IEEE Transactions on Audio, Speech & Language Processing, Volume 14

Volume 14, Number 1, January 2006

• Lie Lu, Dan Liu, HongJiang Zhang:
  Automatic mood detection and tracking of music audio signals. 5-18
  
• Ning Ma, Martin Bouchard, Rafik A. Goubran:
  Speech enhancement using a masking threshold constrained Kalman filter and its heuristic implementations. 19-32
  
• James D. Gordy, Rafik A. Goubran:
  On the perceptual performance limitations of echo cancellers in wideband telephony. 33-42
  
• Marcus Holmberg, David Gelbart, Werner Hemmert:
  Automatic speech recognition with an adaptation model motivated by auditory processing. 43-49
  
• Thomas Blumensath, Mike E. Davies:
  Sparse and shift-Invariant representations of music. 50-57
  
• Sue Harding, Jon P. Barker, Guy J. Brown:
  Mask estimation for missing data speech recognition based on statistics of binaural interaction. 58-67
  
• Slim Essid, Gaël Richard, Bertrand David:
  Instrument recognition in polyphonic music based on automatic taxonomies. 68-80
  
• Fabian Mörchen, Alfred Ultsch, Michael Thies, Ingo Lohken:
  Modeling timbre distance with temporal statistics from polyphonic music. 81-90
  
• Emmanuel Vincent:
  Musical source separation using time-frequency source priors. 91-98
  
• Mads Græsbøll Christensen, Søren Holdt Jensen:
  On perceptual distortion minimization and nonlinear least-squares frequency estimation. 99-109
  
• Alberto Gonzalez, Maria de Diego, Miguel Ferrer, Gema Pinero:
  Multichannel active noise equalization of interior noise. 110-122
  
• Y. Hinamoto, H. Sakai:
  Analysis of the filtered-X LMS algorithm and a related new algorithm for active control of multitonal noise. 123-130
  
• Norman H. Adams, Mark A. Bartsch, Gregory H. Wakefield:
  Note segmentation and quantization for music information retrieval. 131-141
  
• Norman D. Cook, Takashi X. Fujisawa, Kazuaki Takami:
  Evaluation of the affective valence of speech using pitch substructure. 142-151
  
• Anand D. Subramaniam, W. R. Gardner, Bhaskar D. Rao:
  Iterative joint source-channel decoding of speech spectrum parameters over an additive white Gaussian noise channel. 152-162
  
• Sriram Srinivasan, Jonas Samuelsson, W. Bastiaan Kleijn:
  Codebook driven short-term predictor parameter estimation for speech enhancement. 163-176
  
• Y. Nagata, T. Fujioka, M. Abe:
  Speech enhancement based on auto gain control. 177-190
  
• Laurent Benaroya, Frédéric Bimbot, Rémi Gribonval:
  Audio source separation with a single sensor. 191-199
  
• Kostas Kokkinakis, Asoke K. Nandi:
  Multichannel blind deconvolution for source separation in convolutive mixtures of speech. 200-212
  
• Narendra K. Gupta, Gökhan Tür, Dilek Hakkani-Tür, Srinivas Bangalore, Giuseppe Riccardi, Mazin Gilbert:
  The AT&T spoken language understanding system. 213-222
  
• B. Milner, A. James:
  Robust speech recognition over mobile and IP networks in burst-like packet loss. 223-231
  
• Ken Chen, Mark Hasegawa-Johnson, Aaron Cohen, Sarah Borys, Sung-Suk Kim, Jennifer Cole, Jeung-Yoon Choi:
  Prosody dependent speech recognition on radio news corpus of American English. 232-245
  
• Néstor Becerra Yoma, Carlos Molina, Jorge Silva, Carlos Busso:
  Modeling, estimating, and compensating low-bit rate coding distortion in speech recognition. 246-255
  
• Li Deng, Dong Yu, Alex Acero:
  A bidirectional target-filtering model of speech coarticulation and reduction: two-stage implementation for phonetic recognition. 256-265
  
• Chung-Hsien Wu, Yu-Hsien Chiu, Chi-Jiun Shia, Chun-Yu Lin:
  Automatic segmentation and identification of mixed-language speech using delta-BIC and LSA-based GMMs. 266-276
  
• Tomi Kinnunen, Evgeny Karpov, Pasi Fränti:
  Real-time speaker identification and verification. 277-288
  
