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ORCIDIEEE Transactions on Audio, Speech & Language Processing, Volume 15
Volume 15, Number 1, January 2007
- Paris Smaragdis:
Convolutive Speech Bases and Their Application to Supervised Speech Separation. 1-12 - Li Deng, Leo J. Lee, Hagai Attias, Alex Acero:
Adaptive Kalman Filtering and Smoothing for Tracking Vocal Tract Resonances Using a Continuous-Valued Hidden Dynamic Model. 13-23 - Ben Milner, Xu Shao:
Prediction of Fundamental Frequency and Voicing From Mel-Frequency Cepstral Coefficients for Unconstrained Speech Reconstruction. 24-33 - Patrick A. Naylor
, Anastasis Kounoudes, Jón Guðnason, Mike Brookes:
Estimation of Glottal Closure Instants in Voiced Speech Using the DYPSA Algorithm. 34-43 - Farshad Lahouti, Amir K. Khandani:
Soft Reconstruction of Speech in the Presence of Noise and Packet Loss. 44-56 - Sean A. Ramprashad:
Sparse Bit-Allocations Based on Partial Ordering Schemes With Application to Speech and Audio Coding. 57-69 - Taesu Kim, Hagai Thomas Attias, Soo-Young Lee, Te-Won Lee:
Blind Source Separation Exploiting Higher-Order Frequency Dependencies. 70-79 - Tomohiro Nakatani, Keisuke Kinoshita, Masato Miyoshi:
Harmonicity-Based Blind Dereverberation for Single-Channel Speech Signals. 80-95 - Bertrand Rivet, Laurent Girin, Christian Jutten:
Mixing Audiovisual Speech Processing and Blind Source Separation for the Extraction of Speech Signals From Convolutive Mixtures. 96-108 - Guangji Shi, Parham Aarabi, Hui Jiang:
Phase-Based Dual-Microphone Speech Enhancement Using A Prior Speech Model. 109-118 - Gwo-hwa Ju, Lin-Shan Lee:
A Perceptually Constrained GSVD-Based Approach for Enhancing Speech Corrupted by Colored Noise. 119-134 - Steven J. Rennie, Parham Aarabi, Brendan J. Frey:
Variational Probabilistic Speech Separation Using Microphone Arrays. 135-149 - Ian R. Lane, Tatsuya Kawahara, Tomoko Matsui, Satoshi Nakamura:
Out-of-Domain Utterance Detection Using Classification Confidences of Multiple Topics. 150-161 - Christian Raymond, Frédéric Béchet, Nathalie Camelin, Renato de Mori, Géraldine Damnati:
Sequential Decision Strategies for Machine Interpretation of Speech. 162-171 - Scott Axelrod, Vaibhava Goel, Ramesh A. Gopinath, Peder A. Olsen, Karthik Visweswariah:
Discriminative Estimation of Subspace Constrained Gaussian Mixture Models for Speech Recognition. 172-189 - Rajesh M. Hegde, Hema A. Murthy, Venkata Ramana Rao Gadde:
Significance of the Modified Group Delay Feature in Speech Recognition. 190-202 - Erik McDermott, Timothy J. Hazen, Jonathan Le Roux, Atsushi Nakamura, Shigeru Katagiri:
Discriminative Training for Large-Vocabulary Speech Recognition Using Minimum Classification Error. 203-223 - Satya Dharanipragada, Umit H. Yapanel, Bhaskar D. Rao:
Robust Feature Extraction for Continuous Speech Recognition Using the MVDR Spectrum Estimation Method. 224-234 - Michael L. Seltzer, Alex Acero:
Training Wideband Acoustic Models Using Mixed-Bandwidth Training Data for Speech Recognition. 235-245 - Joe Frankel, Simon King:
Speech Recognition Using Linear Dynamic Models. 246-256 - Chia-Ping Chen, Jeff A. Bilmes:
MVA Processing of Speech Features. 257-270 - Haizhou Li, Bin Ma, Chin-Hui Lee:
A Vector Space Modeling Approach to Spoken Language Identification. 271-284 - Peter Day, Asoke K. Nandi:
