INTERSPEECH 2010: Makuhari, Japan

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

• Stephen So, Kuldip K. Paliwal:
  Fast converging iterative kalman filtering for speech enhancement using long and overlapped tapered windows with large side lobe attenuation. 1081-1084
  
• Xuejing Sun, Kuan-Chieh Yen, Rogerio Guedes Alves:
  Robust noise estimation using minimum correction with harmonicity control. 1085-1088
  
• Mahdi Triki:
  New insights into subspace noise tracking. 1089-1092
  
• Mahdi Triki, Kees Janse:
  Bias considerations for minimum subspace noise tracking. 1093-1096
  
• Ji Ming, Ramji Srinivasan, Danny Crookes:
  A corpus-based approach to speech enhancement from nonstationary noise. 1097-1100
  
• Zhe Chen, You-Chi Cheng, Fuliang Yin, Chin-Hui Lee:
  Bandwidth expansion of speech based on wavelet transform modulus maxima vector mapping. 1101-1104
  

Speech Synthesis: Miscellaneous Topics

• Kalu U. Ogbureke, Peter Cahill, Julie Carson-Berndsen:
  Hidden Markov models with context-sensitive observations for grapheme-to-phoneme conversion. 1105-1108
  
• Brian Langner, Stephan Vogel, Alan W. Black:
  Evaluating a dialog language generation system: comparing the mountain system to other NLG approaches. 1109-1112
  
• Wesley Mattheyses, Lukas Latacz, Werner Verhelst:
  Active appearance models for photorealistic visual speech synthesis. 1113-1116
  
• Jerome R. Bellegarda:
  Latent affective mapping: a novel framework for the data-driven analysis of emotion in text. 1117-1120
  
• Anna C. Janska, Robert A. J. Clark:
  Native and non-native speaker judgements on the quality of synthesized speech. 1121-1124
  
• Dominic Espinosa, Michael White, Eric Fosler-Lussier, Chris Brew:
  Machine learning for text selection with expressive unit-selection voices. 1125-1128
  

Prosody: Basics & Applications

• Alexei V. Ivanov, Giuseppe Riccardi, Sucheta Ghosh, Sara Tonelli, Evgeny A. Stepanov:
  Acoustic correlates of meaning structure in conversational speech. 1129-1132
  
• Nicolas Obin, Xavier Rodet, Anne Lacheret:
  HMM-based prosodic structure model using rich linguistic context. 1133-1136
  
• Charlotte Wollermann, Bernhard Schröder, Ulrich Schade:
  Audiovisual congruence and pragmatic focus marking. 1137-1140
  
• Margaret Zellers, Michele Gubian, Brechtje Post:
  Redescribing intonational categories with functional data analysis. 1141-1144
  
• Shen Huang, Hongyan Li, Shijin Wang, Jiaen Liang, Bo Xu:
  Exploring goodness of prosody by diverse matching templates. 1145-1148
  
• Mickael Rouvier, Richard Dufour, Georges Linarès, Yannick Estève:
  A language-identification inspired method for spontaneous speech detection. 1149-1152
  
• Gérard Bailly, Amélie Lelong:
  Speech dominoes and phonetic convergence. 1153-1156
  
• Mátyás Brendel, Riccardo Zaccarelli, Laurence Devillers:
  A quick sequential forward floating feature selection algorithm for emotion detection from speech. 1157-1160
  
• Géza Kiss, Jan P. H. van Santen:
  Automated vocal emotion recognition using phoneme class specific features. 1161-1164
  
• Adrian Pass, Jianguo Zhang, Darryl Stewart:
  Feature selection for pose invariant lip biometrics. 1165-1168
  
• Hussein Hussein, Rüdiger Hoffmann:
  Signal-based accent and phrase marking using the fujisaki model. 1169-1172
  
• Jangwon Kim, Sungbok Lee, Shrikanth S. Narayanan:
  A study of interplay between articulatory movement and prosodic characteristics in emotional speech production. 1173-1176
  