• Yang Shao, DeLiang Wang:
  Model-based sequential organization in cochannel speech. 289-298
  
• Christof Faller:
  Parametric multichannel audio coding: synthesis of coherence cues. 299-310
  
• Renat Vafin, W. Bastiaan Kleijn:
  Rate-distortion optimized quantization in multistage audio coding. 311-320
  
• Antti J. Eronen, V. T. Peltonen, J. T. Tuomi, Anssi Klapuri, S. Fagerlund, T. Sorsa, G. Lorho, Jyri Huopaniemi:
  Audio-based context recognition. 321-329
  
• Wei-Ho Tsai, Hsin-Min Wang:
  Automatic singer recognition of popular music recordings via estimation and modeling of solo vocal signals. 330-341
  
• Anssi Klapuri, Antti J. Eronen, Jaakko Astola:
  Analysis of the meter of acoustic musical signals. 342-355
  
• Vaibhava Goel, Shankar Kumar, William Byrne:
  Corrections to "Segmental minimum Bayes-risk decoding for automatic speech recognition". 356-357
  

Volume 14, Number 2, March 2006

• Satoshi Nakamura, Konstantin Markov, Hiromi Nakaiwa, Gen-ichiro Kikui, Hisashi Kawai, Takatoshi Jitsuhiro, Jin-Song Zhang, Hirofumi Yamamoto, Eiichiro Sumita, Seiichi Yamamoto:
  The ATR multilingual speech-to-speech translation system. 365-376
  
• Liang Gu, Yuqing Gao, Fu-Hua Liu, Michael Picheny:
  Concept-based speech-to-speech translation using maximum entropy models for statistical natural concept generation. 377-392
  
• Yasuhiro Akiba, Kenji Imamura, Eiichiro Sumita, Hiromi Nakaiwa, Shun'ichi Yamamoto, Hiroshi G. Okuno:
  Using multiple edit distances to automatically grade outputs from Machine translation systems. 393-402
  
• Tanja Schultz, Alan W. Black, Stephan Vogel, Monika Woszczyna:
  Flexible speech translation systems. 403-411
  
• A. Davis, Sven Nordholm, Roberto Togneri:
  Statistical voice activity detection using low-variance spectrum estimation and an adaptive threshold. 412-424
  
• Li Deng, Alex Acero, Issam Bazzi:
  Tracking vocal tract resonances using a quantized nonlinear function embeddedin a temporal constraint. 425-434
  
• K. Mustafa, Ian C. Bruce:
  Robust formant tracking for continuous speech with speaker variability. 435-444
  
• Huiqun Deng, Rabab K. Ward, Michael P. Beddoes, Murray Hodgson:
  A new method for obtaining accurate estimates of vocal-tract filters and glottal waves from vowel sounds. 445-455
  
• Mike Brookes, Patrick A. Naylor, Jon Gudnason:
  A quantitative assessment of group delay methods for identifying glottal closures in voiced speech. 456-466
  
• Ran D. Zilca, Brian Kingsbury, Jiri Navratil, Ganesh N. Ramaswamy:
  Pseudo pitch synchronous analysis of speech with applications to speaker recognition. 467-478
  
• S. Gazor, R. R. Far:
  Adaptive maximum windowed likelihood multicomponent AM-FM signal decomposition. 479-491
  
• Qiang Fu, P. Murphy:
  Robust glottal source estimation based on joint source-filter model optimization. 492-501
  
• E. Fisher, Joseph Tabrikian, Shlomo Dubnov:
  Generalized likelihood ratio test for voiced-unvoiced decision in noisy speech using the harmonic model. 502-510
  
• Doroteo Torre Toledano, J. Gó Villardebo, Luis A. Hernández Gómez:
  Initialization, training, and context-dependency in HMM-based formant tracking. 511-523
  
• Anand D. Subramaniam, W. R. Gardner, Bhaskar D. Rao:
  Low-complexity source coding using Gaussian mixture models, lattice vector quantization, and recursive coding with application to speech spectrum quantization. 524-532
  
• Thomas F. Quatieri, Kevin Brady, D. Messing, Joseph P. Campbell, William M. Campbell, M. S. Brandstein, Clifford J. Weinstein, J. D. Tardelli, P. D. Gatewood:
  Exploiting nonacoustic sensors for speech encoding. 533-544
  
