Journal: IEEE Transactions on Audio, Speech & Language Processing

Volume 13, Issue 6

1093 -- 1097Broneslav A. Kiselman, Vladimir V. Krylov. Comparative analysis of linear and nonlinear speech signals predictors
1098 -- 1109Iasonas Kokkinos, Petros Maragos. Nonlinear speech analysis using models for chaotic systems
1110 -- 1118B. Yegnanarayana, S. R. Mahadeva Prasanna, Ramani Duraiswami, Dmitry N. Zotkin. Processing of reverberant speech for time-delay estimation
1119 -- 1129Javier Ramírez, José C. Segura, Carmen Benítez, Ángel de la Torre, Antonio Rubio. An effective subband OSF-based VAD with noise reduction for robust speech recognition
1130 -- 1143Geert Rombouts, Marc Moonen. Fast QRD-lattice-based unconstrained optimal filtering for acoustic noise reduction
1144 -- 1160Scott Axelrod, Vaibhava Goel, Ramesh A. Gopinath, Peder A. Olsen, Karthik Visweswariah. Subspace constrained Gaussian mixture models for speech recognition
1161 -- 1172Xiaodong Cui, Abeer Alwan. Noise robust speech recognition using feature compensation based on polynomial regression of utterance SNR
1173 -- 1185Thomas 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
1186 -- 1205Ascensión Gallardo-Antolín, Carmen Peláez-Moreno, Fernando Díaz-de-María. Recognizing GSM digital speech
1206 -- 1209Jae Sik Lee, Jong-Hoon Jeong, Tae-Gyu Chang. An efficient method of Huffman decoding for MPEG-2 AAC and its performance analysis
1210 -- 1216I. Kauppinen, K. Roth. Improved noise reduction in audio signals using spectral resolution enhancement with time-domain signal extrapolation
1217 -- 1230Orlando José Tobias, Rui Seara. Leaky-FXLMS algorithm: stochastic analysis for Gaussian data and secondary path modeling error
1231 -- 1237Per Åhgren. Acoustic echo cancellation and doubletalk detection using estimated loudspeaker impulse responses

