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ORCIDIEEE Transactions on Speech and Audio Processing, Volume 13
Volume 13, Number 1, January 2005
- Muhammad Z. Ikram, Dennis R. Morgan:
Permutation inconsistency in blind speech separation: investigation and solutions. 1-13 - Pere Pujol, Susagna Pol, Climent Nadeu
, Astrid Hagen, Hervé Bourlard:
Comparison and combination of features in a hybrid HMM/MLP and a HMM/GMM speech recognition system. 14-22 - Frank Wessel, Hermann Ney:
Unsupervised training of acoustic models for large vocabulary continuous speech recognition. 23-31 - Michael M. Goodwin, A. J. Hipple, B. Link:
Predicting and preventing unmasking incurred in coded audio post-processing. 32-41 - Joshua M. Sachar, Harvey F. Silverman, William R. Patterson III:
Microphone position and gain calibration for a large-aperture microphone array. 42-52 - Simon Doclo, Marc Moonen:
Multimicrophone noise reduction using recursive GSVD-based optimal filtering with ANC postprocessing stage. 53-69 - Vikas C. Raykar, Igor Kozintsev, Rainer Lienhart:
Position calibration of microphones and loudspeakers in distributed computing platforms. 70-83 - Stuart N. Wrigley, Guy J. Brown, Vincent Wan, Steve Renals:
Speech and crosstalk detection in multichannel audio. 84-91 - Scott C. Douglas, Hiroshi Sawada, Shoji Makino:
Natural gradient multichannel blind deconvolution and speech separation using causal FIR filters. 92-104 - Ville Pulkki, Toni Hirvonen:
Localization of virtual sources in multichannel audio reproduction. 105-119 - Herbert Buchner, Robert Aichner, Walter Kellermann:
A generalization of blind source separation algorithms for convolutive mixtures based on second-order statistics. 120-134 - Boaz Rafaely:
Analysis and design of spherical microphone arrays. 135-143
Volume 13, Number 2, March 2005
- Ted Painter, Andreas Spanias:
Perceptual segmentation and component selection for sinusoidal representations of audio. 149-162 - Fredrik Nordén, Per Hedelin:
Companded quantization of speech MDCT coefficients. 163-173 - Robert E. Schapire, Marie Rochery, Mazin G. Rahim, Narendra K. Gupta:
Boosting with prior knowledge for call classification. 174-181 - Jen-Tzung Chien:
Decision tree State tying using cluster validity criteria. 182-193 - Dong Kook Kim, Nam Soo Kim:
Rapid online adaptation based on transformation space model evolution. 194-202 - Vincent Wan, Steve Renals:
Speaker verification using sequence discriminant support vector machines. 203-210 - William J. J. Roberts, Yariv Ephraim, Howard W. Sabrin:
Speaker classification using composite hypothesis testing and list decoding. 211-219 - Renat Vafin, W. Bastiaan Kleijn:
Entropy-constrained polar quantization and its application to audio coding. 220-232 - Dietrich Fränken, Klaus Meerkötter, Joachim Wassmuth:
Observer-based feedback linearization of dynamic loudspeakers with Ac amplifiers. 233-242 - Lorenzo Turicchia, Rahul Sarpeshkar:
A bio-inspired companding strategy for spectral enhancement. 243-253 - H. K. Jang, Ju Sung Park:
Multiresolution sinusoidal model with dynamic segmentation for timescale modification of polyphonic audio signals. 254-262 - Athanasios Mouchtaris, Shrikanth S. Narayanan, Chris Kyriakakis:
Multichannel audio synthesis by subband-based spectral conversion and parameter adaptation. 263-274 - William A. Sethares, Robin D. Morris, James C. Sethares:
Beat tracking of musical performances using low-level audio features. 275-285 - Mrityunjoy Chakraborty, Hideaki Sakai:
Convergence analysis of a complex LMS algorithm with tonal reference signals. 286-292 - Chul Min Lee, Shrikanth S. Narayanan:
Toward detecting emotions in spoken dialogs. 293-303
Volume 13, Number 3, May 2005
- Tetsuya Hoya, Toshihisa Tanaka, Andrzej Cichocki, Takahiro Murakami, Gen Hori, Jonathon A. Chambers:
Stereophonic noise reduction using a combined sliding subspace projection and adaptive signal enhancement. 309-320 - Alexandros Potamianos, Shrikanth S. Narayanan, Giuseppe Riccardi:
Adaptive categorical understanding for spoken dialogue systems. 321-329 - Chung-Hsien Wu, Gwo-Lang Yan:
Speech act modeling and verification of spontaneous speech with disfluency in a spoken dialogue system. 330-344 - Patrick Kenny, Gilles Boulianne, Pierre Dumouchel:
Eigenvoice modeling with sparse training data. 345-354 - Ángel de la Torre, Antonio M. Peinado, José C. Segura, José L. Pérez-Córdoba, M. Carmen Benítez, Antonio J. Rubio:
Histogram equalization of speech representation for robust speech recognition. 355-366 - Stavros Tsakalidis, Vlasios Doumpiotis, William J. Byrne:
Discriminative linear transforms for feature normalization and speaker adaptation in HMM estimation. 367-376 - Jen-Tzung Chien, Sadaoki Furui:
Predictive hidden Markov model selection for speech recognition. 377-387 - Mohamed Afify:
Accurate compensation in the log-spectral domain for noisy speech recognition. 388-398 - Yu Tsao, Shang-Ming Lee, Lin-Shan Lee:
Segmental eigenvoice with delicate eigenspace for improved speaker adaptation. 399-411 - Li Deng, Jasha Droppo, Alex Acero:
Dynamic compensation of HMM variances using the feature enhancement uncertainty computed from a parametric model of speech distortion. 412-421 - Charles D. Creusere:
Understanding perceptual distortion in MPEG scalable audio coding. 422-431 - Mohamed F. Mansour, Ahmed H. Tewfik:
Data embedding in audio using time-scale modification. 432-440 - Changsheng Xu, Namunu Chinthaka Maddage, Xi Shao:
Automatic music classification and summarization. 441-450
Volume 13, Number 4, July 2005
- Isabel Trancoso:
Editorial. 457 - Michael T. Johnson, Richard J. Povinelli, Andrew C. Lindgren, Jinjin Ye, Xiaolin Liu, Kevin M. Indrebo:
Time-domain isolated phoneme classification using reconstructed phase spaces. 458-466 - Bowen Zhou, John H. L. Hansen:
Efficient audio stream segmentation via the combined T2 statistic and Bayesian information criterion. 467-474 - Chang Huai You, Soo Ngee Koh, Susanto Rahardja:
beta-order MMSE spectral amplitude estimation for speech enhancement. 475-486 - Ann Spriet, Marc Moonen, Jan Wouters:
Robustness analysis of multichannel Wiener filtering and generalized sidelobe cancellation for multimicrophone noise reduction in hearing aid applications. 487-503 - Giuseppe Riccardi, Dilek Hakkani-Tür:
Active learning: theory and applications to automatic speech recognition. 504-511 - Mukund Padmanabhan, Satya Dharanipragada:
Maximizing information content in feature extraction. 512-519 - Frank Seide:
The use of virtual hypothesis copies in decoding of large-vocabulary continuous speech. 520-533 - Ruhi Sarikaya, Yuqing Gao, Michael Picheny, Hakan Erdogan:
Semantic confidence measurement for spoken dialog systems. 534-545 - Mohamed Afify, Feng Liu, Hui Jiang, Olivier Siohan:
A new verification-based fast-match for large vocabulary continuous speech recognition. 546-553 - Bowen Zhou, John H. L. Hansen:
Rapid discriminative acoustic model based on eigenspace mapping for fast speaker adaptation. 554-564 - Kuo-Hwei Yuo, Tai-Hwei Hwang, Hsiao-Chuan Wang:
Combination of autocorrelation-based features and projection measure technique for speaker identification. 565-574 - B. Yegnanarayana, S. R. Mahadeva Prasanna, Jinu Mariam Zachariah, Cheedella S. Gupta:
Combining evidence from source, suprasegmental and spectral features for a fixed-text speaker verification system. 575-582 - Masafumi Nishida, Tatsuya Kawahara:
Speaker model selection based on the Bayesian information criterion applied to unsupervised speaker indexing. 583-592 - Harvey F. Silverman, Ying Yu, Joshua M. Sachar, William R. Patterson III:
Performance of real-time source-location estimators for a large-aperture microphone array. 593-606 - Ying Song, Yu Gong, Sen M. Kuo:
A robust hybrid feedback active noise cancellation headset. 607-617 - Ming Zhang, Hui Lan, Wee Ser:
On comparison of online secondary path modeling methods with auxiliary noise. 618-628
Volume 13, Numbers 5-1, September 2005
- Mazin Gilbert, Roger K. Moore, Geoffrey Zweig:
Introduction to the Special Issue on Data Mining of Speech, Audio, and Dialog. 633-634 - Peng Yu, Kaijiang Chen, Chengyuan Ma, Frank Seide:
Vocabulary-Independent Indexing of Spontaneous Speech. 635-643 - Chien-Chang Lin, Shi-Huang Chen, Trieu-Kien Truong, Yukon Chang:
Audio Classification and Categorization Based on Wavelets and Support Vector Machine. 644-651 - Shona Douglas, Deepak Agarwal, Tirso Alonso, Robert M. Bell, Mazin Gilbert, Deborah F. Swayne, Chris Volinsky:
Mining Customer Care Dialogs for "Daily News". 652-660 - Dong Yu, Alex Acero:
Semiautomatic Improvements of System-Initiative Spoken Dialog Applications Using Interactive Clustering. 661-671 - Lee Begeja, Harris Drucker, David C. Gibbon, Patrick Haffner, Zhu Liu, Bernard Renger, Behzad Shahraray:
Semantic Data Mining of Short Utterances. 672-680 - Lina Zhou, Yongmei Shi, Jinjuan Feng, Andrew Sears:
Data Mining for Detecting Errors in Dictation Speech Recognition. 681-688 - Chaug-Ching Huang, Jhing-Fa Wang, Dian-Jia Wu:
Automatic Scene Change Detection for Composed Speech and Music Sound Under Low SNR Noisy Environment. 689-699 - John Grothendieck:
Tracking Changes in Language. 700-711 - John H. L. Hansen, Rongqing Huang, Bowen Zhou, Michael S. Seadle, John R. Deller Jr., Aparna Gurijala, Mikko Kurimo, Pongtep Angkititrakul:
SpeechFind: Advances in Spoken Document Retrieval for a National Gallery of the Spoken Word. 712-730
Volume 13, Numbers 5-2, September 2005
- S. S. Yedlapalli:
Transforming Real Linear Prediction Coefficients to Line Spectral Representations With a Real FFT. 733-740 - Minkyu Lee, Jan P. H. van Santen, Bernd Möbius, Joseph P. Olive:
Formant Tracking Using Context-Dependent Phonemic Information. 741-750 - Vikas C. Raykar, B. Yegnanarayana, S. R. Mahadeva Prasanna, Ramani Duraiswami:
Speaker Localization Using Excitation Source Information in Speech. 751-761 - B.-F. Wu, K.-C. Wang:
Robust Endpoint Detection Algorithm Based on the Adaptive Band-Partitioning Spectral Entropy in Adverse Environments. 762-775 - Om Deshmukh, Carol Y. Espy-Wilson, Ariel Salomon, Jawahar Singh:
Use of Temporal Information: Detection of Periodicity, Aperiodicity, and Pitch in Speech. 776-786 - Jonas Lindblom:
A Sinusoidal Voice Over Packet Coder Tailored for the Frame-Erasure Channel. 787-798 - Mikko Tammi, Milan Jelinek, Vesa T. Ruoppila:
Signal Modification Method for Variable Bit Rate Wide-band Speech Coding. 799-810 - Turaj Zakizadeh Shabestary, Per Hedelin:
LSP Quantization by a Union of Locally Trained Codebooks. 811-820 - Doh-Suk Kim:
ANIQUE: An Auditory Model for Single-Ended Speech Quality Estimation. 821-831 - Kamran Rahbar, James P. Reilly:
A Frequency Domain Method for Blind Source Separation of Convolutive Audio Mixtures. 832-844 - Rainer Martin:
Speech Enhancement Based on Minimum Mean-Square Error Estimation and Supergaussian Priors. 845-856 - Philipos C. Loizou:
Speech Enhancement Based on Perceptually Motivated Bayesian Estimators of the Magnitude Spectrum. 857-869 - Israel Cohen:
Relaxed Statistical Model for Speech Enhancement and a Priori SNR Estimation. 870-881 - Yiteng Huang, Jacob Benesty, Jingdong Chen:
A Blind Channel Identification-Based Two-Stage Approach to Separation and Dereverberation of Speech Signals in a Reverberant Environment. 882-895 - Saeed Gazor, Wei Zhang:
Speech enhancement employing Laplacian-Gaussian mixture. 896-904 - Ting Liu, Saeed Gazor:
A Variable Step-Size Pre-Filter-Bank Adaptive Algorithm. 905-916 - Alfonso Ortega, Eduardo Lleida, Enrique Masgrau:
Speech Reinforcement System for Car Cabin Communications. 917-929 - Michael Pitz, Hermann Ney:
Vocal Tract Normalization Equals Linear Transformation in Cepstral Space. 930-944 - Hui Jiang, Frank K. Soong, Chin-Hui Lee:
A Dynamic In-Search Data Selection Method With Its Applications to Acoustic Modeling and Utterance Verification. 945-955 - James McAuley, Ji Ming, Darryl Stewart, Philip Hanna:
Subband Correlation and Robust Speech Recognition. 956-964 - K. Li, M. N. S. Swamy, M. Omair Ahmad:
An Improved Voice Activity Detection Using Higher Order Statistics. 965-974 - Yifan Gong:
A Method of Joint Compensation of Additive and Convolutive Distortions for Speaker-Independent Speech Recognition. 975-983 - Brian Mak, James Tin-Yau Kwok, Simon Ka-Lung Ho:
Kernel Eigenvoice Speaker Adaptation. 984-992 - Brian Kan-Wing Mak, Kin-Wah Chan:
Pruning Hidden Markov Models With Optimal Brain Surgeon. 993-1003 - S. Kwon, Shri Narayanan:
Unsupervised Speaker Indexing Using Generic Models. 1004-1013 - Gerald Schuller, Jelena Kovacevic, F. Masson, Vivek K. Goyal:
Robust Low-Delay Audio Coding Using Multiple Descriptions. 1014-1024 - Stanley T. Birchfield, Amarnag Subramanya:
Microphone Array Position Calibration by Basis-Point Classical Multidimensional Scaling. 1025-1034 - Juan Pablo Bello, Laurent Daudet, Samer A. Abdallah, Chris Duxbury, Mike E. Davies, Mark B. Sandler:
A Tutorial on Onset Detection in Music Signals. 1035-1047 - Christof Faller, Jingdong Chen:
Suppressing Acoustic Echo in a Spectral Envelope Space. 1048-1062 - G. R. Campos, David M. Howard:
On the Computational Efficiency of Different Waveguide Mesh Topologies for Room Acoustic Simulation. 1063-1072 - Federico Avanzini, Stefania Serafin, Davide Rocchesso:
Interactive Simulation of Rigid Body Interaction With Friction-Induced Sound Generation. 1073-1081 - Muhammad Tahir Akhtar, Masahide Abe, Masayuki Kawamata:
A New Structure for Feedforward Active Noise Control Systems With Improved Online Secondary Path Modeling. 1082-1088
Volume 13, Number 6, November 2005
- Broneslav A. Kiselman, Vladimir V. Krylov:
Comparative analysis of linear and nonlinear speech signals predictors. 1093-1097 - Iasonas Kokkinos, Petros Maragos:
Nonlinear speech analysis using models for chaotic systems. 1098-1109 - B. Yegnanarayana, S. R. Mahadeva Prasanna, Ramani Duraiswami, Dmitry N. Zotkin:
Processing of reverberant speech for time-delay estimation. 1110-1118 - Javier Ramírez, José C. Segura, M. Carmen Benítez, Ángel de la Torre, Antonio J. Rubio:
An effective subband OSF-based VAD with noise reduction for robust speech recognition. 1119-1129 - Geert Rombouts, Marc Moonen:
Fast QRD-lattice-based unconstrained optimal filtering for acoustic noise reduction. 1130-1143 - Scott Axelrod, Vaibhava Goel, Ramesh A. Gopinath, Peder A. Olsen, Karthik Visweswariah:
Subspace constrained Gaussian mixture models for speech recognition. 1144-1160 - Xiaodong Cui, Abeer Alwan:
Noise robust speech recognition using feature compensation based on polynomial regression of utterance SNR. 1161-1172 - Thomas Hain, Philip C. Woodland, Gunnar Evermann, Mark J. F. Gales, Xunying Liu, Gareth L. Moore, Daniel Povey, Lan Wang:
Automatic transcription of conversational telephone speech. 1173-1185 - Ascensión Gallardo-Antolín, Carmen Peláez-Moreno, Fernando Díaz-de-María:
Recognizing GSM digital speech. 1186-1205 - Jae-Sik Lee, Jong-Hoon Jeong, Tae-Gyu Chang:
An efficient method of Huffman decoding for MPEG-2 AAC and its performance analysis. 1206-1209 - I. Kauppinen, K. Roth:
Improved noise reduction in audio signals using spectral resolution enhancement with time-domain signal extrapolation. 1210-1216 - Orlando José Tobias, Rui Seara:
Leaky-FXLMS algorithm: stochastic analysis for Gaussian data and secondary path modeling error. 1217-1230 - Per Åhgren:
Acoustic echo cancellation and doubletalk detection using estimated loudspeaker impulse responses. 1231-1237
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