniques
195 -- 199Hong Kook Kim, Seung Ho Choi, Hwang Soo Lee. On approximating line spectral frequencies to LPC cepstral coefficients
200 -- 204Qiang Huo, Chin-Hui Lee. A Bayesian predictive classification approach to robust speech recognition
205 -- 208M. Padmanabhan, L. R. Ban. Model complexity adaptation using a discriminant measure
208 -- 211Lúcio Martins da Silva, Abraham Alcaim. Differential coding of speech LSF parameters using hybrid vector quantization and bidirectional prediction
211 -- 215Darren B. Ward. Joint least squares optimization for robust acoustic crosstalk cancellation

Volume 8, Issue 1

1 -- 2James R. Glass, Ronald Rosenfeld. Guest editorial introduction to the special issue on language modeling and dialogue systems
3 -- 10Giuseppe Riccardi, Allen L. Gorin. Stochastic language adaptation over time and state in natural spoken dialog systems
11 -- 23Esther Levin, Roberto Pieraccini, Wieland Eckert. A stochastic model of human-machine interaction for learning dialog strategies
24 -- 36Hermann Ney, Sonja Nießen, Franz Josef Och, Hassan Sawaf, Christoph Tillmann, Stephan Vogel. Algorithms for statistical translation of spoken language
37 -- 50Stanley F. Chen, Ronald Rosenfeld. A survey of smoothing techniques for ME models
51 -- 62Bernd Souvignier, Andreas Kellner, Bernhard Rueber, Hauke Schramm, Frank Seide. The thoughtful elephant: strategies for spoken dialog systems
63 -- 75Man-Hung Siu, Mari Ostendorf. Variable n-grams and extensions for conversational speech language modeling
76 -- 84Jerome R. Bellegarda. Large vocabulary speech recognition with multispan statistical language models
85 -- 96Victor 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