Keynotes

• 
  Steve Young: Still talking to machines (cognitively speaking). 1-10
• 
  Tohru Ifukube: Sound-based assistive technology supporting "seeing", "hearing" and "speaking" for the disabled and the elderly. 11-19
• 
  Chiu-yu Tseng: Beyond sentence prosody. 20-29

Special Session: Models of Speech - In Search of Better Representations

• 
  Hosung Nam, Vikramjit Mitra, Mark Tiede, Elliot Saltzman, Louis Goldstein, Carol Y. Espy-Wilson, Mark Hasegawa-Johnson: A procedure for estimating gestural scores from natural speech. 30-33
• 
  Yen-Liang Shue, Gang Chen, Abeer Alwan: On the interdependencies between voice quality, glottal gaps, and voice-source related acoustic measures. 34-37
• 
  Hideki Kawahara, Masanori Morise, Toru Takahashi, Hideki Banno, Ryuichi Nisimura, Toshio Irino: Simplification and extension of non-periodic excitation source representations for high-quality speech manipulation systems. 38-41
• 
  Sadao Hiroya, Takemi Mochida: Phase equalization-based autoregressive model of speech signals. 42-45
• 
  Yi Xu, Santitham Prom-on: Articulatory-functional modeling of speech prosody: a review. 46-49
• 
  Humberto M. Torres, Hansjörg Mixdorff, Jorge A. Gurlekian, Hartmut R. Pfitzinger: Two new estimation methods for a superpositional intonation model. 50-53

ASR: Acoustic Models I-III

• 
  Simon Wiesler, Georg Heigold, Markus Nußbaum-Thom, Ralf Schlüter, Hermann Ney: A discriminative splitting criterion for phonetic decision trees. 54-57
• 
  Mark J. F. Gales, Kai Yu: Canonical state models for automatic speech recognition. 58-61
• 
  Pierre L. Dognin, John R. Hershey, Vaibhava Goel, Peder A. Olsen: Restructuring exponential family mixture models. 62-65
• 
  Françoise Beaufays, Vincent Vanhoucke, Brian Strope: Unsupervised discovery and training of maximally dissimilar cluster models. 66-69
• 
  Khe Chai Sim: Probabilistic state clustering using conditional random field for context-dependent acoustic modelling. 70-73
• 
  Xie Sun, Yunxin Zhao: Integrate template matching and statistical modeling for speech recognition. 74-77
• 
  George Saon, Hagen Soltau: Boosting systems for LVCSR. 1341-1344
• 
  Vaibhava Goel, Tara N. Sainath, Bhuvana Ramabhadran, Peder A. Olsen, David Nahamoo, Dimitri Kanevsky: Incorporating sparse representation phone identification features in automatic speech recognition using exponential families. 1345-1348
• 
  Xin Chen, Yunxin Zhao: Integrating MLP features and discriminative training in data sampling based ensemble acoustic modeling. 1349-1352
• 
  Jui-Ting Huang, Mark Hasegawa-Johnson: Semi-supervised training of Gaussian mixture models by conditional entropy minimization. 1353-1356
• 
  Guangchuan Shi, Yu Shi, Qiang Huo: A study of irrelevant variability normalization based training and unsupervised online adaptation for LVCSR. 1357-1360
• 
  Roger Hsiao, Florian Metze, Tanja Schultz: Improvements to generalized discriminative feature transformation for speech recognition. 1361-1364
• 
  Karel Veselý, Lukas Burget, Frantisek Grézl: Parallel training of neural networks for speech recognition. 2934-2937
• 
  Rita Singh, Benjamin Lambert, Bhiksha Raj: The use of sense in unsupervised training of acoustic models for ASR systems. 2938-2941
• 
  Jun Du, Yu Hu, Hui Jiang: Boosted mixture learning of Gaussian mixture HMMs for speech recognition. 2942-2945
• 
  Volker Leutnant, Reinhold Haeb-Umbach: On the exploitation of hidden Markov models and linear dynamic models in a hybrid decoder architecture for continuous speech recognition. 2946-2949
• 
  Alberto Abad, Thomas Pellegrini, Isabel Trancoso, João Paulo Neto: Context dependent modelling approaches for hybrid speech recognizers. 2950-2953
• 
  Yotaro Kubo, Shinji Watanabe, Atsushi Nakamura, Tetsunori Kobayashi: A regularized discriminative training method of acoustic models derived by minimum relative entropy discrimination. 2954-2957
• 
  Hank Liao, Christopher Alberti, Michiel Bacchiani, Olivier Siohan: Decision tree state clustering with word and syllable features. 2958-2961
• 
  Hiroshi Fujimura, Takashi Masuko, Mitsuyoshi Tachimori: A duration modeling technique with incremental speech rate normalization. 2962-2965
• 
  Martin Wöllmer, Yang Sun, Florian Eyben, Björn Schuller: Long short-term memory networks for noise robust speech recognition. 2966-2969
• 
  Tsuneo Nitta, Takayuki Onoda, Masashi Kimura, Yurie Iribe, Kouichi Katsurada: One-model speech recognition and synthesis based on articulatory movement HMMs. 2970-2973
• 
  Xiaodong Cui, Jian Xue, Pierre L. Dognin, Upendra V. Chaudhari, Bowen Zhou: Acoustic modeling with bootstrap and restructuring for low-resourced languages. 2974-2977
• 
  Tetsuo Kosaka, Keisuke Goto, Takashi Ito, Masaharu Katoh: Lecture speech recognition by combining word graphs of various acoustic models. 2978-2981
• 
  Khe Chai Sim, Shilin Liu: Semi-parametric trajectory modelling using temporally varying feature mapping for speech recognition. 2982-2985
• 
  Dong Yu, Li Deng: Deep-structured hidden conditional random fields for phonetic recognition. 2986-2989
• 
  Jonathan Malkin, Jeff A. Bilmes: Semi-supervised learning for improved expression of uncertainty in discriminative classifiers. 2990-2993
• 
  Peder A. Olsen, Vaibhava Goel, Charles A. Micchelli, John R. Hershey: Modeling posterior probabilities using the linear exponential family. 2994-2997

Spoken Dialogue Systems I, II

• 
  Fabrice Lefèvre, François Mairesse, Steve Young: Cross-lingual spoken language understanding from unaligned data using discriminative classification models and machine translation. 78-81
• 
  Rajesh Balchandran, Leonid Rachevsky, Bhuvana Ramabhadran, Miroslav Novak: Techniques for topic detection based processing in spoken dialog systems. 82-85
• 
  Senthilkumar Chandramohan, Matthieu Geist, Olivier Pietquin: Optimizing spoken dialogue management with fitted value iteration. 86-89
• 
  Filip Jurcícek, Blaise Thomson, Simon Keizer, François Mairesse, Milica Gasic, Kai Yu, Steve Young: Natural belief-critic: a reinforcement algorithm for parameter estimation in statistical spoken dialogue systems. 90-93
• 
  Alexander Schmitt, Michael Scholz, Wolfgang Minker, Jackson Liscombe, David Suendermann: Is it possible to predict task completion in automated troubleshooters?. 94-97
• 
  David Suendermann, Jackson Liscombe, Roberto Pieraccini: Minimally invasive surgery for spoken dialog systems. 98-101