Volume 13, Issue 5-2

733 -- 740S. S. Yedlapalli. Transforming Real Linear Prediction Coefficients to Line Spectral Representations With a Real FFT
741 -- 750Minkyu Lee, Jan P. H. van Santen, Bernd Möbius, Joseph Olive. Formant Tracking Using Context-Dependent Phonemic Information
751 -- 761Vikas C. Raykar, B. Yegnanarayana, S. R. Mahadeva Prasanna, Ramani Duraiswami. Speaker Localization Using Excitation Source Information in Speech
762 -- 775B.-F. Wu, K.-C. Wang. Robust Endpoint Detection Algorithm Based on the Adaptive Band-Partitioning Spectral Entropy in Adverse Environments
776 -- 786Om Deshmukh, Carol Y. Espy-Wilson, Ariel Salomon, Jawahar Singh. Use of Temporal Information: Detection of Periodicity, Aperiodicity, and Pitch in Speech
787 -- 798J. Lindblom. A Sinusoidal Voice Over Packet Coder Tailored for the Frame-Erasure Channel
799 -- 810Mikko Tammi, Milan Jelinek, Vesa T. Ruoppila. Signal Modification Method for Variable Bit Rate Wide-band Speech Coding
811 -- 820Turaj Zakizadeh Shabestary, Per Hedelin. LSP Quantization by a Union of Locally Trained Codebooks
821 -- 831Doh-Suk Kim. ANIQUE: An Auditory Model for Single-Ended Speech Quality Estimation
832 -- 844Kamran Rahbar, James P. Reilly. A Frequency Domain Method for Blind Source Separation of Convolutive Audio Mixtures
845 -- 856Rainer Martin. Speech Enhancement Based on Minimum Mean-Square Error Estimation and Supergaussian Priors
857 -- 869Philipos C. Loizou. Speech Enhancement Based on Perceptually Motivated Bayesian Estimators of the Magnitude Spectrum
870 -- 881Israel Cohen. Relaxed Statistical Model for Speech Enhancement and a Priori SNR Estimation
882 -- 895Yiteng Huang, Jacob Benesty, Jingdong Chen. A Blind Channel Identification-Based Two-Stage Approach to Separation and Dereverberation of Speech Signals in a Reverberant Environment
896 -- 904Saeed Gazor, Wei Zhang. Speech enhancement employing Laplacian-Gaussian mixture
905 -- 916Ting Liu, Saeed Gazor. A Variable Step-Size Pre-Filter-Bank Adaptive Algorithm
917 -- 929Alfonso Ortega, Eduardo Lleida, Enrique Masgrau. Speech Reinforcement System for Car Cabin Communications
930 -- 944Michael Pitz, Hermann Ney. Vocal Tract Normalization Equals Linear Transformation in Cepstral Space
945 -- 955Hui Jiang, Frank K. Soong, C.-H. Lee. A Dynamic In-Search Data Selection Method With Its Applications to Acoustic Modeling and Utterance Verification
956 -- 964James McAuley, Ji Ming, Darryl Stewart, Philip Hanna. Subband Correlation and Robust Speech Recognition
965 -- 974K. Li, M. N. S. Swamy, M. Omair Ahmad. An Improved Voice Activity Detection Using Higher Order Statistics
975 -- 983Y. Gong. A Method of Joint Compensation of Additive and Convolutive Distortions for Speaker-Independent Speech Recognition
984 -- 992Brian Mak, James Tin-Yau Kwok, Simon Ka-Lung Ho. Kernel Eigenvoice Speaker Adaptation
993 -- 1003Brian Kan-Wing Mak, Kin-Wah Chan. Pruning Hidden Markov Models With Optimal Brain Surgeon
1004 -- 1013S. Kwon, S. Narayanan. Unsupervised Speaker Indexing Using Generic Models
1014 -- 1024Gerald Schuller, Jelena Kovacevic, F. Masson, Vivek K. Goyal. Robust Low-Delay Audio Coding Using Multiple Descriptions
1025 -- 1034Stanley T. Birchfield, Amarnag Subramanya. Microphone Array Position Calibration by Basis-Point Classical Multidimensional Scaling
1035 -- 1047Juan Pablo Bello, Laurent Daudet, Samer A. Abdallah, Chris Duxbury, Mike E. Davies, Mark B. Sandler. A Tutorial on Onset Detection in Music Signals
1048 -- 1062Christof Faller, Jingdong Chen. Suppressing Acoustic Echo in a Spectral Envelope Space
1063 -- 1072G. R. Campos, D. M. Howard. On the Computational Efficiency of Different Waveguide Mesh Topologies for Room Acoustic Simulation
1073 -- 1081Federico Avanzini, Stefania Serafin, Davide Rocchesso. Interactive Simulation of Rigid Body Interaction With Friction-Induced Sound Generation
1082 -- 1088Muhammad Tahir Akhtar, Masahide Abe, Masayuki Kawamata. A New Structure for Feedforward Active Noise Control Systems With Improved Online Secondary Path Modeling

Volume 13, Issue 5-1

633 -- 634M. Gilbert, R. Moore, G. Zweig. Introduction to the Special Issue on Data Mining of Speech, Audio, and Dialog
635 -- 643Peng Yu, Kaijiang Chen, Chengyuan Ma, Frank Seide. Vocabulary-Independent Indexing of Spontaneous Speech
644 -- 651Chien-Chang Lin, Shi-Huang Chen, Trieu-Kien Truong, Yukon Chang. Audio Classification and Categorization Based on Wavelets and Support Vector Machine
652 -- 660Shona Douglas, Deepak Agarwal, Tirso Alonso, Robert M. Bell, Mazin Gilbert, Deborah F. Swayne, Chris Volinsky. Mining Customer Care Dialogs for Daily News
661 -- 671D. Yu, A. Acero. Semiautomatic Improvements of System-Initiative Spoken Dialog Applications Using Interactive Clustering
672 -- 680Lee Begeja, Harris Drucker, David C. Gibbon, Patrick Haffner, Zhu Liu, Bernard Renger, Behzad Shahraray. Semantic Data Mining of Short Utterances
681 -- 688L. Zhou, Y. Shi, J. Feng, A. Sears. Data Mining for Detecting Errors in Dictation Speech Recognition
689 -- 699C.-C. Huang, J. F. Wang, D. J. Wu. Automatic Scene Change Detection for Composed Speech and Music Sound Under Low SNR Noisy Environment
700 -- 711J. Grothendieck. Tracking Changes in Language
712 -- 730John 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