• Hui Dong, Jerry D. Gibson:
  Structures for SNR scalable speech coding. 545-557
  
• U. Bhaskar, K. Swaminathan:
  Low bit-rate voice compression based on frequency domain interpolative techniques. 558-576
  
• Harald Gustafsson, Ulf A. Lindgren, Ingvar Claesson:
  Low-complexity feature-mapped speech bandwidth extension. 577-588
  
• Olivier Pietquin, Thierry Dutoit:
  A probabilistic framework for dialog simulation and optimal strategy learning. 589-599
  
• Bojana Gajic, Kuldip K. Paliwal:
  Robust speech recognition in noisy environments based on subband spectral centroid histograms. 600-608
  
• Hossein Najaf-Zadeh, Peter Kabal:
  Perceptual coding of narrow-band audio signals at low rates. 609-622
  
• Ashish Aggarwal, Shankar L. Regunathan, Kenneth Rose:
  A trellis-based optimal parameter value selection for audio coding. 623-633
  
• Pongtep Angkititrakul, John H. L. Hansen:
  Advances in phone-based modeling for automatic accent classification. 634-646
  
• Chung-Hsien Wu, Chia-Hsin Hsieh:
  Multiple change-point audio segmentation and classification using an MDL-based Gaussian model. 647-657
  
• Ngwa A. Shusina, Boaz Rafaely:
  Unbiased adaptive feedback cancellation in hearing aids by closed-loop identification. 658-665
  
• Hiroshi Saruwatari, Toshiya Kawamura, Tsuyoki Nishikawa, Akinobu Lee, Kiyohiro Shikano:
  Blind source separation based on a fast-convergence algorithm combining ICA and beamforming. 666-678
  
• Ali Taylan Cemgil, Hilbert J. Kappen, D. Barber:
  A generative model for music transcription. 679-694
  
• Mitsuko Aramaki, Richard Kronland-Martinet:
  Analysis-synthesis of impact sounds by real-time dynamic filtering. 695-705
  
• Kelvin Chee-Mun Lee, Woon-Seng Gan:
  Bandwidth-efficient recursive pth-order equalization for correcting baseband distortion in parametric loudspeakers. 706-710
  
• L. E. Rees, S. J. Elliott:
  Adaptive algorithms for active sound-profiling. 711-719
  
• Muhammad Tahir Akhtar, Masahide Abe, Masayuki Kawamata:
  A new variable step size LMS algorithm-based method for improved online secondary path modeling in active noise control systems. 720-726
  
• Thomas Hain, Philip C. Woodland, Gunnar Evermann, M. J. F. Gales, Xunying Liu, G. L. Moore, Daniel Povey, Lan Wang:
  Corrections to "Automatic Transcription of Conversational Telephone Speech". 727-727
  

Volume 14, Number 3, May 2006

• S. Ramamohan, S. Dandapat:
  Sinusoidal model-based analysis and classification of stressed speech. 737-746
  
• Joon-Hyuk Chang, Nam Soo Kim:
  A new structural approach in system identification with generalized analysis-by-synthesis for robust speech coding. 747-751
  
• C. A. Rodbro, Jesper Jensen, Richard Heusdens:
  Rate-distortion optimal time-segmentation and redundancy selection for VoIP. 752-763
  
• Volodya Grancharov, Jonas Samuelsson, W. Bastiaan Kleijn:
  On causal algorithms for speech enhancement. 764-773
  
• Mingyang Wu, DeLiang Wang:
  A two-stage algorithm for one-microphone reverberant speech enhancement. 774-784
  
• Andy W. H. Khong, Patrick A. Naylor:
  Stereophonic acoustic echo cancellation employing selective-tap adaptive algorithms. 785-796
  
• Jen-Tzung Chien, Chih-Hsien Huang:
  Aggregate a posteriori linear regression adaptation. 797-807
  
• Jeih-Weih Hung, Lin-Shan Lee:
  Optimization of temporal filters for constructing robust features in speech recognition. 808-832
  
• Ji Ming:
  Noise compensation for speech recognition with arbitrary additive noise. 833-844
  
• Florian Hilger, Hermann Ney:
  Quantile based histogram equalization for noise robust large vocabulary speech recognition. 845-854
  
• Shinji Watanabe, Atsushi Sako, Atsushi Nakamura:
  Automatic determination of acoustic model topology using variational Bayesian estimation and clustering for large vocabulary continuous speech recognition. 855-872
  
