IEEE Transactions on Audio, Speech & Language Processing, Volume 17

Volume 17, Number 1, January 2009

• Serdar Yildirim, Shrikanth Narayanan:
  Automatic Detection of Disfluency Boundaries in Spontaneous Speech of Children Using Audio-Visual Information. 2-12
  
• Abhinav Sethy, Panayiotis G. Georgiou, Bhuvana Ramabhadran, Shrikanth S. Narayanan:
  An Iterative Relative Entropy Minimization-Based Data Selection Approach for n-Gram Model Adaptation. 13-23
  
• Jiucang Hao, Hagai Attias, Srikantan S. Nagarajan, Te-Won Lee, Terrence J. Sejnowski:
  Speech Enhancement, Gain, and Noise Spectrum Adaptation Using Approximate Bayesian Estimation. 24-37
  
• Simon Doclo, Marc Moonen, Tim Van den Bogaert, Jan Wouters:
  Reduced-Bandwidth and Distributed MWF-Based Noise Reduction Algorithms for Binaural Hearing Aids. 38-51
  
• Y. Nagata, S. Iwasaki, T. Hariyama, T. Fujioka, T. Obara, T. Wakatake, M. Abe:
  Binaural Localization Based on Weighted Wiener Gain Improved by Incremental Source Attenuation. 52-65
  
• J. Yamagishi, T. Kobayashi, Y. Nakano, K. Ogata, J. Isogai:
  Analysis of Speaker Adaptation Algorithms for HMM-Based Speech Synthesis and a Constrained SMAPLR Adaptation Algorithm. 66-83
  
• Shih-Hsiang Lin, Berlin Chen, Yao-Ming Yeh:
  Exploring the Use of Speech Features and Their Corresponding Distribution Characteristics for Robust Speech Recognition. 84-94
  
• Yi-Ting Chen, Berlin Chen, Hsin-Min Wang:
  A Probabilistic Generative Framework for Extractive Broadcast News Speech Summarization. 95-106
  
• Y. J. Wu, T. D. Abhayapala:
  Theory and Design of Soundfield Reproduction Using Continuous Loudspeaker Concept. 107-116
  
• Radoslaw Mazur, Alfred Mertins:
  An Approach for Solving the Permutation Problem of Convolutive Blind Source Separation Based on Statistical Signal Models. 117-126
  
• Chung-Hsien Wu, Chung-Han Lee, Chung-Hau Liang:
  Idiolect Extraction and Generation for Personalized Speaking Style Modeling. 127-137
  
• S. Ananthakrishnan, Shrikanth S. Narayanan:
  Unsupervised Adaptation of Categorical Prosody Models for Prosody Labeling and Speech Recognition. 138-149
  
• R. M. M. Derkx, K. Janse:
  Theoretical Analysis of a First-Order Azimuth-Steerable Superdirective Microphone Array. 150-162
  
• Ebru Arisoy, Murat Saraclar:
  Lattice Extension and Vocabulary Adaptation for Turkish LVCSR. 163-173
  
• Cyril Joder, Slim Essid, Gaël Richard:
  Temporal Integration for Audio Classification With Application to Musical Instrument Classification. 174-186
  
• Man-Hung Siu, Xi Yang, Herbert Gish:
  Discriminatively Trained GMMs for Language Classification Using Boosting Methods. 187-197
  

Volume 17, Number 2, February 2009

• Chang-Wen Hsu, Lin-Shan Lee:
  Higher Order Cepstral Moment Normalization for Improved Robust Speech Recognition. 205-220
  
• Rade Kutil:
  Optimized Sinusoid Synthesis via Inverse Truncated Fourier Transform. 221-230
  
• T. Yoshioka, T. Nakatani, M. Miyoshi:
  Integrated Speech Enhancement Method Using Noise Suppression and Dereverberation. 231-246
  
• G. Wersenyi:
  Effect of Emulated Head-Tracking for Reducing Localization Errors in Virtual Audio Simulation. 247-252
  
• P. Krishnamoorthy, S. Prasanna:
  Reverberant Speech Enhancement by Temporal and Spectral Processing. 253-266
  