Spoken Dialogue Systems II

• 
  Ramón López-Cózar, David Griol: New technique to enhance the performance of spoken dialogue systems based on dialogue states-dependent language models and grammatical rules. 2998-3001
• 
  Lluís F. Hurtado, Joaquin Planells, Encarna Segarra, Emilio Sanchis, David Griol: A stochastic finite-state transducer approach to spoken dialog management. 3002-3005
• 
  Romain Laroche, Philippe Bretier, Ghislain Putois: Enhanced monitoring tools and online dialogue optimisation merged into a new spoken dialogue system design experience. 3006-3009
• 
  Romain Laroche, Ghislain Putois, Philippe Bretier: Optimising a handcrafted dialogue system design. 3010-3013
• 
  Felix Putze, Tanja Schultz: Utterance selection for speech acts in a cognitive tourguide scenario. 3014-3017
• 
  Gabriel Parent, Maxine Eskenazi: Lexical entrainment of real users in the let's go spoken dialog system. 3018-3021
• 
  Silvia Quarteroni, Meritxell González, Giuseppe Riccardi, Sebastian Varges: Combining user intention and error modeling for statistical dialog simulators. 3022-3025
• 
  Jaakko Hakulinen, Markku Turunen, Raul Santos de la Camara, Nigel Crook: Parallel processing of interruptions and feedback in companions affective dialogue system. 3026-3029
• 
  Antoine Raux, Neville Mehta, Deepak Ramachandran, Rakesh Gupta: Dynamic language modeling using Bayesian networks for spoken dialog systems. 3030-3033
• 
  Sunao Hara, Norihide Kitaoka, Kazuya Takeda: Automatic detection of task-incompleted dialog for spoken dialog system based on dialog act n-gram. 3034-3037
• 
  Wei-Bin Liang, Chung-Hsien Wu, Yu-Cheng Hsiao: Dialogue act detection in error-prone spoken dialogue systems using partial sentence tree and latent dialogue act matrix. 3038-3041
• 
  Tatsuya Kawahara, Kouhei Sumi, Zhi-Qiang Chang, Katsuya Takanashi: Detection of hot spots in poster conversations based on reactive tokens of audience. 3042-3045
• 
  Yoichi Matsuyama, Shinya Fujie, Hikaru Taniyama, Tetsunori Kobayashi: Psychological evaluation of a group communication activation robot in a party game. 3046-3049
• 
  Kyoko Matsuyama, Kazunori Komatani, Ryu Takeda, Toru Takahashi, Tetsuya Ogata, Hiroshi G. Okuno: Analyzing user utterances in barge-in-able spoken dialogue system for improving identification accuracy. 3050-3053
• 
  Mattias Heldner, Jens Edlund, Julia Hirschberg: Pitch similarity in the vicinity of backchannels. 3054-3057
• 
  Khiet P. Truong, Ronald Poppe, Dirk Heylen: A rule-based backchannel prediction model using pitch and pause information. 3058-3061

Speech Perception: Factors Influencing Perception

• 
  Paul Boersma, Katerina Chládková: Detecting categorical perception in continuous discrimination data. 102-105
• 
  Titia Benders, Paola Escudero: The interrelation between the stimulus range and the number of response categories in vowel categorization. 106-109
• 
  Marie Nilsenová, Martijn Goudbeek, Luuk Kempen: The relation between pitch perception preference and emotion identification. 110-113
• 
  Takashi Otake, James M. McQueen, Anne Cutler: Competition in the perception of spoken Japanese words. 114-117
• 
  Makiko Sadakata, Lotte van der Zanden, Kaoru Sekiyama: Influence of musical training on perception of L2 speech. 118-121
• 
  Donald Derrick, Bryan Gick: Full body aero-tactile integration in speech perception. 122-125

Prosody: Models

• 
  Tomás Dubeda, Katalin Mády: Nucleus position within the intonation phrase: a typological study of English, Czech and Hungarian. 126-129
• 
  Yong-cheol Lee, Satoshi Nambu: Focus-sensitive operator or focus inducer: always and only. 130-133
• 
  Jiahong Yuan, Mark Liberman: F₀ declination in English and Mandarin broadcast news speech. 134-137
• 
  Katrin Schweitzer, Michael Walsh, Bernd Möbius, Hinrich Schütze: Frequency of occurrence effects on pitch accent realisation. 138-141
• 
  César González Ferreras, Carlos Vivaracho-Pascual, David Escudero Mancebo, Valentín Cardeñoso-Payo: On the automatic toBI accent type identification from data. 142-145
• 
  Andrew Rosenberg: AutoBI - a tool for automatic toBI annotation. 146-149

Speech Synthesis: Unit Selection and Others

• 
  Volker Strom, Simon King: A classifier-based target cost for unit selection speech synthesis trained on perceptual data. 150-153
• 
  Wei Zhang, Xiaodong Cui: Applying scalable phonetic context similarity in unit selection of concatenative text-to-speech. 154-157
• 
  Mitsuaki Isogai, Hideyuki Mizuno: Speech database reduction method for corpus-based TTS system. 158-161
• 
  Heng Lu, Zhen-Hua Ling, Si Wei, Li-Rong Dai, Ren-Hua Wang: Automatic error detection for unit selection speech synthesis using log likelihood ratio based SVM classifier. 162-165
• 
  Hanna Silén, Elina Helander, Jani Nurminen, Konsta Koppinen, Moncef Gabbouj: Using robust viterbi algorithm and HMM-modeling in unit selection TTS to replace units of poor quality. 166-169
• 
  Yeon-Jun Kim, Marc C. Beutnagel: Automatic detection of abnormal stress patterns in unit selection synthesis. 170-173
• 
  Daniel Tihelka, Jirí Kala, Jindrich Matousek: Enhancements of viterbi search for fast unit selection synthesis. 174-177
• 
  Thomas Ewender, Beat Pfister: Accurate pitch marking for prosodic modification of speech segments. 178-181
• 
  Shifeng Pan, Meng Zhang, Jianhua Tao: A novel hybrid approach for Mandarin speech synthesis. 182-185
• 
  Josafá de Jesus Aguiar Pontes, Sadaoki Furui: Modeling liaison in French by using decision trees. 186-189
• 
  Jian Luan, Jian Li: Improvement on plural unit selection and fusion. 190-193
• 
  Alok Parlikar, Alan W. Black, Stephan Vogel: Improving speech synthesis of machine translation output. 194-197
• 
  Ghislain Putois, Jonathan Chevelu, Cédric Boidin: Paraphrase generation to improve text-to-speech synthesis. 198-201