• Hong-Kwang Jeff Kuo, Yuqing Gao:
  Maximum entropy direct models for speech recognition. 873-881
  
• K. C. Sim, M. J. F. Gales:
  Minimum phone error training of precision matrix models. 882-889
  
• J. Silva, S. Narayanan:
  Average divergence distance as a statistical discrimination measure for hidden Markov models. 890-906
  
• Rongqing Huang, John H. L. Hansen:
  Advances in unsupervised audio classification and segmentation for the broadcast news and NGSW corpora. 907-919
  
• Nima Mesgarani, M. Slaney, Shihab A. Shamma:
  Discrimination of speech from nonspeech based on multiscale spectro-temporal Modulations. 920-930
  
• R. Sant'Ana, Rosangela Coelho, Abraham Alcaim:
  Text-independent speaker recognition based on the Hurst parameter and the multidimensional fractional Brownian motion model. 931-940
  
• Enrique Vidal, Francisco Casacuberta, Luis Rodríguez, Jorge Civera, Carlos D. Martínez-Hinarejos:
  Computer-assisted translation using speech recognition. 941-951
  
• Athanasios Mouchtaris, Jan Van der Spiegel, Paul Mueller:
  Nonparallel training for voice conversion based on a parameter adaptation approach. 952-963
  
• J. Mullen, David M. Howard, Damian T. Murphy:
  Waveguide physical modeling of vocal tract acoustics: flexible formant bandwidth control from increased model dimensionality. 964-971
  
• K. Sreenivasa Rao, B. Yegnanarayana:
  Prosody modification using instants of significant excitation. 972-980
  
• Ki-Seung Lee:
  MLP-based phone boundary refining for a TTS database. 981-989
  
• Jerome R. Bellegarda:
  A global, boundary-centric framework for unit selection text-to-speech synthesis. 990-997
  
• Cheng-Han Yang, Hsueh-Ming Hang:
  Cascaded trellis-based rate-distortion control algorithm for MPEG-4 advanced audio coding. 998-1007
  
• B. Supper, T. Brookes, F. Rumsey:
  An auditory onset detection algorithm for improved automatic source localization. 1008-1017
  
• Woon-Seng Gan, Jun Yang, K.-S. Tan, Meng Hwa Er:
  A digital beamsteerer for difference frequency in a parametric array. 1018-1025
  
• Rui Cai, Lie Lu, Alan Hanjalic, HongJiang Zhang, Lian-Hong Cai:
  A flexible framework for key audio effects detection and auditory context inference. 1026-1039
  
• Dimitrios K. Fragoulis, Constantin Papaodysseus, Mihalis Exarhos, George Roussopoulos, Thanasis Panagopoulos, D. Kamarotos:
  Automated classification of piano-guitar notes. 1040-1050
  
• Harald Viste, Gianpaolo Evangelista:
  A method for separation of overlapping partials based on similarity of temporal envelopes in multichannel mixtures. 1051-1061
  
• Serkan Kiranyaz, Ahmad Farooq Qureshi, Moncef Gabbouj:
  A generic audio classification and segmentation approach for multimedia indexing and retrieval. 1062-1081
  
• Timothy J. Hazen:
  Visual model structures and synchrony constraints for audio-visual speech recognition. 1082-1089
  

Volume 14, Number 4, July 2006

• John F. Pitrelli, Raimo Bakis, Ellen Eide, Raul Fernandez, Wael Hamza, M. A. Picheny:
  The IBM expressive text-to-speech synthesis system for American English. 1099-1108
  
• Chung-Hsien Wu, Chi-Chun Hsia, Te-Hsien Liu, Jhing-Fa Wang:
  Voice conversion using duration-embedded bi-HMMs for expressive speech synthesis. 1109-1116
  
• Eva Navas, Inma Hernáez, Iker Luengo:
  An objective and subjective study of the role of semantics and prosodic features in building corpora for emotional TTS. 1117-1127
  
• M. Schroder:
  Expressing degree of activation in synthetic speech. 1128-1136
  
• Mariët Theune, K. Meijs, Dirk Heylen, R. Ordelman:
  Generating expressive speech for storytelling applications. 1137-1144
  
• Jianhua Tao, Yongguo Kang, Aijun Li:
  Prosody conversion from neutral speech to emotional speech. 1145-1154
  