• Jen-Tzung Chien, Meng-Sung Wu:
  Minimum Rank Error Language Modeling. 267-276
  
• P. C. Pandey, M. S. Shah:
  Estimation of Place of Articulation During Stop Closures of Vowel-Consonant-Vowel Utterances. 277-286
  
• George Almpanidis, Margarita Kotti, Constantine Kotropoulos:
  Robust Detection of Phone Boundaries Using Model Selection Criteria With Few Observations. 287-298
  
• Leandro E. Di Persia, Diego H. Milone, Masuzo Yanagida:
  Indeterminacy Free Frequency-Domain Blind Separation of Reverberant Audio Sources. 299-311
  
• Matthias Wölfel:
  Enhanced Speech Features by Single-Channel Joint Compensation of Noise and Reverberation. 312-323
  
• Marc Delcroix, Tomohiro Nakatani, S. Watanabe:
  Static and Dynamic Variance Compensation for Recognition of Reverberant Speech With Dereverberation Preprocessing. 324-334
  
• François Pachet, Pierre Roy:
  Improving Multilabel Analysis of Music Titles: A Large-Scale Validation of the Correction Approach. 335-343
  
• R. Saeidi, H. R. S. Mohammadi, T. Ganchev, R. D. Rodman:
  Particle Swarm Optimization for Sorted Adapted Gaussian Mixture Models. 344-353
  
• Yasser Hifny, Steve Renals:
  Speech Recognition Using Augmented Conditional Random Fields. 354-365
  
• J. Hansen, V. Varadarajan:
  Analysis and Compensation of Lombard Speech Across Noise Type and Levels With Application to In-Set/Out-of-Set Speaker Recognition. 366-378
  
• S. Subasingha, M. N. Murthi, Søren Vang Andersen:
  Gaussian Mixture Kalman Predictive Coding of Line Spectral Frequencies. 379-391
  
• Jacek Dmochowski, Jacob Benesty, Sofiène Affes:
  An Information-Theoretic Viewof ArrayProcessing. 392-401
  

Volume 17, Number 3, March 2009

• A. Katsamanis, George Papandreou, Petros Maragos:
  Face Active Appearance Modeling and Speech Acoustic Information to Recover Articulation. 411-422
  
• George Papandreou, A. Katsamanis, V. Pitsikalis, Petros Maragos:
  Adaptive Multimodal Fusion by Uncertainty Compensation With Application to Audiovisual Speech Recognition. 423-435
  
• E. Sanchez-Soto, A. Potamianos, K. Daoudi:
  Unsupervised Stream-Weights Computation in Classification and Recognition Tasks. 436-445
  
• Jon Barker, Xu Shao:
  Energetic and Informational Masking Effects in an Audiovisual Speech Recognition System. 446-458
  
• Javier Melenchon, Elisa Martinez, Fernando De La Torre, José A. Montero:
  Emphatic Visual Speech Synthesis. 459-468
  
• Jianhua Tao, Le Xin, Panrong Yin:
  Realistic Visual Speech Synthesis Based on Hybrid Concatenation Method. 469-477
  
• Peng Liu, Frank K. Soong:
  Graph-Based Partial Hypothesis Fusion for Pen-Aided Speech Input. 478-485
  
• Pui-Yu Hui, Helen M. Meng:
  Cross-Modality Semantic Integration With Hypothesis Rescoring for Robust Interpretation of Multimodal User Interactions. 486-500
  
• Dinesh Babu Jayagopi, Hayley Hung, Chuohao Yeo, Daniel Gatica-Perez:
  Modeling Dominance in Group Conversations Using Nonverbal Activity Cues. 501-513
  

Volume 17, Number 4, May 2009

• A. Homayoun Kamkar-Parsi, Martin Bouchard:
  Improved Noise Power Spectrum Density Estimation for Binaural Hearing Aids Operating in a Diffuse Noise Field Environment. 521-533
  
• Keisuke Kinoshita, Marc Delcroix, Tomohiro Nakatani, Masato Miyoshi:
  Suppression of Late Reverberation Effect on Speech Signal Using Long-Term Multiple-step Linear Prediction. 534-545
  
• Ronen Talmon, Israel Cohen, Sharon Gannot:
  Relative Transfer Function Identification Using Convolutive Transfer Function Approximation. 546-555
  