ASR: Search, Decoding and Confidence Measures I, II

• 
  Chang Woo Han, Shin Jae Kang, Chul Min Lee, Nam Soo Kim: Phone mismatch penalty matrices for two-stage keyword spotting via multi-pass phone recognizer. 202-205
• 
  Petr Motlícek, Fabio Valente, Philip N. Garner: English spoken term detection in multilingual recordings. 206-209
• 
  Icksang Han, Chiyoun Park, Jeongmi Cho, Jeongsu Kim: A hybrid approach to robust word lattice generation via acoustic-based word detection. 210-213
• 
  Volker Steinbiss, Martin Sundermeyer, Hermann Ney: Direct observation of pruning errors (DOPE): a search analysis tool. 214-217
• 
  David Rybach, Michael Riley: Direct construction of compact context-dependency transducers from data. 218-221
• 
  Miroslav Novak: Incremental composition of static decoding graphs with label pushing. 222-225
• 
  Zhanlei Yang, Wenju Liu: A novel path extension framework using steady segment detection for Mandarin speech recognition. 226-229
• 
  Ralf Schlüter, Markus Nußbaum-Thom, Hermann Ney: On the relation of Bayes risk, word error, and word posteriors in ASR. 230-233
• 
  David Nolden, Hermann Ney, Ralf Schlüter: Time conditioned search in automatic speech recognition reconsidered. 234-237
• 
  Satoshi Kobashikawa, Taichi Asami, Yoshikazu Yamaguchi, Hirokazu Masataki, Satoshi Takahashi: Efficient data selection for speech recognition based on prior confidence estimation using speech and context independent models. 238-241
• 
  Atsunori Ogawa, Atsushi Nakamura: A novel confidence measure based on marginalization of jointly estimated error cause probabilities. 242-245
• 
  Julien Fayolle, Fabienne Moreau, Christian Raymond, Guillaume Gravier, Patrick Gros: CRF-based combination of contextual features to improve a posteriori word-level confidence measures. 1942-1945
• 
  Martin Wöllmer, Florian Eyben, Björn Schuller, Gerhard Rigoll: Recognition of spontaneous conversational speech using long short-term memory phoneme predictions. 1946-1949
• 
  Thomas Pellegrini, Isabel Trancoso: Improving ASR error detection with non-decoder based features. 1950-1953
• 
  Ladan Golipour, Douglas D. O'Shaughnessy: Phoneme classification and lattice rescoring based on a k-NN approach. 1954-1957
• 
  Jeff Bilmes, Hui Lin: Online adaptive learning for speech recognition decoding. 1958-1961
• 
  Takaaki Hori, Shinji Watanabe, Atsushi Nakamura: Improvements of search error risk minimization in viterbi beam search for speech recognition. 1962-1965

Special-Purpose Speech Applications

• 
  Robin Hofe, Stephen R. Ell, Michael J. Fagan, James M. Gilbert, Phil D. Green, Roger K. Moore, Sergey I. Rybchenko: Evaluation of a silent speech interface based on magnetic sensing. 246-249
• 
  Rubén San Segundo, Verónica López, Raquel Martín, Syaheerah L. Lutfi, Javier Ferreiros, Ricardo de Córdoba, José Manuel Pardo: Advanced speech communication system for deaf people. 250-253
• 
  Sethserey Sam, Eric Castelli, Laurent Besacier: Unsupervised acoustic model adaptation for multi-origin non native ASR. 254-257
• 
  Dilek Hakkani-Tür, Dimitra Vergyri, Gökhan Tür: Speech-based automated cognitive status assessment. 258-261
• 
  Toru Imai, Shinichi Homma, Akio Kobayashi, Takahiro Oku, Shoei Sato: Speech recognition with a seamlessly updated language model for real-time closed-captioning. 262-265
• 
  Takuya Nishimoto, Takayuki Watanabe: The comparison between the deletion-based methods and the mixing-based methods for audio CAPTCHA systems. 266-269
• 
  Martine Adda-Decker, Lori Lamel, Natalie D. Snoeren: Comparing mono- & multilingual acoustic seed models for a low e-resourced language: a case-study of luxembourgish. 270-273
• 
  R. J. J. H. van Son, Irene Jacobi, Frans J. M. Hilgers: Manipulating treacheoesophageal speech. 274-277
• 
  David Imseng, Hervé Bourlard, Mathew Magimai-Doss: Towards mixed language speech recognition systems. 278-281
• 
  Etienne Barnard, Johan Schalkwyk, Charl Johannes van Heerden, Pedro J. Moreno: Voice search for development. 282-285
• 
  Gina-Anne Levow, Susan Duncan, Edward T. King: Cross-cultural investigation of prosody in verbal feedback in interactional rapport. 286-289
• 
  Mary Tai Knox, Gerald Friedland: Multimodal speaker diarization using oriented optical flow histograms. 290-293
• 
  Catherine Middag, Yvan Saeys, Jean-Pierre Martens: Towards an ASR-free objective analysis of pathological speech. 294-297

Speech Analysis

• 
  Keith W. Godin, John H. L. Hansen: Session variability contrasts in the MARP corpus. 298-301
• 
  Kazuhiro Kondo, Yusuke Takano: Estimation of two-to-one forced selection intelligibility scores by speech recognizers using noise-adapted models. 302-305
• 
  Thomas Schaaf, Florian Metze: Analysis of gender normalization using MLP and VTLN features. 306-309
• 
  Guillaume Aimetti, Roger K. Moore, Louis ten Bosch: Discovering an optimal set of minimally contrasting acoustic speech units: a point of focus for whole-word pattern matching. 310-313
• 
  Themos Stafylakis, Xavier Anguera: Improvements to the equal-parameter BIC for speaker diarization. 314-317
• 
  Nima Mesgarani, Samuel Thomas, Hynek Hermansky: A multistream multiresolution framework for phoneme recognition. 318-321
• 
  Giampiero Salvi, Fabio Tesser, Enrico Zovato, Piero Cosi: Cluster analysis of differential spectral envelopes on emotional speech. 322-325
• 
  Sam Bowman, Karen Livescu: Modeling pronunciation variation with context-dependent articulatory feature decision trees. 326-329
• 
  Bhiksha Raj, Kevin W. Wilson, Alexander Krueger, Reinhold Haeb-Umbach: Ungrounded independent non-negative factor analysis. 330-333
• 
  John R. Hershey, Peder A. Olsen, Steven J. Rennie: Signal interaction and the devil function. 334-337

Systems for LVCSR

• 
  Yuya Akita, Masato Mimura, Graham Neubig, Tatsuya Kawahara: Semi-automated update of automatic transcription system for the Japanese national congress. 338-341
• 
  Xunying Liu, Mark J. F. Gales, Philip C. Woodland: Language model cross adaptation for LVCSR system combination. 342-345
• 
  Shinji Watanabe, Takaaki Hori, Atsushi Nakamura: Large vocabulary continuous speech recognition using WFST-based linear classifier for structured data. 346-349
• 
  Pavel Kveton, Miroslav Novak: Accelerating hierarchical acoustic likelihood computation on graphics processors. 350-353
• 
  Jiulong Shan, Genqing Wu, Zhihong Hu, Xiliu Tang, Martin Jansche, Pedro J. Moreno: Search by voice in Mandarin Chinese. 354-357
• 
  Thomas Hain, Lukas Burget, John Dines, Philip N. Garner, Asmaa El Hannani, Marijn Huijbregts, Martin Karafiát, Mike Lincoln, Vincent Wan: The AMIDA 2009 meeting transcription system. 358-361