Volume 13, Issue 4

457 -- 0Isabel Trancoso. Editorial
458 -- 466Michael T. Johnson, Richard J. Povinelli, Andrew C. Lindgren, Jinjin Ye, Xiaolin Liu, Kevin M. Indrebo. Time-domain isolated phoneme classification using reconstructed phase spaces
467 -- 474Bowen Zhou, John H. L. Hansen. Efficient audio stream segmentation via the combined T:::2::: statistic and Bayesian information criterion
475 -- 486Chang Huai You, Soo Ngee Koh, Susanto Rahardja. beta-order MMSE spectral amplitude estimation for speech enhancement
487 -- 503Ann Spriet, Marc Moonen, Jan Wouters. Robustness analysis of multichannel Wiener filtering and generalized sidelobe cancellation for multimicrophone noise reduction in hearing aid applications
504 -- 511Giuseppe Riccardi, Dilek Hakkani-Tür. Active learning: theory and applications to automatic speech recognition
512 -- 519Mukund Padmanabhan, Satya Dharanipragada. Maximizing information content in feature extraction
520 -- 533Frank Seide. The use of virtual hypothesis copies in decoding of large-vocabulary continuous speech
534 -- 545Ruhi Sarikaya, Yuqing Gao, Michael Picheny, Hakan Erdogan. Semantic confidence measurement for spoken dialog systems
546 -- 553Mohamed Afify, Feng Liu, Hui Jiang, Olivier Siohan. A new verification-based fast-match for large vocabulary continuous speech recognition
554 -- 564Bowen Zhou, John H. L. Hansen. Rapid discriminative acoustic model based on eigenspace mapping for fast speaker adaptation
565 -- 574Kuo-Hwei Yuo, Tai-Hwei Hwang, Hsiao-Chuan Wang. Combination of autocorrelation-based features and projection measure technique for speaker identification
575 -- 582B. 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
583 -- 592Masafumi Nishida, Tatsuya Kawahara. Speaker model selection based on the Bayesian information criterion applied to unsupervised speaker indexing
593 -- 606Harvey F. Silverman, Ying Yu, Joshua M. Sachar, William R. Patterson III. Performance of real-time source-location estimators for a large-aperture microphone array
607 -- 617Ying Song, Yu Gong, S. M. Kuo. A robust hybrid feedback active noise cancellation headset
618 -- 628Ming Zhang, Hui Lan, Wee Ser. On comparison of online secondary path modeling methods with auxiliary noise

Volume 13, Issue 3

309 -- 320Tetsuya 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
321 -- 329Alexandros Potamianos, Shrikanth S. Narayanan, Giuseppe Riccardi. Adaptive categorical understanding for spoken dialogue systems
330 -- 344Chung-Hsien Wu, Gwo-Lang Yan. Speech act modeling and verification of spontaneous speech with disfluency in a spoken dialogue system
345 -- 354Patrick Kenny, Gilles Boulianne, Pierre Dumouchel. Eigenvoice modeling with sparse training data
355 -- 366Ángel 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
367 -- 376S. Tsakalidis, Vlasios Doumpiotis, William J. Byrne. Discriminative linear transforms for feature normalization and speaker adaptation in HMM estimation
377 -- 387Jen-Tzung Chien, Sadaoki Furui. Predictive hidden Markov model selection for speech recognition
388 -- 398M. Afify. Accurate compensation in the log-spectral domain for noisy speech recognition
399 -- 411Yu Tsao, Shang-Ming Lee, Lin-Shan Lee. Segmental eigenvoice with delicate eigenspace for improved speaker adaptation
412 -- 421Li Deng, James Droppo, Alex Acero. Dynamic compensation of HMM variances using the feature enhancement uncertainty computed from a parametric model of speech distortion
422 -- 431C. D. Creusere. Understanding perceptual distortion in MPEG scalable audio coding
432 -- 440Mohamed F. Mansour, Ahmed H. Tewfik. Data embedding in audio using time-scale modification
441 -- 450Changsheng Xu, Namunu Chinthaka Maddage, Xi Shao. Automatic music classification and summarization