Robust Text-Independent Speaker Verification Using Genetic Programming. 285-295 - Youngim Jung, Ae-sun Yoon, Hyuk-Chul Kwon:
Grapheme-to-Phoneme Conversion of Arabic Numeral Expressions for Embedded TTS Systems. 296-309 - Jan H. Plasberg, W. Bastiaan Kleijn:
The Sensitivity Matrix: Using Advanced Auditory Models in Speech and Audio Processing. 310-319 - Ixone Arroabarren, Alfonso Carlosena:
Voice Production Mechanisms of Vocal Vibrato in Male Singers. 320-332 - Kazuyoshi Yoshii, Masataka Goto, Hiroshi G. Okuno:
Drum Sound Recognition for Polyphonic Audio Signals by Adaptation and Matching of Spectrogram Templates With Harmonic Structure Suppression. 333-345 - Kishan Thambiratnam, Sridha Sridharan:
Rapid Yet Accurate Speech Indexing Using Dynamic Match Lattice Spotting. 346-357 - Paris Smaragdis, Petros Boufounos:
Position and Trajectory Learning for Microphone Arrays. 358-368
Volume 15, Number 2, February 2007
- Yannis Agiomyrgiannakis, Yannis Stylianou:
Conditional Vector Quantization for Speech Coding. 377-386 - Sorin Dusan, James L. Flanagan, Amod Karve, Mridul Balaraman:
Speech Compression by Polynomial Approximation. 387-395 - Guoning Hu, DeLiang Wang:
Auditory Segmentation Based on Onset and Offset Analysis. 396-405 - Richard C. Hendriks, Richard Heusdens, Jesper Jensen:
An MMSE Estimator for Speech Enhancement Under a Combined Stochastic-Deterministic Speech Model. 406-415 - Yoshifumi Nagata, Toyota Fujioka, Masato Abe:
Two-Dimensional DOA Estimation of Sound Sources Based on Weighted Wiener Gain Exploiting Two-Directional Microphones. 416-429 - Marc Delcroix, Takafumi Hikichi, Masato Miyoshi:
Precise Dereverberation Using Multichannel Linear Prediction. 430-440 - Sriram Srinivasan, Jonas Samuelsson, W. Bastiaan Kleijn:
Codebook-Based Bayesian Speech Enhancement for Nonstationary Environments. 441-452 - Rongqing Huang, John H. L. Hansen, Pongtep Angkititrakul:
Dialect/Accent Classification Using Unrestricted Audio. 453-464 - Murat Akbacak, John H. L. Hansen:
Environmental Sniffing: Noise Knowledge Estimation for Robust Speech Systems. 465-477 - Jian Wu, Qiang Huo:
A Study of Minimum Classification Error (MCE) Linear Regression for Supervised Adaptation of MCE-Trained Continuous-Density Hidden Markov Models. 478-488 - Paul D. Teal:
Tracking Wide-Band Targets Having Significant Doppler Shift. 489-497 - Pongtep Angkititrakul, John H. L. Hansen:
Discriminative In-Set/Out-of-Set Speaker Recognition. 498-508 - Darko Kirovski, Zeph Landau:
Generalized Lempel-Ziv Compression for Audio. 509-518 - Tin Lay Nwe, Haizhou Li:
Exploring Vibrato-Motivated Acoustic Features for Singer Identification. 519-530 - Nicola Laurenti, Giovanni De Poli, Daniele Montagner:
A Nonlinear Method for Stochastic Spectrum Estimation in the Modeling of Musical Sounds. 531-541 - Sunil Bharitkar, Chris Kyriakakis:
Visualization of Multiple Listener Room Acoustic Equalization With the Sammon Map. 542-551 - Damian T. Murphy, Mark Beeson:
The KW-Boundary Hybrid Digital Waveguide Mesh for Room Acoustics Applications. 552-564 - Ramani Duraiswami, Dmitry N. Zotkin, Nail A. Gumerov:
Fast Evaluation of the Room Transfer Function Using Multipole Expansion. 565-576 - Jack Mullen, David M. Howard, Damian T. Murphy:
Real-Time Dynamic Articulations in the 2-D Waveguide Mesh Vocal Tract Model. 577-585 - Xu Sun, Sen M. Kuo:
Active Narrowband Noise Control Systems Using Cascading Adaptive Filters. 586-592 - Muhammad Tahir Akhtar, Masahide Abe, Masayuki Kawamata:
On Active Noise Control Systems With Online Acoustic Feedback Path Modeling. 593-600 - Daniel Gatica-Perez, Guillaume Lathoud, Jean-Marc Odobez, Iain McCowan:
Audiovisual Probabilistic Tracking of Multiple Speakers in Meetings. 601-616 - Simon Doclo, Marc Moonen:
Superdirective Beamforming Robust Against Microphone Mismatch. 617-631 - Chang-Heon Lee, Sung-Kyo Jung, Hong-Goo Kang:
Applying a Speaker-Dependent Speech Compression Technique to Concatenative TTS Synthesizers. 632-640 - K.-S. Lee:
Statistical Approach for Voice Personality Transformation. 641-651 - Xiaodong Cui, Abeer Alwan:
Robust Speaker Adaptation by Weighted Model Averaging Based on the Minimum Description Length Criterion. 652-660 - M.-Y. Tsai, F.-C. Chou, L.-S. Lee:
Pronunciation Modeling With Reduced Confusion for Mandarin Chinese Using a Three-Stage Framework. 661-675 - Qin Yan, Saeed Vaseghi, Dimitrios Rentzos, Ching-Hsiang Ho:
Analysis and Synthesis of Formant Spaces of British, Australian, and American Accents. 676-689 - Dagen Wang, Shrikanth S. Narayanan:
An Acoustic Measure for Word Prominence in Spontaneous Speech. 690-701 - Zhiyun Li, Ramani Duraiswami:
Flexible and Optimal Design of Spherical Microphone Arrays for Beamforming. 702-714 - Mirko Knaak, Shoko Araki, Shoji Makino:
Geometrically Constrained Independent Component Analysis. 715-726 - I. Balmages, Boaz Rafaely:
Open-Sphere Designs for Spherical Microphone Arrays. 727-732 - Peter Jancovic:
Fast Algorithm for Calculation of the Union-Based Probability. 732-734 - Young-Ik Kim, Rhee Man Kil:
Estimation of Interaural Time Differences Based on Zero-Crossings in Noisy Multisource Environments. 734-743
Volume 15, Number 3, March 2007
- Pradeepa Yahampath, Paul Rondeau:
Multiple-Description Predictive-Vector Quantization With Applications to Low Bit-Rate Speech Coding Over Networks. 749-755 - Ethan Robert Duni, Bhaskar D. Rao:
High-Rate Optimized Recursive Vector Quantization Structures Using Hidden Markov Models. 756-769 - Ethan Robert Duni, Bhaskar D. Rao:
A High-Rate Optimal Transform Coder With Gaussian Mixture Companders. 770-783 - Brian Kan-Wing Mak, Roger Wend-Huu Hsiao:
Kernel Eigenspace-Based MLLR Adaptation. 784-795 - Bertrand Rivet, Laurent Girin, Christian Jutten:
Log-Rayleigh Distribution: A Simple and Efficient Statistical Representation of Log-Spectral Coefficients. 796-802 - Patricia Scanlon, Daniel P. W. Ellis, Richard B. Reilly:
Using Broad Phonetic Group Experts for Improved Speech Recognition. 803-812 - Barbara Resch, Mattias Nilsson, L. Anders Ekman, W. Bastiaan Kleijn:
Estimation of the Instantaneous Pitch of Speech. 813-822 - Francesco Gianfelici, Giorgio Biagetti, Paolo Crippa, Claudio Turchetti:
Multicomponent AM-FM Representations: An Asymptotically Exact Approach. 823-837 - Dima Ruinskiy, Yizhar Lavner:
An Effective Algorithm for Automatic Detection and Exact Demarcation of Breath Sounds in Speech and Song Signals. 838-850 - Laurent Girin, Mohammad Firouzmand, Sylvain Marchand:
Perceptual Long-Term Variable-Rate Sinusoidal Modeling of Speech. 851-861 - Jesper Jensen, Richard Heusdens:
Improved Subspace-Based Single-Channel Speech Enhancement Using Generalized Super-Gaussian Priors. 862-872 - Juho Kontio, Laura Laaksonen, Paavo Alku:
Neural Network-Based Artificial Bandwidth Expansion of Speech. 873-881 - David Yuheng Zhao, W. Bastiaan Kleijn:
HMM-Based Gain Modeling for Enhancement of Speech in Noise. 882-892 - M. Khademul Islam Molla, Keikichi Hirose:
Single-Mixture Audio Source Separation by Subspace Decomposition of Hilbert Spectrum. 893-900 - Karsten Vandborg Sørensen, Søren Vang Andersen:
Rayleigh Mixture Model-Based Hidden Markov Modeling and Estimation of Noise in Noisy Speech Signals. 901-917 - Richard C. Hendriks, Rainer Martin:
MAP Estimators for Speech Enhancement Under Normal and Rayleigh Inverse Gaussian Distributions. 918-927 - Nikos Chatzichrisafis, Vassilios Diakoloukas, Vassilios Digalakis, Costas Harizakis:
Gaussian Mixture Clustering and Language Adaptation for the Development of a New Language Speech Recognition System. 928-938 - Ghinwa F. Choueiter, James R. Glass:
An Implementation of Rational Wavelets and Filter Design for Phonetic Classification. 939-948 - Esther Klabbers, Jan P. H. van Santen, Alexander Kain:
The Contribution of Various Sources of Spectral Mismatch to Audible Discontinuities in a Diphone Database. 949-956 - Jerome R. Bellegarda:
Globally Optimal Training of Unit Boundaries in Unit Selection Text-to-Speech Synthesis. 957-965 - Pim Korten, Jesper Jensen, Richard Heusdens:
High-Resolution Spherical Quantization of Sinusoidal Parameters. 966-981 - Hirokazu Kameoka, Takuya Nishimoto, Shigeki Sagayama:
A Multipitch Analyzer Based on Harmonic Temporal Structured Clustering. 982-994 - Johannes Nix, Volker Hohmann:
Combined Estimation of Spectral Envelopes and Sound Source Direction of Concurrent Voices by Multidimensional Statistical Filtering. 995-1008 - Matthew E. P. Davies, Mark D. Plumbley:
Context-Dependent Beat Tracking of Musical Audio. 1009-1020 - Leevi Peltola, Cumhur Erkut, Perry R. Cook, Vesa Välimäki:
Synthesis of Hand Clapping Sounds. 1021-1029 - Jean-Marc Valin:
On Adjusting the Learning Rate in Frequency Domain Echo Cancellation With Double-Talk. 1030-1034 - James D. Gordy, Rafik A. Goubran:
Statistical Analysis of Doubletalk Detection for Calibration and Performance Evaluation. 1035-1043 - Felix Albu, Martin Bouchard, Yuriy V. Zakharov:
Pseudo-Affine Projection Algorithms for Multichannel Active Noise Control. 1044-1052 - Jacob Benesty, Jingdong Chen, Yiteng Huang, Jacek Dmochowski:
On Microphone-Array Beamforming From a MIMO Acoustic Signal Processing Perspective. 1053-1065 - Tuomas Virtanen:
Monaural Sound Source Separation by Nonnegative Matrix Factorization With Temporal Continuity and Sparseness Criteria. 1066-1074 - Carlos Busso, Zhigang Deng, Michael Grimm, Ulrich Neumann, Shrikanth S. Narayanan:
Rigid Head Motion in Expressive Speech Animation: Analysis and Synthesis. 1075-1086 - Chen Yang, Frank K. Soong, Tan Lee:
Static and Dynamic Spectral Features: Their Noise Robustness and Optimal Weights for ASR. 1087-1097 - Luis Buera, Eduardo Lleida, Antonio Miguel, Alfonso Ortega, Oscar Saz:
Cepstral Vector Normalization Based on Stereo Data for Robust Speech Recognition. 1098-1113 - Xianyu Zhao, Zhijian Ou:
Closely Coupled Array Processing and Model-Based Compensation for Microphone Array Speech Recognition. 1114-1122
Volume 15, Number 4, May 2007