• S. R. Mahadeva Prasanna, B. V. Sandeep Reddy, P. Krishnamoorthy:
  Vowel Onset Point Detection Using Source, Spectral Peaks, and Modulation Spectrum Energies. 556-565
  
• Liang Wang, Woon-Seng Gan:
  Convergence Analysis of Narrowband Active Noise Equalizer System Under Imperfect Secondary Path Estimation. 566-571
  
• Leonardo Rey Vega, Hernan Rey, Jacob Benesty, Sara Tressens:
  A Family of Robust Algorithms Exploiting Sparsity in Adaptive Filters. 572-581
  
• Carlos Busso, Sungbok Lee, Shrikanth Narayanan:
  Analysis of Emotionally Salient Aspects of Fundamental Frequency for Emotion Detection. 582-596
  
• M. Kuster:
  Multichannel Room Impulse Response Generation With Coherence Control. 597-606
  
• S. Sen, Arye Nehorai:
  Performance Analysis of 3-D Direction Estimation Based on Head-Related Transfer Function. 607-613
  
• B. Yegnanarayana, K. S. R. Murty:
  Event-Based Instantaneous Fundamental Frequency Estimation From Speech Signals. 614-624
  
• Zhaozhang Jin, DeLiang Wang:
  A Supervised Learning Approach to Monaural Segregation of Reverberant Speech. 625-638
  
• Hiroko Kato Solvang, Y. Nagahara, Shoko Araki, Hiroshi Sawada, Shoji Makino:
  Frequency-Domain Pearson Distribution Approach for Independent Component Analysis (FD-Pearson-ICA) in Blind Source Separation. 639-649
  
• Yu Takahashi, Tomoya Takatani, Keiichi Osako, Hiroshi Saruwatari, Kiyohiro Shikano:
  Blind Spatial Subtraction Array for Speech Enhancement in Noisy Environment. 650-664
  
• Huawei Chen, Wee Ser:
  Design of Robust Broadband Beamformers With Passband Shaping Characteristics Using Tikhonov Regularization. 665-681
  
• Jwu-Sheng Hu, Wei-Han Liu:
  Location Classification of Nonstationary Sound Sources Using Binaural Room Distribution Patterns. 682-692
  
• Jesper Højvang Jensen, Mads Græsbøll Christensen, D. P. W. Ellis, Søren Holdt Jensen:
  Quantitative Analysis of a Common Audio Similarity Measure. 693-703
  
• Hong Kook Kim, R. C. Rose:
  Cepstrum-Domain Model Combination Based on Decomposition of Speech and Noise Using MMSE-LSA for ASR in Noisy Environments. 704-713
  
• Kai Yu, M. J. F. Gales, Philip C. Woodland:
  Unsupervised Adaptation With Discriminative Mapping Transforms. 714-723
  
• Teemu Hirsimäki, Janne Pylkkönen, Mikko Kurimo:
  Importance of High-Order N-Gram Models in Morph-Based Speech Recognition. 724-732
  
• Jost Schatzmann, S. Young:
  The Hidden Agenda User Simulation Model. 733-747
  
• C. Longworth, M. J. F. Gales:
  Combining Derivative and Parametric Kernels for Speaker Verification. 748-757
  
• Nicolás Morales, Doroteo T. Toledano, John H. L. Hansen, Javier Garrido:
  Feature Compensation Techniques for ASR on Band-Limited Speech. 758-774
  
• Y. Agiomyrgiannakis, Yannis Stylianou:
  Wrapped Gaussian Mixture Models for Modeling and High-Rate Quantization of Phase Data of Speech. 775-786
  
• Jingdong Chen, Jacob Benesty, Yiteng Huang:
  Study of the Noise-Reduction Problem in the Karhunen-LoÈve Expansion Domain. 787-802
  
• Klaus Macherey, Oliver Bender, Hermann Ney:
  Applications of Statistical Machine Translation Approaches to Spoken Language Understanding. 803-818
  
• Wen Zhang, Rodney A. Kennedy, Thushara D. Abhayapala:
  Efficient Continuous HRTF Model Using Data Independent Basis Functions: Experimentally Guided Approach. 819-829
  
• Malay Gupta, Scott C. Douglas:
  A Spatio-Temporal Speech Enhancement Technique Based on Generalized Eigenvalue Decomposition. 830-839
  