Volume 13, Issue 2

149 -- 162Ted Painter, Andreas Spanias. Perceptual segmentation and component selection for sinusoidal representations of audio
163 -- 173Fredrik Norden, Per Hedelin. Companded quantization of speech MDCT coefficients
174 -- 181Robert E. Schapire, Marie Rochery, Mazin G. Rahim, Narendra Gupta. Boosting with prior knowledge for call classification
182 -- 193Jen-Tzung Chien. Decision tree State tying using cluster validity criteria
194 -- 202Dong Kook Kim, Nam Soo Kim. Rapid online adaptation based on transformation space model evolution
203 -- 210Vincent Wan, Steve Renals. Speaker verification using sequence discriminant support vector machines
211 -- 219William J. J. Roberts, Yariv Ephraim, Howard W. Sabrin. Speaker classification using composite hypothesis testing and list decoding
220 -- 232Renat Vafin, W. Bastiaan Kleijn. Entropy-constrained polar quantization and its application to audio coding
233 -- 242Dietrich Fränken, Klaus Meerkotter, Joachim Wassmuth. Observer-based feedback linearization of dynamic loudspeakers with Ac amplifiers
243 -- 253Lorenzo Turicchia, Rahul Sarpeshkar. A bio-inspired companding strategy for spectral enhancement
254 -- 262H. K. Jang, Ju Sung Park. Multiresolution sinusoidal model with dynamic segmentation for timescale modification of polyphonic audio signals
263 -- 274Athanasios Mouchtaris, S. S. Narayanan, Chris Kyriakakis. Multichannel audio synthesis by subband-based spectral conversion and parameter adaptation
275 -- 285William A. Sethares, Robin D. Morris, James C. Sethares. Beat tracking of musical performances using low-level audio features
286 -- 292Mrityunjoy Chakraborty, Hideaki Sakai. Convergence analysis of a complex LMS algorithm with tonal reference signals
293 -- 303Chul-Min Lee, Shrikanth S. Narayanan. Toward detecting emotions in spoken dialogs

Volume 13, Issue 1

1 -- 13Muhammad Z. Ikram, Dennis R. Morgan. Permutation inconsistency in blind speech separation: investigation and solutions
14 -- 22Pere 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
23 -- 31Frank Wessel, Hermann Ney. Unsupervised training of acoustic models for large vocabulary continuous speech recognition
32 -- 41Michael M. Goodwin, A. J. Hipple, B. Link. Predicting and preventing unmasking incurred in coded audio post-processing
42 -- 52Joshua M. Sachar, Harvey F. Silverman, William R. Patterson III. Microphone position and gain calibration for a large-aperture microphone array
53 -- 69Simon Doclo, Marc Moonen. Multimicrophone noise reduction using recursive GSVD-based optimal filtering with ANC postprocessing stage
70 -- 83Vikas C. Raykar, Igor Kozintsev, Rainer Lienhart. Position calibration of microphones and loudspeakers in distributed computing platforms
84 -- 91Stuart N. Wrigley, Guy J. Brown, Vincent Wan, Steve Renals. Speech and crosstalk detection in multichannel audio
92 -- 104Scott C. Douglas, Hiroshi Sawada, Shoji Makino. Natural gradient multichannel blind deconvolution and speech separation using causal FIR filters
105 -- 119Ville Pulkki, Toni Hirvonen. Localization of virtual sources in multichannel audio reproduction
120 -- 134Herbert Buchner, Robert Aichner, Walter Kellermann. A generalization of blind source separation algorithms for convolutive mixtures based on second-order statistics
135 -- 143Boaz Rafaely. Analysis and design of spherical microphone arrays