ASR: Feature Extraction I, II

• Shang-wen Li, Liang-Che Sun, Lin-Shan Lee:
  Improved phoneme recognition by integrating evidence from spectro-temporal and cepstral features. 1177-1180
  
• Suman V. Ravuri, Nelson Morgan:
  Using spectro-temporal features to improve AFE feature extraction for ASR. 1181-1184
  
• Ibon Saratxaga, Inma Hernáez, Igor Odriozola, Eva Navas, Iker Luengo, Daniel Erro:
  Using harmonic phase information to improve ASR rate. 1185-1188
  
• Kazumasa Yamamoto, Eiichi Sueyoshi, Seiichi Nakagawa:
  Speech recognition using long-term phase information. 1189-1192
  
• Jan Zelinka, Jan Trmal, Ludek Müller:
  Low-dimensional space transforms of posteriors in speech recognition. 1193-1196
  
• Christian Plahl, Ralf Schlüter, Hermann Ney:
  Hierarchical bottle neck features for LVCSR. 1197-1200
  
• Frantisek Grézl, Martin Karafiát:
  Hierarchical neural net architectures for feature extraction in ASR. 1201-1204
  
• Vivek Kumar Rangarajan Sridhar, Rohit Prasad, Prem Natarajan:
  Mutual information analysis for feature and sensor subset selection in surface electromyography based speech recognition. 1205-1208
  
• Bernd T. Meyer, Birger Kollmeier:
  Learning from human errors: prediction of phoneme confusions based on modified ASR training. 1209-1212
  
• Bo Li, Khe Chai Sim:
  Hidden logistic linear regression for support vector machine based phone verification. 2614-2617
  
• Tim Ng, Bing Zhang, Long Nguyen:
  Jointly optimized discriminative features for speech recognition. 2618-2621
  
• Florian Müller, Alfred Mertins:
  Invariant integration features combined with speaker-adaptation methods. 2622-2625
  
• Mark Raugas, Vivek Kumar Rangarajan Sridhar, Rohit Prasad, Prem Natarajan:
  Multi resolution discriminative models for subvocalic speech recognition. 2626-2629
  
• Fabio Valente, Mathew Magimai-Doss, Christian Plahl, Suman V. Ravuri, Wen Wang:
  A comparative large scale study of MLP features for Mandarin ASR. 2630-2633
  
• Cong-Thanh Do, Dominique Pastor, Gaël Le Lan, André Goalic:
  Recognizing cochlear implant-like spectrally reduced speech with HMM-based ASR: experiments with MFCCs and PLP coefficients. 2634-2637
  

Speech Perception: Cross Language and Age

• Kazuhiro Kondo, Takayuki Kanda, Yosuke Kobayashi, Hiroyuki Yagyu:
  Speech intelligibility of diagonally localized speech with competing noise using bone-conduction headphones. 1213-1216
  
• Pierre L. Divenyi:
  Masking of vowel-analog transitions by vowel-analog distracters. 1217-1220
  
• François Pellegrino, Emmanuel Ferragne, Fanny Meunier:
  2010, a speech oddity: phonetic transcription of reversed speech. 1221-1224
  
• Hsin-Yi Lin, Janice Fon:
  Perception on pitch reset at discourse boundaries. 1225-1228
  
• Marjorie Dole, Michel Hoen, Fanny Meunier:
  Effect of spatial separation on speech-in-noise comprehension in dyslexic adults. 1229-1232
  
• Ellen Marklund, Francisco Lacerda, Anna Ericsson:
  Speech categorization context effects in seven- to nine-month-old infants. 1233-1236
  
• Diane Kewley-Port, Larry E. Humes, Daniel Fogerty:
  Changes in temporal processing of speech across the adult lifespan. 1237-1240
  
• Jared Bernstein, Jian Cheng, Masanori Suzuki:
  Fluency and structural complexity as predictors of L2 oral proficiency. 1241-1244
  
• Marco van de Ven, Benjamin V. Tucker, Mirjam Ernestus:
  Semantic facilitation in bilingual everyday speech comprehension. 1245-1248
  