• Pavel Ircing, Josef V. Psutka, Josef Psutka:
  Using Morphological Information for Robust Language Modeling in Czech ASR System. 840-847
  
• V. R. Apsingekar, P. L. De Leon:
  Speaker Model Clustering for Efficient Speaker Identification in Large Population Applications. 848-853
  

Volume 17, Number 5, July 2009

• Ruhi Sarikaya, Katrin Kirchhoff, Tanja Schultz, Dilek Z. Hakkani-Tür:
  Introduction to the Special Issue on Processing Morphologically Rich Languages. 861-862
  
• Thomas Pellegrini, Lori Lamel:
  Automatic Word Decompounding for ASR in a Morphologically Rich Language: Application to Amharic. 863-873
  
• Ebru Arisoy, Dogan Can, Siddika Parlak, Hasim Sak, Murat Saraclar:
  Turkish Broadcast News Transcription and Retrieval. 874-883
  
• Hagen Soltau, George Saon, Brian Kingsbury, Hong-Kwang Jeff Kuo, Lidia Mangu, Daniel Povey, Ahmad Emami:
  Advances in Arabic Speech Transcription at IBM Under the DARPA GALE Program. 884-894
  
• U. Guz, Benoît Favre, Dilek Z. Hakkani-Tür, Gökhan Tür:
  Generative and Discriminative Methods Using Morphological Information for Sentence Segmentation of Turkish. 895-903
  
• Xabier Artola, Arantza Díaz de Ilarraza, Aitor Soroa, A. Sologaistoa:
  Dealing With Complex Linguistic Annotations Within a Language Processing Framework. 904-915
  
• Mohamed Attia, Mohsen Rashwan, Mohamed Al-Badrashiny:
  Fassieh-, a Semi-Automatic Visual Interactive Tool for Morphological, PoS-Tags, Phonetic, and Semantic Annotation of Arabic Text Corpora. 916-925
  
• Yassine Benajiba, Mona T. Diab, Paolo Rosso:
  Arabic Named Entity Recognition: A Feature-Driven Study. 926-934
  
• Imed Zitouni, Xiaoqiang Luo, Radu Florian:
  A Cascaded Approach to Mention Detection and Chaining in Arabic. 935-944
  
• Do-Gil Lee, Hae-Chang Rim:
  Probabilistic Modeling of Korean Morphology. 945-955
  
• Kseniya B. Shalonova, Bruno Golenia, Peter Flach:
  Towards Learning Morphology for Under-Resourced Fusional and Agglutinating Languages. 956-965
  
• Paris Smaragdis:
  Dynamic Range Extension Using Interleaved Gains. 966-973
  
• Stefan Windmann, Reinhold Haeb-Umbach:
  Approaches to Iterative Speech Feature Enhancement and Recognition. 974-984
  
• Gerald Friedland, Oriol Vinyals, Yan Huang, Christian Müller:
  Prosodic and other Long-Term Features for Speaker Diarization. 985-993
  
• Ken'ichi Kumatani, John W. McDonough, Barbara Rauch, Dietrich Klakow, Philip N. Garner, Weifeng Li:
  Beamforming With a Maximum Negentropy Criterion. 994-1008
  
• Ozlem Kalinli, Shrikanth S. Narayanan:
  Prominence Detection Using Auditory Attention Cues and Task-Dependent High Level Information. 1009-1024
  
• Yu Tsao, Chin-Hui Lee:
  An Ensemble Speaker and Speaking Environment Modeling Approach to Robust Speech Recognition. 1025-1037
  
• Ali A. Milani, Issa M. S. Panahi, Philipos C. Loizou:
  A New Delayless Subband Adaptive Filtering Algorithm for Active Noise Control Systems. 1038-1045
  
• Arie Livshin, Xavier Rodet:
  Purging Musical Instrument Sample Databases Using Automatic Musical Instrument Recognition Methods. 1046-1051
  

Volume 17, Number 6, August 2009

• Nikolay D. Gaubitch, Patrick A. Naylor:
  Equalization of Multichannel Acoustic Systems in Oversampled Subbands. 1061-1070
  
• Shmulik Markovich, Sharon Gannot, Israel Cohen:
  Multichannel Eigenspace Beamforming in a Reverberant Noisy Environment With Multiple Interfering Speech Signals. 1071-1086
  