Speaker Characterization and Recognition I-IV

• 
  William M. Campbell, Zahi N. Karam: Simple and efficient speaker comparison using approximate KL divergence. 362-365
• 
  Hanwu Sun, Bin Ma, Chien-Lin Huang, Trung Hieu Nguyen, Haizhou Li: The IIR NIST SRE 2008 and 2010 summed channel speaker recognition systems. 366-369
• 
  Chien-Lin Huang, Hanwu Sun, Bin Ma, Haizhou Li: Speaker characterization using long-term and temporal information. 370-373
• 
  Sergio Perez-Gomez, Daniel Ramos, Javier Gonzalez-Dominguez, Joaquin Gonzalez-Rodriguez: Score-level compensation of extreme speech duration variability in speaker verification. 374-377
• 
  Alberto Abad, Isabel Trancoso: Speaker recognition experiments using connectionist transformation network features. 378-381
• 
  Yun Lei, John H. L. Hansen: Speaker recognition using supervised probabilistic principal component analysis. 382-385
• 
  Benjamin Bigot, Julien Pinquier, Isabelle Ferrane, Régine André-Obrecht: Looking for relevant features for speaker role recognition. 1057-1060
• 
  Marcel Kockmann, Lukas Burget, Ondrej Glembek, Luciana Ferrer, Jan Cernocký: Prosodic speaker verification using subspace multinomial models with intersession compensation. 1061-1064
• 
  Eryu Wang, Kong-Aik Lee, Bin Ma, Haizhou Li, Wu Guo, Li-Rong Dai: The estimation and kernel metric of spectral correlation for text-independent speaker verification. 1065-1068
• 
  Rahim Saeidi, Pejman Mowlaee, Tomi Kinnunen, Zheng-Hua Tan, Mads Græsbøll Christensen, Søren Holdt Jensen, Pasi Fränti: Improving monaural speaker identification by double-talk detection. 1069-1072
• 
  B. Avinash, S. Guruprasad, B. Yegnanarayana: Exploring subsegmental and suprasegmental features for a text-dependent speaker verification in distant speech signals. 1073-1076
• 
  Qingsong Liu, Wei Huang, Dongxing Xu, Hongbin Cai, Beiqian Dai: A fast implementation of factor analysis for speaker verification. 1077-1080
• 
  Ce Zhang, Rong Zheng, Bo Xu: An investigation into direct scoring methods without SVM training in speaker verification. 1437-1440
• 
  Reda Jourani, Khalid Daoudi, Régine André-Obrecht, Driss Aboutajdine: Large margin Gaussian mixture models for speaker identification. 1441-1444
• 
  Rong Zheng, Bo Xu: On the use of Gaussian component information in the generative likelihood ratio estimation for speaker verification. 1445-1448
• 
  Man-Wai Mak, Wei Rao: Acoustic vector resampling for GMMSVM-based speaker verification. 1449-1452
• 
  Konstantin Biatov: A fast speaker indexing using vector quantization and second order statistics with adaptive threshold computation. 1453-1456
• 
  Gang Wang, Xiaojun Wu, Thomas Fang Zheng: Using phoneme recognition and text-dependent speaker verification to improve speaker segmentation for Chinese speech. 1457-1460
• 
  Claudio Garretón, Néstor Becerra Yoma: On enhancing feature sequence filtering with filter-bank energy transformation in speaker verification with telephone speech. 1461-1464
• 
  Donglai Zhu, Bin Ma, Kong-Aik Lee, Cheung-Chi Leung, Haizhou Li: MAP estimation of subspace transform for speaker recognition. 1465-1468
• 
  Ayeh Jafari, Ramji Srinivasan, Danny Crookes, Ji Ming: A longest matching segment approach for text-independent speaker recognition. 1469-1472
• 
  Ville Hautamäki, Tomi Kinnunen, Mohaddeseh Nosratighods, Kong-Aik Lee, Bin Ma, Haizhou Li: Approaching human listener accuracy with modern speaker verification. 1473-1476
• 
  Jouni Pohjalainen, Rahim Saeidi, Tomi Kinnunen, Paavo Alku: Extended weighted linear prediction (XLP) analysis of speech and its application to speaker verification in adverse conditions. 1477-1480
• 
  Guoli Ye, Brian Mak: The use of subvector quantization and discrete densities for fast GMM computation for speaker verification. 1481-1484
• 
  Fred S. Richardson, Joseph P. Campbell: Transcript-dependent speaker recognition using mixer 1 and 2. 2102-2105
• 
  Thomas Drugman, Thierry Dutoit: On the potential of glottal signatures for speaker recognition. 2106-2109
• 
  R. Padmanabhan, Hema A. Murthy: Acoustic feature diversity and speaker verification. 2110-2113
• 
  Omid Dehzangi, Bin Ma, Engsiong Chng, Haizhou Li: A discriminative performance metric for GMM-UBM speaker identification. 2114-2117
• 
  Xavier Anguera, Jean-François Bonastre: A novel speaker binary key derived from anchor models. 2118-2121
• 
  Weiqiang Zhang, Yan Deng, Liang He, Jia Liu: Variant time-frequency cepstral features for speaker recognition. 2122-2125
• 
  Ning Wang, P. C. Ching, Tan Lee: Exploitation of phase information for speaker recognition. 2126-2129
• 
  Yanhua Long, Li-Rong Dai, Bin Ma, Wu Guo: Effects of the phonological relevance in speaker verification. 2130-2133
• 
  Gabriel Hernández Sierra, Jean-François Bonastre, Driss Matrouf, José R. Calvo: Topological representation of speech for speaker recognition. 2134-2137
• 
  Seyed Omid Sadjadi, John H. L. Hansen: Assessment of single-channel speech enhancement techniques for speaker identification under mismatched conditions. 2138-2141
• 
  Xiang Zhang, Chuan Cao, Lin Yang, Hongbin Suo, Jianping Zhang, Yonghong Yan: Speaker recognition using the resynthesized speech via spectrum modeling. 2142-2145

Source Separation

• 
  Robert Peharz, Michael Stark, Franz Pernkopf, Yannis Stylianou: A factorial sparse coder model for single channel source separation. 386-389
• 
  Yasmina Benabderrahmane, Sid-Ahmed Selouani, Douglas D. O'Shaughnessy: Oriented PCA method for blind speech separation of convolutive mixtures. 390-393
• 
  Hsin-Lung Hsieh, Jen-Tzung Chien: Online Gaussian process for nonstationary speech separation. 394-397
• 
  Meng Yu, Wenye Ma, Jack Xin, Stanley Osher: Convexity and fast speech extraction by split bregman method. 398-401
• 
  Wenye Ma, Meng Yu, Jack Xin, Stanley Osher: Reducing musical noise in blind source separation by time-domain sparse filters and split bregman method. 402-405
• 
  John Woodruff, Rohit Prabhavalkar, Eric Fosler-Lussier, DeLiang Wang: Combining monaural and binaural evidence for reverberant speech segregation. 406-409