• Bo-ren Hsieh, Ho-hsien Pan:
  L2 experience and non-native vowel categorization of L1-Mandarin speakers. 1249-1252
  
• Mirjam Wester:
  Cross-lingual talker discrimination. 1253-1256
  
• Takashi Otake:
  Dajare is not the lowest form of wit. 1257-1260
  

SLP Systems

• Rafael Torres, Shota Takeuchi, Hiromichi Kawanami, Tomoko Matsui, Hiroshi Saruwatari, Kiyohiro Shikano:
  Comparison of methods for topic classification in a speech-oriented guidance system. 1261-1264
  
• Pere Comas, Jordi Turmo, Lluís Màrquez:
  Using dependency parsing and machine learning for factoid question answering on spoken documents. 1265-1268
  
• Carolina Parada, Abhinav Sethy, Mark Dredze, Frederick Jelinek:
  A spoken term detection framework for recovering out-of-vocabulary words using the web. 1269-1272
  
• Hung-yi Lee, Chia-Ping Chen, Ching-feng Yeh, Lin-Shan Lee:
  Improved spoken term detection by discriminative training of acoustic models based on user relevance feedback. 1273-1276
  
• Sebastian Tschöpel, Daniel Schneider:
  A lightweight keyword and tag-cloud retrieval algorithm for automatic speech recognition transcripts. 1277-1280
  
• Noboru Kanedera, Tetsuo Funada, Seiichi Nakagawa:
  Lecture subtopic retrieval by retrieval keyword expansion using subordinate concept. 1281-1284
  
• Hiroaki Nanjo, Yusuke Iyonaga, Takehiko Yoshimi:
  Spoken document retrieval for oral presentations integrating global document similarities into local document similarities. 1285-1288
  
• Joseph Polifroni, Stephanie Seneff:
  Combining word-based features, statistical language models, and parsing for named entity recognition. 1289-1292
  
• Azeddine Zidouni, Sophie Rosset, Hervé Glotin:
  Efficient combined approach for named entity recognition in spoken language. 1293-1296
  
• Sree Harsha Yella, Vasudeva Varma, Kishore Prahallad:
  Prominence based scoring of speech segments for automatic speech-to-speech summarization. 1297-1300
  
• Zihan Liu, Lei Xie, Wei Feng:
  Maximum lexical cohesion for fine-grained news story segmentation. 1301-1304
  
• Xiaoxuan Wang, Lei Xie, Bin Ma, Engsiong Chng, Haizhou Li:
  Phoneme lattice based texttiling towards multilingual story segmentation. 1305-1308
  

Quality of Experiencing Speech Services (Special Session)

• Anton Schlesinger, Marinus M. Boone:
  The characterization of the relative information content by spectral features for the objective intelligibility assessment of nonlinearly processed speech. 1309-1312
  
• Marcel Wältermann, Alexander Raake, Sebastian Möller:
  Analytical assessment and distance modeling of speech transmission quality. 1313-1316
  
• Nicolas Côté, Vincent Koehl, Valérie Gautier-Turbin, Alexander Raake, Sebastian Möller:
  An intrusive super-wideband speech quality model: DIAL. 1317-1320
  
• Sebastian Egger, Raimund Schatz, Stefan Scherer:
  It takes two to tango - assessing the impact of delay on conversational interactivity on perceived speech quality. 1321-1324
  
• Sebastian Möller, Florian Hinterleitner, Tiago H. Falk, Tim Polzehl:
  Comparison of approaches for instrumentally predicting the quality of text-to-speech systems. 1325-1328
  
• Imre Kiss, Joseph Polifroni, Chao Wang, Ghinwa F. Choueiter, Mike Phillips:
  A hybrid architecture for mobile voice user interfaces. 1329-1332
  
• Markku Turunen, Jaakko Hakulinen, Tomi Heimonen:
  Assessment of spoken and multimodal applications: lessons learned from laboratory and field studies. 1333-1336
  
• Klaus-Peter Engelbrecht, Hamed Ketabdar, Sebastian Möller:
  Improving cross database prediction of dialogue quality using mixture of experts. 1337-1340