• Jerónimo Arenas-García, Aníbal R. Figueiras-Vidal:
  Adaptive Combination of Proportionate Filters for Sparse Echo Cancellation. 1087-1098
  
• Tiemin Mei, Fuliang Yin, Jun Wang:
  Blind Source Separation Based on Cumulants With Time and Frequency Non-Properties. 1099-1108
  
• Jacob Benesty, Jingdong Chen, Yiteng Arden Huang:
  Noise Reduction Algorithms in a Generalized Transform Domain. 1109-1123
  
• Tianshu Qu, Zheng Xiao, Mei Gong, Ying Huang, Xiaodong Li, Xihong Wu:
  Distance-Dependent Head-Related Transfer Functions Measured With High Spatial Resolution Using a Spark Gap. 1124-1132
  
• Nima Khademi Kalantari, Mohannad Ali Akhaee, Seyed Mohammad Ahadi, Hamidreza Amindavar:
  Robust Multiplicative Patchwork Method for Audio Watermarking. 1133-1141
  
• Selina Chu, Shrikanth S. Narayanan, C. C. Jay Kuo:
  Environmental Sound Recognition With Time-Frequency Audio Features. 1142-1158
  
• Jouni Paulus, Anssi Klapuri:
  Music Structure Analysis Using a Probabilistic Fitness Measure and a Greedy Search Algorithm. 1159-1170
  
• Zhen-Hua Ling, Korin Richmond, Junichi Yamagishi, Ren-Hua Wang:
  Integrating Articulatory Features Into HMM-Based Parametric Speech Synthesis. 1171-1185
  
• Patricia Henriquez, Jesús B. Alonso, Miguel A. Ferrer, Carlos M. Travieso, Juan Ignacio Godino-Llorente, Fernando Díaz-de-María:
  Characterization of Healthy and Pathological Voice Through Measures Based on Nonlinear Dynamics. 1186-1195
  
• B. Yegnanarayana, R. Kumara Swamy, K. S. R. Murty:
  Determining Mixing Parameters From Multispeaker Data Using Speech-Specific Information. 1196-1207
  
• Junichi Yamagishi, Takashi Nose, Heiga Zen, Zhen-Hua Ling, Tomoki Toda, Keiichi Tokuda, Simon King, Steve Renals:
  Robust Speaker-Adaptive HMM-Based Text-to-Speech Synthesis. 1208-1230
  
• Yao Qian, Hui Liang, Frank K. Soong:
  A Cross-Language State Sharing and Mapping Approach to Bilingual (Mandarin-English) TTS. 1231-1239
  

Volume 17, Number 7, September 2009

• Mei-Yuh Hwang, Gang Peng, Mari Ostendorf, Wen Wang, Arlo Faria, Aaron Heidel:
  Building A Highly Accurate Mandarin Speech Recognizer With Language-Independent Technologies and Language-Dependent Modules. 1253-1262
  
• Che-Kuang Lin, Lin-Shan Lee:
  Improved Features and Models for Detecting Edit Disfluencies in Transcribing Spontaneous Mandarin Speech. 1263-1278
  
• Jen-Tzung Chien, Chuan-Wei Ting:
  Acoustic Factor Analysis for Streamed Hidden Markov Modeling. 1279-1291
  
• Wooil Kim, John H. L. Hansen:
  Time-Frequency Correlation-Based Missing-Feature Reconstruction for Robust Speech Recognition in Band-Restricted Conditions. 1292-1304
  
• Hung-Yu Su, Chung-Hsien Wu:
  Improving Structural Statistical Machine Translation for Sign Language With Small Corpus Using Thematic Role Templates as Translation Memory. 1305-1315
  
• Engin Erzin:
  Improving Throat Microphone Speech Recognition by Joint Analysis of Throat and Acoustic Microphone Recordings. 1316-1324
  
• M. Afify, X. Cui, Y. Gao:
  Stereo-Based Stochastic Mapping for Robust Speech Recognition. 1325-1334
  
• Rong Tong, Bin Ma, Haizhou Li, Chng Eng Siong:
  A Target-Oriented Phonotactic Front-End for Spoken Language Recognition. 1335-1347
  