Speech Synthesis: HMM-Based Speech Synthesis I, II

• 
  Heiga Zen: Speaker and language adaptive training for HMM-based polyglot speech synthesis. 410-413
• 
  Kai Yu, Heiga Zen, François Mairesse, Steve Young: Context adaptive training with factorized decision trees for HMM-based speech synthesis. 414-417
• 
  Junichi Yamagishi, Oliver Watts, Simon King, Bela Usabaev: Roles of the average voice in speaker-adaptive HMM-based speech synthesis. 418-421
• 
  Yao Qian, Zhi-Jie Yan, Yi-Jian Wu, Frank K. Soong, Xin Zhuang, Shengyi Kong: An HMM trajectory tiling (HTT) approach to high quality TTS. 422-425
• 
  Yining Chen, Zhi-Jie Yan, Frank K. Soong: A perceptual study of acceleration parameters in HMM-based TTS. 426-429
• 
  Shuji Yokomizo, Takashi Nose, Takao Kobayashi: Evaluation of prosodic contextual factors for HMM-based speech synthesis. 430-433
• 
  Slava Shechtman, Alexander Sorin: Sinusoidal model parameterization for HMM-based TTS system. 805-808
• 
  Yoshinori Shiga, Tomoki Toda, Shinsuke Sakai, Hisashi Kawai: Improved training of excitation for HMM-based parametric speech synthesis. 809-812
• 
  June Sig Sung, Doo Hwa Hong, Kyung Hwan Oh, Nam Soo Kim: Excitation modeling based on waveform interpolation for HMM-based speech synthesis. 813-816
• 
  Xin Zhuang, Yao Qian, Frank K. Soong, Yi-Jian Wu, Bo Zhang: Formant-based frequency warping for improving speaker adaptation in HMM TTS. 817-820
• 
  Hongwei Hu, Martin J. Russell: Improved modelling of speech dynamics using non-linear formant trajectories for HMM-based speech synthesis. 821-824
• 
  Zhen-Hua Ling, Yu Hu, Li-Rong Dai: Global variance modeling on the log power spectrum of LSPs for HMM-based speech synthesis. 825-828
• 
  Matt Shannon, William Byrne: Autoregressive clustering for HMM speech synthesis. 829-832
• 
  Nicholas Pilkington, Heiga Zen: An implementation of decision tree-based context clustering on graphics processing units. 833-836
• 
  Alexander Gutkin, Xavi Gonzalvo, Stefan Breuer, Paul Taylor: Quantized HMMs for low footprint text-to-speech synthesis. 837-840
• 
  Oliver Watts, Junichi Yamagishi, Simon King: The role of higher-level linguistic features in HMM-based speech synthesis. 841-844
• 
  Ayami Mase, Keiichiro Oura, Yoshihiko Nankaku, Keiichi Tokuda: HMM-based singing voice synthesis system using pitch-shifted pseudo training data. 845-848
• 
  Jinfu Ni, Hisashi Kawai: An unsupervised approach to creating web audio contents-based HMM voices. 849-852
• 
  Tomoki Koriyama, Takashi Nose, Takao Kobayashi: Conversational spontaneous speech synthesis using average voice model. 853-856

Multi-Modal Signal Processing

• 
  Jonas Hörnstein, José Santos-Victor: Learning words and speech units through natural interactions. 434-437
• 
  Qingju Liu, Wenwu Wang, Philip J. B. Jackson: Bimodal coherence based scale ambiguity cancellation for target speech extraction and enhancement. 438-441
• 
  Hiroaki Kawashima, Yu Horii, Takashi Matsuyama: Speech estimation in non-stationary noise environments using timing structures between mouth movements and sound signals. 442-445
• 
  Lijuan Wang, Xiaojun Qian, Wei Han, Frank K. Soong: Synthesizing photo-real talking head via trajectory-guided sample selection. 446-449
• 
  Victoria M. Florescu, Lise Crevier-Buchman, Bruce Denby, Thomas Hueber, Antonia Colazo-Simon, Claire Pillot-Loiseau, Pierre Roussel-Ragot, Cédric Gendrot, Sophie Quattrocchi: Silent vs vocalized articulation for a portable ultrasound-based silent speech interface. 450-453
• 
  Gregor Hofer, Korin Richmond: Comparison of HMM and TMDN methods for lip synchronisation. 454-457

Paralanguage

• 
  Florian Schiel, Christian Heinrich, Veronika Neumeyer: Rhythm and formant features for automatic alcohol detection. 458-461
• 
  Irena Yanushevskaya, Christer Gobl, John Kane, Ailbhe Ní Chasaide: An exploration of voice source correlates of focus. 462-465
• 
  James D. Harnsberger, Rahul Shrivastav, W. S. Brown Jr.: Modeling perceived vocal age in american English. 466-469
• 
  Marie-José Caraty, Claude Montacié: Multivariate analysis of vocal fatigue in continuous reading. 470-473
• 
  Alexander Kain, Jan P. H. van Santen: Frequency-domain delexicalization using surrogate vowels. 474-477
• 
  Florian Metze, Anton Batliner, Florian Eyben, Tim Polzehl, Björn Schuller, Stefan Steidl: Emotion recognition using imperfect speech recognition. 478-481
• 
  Gang Liu, Yun Lei, John H. L. Hansen: A novel feature extraction strategy for multi-stream robust emotion identification. 482-485
• 
  Asterios Toutios, Utpala Musti, Slim Ouni, Vincent Colotte, Brigitte Wrobel-Dautcourt, Marie-Odile Berger: Setup for acoustic-visual speech synthesis by concatenating bimodal units. 486-489
• 
  Bart Jochems, Martha Larson, Roeland Ordelman, Ronald Poppe, Khiet P. Truong: Towards affective state modeling in narrative and conversational settings. 490-493
• 
  Narichika Nomoto, Hirokazu Masataki, Osamu Yoshioka, Satoshi Takahashi: Detection of anger emotion in dialog speech using prosody feature and temporal relation of utterances. 494-497
• 
  Benjamin Roustan, Marion Dohen: Gesture and speech coordination: the influence of the relationship between manual gesture and speech. 498-501
• 
  Hynek Boril, Seyed Omid Sadjadi, Tristan Kleinschmidt, John H. L. Hansen: Analysis and detection of cognitive load and frustration in drivers' speech. 502-505
• 
  Akira Sasou, Yasuharu Hashimoto, Katsuhiko Sakaue: Acoustic-based recognition of head gestures accompanying speech. 506-509
• 
  Sandro Castronovo, Angela Mahr, Margarita Pentcheva, Christian A. Müller: Multimodal dialog in the car: combining speech and turn-and-push dial to control comfort functions. 510-513
• 
  Danil Korchagin, Philip N. Garner, Petr Motlícek: Hands free audio analysis from home entertainment. 514-517
• 
  Shaikh Mostafa Al Masum, Antonio Rui Ferreira Rebordão, Keikichi Hirose: Affective story teller: a TTS system for emotional expressivity. 518-521

ASR: Speaker Adaptation, Robustness Against Reverberation

• 
  Shweta Ghai, Rohit Sinha: Enhancing children's speech recognition under mismatched condition by explicit acoustic normalization. 522-525
• 
  Bo Li, Khe Chai Sim: Comparison of discriminative input and output transformations for speaker adaptation in the hybrid NN/HMM systems. 526-529
• 
  Ravichander Vipperla, Steve Renals, Joe Frankel: Augmentation of adaptation data. 530-533
• 
  Lukás Machlica, Zbynek Zajíc, Ludek Müller: Discriminative adaptation based on fast combination of DMAP and dfMLLR. 534-537
• 
  Doddipatla Rama Sanand, Ralf Schlüter, Hermann Ney: Revisiting VTLN using linear transformation on conventional MFCC. 538-541
• 
  Toyohiro Hayashi, Yoshihiko Nankaku, Akinobu Lee, Keiichi Tokuda: Speaker adaptation based on nonlinear spectral transform for speech recognition. 542-545
• 
  Tetsuo Kosaka, Takashi Ito, Masaharu Katoh, Masaki Kohda: Speaker adaptation based on system combination using speaker-class models. 546-549
• 
  Yongwon Jeong, Young Rok Song, Hyung Soon Kim: Speaker adaptation in transformation space using two-dimensional PCA. 550-553
• 
  Jan Trmal, Jan Zelinka, Ludek Müller: On speaker adaptive training of artificial neural networks. 554-557
• 
  Yongjun He, Jiqing Han: Model synthesis for band-limited speech recognition. 558-561
• 
  Takahiro Fukumori, Masanori Morise, Takanobu Nishiura: Performance estimation of reverberant speech recognition based on reverberant criteria RSR-d_n with acoustic parameters. 562-565
• 
  Armin Sehr, Christian Hofmann, Roland Maas, Walter Kellermann: A novel approach for matched reverberant training of HMMs using data pairs. 566-569
• 
  Hari Krishna Maganti, Marco Matassoni: An auditory based modulation spectral feature for reverberant speech recognition. 570-573
• 
  Martin Wolf, Climent Nadeu: On the potential of channel selection for recognition of reverberated speech with multiple microphones. 574-577
• 
  Randy Gomez, Tatsuya Kawahara: An improved wavelet-based dereverberation for robust automatic speech recognition. 578-581
• 
  Rico Petrick, Thomas Fehér, Masashi Unoki, Rüdiger Hoffmann: Methods for robust speech recognition in reverberant environments: a comparison. 582-585