• Wentao Gu, Keikichi Hirose, Hiroya Fujisaki:
  Modeling the effects of emphasis and question on fundamental frequency contours of Cantonese utterances. 1155-1170
  
• N. Campbell:
  Conversational speech synthesis and the need for some laughter. 1171-1178
  
• Taishih Chi, Shihab A. Shamma:
  Spectrum restoration from multiscale auditory phase singularities by generalized projections. 1179-1192
  
• A. Watanabe, T. Sakata:
  Reliable methods for estimating relative vocal tract lengths from formant trajectories of common words. 1193-1204
  
• W. C. Chu:
  Embedded quantization of line spectral frequencies using a multistage tree-structured vector quantizer. 1205-1217
  
• Jingdong Chen, Jacob Benesty, Yiteng Arden Huang, Simon Doclo:
  New insights into the noise reduction Wiener filter. 1218-1234
  
• Yunxin Zhao, Rong Hu, Xiaolong Li:
  Speedup convergence and reduce noise for enhanced speech separation and recognition. 1235-1244
  
• Jen-Tzung Chien, Bo-Cheng Chen:
  A new independent component analysis for speech recognition and separation. 1245-1254
  
• Satya Dharanipragada, K. Visweswariah:
  Gaussian mixture models with covariances or precisions in shared multiple subspaces. 1255-1266
  
• Brian Kan-Wing Mak, Roger Wend-Huu Hsiao, Simon Ka-Lung Ho, James T. Kwok:
  Embedded kernel eigenvoice speaker adaptation and its implication to reference speaker weighting. 1267-1280
  
• Diamantino Caseiro, Isabel Trancoso:
  A specialized on-the-fly algorithm for lexicon and language model composition. 1281-1291
  
• Toshihiko Abe, Masaaki Honda:
  Sinusoidal model based on instantaneous frequency attractors. 1292-1300
  
• Hui Ye, S. Young:
  Quality-enhanced voice morphing using maximum likelihood transformations. 1301-1312
  
• Ashish Aggarwal, Shankar L. Regunathan, Kenneth Rose:
  Efficient bit-rate scalability for weighted squared error optimization in audio coding. 1313-1327
  
• Olivier Derrien, Pierre Duhamel, Maurice Charbit, Gaël Richard:
  A new quantization optimization algorithm for the MPEG advanced audio coder using a statistical subband model of the quantization noise. 1328-1339
  
• M. G. Christensen, Steven van de Par:
  Efficient parametric coding of transients. 1340-1351
  
• Rongshan Yu, Susanto Rahardja, Lin Xiao, Chi Chung Ko:
  A fine granular scalable to lossless audio coder. 1352-1363
  
• T. Umayahara, Haruhide Hokari, Shoji Shimada:
  Stereo width control using interpolation and extrapolation of time-frequency representation. 1364-1377
  
• F. Talantzis, Darren B. Ward, Patrick A. Naylor:
  Performance analysis of dynamic acoustic source separation in reverberant rooms. 1378-1390
  
• Paulo A. A. Esquef, Luiz W. P. Biscainho:
  An efficient model-based multirate method for reconstruction of audio signals across long gaps. 1391-1400
  
• Slim Essid, Gaël Richard, Bertrand David:
  Musical instrument recognition by pairwise classification strategies. 1401-1412
  
• Ixone Arroabarren, Xavier Rodet, Alfonso Carlosena:
  On the measurement of the instantaneous frequency and amplitude of partials in vocal vibrato. 1413-1421
  
• Ixone Arroabarren, Alfonso Carlosena:
  Inverse filtering in singing voice: a critical analysis. 1422-1431
  
• Saman S. Abeysekera, Kabi Prakash Padhi:
  An investigation of window effects on the frequency estimation using the phase vocoder. 1432-1439
  
• Axel Röbel:
  Adaptive additive modeling with continuous parameter trajectories. 1440-1453
  
• Crispin H. V. Cooper, Damian T. Murphy, David M. Howard, Alexander Tyrrell:
  Singing synthesis with an evolved physical model. 1454-1461
  
• Emmanuel Vincent, Rémi Gribonval, Cédric Févotte:
  Performance measurement in blind audio source separation. 1462-1469
  
• Panayiotis G. Georgiou, Chris Kyriakakis:
  Robust maximum likelihood source localization: the case for sub-Gaussian versus Gaussian. 1470-1480
  