• Dong Yu, Li Deng, Yifan Gong, Alex Acero:
  A Novel Framework and Training Algorithm for Variable-Parameter Hidden Markov Models. 1348-1360
  
• Yipeng Li, John Woodruff, DeLiang Wang:
  Monaural Musical Sound Separation Based on Pitch and Common Amplitude Modulation. 1361-1371
  
• Brian Kan-Wing Mak, Tsz-Chung Lai, Ivor W. Tsang, James Tin-Yau Kwok:
  Maximum Penalized Likelihood Kernel Regression for Fast Adaptation. 1372-1381
  
• Deepu Vijayasenan, Fabio Valente, Hervé Bourlard:
  An Information Theoretic Approach to Speaker Diarization of Meeting Data. 1382-1393
  
• Nitish Krishnamurthy, John H. L. Hansen:
  Babble Noise: Modeling, Analysis, and Applications. 1394-1407
  
• Giso Grimm, Volker Hohmann, Birger Kollmeier:
  Increase and Subjective Evaluation of Feedback Stability in Hearing Aids by a Binaural Coherence-Based Noise Reduction Scheme. 1408-1419
  
• Ronen Talmon, Israel Cohen, Sharon Gannot:
  Convolutive Transfer Function Generalized Sidelobe Canceler. 1420-1434
  
• J. Barbedo, A. Lopes, P. J. Wolfe:
  Empirical Methods to Determine the Number of Sources in Single-Channel Musical Signals. 1435-1444
  

Volume 17, Number 8, November 2009

• Aren Jansen, Partha Niyogi:
  Point Process Models for Spotting Keywords in Continuous Speech. 1457-1470
  
• Viet Bac Le, Laurent Besacier:
  Automatic Speech Recognition for Under-Resourced Languages: Application to Vietnamese Language. 1471-1482
  
• Christos Tzagkarakis, Athanasios Mouchtaris, Panagiotis Tsakalides:
  A Multichannel Sinusoidal Model Applied to Spot Microphone Signals for Immersive Audio. 1483-1497
  
• Sampo Vesa:
  Binaural Sound Source Distance Learning in Rooms. 1498-1507
  
• Ioannis Andrianakis, Paul R. White:
  A Speech Enhancement Algorithm Based on a Chi MRF Model of the Speech STFT Amplitudes. 1508-1517
  
• Emre Özkan, I. Yücel Özbek, Mübeccel Demirekler:
  Dynamic Speech Spectrum Representation and Tracking Variable Number of Vocal Tract Resonance Frequencies With Time-Varying Dirichlet Process Mixture Models. 1518-1532
  
• Ilana Heintz, Eric Fosler-Lussier, Chris Brew:
  Discriminative Input Stream Combination for Conditional Random Field Phone Recognition. 1533-1546
  
• Zhi-Sheng Chen, J.-S. R. Jang:
  On the Use of Anti-Word Models for Audio Music Annotation and Retrieval. 1547-1556
  
• M. R. P. Thomas, P. A. Naylor:
  The SIGMA Algorithm: A Glottal Activity Detector for Electroglottographic Signals. 1557-1566
  
• Zhiyong Wu, Helen M. Meng, Hongwu Yang, Lianhong Cai:
  Modeling the Expressivity of Input Text Semantics for Chinese Text-to-Speech Synthesis in a Spoken Dialog System. 1567-1576
  
• Stefan Windmann, Reinhold Haeb-Umbach:
  Parameter Estimation of a State-Space Model of Noise for Robust Speech Recognition. 1577-1590
  
• P. Loganathan, A. W. H. Khong, Patrick A. Naylor:
  A Class of Sparseness-Controlled Algorithms for Echo Cancellation. 1591-1601
  
• Bo Shao, Mitsunori Ogihara, Dingding Wang, Tao Li:
  Music Recommendation Based on Acoustic Features and User Access Patterns. 1602-1611
  
• Chung-Hsien Wu, Chia-Hsin Hsieh:
  Story Segmentation and Topic Classification of Broadcast News via a Topic-Based Segmental Model and a Genetic Algorithm. 1612-1623
  
• Konrad Hofbauer, Gernot Kubin, W. Bastiaan Kleijn:
  Speech Watermarking for Analog Flat-Fading Bandpass Channels. 1624-1637