Language Learning, TTS, and Other Applications

• 
  Masayuki Suzuki, Yu Qiao, Nobuaki Minematsu, Keikichi Hirose: Integration of multilayer regression analysis with structure-based pronunciation assessment. 586-589
• 
  Joost van Doremalen, Catia Cucchiarini, Helmer Strik: Using non-native error patterns to improve pronunciation verification. 590-593
• 
  Dean Luo, Yu Qiao, Nobuaki Minematsu, Yutaka Yamauchi, Keikichi Hirose: Regularized-MLLR speaker adaptation for computer-assisted language learning system. 594-597
• 
  Kuniaki Hirabayashi, Seiichi Nakagawa: Automatic evaluation of English pronunciation by Japanese speakers using various acoustic features and pattern recognition techniques. 598-601
• 
  Hsien-Cheng Liao, Jiang-Chun Chen, Sen-Chia Chang, Ying-Hua Guan, Chin-Hui Lee: Decision tree based tone modeling with corrective feedbacks for automatic Mandarin tone assessment. 602-605
• 
  Jingli Lu, Ruili Wang, Liyanage C. De Silva, Yang Gao, Jia Liu: CASTLE: a computer-assisted stress teaching and learning environment for learners of English as a second language. 606-609
• 
  Shen Huang, Hongyan Li, Shijin Wang, Jiaen Liang, Bo Xu: Automatic reference independent evaluation of prosody quality using multiple knowledge fusions. 610-613
• 
  Su-Youn Yoon, Mark Hasegawa-Johnson, Richard Sproat: Landmark-based automated pronunciation error detection. 614-617
• 
  Zhiwei Shuang, Shiyin Kang, Yong Qin, Li-Rong Dai, Lianhong Cai: HMM based TTS for mixed language text. 618-621
• 
  Hui Liang, John Dines: An analysis of language mismatch in HMM state mapping-based cross-lingual speaker adaptation. 622-625
• 
  Tatsuya Kawahara, Norihiro Katsumaru, Yuya Akita, Shinsuke Mori: Classroom note-taking system for hearing impaired students using automatic speech recognition adapted to lectures. 626-629
• 
  Paul R. Dixon, Sadaoki Furui: Exploring web-browser based runtimes engines for creating ubiquitous speech interfaces. 630-632

Pitch and Glottal-Waveform Estimation and Modeling I, II

• 
  Xuejing Sun, Sameer Gadre: Efficient three-stage pitch estimation for packet loss concealment. 633-636
• 
  Keiichi Funaki: On evaluation of the f₀ estimation based on time-varying complex speech analysis. 637-640
• 
  Feng Huang, Tan Lee: Pitch estimation in noisy speech based on temporal accumulation of spectrum peaks. 641-644
• 
  Tianyu T. Wang, Thomas F. Quatieri: Multi-pitch estimation by a joint 2-d representation of pitch and pitch dynamics. 645-648
• 
  Pirros Tsiakoulis, Alexandros Potamianos: On the effect of fundamental frequency on amplitude and frequency modulation patterns in speech resonances. 649-652
• 
  M. Shahidur Rahman, Tetsuya Shimamura: Pitch determination using autocorrelation function in spectral domain. 653-656
• 
  Thomas Drugman, Thierry Dutoit: Chirp complex cepstrum-based decomposition for asynchronous glottal analysis. 657-660
• 
  Alan Ó Cinnéide, David Dorran, Mikel Gainza, Eugene Coyle: Exploiting glottal formant parameters for glottal inverse filtering and parameterization. 661-664
• 
  Nicolas Sturmel, Christophe d'Alessandro, Boris Doval: Glottal parameters estimation on speech using the zeros of the z-transform. 665-668
• 
  Sri Harish Reddy Mallidi, Kishore Prahallad, Suryakanth V. Gangashetty, B. Yegnanarayana: Significance of pitch synchronous analysis for speaker recognition using AANN models. 669-672
• 
  Gang Chen, Xue Feng, Yen-Liang Shue, Abeer Alwan: On using voice source measures in automatic gender classification of children's speech. 673-676
• 
  Wei Chu, Abeer Alwan: SAFE: a statistical algorithm for F0 estimation for both clean and noisy speech. 2590-2593
• 
  Jung Ook Hong, Patrick J. Wolfe: Robust and efficient pitch estimation using an iterative ARMA technique. 2594-2597
• 
  Yasunori Ohishi, Hirokazu Kameoka, Daichi Mochihashi, Hidehisa Nagano, Kunio Kashino: Statistical modeling of F0 dynamics in singing voices based on Gaussian processes with multiple oscillation bases. 2598-2601
• 
  Martin Heckmann, Claudius Gläser, Frank Joublin, Kazuhiro Nakadai: Applying geometric source separation for improved pitch extraction in human-robot interaction. 2602-2605
• 
  John Kane, Mark Kane, Christer Gobl: A spectral LF model based approach to voice source parameterisation. 2606-2609
• 
  Thomas Drugman, Thierry Dutoit: Glottal-based analysis of the lombard effect. 2610-2613

Open Vocabulary Spoken Document Retrieval (Special Session)

• 
  Yoshiaki Itoh, Hiromitsu Nishizaki, Xinhui Hu, Hiroaki Nanjo, Tomoyosi Akiba, Tatsuya Kawahara, Seiichi Nakagawa, Tomoko Matsui, Yoichi Yamashita, Kiyoaki Aikawa: Constructing Japanese test collections for spoken term detection. 677-680
• 
  Satoshi Natori, Hiromitsu Nishizaki, Yoshihiro Sekiguchi: Japanese spoken term detection using syllable transition network derived from multiple speech recognizers' outputs. 681-684
• 
  Sha Meng, Weiqiang Zhang, Jia Liu: Combining Chinese spoken term detection systems via side-information conditioned linear logistic regression. 685-688
• 
  Taisuke Kaneko, Tomoyosi Akiba: Metric subspace indexing for fast spoken term detection. 689-692
• 
  Chun-an Chan, Lin-Shan Lee: Unsupervised spoken-term detection with spoken queries using segment-based dynamic time warping. 693-696
• 
  Daniel Schneider, Timo Mertens, Martha Larson, Joachim Köhler: Contextual verification for open vocabulary spoken term detection. 697-700
• 
  Javier Tejedor, Doroteo Torre Toledano, Miguel Bautista, Simon King, Dong Wang, José Colás: Augmented set of features for confidence estimation in spoken term detection. 701-704
• 
  Xinhui Hu, Ryosuke Isotani, Hisashi Kawai, Satoshi Nakamura: Cluster-based language model for spoken document retrieval using NMF-based document clustering. 705-708