Volume 14, Number 5, September 2006

• Li Deng, Dong Yu, Alex Acero:
  Structured speech modeling. 1492-1504
  
• Claude Barras, Xuan Zhu, Sylvain Meignier, Jean-Luc Gauvain:
  Multistage speaker diarization of broadcast news. 1505-1512
  
• M. J. F. Gales, Do Yeong Kim, Philip C. Woodland, Ho Yin Chan, David Mrva, R. Sinha, S. E. Tranter:
  Progress in the CU-HTK broadcast news transcription system. 1513-1525
  
• Yang Liu, Elizabeth Shriberg, Andreas Stolcke, Dustin Hillard, Mari Ostendorf, Mary P. Harper:
  Enriching speech recognition with automatic detection of sentence boundaries and disfluencies. 1526-1540
  
• Spyridon Matsoukas, Jean-Luc Gauvain, Gilles Adda, Thomas Colthurst, Chia-Lin Kao, Owen Kimball, Lori Lamel, Fabrice Lefevre, Jeff Z. Ma, John Makhoul, Long Nguyen, Rohit Prasad, Richard M. Schwartz, Holger Schwenk, Bing Xiang:
  Advances in transcription of broadcast news and conversational telephone speech within the combined EARS BBN/LIMSI system. 1541-1556
  
• S. E. Tranter, Douglas A. Reynolds:
  An overview of automatic speaker diarization systems. 1557-1565
  
• Matthew Lease, Mark Johnson, Eugene Charniak:
  Recognizing disfluencies in conversational speech. 1566-1573
  
• Jui-Feng Yeh, Chung-Hsien Wu:
  Edit disfluency detection and correction using a cleanup language model and an alignment model. 1574-1583
  
• Hui Jiang, Xinwei Li, Chaojun Liu:
  Large margin hidden Markov models for speech recognition. 1584-1595
  
• Stanley F. Chen, Brian Kingsbury, Lidia Mangu, Daniel Povey, George Saon, Hagen Soltau, Geoffrey Zweig:
  Advances in speech transcription at IBM under the DARPA EARS program. 1596-1608
  
• C. A. Rodbro, Manohar N. Murthi, Søren Vang Andersen, Søren Holdt Jensen:
  Hidden Markov model-based packet loss concealment for voice over IP. 1609-1623
  
• Farshad Lahouti, A. R. Fazel, A. H. Safavi-Naeini, Amir K. Khandani:
  Single and double frame coding of speech LPC parameters using a lattice-based quantization scheme. 1624-1632
  
• Herbert Buchner, Jacob Benesty, Tomas Gänsler, Walter Kellermann:
  Robust extended multidelay filter and double-talk detector for acoustic echo cancellation. 1633-1644
  
• Marcin Kuropatwinski, W. Bastiaan Kleijn:
  Estimation of the short-term predictor parameters of speech under noisy conditions. 1645-1655
  
• Rile Hu, Chengqing Zong, Bo Xu:
  An approach to automatic acquisition of translation templates based on phrase structure extraction and alignment. 1656-1663
  
• Alon Lavie, Fabio Pianesi, Lori S. Levin:
  The NESPOLE! System for multilingual speech communication over the Internet. 1664-1673
  
• Gen-ichiro Kikui, Seiichi Yamamoto, Toshiyuki Takezawa, Eiichiro Sumita:
  Comparative study on corpora for speech translation. 1674-1682
  
• Xiao Li, Jonathan Malkin, Jeff A. Bilmes:
  A high-speed, low-resource ASR back-end based on custom arithmetic. 1683-1693
  
• Kai Yu, M. J. F. Gales:
  Discriminative cluster adaptive training. 1694-1703
  
• Chak-Fai Li, Man-Hung Siu, Jeff Siu-Kei Au-Yeung:
  Recursive likelihood evaluation and fast search algorithm for polynomial segment model with application to speech recognition. 1704-1718
  
• Jen-Tzung Chien:
  Association pattern language modeling. 1719-1728
  
• Andreas Stolcke, Barry Y. Chen, Horacio Franco, Venkata Ramana Rao Gadde, Martin Graciarena, Mei-Yuh Hwang, Katrin Kirchhoff, Arindam Mandal, Nelson Morgan, Xin Lei, Tim Ng, Mari Ostendorf, M. Kemal Sönmez, Anand Venkataraman, Dimitra Vergyri, Wen Wang, Jing Zheng, Qifeng Zhu:
  Recent innovations in speech-to-text transcription at SRI-ICSI-UW. 1729-1744
  