- Rasool Tahmasbi, Sadegh Rezaei:
A Soft Voice Activity Detection Using GARCH Filter and Variance Gamma Distribution. 1129-1134 - Jonathan Le Roux, Hirokazu Kameoka, Nobutaka Ono, Alain de Cheveigné, Shigeki Sagayama:
Single and Multiple F0 Contour Estimation Through Parametric Spectrogram Modeling of Speech in Noisy Environments. 1135-1145 - Thomas Eriksson, Frank Norden:
Memory-Based Vector Quantization of LSF Parameters by a Power Series Approximation. 1146-1155 - Bengt J. Borgstrom, Mihaela van der Schaar, Abeer Alwan:
Rate Allocation for Noncollaborative Multiuser Speech Communication Systems Based on Bargaining Theory. 1156-1166 - Milan Jelinek, Redwan Salami:
Wideband Speech Coding Advances in VMR-WB Standard. 1167-1179 - Athanasios Mouchtaris, Jan Van der Spiegel, Paul Mueller, Panagiotis Tsakalides:
A Spectral Conversion Approach to Single-Channel Speech Enhancement. 1180-1193 - Esfandiar Zavarehei, Saeed Vaseghi, Qin Yan:
Noisy Speech Enhancement Using Harmonic-Noise Model and Codebook-Based Post-Processing. 1194-1203 - Xuechuan Wang, Douglas D. O'Shaughnessy:
Environmental Independent ASR Model Adaptation/Compensation by Bayesian Parametric Representation. 1204-1217 - Peter Birkholz, Dietmar Jackèl, Bernd J. Kröger:
Simulation of Losses Due to Turbulence in the Time-Varying Vocal System. 1218-1226 - Chung-Hsien Wu, Chi-Chun Hsia, Jiun-Fu Chen, Jhing-Fa Wang:
Variable-Length Unit Selection in TTS Using Structural Syntactic Cost. 1227-1235 - Karthikeyan Umapathy, Sridhar Krishnan, R. K. Rao:
Audio Signal Feature Extraction and Classification Using Local Discriminant Bases. 1236-1246 - Graham E. Poliner, Daniel P. W. Ellis, Andreas F. Ehmann, Emilia Gómez, Sebastian Streich, Beesuan Ong:
Melody Transcription From Music Audio: Approaches and Evaluation. 1247-1256 - Harvey D. Thornburg, Randal J. Leistikow, Jonathan Berger:
Melody Extraction and Musical Onset Detection via Probabilistic Models of Framewise STFT Peak Data. 1257-1272 - Emmanuel Vincent, Mark D. Plumbley:
Low Bit-Rate Object Coding of Musical Audio Using Bayesian Harmonic Models. 1273-1282 - Corentin Dubois, Manuel Davy:
Joint Detection and Tracking of Time-Varying Harmonic Components: A Flexible Bayesian Approach. 1283-1295 - H. M. A. Malik, Rashid Ansari, Ashfaq A. Khokhar:
Robust Data Hiding in Audio Using Allpass Filters. 1296-1304 - Yekutiel Avargel, Israel Cohen:
System Identification in the Short-Time Fourier Transform Domain With Crossband Filtering. 1305-1319 - Fredric Lindström, Christian Schüldt, Ingvar Claesson:
An Improvement of the Two-Path Algorithm Transfer Logic for Acoustic Echo Cancellation. 1320-1326 - Jacek Dmochowski, Jacob Benesty, Sofiène Affes:
Direction of Arrival Estimation Using the Parameterized Spatial Correlation Matrix. 1327-1339 - Wolfgang Herbordt, Herbert Buchner, Satoshi Nakamura, Walter Kellermann:
Multichannel Bin-Wise Robust Frequency-Domain Adaptive Filtering and Its Application to Adaptive Beamforming. 1340-1351 - Takaaki Hori, Chiori Hori, Yasuhiro Minami, Atsushi Nakamura:
Efficient WFST-Based One-Pass Decoding With On-The-Fly Hypothesis Rescoring in Extremely Large Vocabulary Continuous Speech Recognition. 1352-1365 - Xiaodong Cui, Yifan Gong:
A Study of Variable-Parameter Gaussian Mixture Hidden Markov Modeling for Noisy Speech Recognition. 1366-1376