Robust ASR

• 
  Rogier C. van Dalen, Mark J. F. Gales: Asymptotically exact noise-corrupted speech likelihoods. 709-712
• 
  Ramón Fernandez Astudillo, Reinhold Orglmeister: A MMSE estimator in mel-cepstral domain for robust large vocabulary automatic speech recognition using uncertainty propagation. 713-716
• 
  Bhiksha Raj, Tuomas Virtanen, Sourish Chaudhuri, Rita Singh: Non-negative matrix factorization based compensation of music for automatic speech recognition. 717-720
• 
  Kris Demuynck, Xueru Zhang, Dirk Van Compernolle, Hugo Van Hamme: Feature versus model based noise robustness. 721-724
• 
  Ji Hun Park, Seon Man Kim, Jae Sam Yoon, Hong Kook Kim, Sung Joo Lee, Yunkeun Lee: SNR-based mask compensation for computational auditory scene analysis applied to speech recognition in a car environment. 725-728
• 
  Chanwoo Kim, Richard M. Stern, Kiwan Eom, Jaewon Lee: Automatic selection of thresholds for signal separation algorithms based on interaural delay. 729-732

Language and Dialect Identification

• 
  Florian Verdet, Driss Matrouf, Jean-François Bonastre, Jean Hennebert: Channel detectors for system fusion in the context of NIST LRE 2009. 733-736
• 
  Rong Tong, Bin Ma, Haizhou Li, Engsiong Chng: Selecting phonotactic features for language recognition. 737-740
• 
  Abualsoud Hanani, Michael J. Carey, Martin J. Russell: Improved language recognition using mixture components statistics. 741-744
• 
  Mikel Peñagarikano, Amparo Varona, Luis Javier Rodríguez-Fuentes, Germán Bordel: Using cross-decoder co-occurrences of phone n-grams in SVM-based phonotactic language recognition. 745-748
• 
  Oscar Koller, Alberto Abad, Isabel Trancoso, Céu Viana: Exploiting variety-dependent phones in portuguese variety identification applied to broadcast news transcription. 749-752
• 
  Fadi Biadsy, Julia Hirschberg, Michael Collins: Dialect recognition using a phone-GMM-supervector-based SVM kernel. 753-756

Technologies for Learning and Education

• 
  Xiaojun Qian, Frank K. Soong, Helen M. Meng: Discriminative acoustic model for improving mispronunciation detection and diagnosis in computer-aided pronunciation training (CAPT). 757-760
• 
  Liang-Yu Chen, Jyh-Shing Roger Jang: Automatic pronunciation scoring using learning to rank and DP-based score segmentation. 761-764
• 
  Wai Kit Lo, Shuang Zhang, Helen M. Meng: Automatic derivation of phonological rules for mispronunciation detection in a computer-assisted pronunciation training system. 765-768
• 
  Minh Duong, Jack Mostow: Adapting a duration synthesis model to rate children's oral reading prosody. 769-772
• 
  Su-Youn Yoon, Lei Chen, Klaus Zechner: Predicting word accuracy for the automatic speech recognition of non-native speech. 773-776
• 
  Taotao Zhu, Dengfeng Ke, Zhenbiao Chen, Bo Xu: A new approach for automatic tone error detection in strong accented Mandarin based on dominant set. 777-780

Emotional Speech

• 
  S. R. Mahadeva Prasanna, D. Govind: Analysis of excitation source information in emotional speech. 781-784
• 
  Dongrui Wu, Thomas D. Parsons, Shrikanth S. Narayanan: Acoustic feature analysis in speech emotion primitives estimation. 785-788
• 
  Lan-Ying Yeh, Tai-Shih Chi: Spectro-temporal modulations for robust speech emotion recognition. 789-792
• 
  Chi-Chun Lee, Matthew Black, Athanasios Katsamanis, Adam C. Lammert, Brian R. Baucom, Andrew Christensen, Panayiotis G. Georgiou, Shrikanth S. Narayanan: Quantification of prosodic entrainment in affective spontaneous spoken interactions of married couples. 793-796
• 
  Emily Mower, Kyu Jeong Han, Sungbok Lee, Shrikanth S. Narayanan: A cluster-profile representation of emotion using agglomerative hierarchical clustering. 797-800
• 
  Björn Schuller, Laurence Devillers: Incremental acoustic valence recognition: an inter-corpus perspective on features, matching, and performance in a gating paradigm. 801-804

New Paradigms in ASR I, II

• 
  Xiaodong Wang, Kunihiko Owa, Makoto Shozakai: Mandarin digit recognition assisted by selective tone distinction. 857-860
• 
  Kazuhiko Abe, Sakriani Sakti, Ryosuke Isotani, Hisashi Kawai, Satoshi Nakamura: Brazilian portuguese acoustic model training based on data borrowing from other language. 861-864
• 
  Ngoc Thang Vu, Tim Schlippe, Franziska Kraus, Tanja Schultz: Rapid bootstrapping of five eastern european languages using the rapid language adaptation toolkit. 865-868
• 
  Houwei Cao, Tan Lee, P. C. Ching: Cross-lingual speaker adaptation via Gaussian component mapping. 869-872
• 
  Mohamed Elmahdy, Rainer Gruhn, Wolfgang Minker, Slim Abdennadher: Cross-lingual acoustic modeling for dialectal Arabic speech recognition. 873-876
• 
  Samuel Thomas, Sriram Ganapathy, Hynek Hermansky: Cross-lingual and multi-stream posterior features for low resource LVCSR systems. 877-880
• 
  Shiva Sundaram, Jerome R. Bellegarda: Latent perceptual mapping: a new acoustic modeling framework for speech recognition. 881-884
• 
  Richard Dufour, Fethi Bougares, Yannick Estève, Paul Deléglise: Unsupervised model adaptation on targeted speech segments for LVCSR system combination. 885-888
• 
  Irene Ayllón Clemente, Martin Heckmann, Alexander Denecke, Britta Wrede, Christian Goerick: Incremental word learning using large-margin discriminative training and variance floor estimation. 889-892
• 
  Tuomas Virtanen, Jort F. Gemmeke, Antti Hurmalainen: State-based labelling for a sparse representation of speech and its application to robust speech recognition. 893-896
• 
  Mirko Hannemann, Stefan Kombrink, Martin Karafiát, Lukas Burget: Similarity scoring for recognizing repeated out-of-vocabulary words. 897-900
• 
  Dino Seppi, Dirk Van Compernolle: Data pruning for template-based automatic speech recognition. 901-904
• 
  Man-Hung Siu, Herbert Gish, Arthur Chan, William Belfield: Improved topic classification and keyword discovery using an HMM-based speech recognizer trained without supervision. 2838-2841
• 
  Dimitri Kanevsky, Tara N. Sainath, Bhuvana Ramabhadran, David Nahamoo: An analysis of sparseness and regularization in exemplar-based methods for speech classification. 2842-2845
• 
  Abdel-rahman Mohamed, Dong Yu, Li Deng: Investigation of full-sequence training of deep belief networks for speech recognition. 2846-2849
• 
  Yow-Bang Wang, Lin-Shan Lee: Mandarin tone recognition using affine-invariant prosodic features and tone posteriorgram. 2850-2853
• 
  Geoffrey Zweig, Patrick Nguyen, Jasha Droppo, Alex Acero: Continuous speech recognition with a TF-IDF acoustic model. 2854-2857
• 
  Geoffrey Zweig, Patrick Nguyen: SCARF: a segmental conditional random field toolkit for speech recognition. 2858-2861