• N. Duta, Richard M. Schwartz, John Makhoul:
  Analysis of the errors produced by the 2004 BBN speech recognition system in the DARPA EARS evaluations. 1745-1753
  
• S. Mathur, B. H. Story, J. J. Rodriguez:
  Vocal-tract modeling: fractional elongation of segment lengths in a waveguide model with half-sample delays. 1754-1762
  
• Jithendra Vepa, Simon King:
  Subjective evaluation of join cost and smoothing methods for unit selection speech synthesis. 1763-1771
  
• Cléo Baras, Nicolas Moreau, P. Dymarski:
  Controlling the inaudibility and maximizing the robustness in an audio annotation watermarking system. 1772-1782
  
• Masataka Goto:
  A chorus section detection method for musical audio signals and its application to a music listening station. 1783-1794
  
• Aggelos Pikrakis, Sergios Theodoridis, Dimitris Kamarotos:
  Classification of musical patterns using variable duration hidden Markov models. 1795-1807
  
• Laurent Daudet:
  Sparse and structured decompositions of signals with the molecular matching pursuit. 1808-1816
  
• V. Verfaille, Udo Zölzer, Dabiel Arfib:
  Adaptive digital audio effects (a-DAFx): a new class of sound transformations. 1817-1831
  
• Fabien Gouyon, Anssi Klapuri, Simon Dixon, M. Alonso, G. Tzanetakis, C. Uhle, Pedro Cano:
  An experimental comparison of audio tempo induction algorithms. 1832-1844
  
• M. R. Every, J. E. Szymanski:
  Separation of synchronous pitched notes by spectral filtering of harmonics. 1845-1856
  
• S. M. Kuo, A. B. Puvvala:
  Effects of frequency separation in periodic active noise control systems. 1857-1866
  
• Guangji Shi, Maryam Modir Shanechi, Parham Aarabi:
  On the importance of phase in human speech recognition. 1867-1874
  
• D. P. Das, S. R. Mohapatra, A. Routray, T. K. Basu:
  Filtered-s LMS algorithm for multichannel active control of nonlinear noise processes. 1875-1880
  

Volume 14, Number 6, November 2006

• A. W. Rix, J. G. Beerends, D.-S. Kim, P. Kroon, O. Ghitza:
  Objective Assessment of Speech and Audio Quality - Technology and Applications. 1890-1901
  
• Rainer Huber, Birger Kollmeier:
  PEMO-Q - A New Method for Objective Audio Quality Assessment Using a Model of Auditory Perception. 1902-1911
  
• Abhijit Karmakar, Arun Kumar, R. K. Patney:
  A Multiresolution Model of Auditory Excitation Pattern and Its Application to Objective Evaluation of Perceived Speech Quality. 1912-1923
  
• Ludovic Malfait, J. Berger, M. Kastner:
  P.563 - The ITU-T Standard for Single-Ended Speech Quality Assessment. 1924-1934
  
• Tiago H. Falk, Wai-Yip Chan:
  Single-Ended Speech Quality Measurement Using Machine Learning Methods. 1935-1947
  
• Volodya Grancharov, David Y. Zhao, Jonas Lindblom, W. Bastiaan Kleijn:
  Low-Complexity, Nonintrusive Speech Quality Assessment. 1948-1956
  
• Alexander Raake:
  Short- and Long-Term Packet Loss Behavior: Towards Speech Quality Prediction for Arbitrary Loss Distributions. 1957-1968
  
• Sebastian Möller, Alexander Raake, Nobuhiko Kitawaki, Akira Takahashi, Marcel Wältermann:
  Impairment Factor Framework for Wide-Band Speech Codecs. 1969-1976
  
• S. R. Broom:
  VoIP Quality Assessment: Taking Account of the Edge-Device. 1977-1983
  
• Akira Takahashi, Atsuko Kurashima, Hideaki Yoshino:
  Objective Assessment Methodology for Estimating Conversational Quality in VoIP. 1984-1993
  
• S. George, S. Zielinski, F. Rumsey:
  Feature Extraction for the Prediction of Multichannel Spatial Audio Fidelity. 1994-2005
  
