IEEE 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 Norden, 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 Meerkotter, 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, S. 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

• M. Gilbert, R. Moore, G. 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
  
• C.-C. Huang, J.-F. Wang, D.-J. 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, J. R. Deller, 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
  
• Y. 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, S. 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, G. 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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