Speech Production: Various Approaches

• 
  Akiko Amano-Kusumoto, John-Paul Hosom, Alexander Kain: Speaking style dependency of formant targets. 905-908
• 
  Tatsuya Kitamura: Similarity of effects of emotions on the speech organ configuration with and without speaking. 909-912
• 
  Daniel Bone, Samuel Kim, Sungbok Lee, Shrikanth S. Narayanan: A study of intra-speaker and inter-speaker affective variability using electroglottograph and inverse filtered glottal waveforms. 913-916
• 
  Ken-Ichi Sakakibara, Hiroshi Imagawa, Miwako Kimura, Hisayuki Yokonishi, Niro Tayama: Modal analysis of vocal fold vibrations using laryngotopography. 917-920
• 
  Martti Vainio, Matti Airas, Juhani Järvikivi, Paavo Alku: Laryngeal voice quality in the expression of focus. 921-924
• 
  Masako Fujimoto, Kikuo Maekawa, Seiya Funatsu: Laryngeal characteristics during the production of geminate consonants. 925-928
• 
  Julien Cisonni, Kazunori Nozaki, Annemie Van Hirtum, Shigeo Wada: Numerical study of turbulent flow-induced sound production in presence of a tooth-shaped obstacle: towards sibilant [s] physical modeling. 929-932
• 
  Iris Hanique, Barbara Schuppler, Mirjam Ernestus: Morphological and predictability effects on schwa reduction: the case of dutch word-initial syllables. 933-936
• 
  Samer Al Moubayed, G. Ananthakrishnan: Acoustic-to-articulatory inversion based on local regression. 937-940
• 
  Mirjam Broersma: Korean lenis, fortis, and aspirated stops: effect of place of articulation on acoustic realization. 941-944
• 
  Toru Nakashika, Ryuki Tachibana, Masafumi Nishimura, Tetsuya Takiguchi, Yasuo Ariki: Speech synthesis by modeling harmonics structure with multiple function. 945-948
• 
  Makoto Otani, Tatsuya Hirahara: Physics of body-conducted silent speech - production, propagation and representation of non-audible murmur. 949-952

Speech Enhancement

• 
  Subhojit Chakladar, Nam Soo Kim, Yu Gwang Jin, Tae Gyoon Kang: Multichannel noise reduction using low order RTF estimate. 953-956
• 
  Inho Lee, Jongsung Yoon, Yoonjae Lee, Hanseok Ko: Reinforced blocking matrix with cross channel projection for speech enhancement. 957-960
• 
  Ning Cheng, Wenju Liu, Lan Wang: Masking property based microphone array post-filter design. 961-964
• 
  Yusuke Sato, Tetsuya Hoya, Hovagim Bakardjian, Andrzej Cichocki: Reduction of broadband noise in speech signals by multilinear subspace analysis. 965-968
• 
  Jungpyo Hong, Seung Ho Han, Sangbae Jeong, Minsoo Hahn: Novel probabilistic control of noise reduction for improved microphone array beamforming. 969-972
• 
  Kai Li, Qiang Fu, Yonghong Yan: Speech enhancement using improved generalized sidelobe canceller in frequency domain with multi-channel postfiltering. 973-976
• 
  Jani Even, Carlos Toshinori Ishi, Hiroshi Saruwatari, Norihiro Hagita: Close speaker cancellation for suppression of non-stationary background noise for hands-free speech interface. 977-980
• 
  Ajay Srinivasamurthy, Thippur V. Sreenivas: Multi-channel iterative dereverberation based on codebook constrained iterative multi-channel wiener filter. 981-984
• 
  Anand Joseph Xavier Medabalimi, Sri Harish Reddy Mallidi, B. Yegnanarayana: Speaker-dependent mapping of source and system features for enhancement of throat microphone speech. 985-988
• 
  Jun Cai, Stefano Marini, Pierre Malarme, Francis Grenez, Jean Schoentgen: An analytic modeling approach to enhancing throat microphone speech commands for keyword spotting. 989-992
• 
  Stephen So, Kamil K. Wójcicki, Kuldip K. Paliwal: Single-channel speech enhancement using kalman filtering in the modulation domain. 993-996
• 
  Miao Yao, Weiqian Liang: Integrated feedback and noise reduction algorithm in digital hearing aids via oscillation detection. 997-1000
• 
  Charles Mercier, Roch Lefebvre: A blind signal-to-noise ratio estimator for high noise speech recordings. 1001-1004

Special Session: Fact and Replica of Speech Production (Special Session)

• 
  Hiroshi Imagawa, Ken-Ichi Sakakibara, Isao T. Tokuda, Mamiko Otsuka, Niro Tayama: Estimation of glottal area function using stereo-endoscopic high-speed digital imaging. 1005-1008
• 
  Kazunori Nozaki, Youhei Ohnishi, Takashi Suda, Shigeo Wada, Shinji Shimojo: Toward aero-acoustical analysis of the sibilant /s/: an oral cavity modeling. 1009-1012
• 
  Kunitoshi Motoki: Effects of wall impedance on transmission and attenuation of higher-order modes in vocal-tract model. 1013-1016
• 
  Peter Birkholz, Bernd J. Kröger, Christiane Neuschaefer-Rube: Articulatory synthesis and perception of plosive-vowel syllables with virtual consonant targets. 1017-1020
• 
  Kotaro Fukui, Toshihiro Kusano, Yoshikazu Mukaeda, Yuto Suzuki, Atsuo Takanishi, Masaaki Honda: Speech robot mimicking human articulatory motion. 1021-1024
• 
  Takayuki Arai: Mechanical vocal-tract models for speech dynamics. 1025-1028
• 
  Michael C. Brady: Prosodic timing analysis for articulatory re-synthesis using a bank of resonators with an adaptive oscillator. 1029-1032

ASR: Language Modeling

• 
  Ahmad Emami, Stanley F. Chen, Abraham Ittycheriah, Hagen Soltau, Bing Zhao: Decoding with shrinkage-based language models. 1033-1036
• 
  Stanley F. Chen, Stephen M. Chu: Enhanced word classing for model M. 1037-1040
• 
  Junho Park, Xunying Liu, Mark J. F. Gales, Philip C. Woodland: Improved neural network based language modelling and adaptation. 1041-1044
• 
  Tomas Mikolov, Martin Karafiát, Lukas Burget, Jan Cernocký, Sanjeev Khudanpur: Recurrent neural network based language model. 1045-1048
• 
  Preethi Jyothi, Eric Fosler-Lussier: Discriminative language modeling using simulated ASR errors. 1049-1052
• 
  Graham Neubig, Masato Mimura, Shinsuke Mori, Tatsuya Kawahara: Learning a language model from continuous speech. 1053-1056

Single-Channel Speech Enhancement