• Takeshi Yamada, Masakazu Kumakura, Nobuhiko Kitawaki:
  Performance Estimation of Speech Recognition System Under Noise Conditions Using Objective Quality Measures and Artificial Voice. 2006-2013
  
• P. Li, Y. Guan, B. Xu, W. Liu:
  Monaural Speech Separation Based on Computational Auditory Scene Analysis and Objective Quality Assessment of Speech. 2014-2023
  
• Georgios Evangelopoulos, Petros Maragos:
  Multiband Modulation Energy Tracking for Noisy Speech Detection. 2024-2038
  
• Mohamed Deriche, Daryl Ning:
  A Novel Audio Coding Scheme Using Warped Linear Prediction Model and the Discrete Wavelet Transform. 2039-2048
  
• John H. L. Hansen, V. Radhakrishnan, Kathryn Hoberg Arehart:
  Speech Enhancement Based on Generalized Minimum Mean Square Error Estimators and Masking Properties of the Auditory System. 2049-2063
  
• Richard C. Hendriks, Richard Heusdens, Jesper Jensen:
  Adaptive Time Segmentation for Improved Speech Enhancement. 2064-2074
  
• Tiemin Mei, Jiangtao Xi, Fuliang Yin, Alfred Mertins, Joe F. Chicharo:
  Blind Source Separation Based on Time-Domain Optimization of a Frequency-Domain Independence Criterion. 2075-2085
  
• Y. Nagata, K. Mitsubori, T. Kagi, T. Fujioka, M. Abe:
  Fast Implementation of KLT-Based Speech Enhancement Using Vector Quantization. 2086-2097
  
• Cyril Plapous, Claude Marro, Pascal Scalart:
  Improved Signal-to-Noise Ratio Estimation for Speech Enhancement. 2098-2108
  
• Michael L. Seltzer, Richard M. Stern:
  Subband Likelihood-Maximizing Beamforming for Speech Recognition in Reverberant Environments. 2109-2121
  
• Man-Hung Siu, Arthur Chan:
  A Robust Viterbi Algorithm Against Impulsive Noise With Application to Speech Recognition. 2122-2133
  
• Y. Tian, J.-L. Zhou, H. Lin, H. Jiang:
  Tree-Based Covariance Modeling of Hidden Markov Models. 2134-2146
  
• Jian Wu, Qiang Huo:
  An Environment-Compensated Minimum Classification Error Training Approach Based on Stochastic Vector Mapping. 2147-2155
  
• K. W. Wilson, Trevor Darrell:
  Learning a Precedence Effect-Like Weighting Function for the Generalized Cross-Correlation Framework. 2156-2164
  
• Hiroshi Sawada, Shoko Araki, Ryo Mukai, Shoji Makino:
  Blind Extraction of Dominant Target Sources Using ICA and Time-Frequency Masking. 2165-2173
  
• Cédric Févotte, Simon J. Godsill:
  A Bayesian Approach for Blind Separation of Sparse Sources. 2174-2188
  
• Yegui Xiao, Liying Ma, Khashayar Khorasani, Akira Ikuta:
  A New Robust Narrowband Active Noise Control System in the Presence of Frequency Mismatch. 2189-2200
  
• Y. Yokotani, Ralf Geiger, G. D. T. Schuller, Soontorn Oraintara, K. R. Rao:
  Lossless Audio Coding Using the IntMDCT and Rounding Error Shaping. 2201-2211
  
• Toshio Irino, Roy D. Patterson, Hideki Kawahara:
  Speech Segregation Using an Auditory Vocoder With Event-Synchronous Enhancements. 2212-2221
  
• Toshio Irino, Roy D. Patterson:
  A Dynamic Compressive Gammachirp Auditory Filterbank. 2222-2232
  
• Wei-Chen Chang, Alvin Wen-Yu Su:
  A Multichannel Recurrent Network Analysis/Synthesis Model for Coupled-String Instruments. 2233-2241
  
• Juan Pablo Bello, Laurent Daudet, Mark B. Sandler:
  Automatic Piano Transcription Using Frequency and Time-Domain Information. 2242-2251
  
• Panu Somervuo, Aki Härmä, S. Fagerlund:
  Parametric Representations of Bird Sounds for Automatic Species Recognition. 2252-2263
  
• Ying Li, Chitra Dorai:
  Instructional Video Content Analysis Using Audio Information. 2264-2274
  

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