IEEE Transactions on Audio, Speech & Language Processing, Volume 20

Volume 20, Number 1, 2012

• Helen Meng:
  Farewell Editorial. 1
  
• Li Deng:
  Inaugural Editorial: Riding the Tidal Wave of Human-Centric Information Processing - Innovate, Outreach, Collaborate, Connect, Expand, and Win. 2-3
  
• Dong Yu, Geoffrey E. Hinton, Nelson Morgan, Jen-Tzung Chien, Shigeki Sagayama:
  Introduction to the Special Section on Deep Learning for Speech and Language Processing. 4-6
  
• Nelson Morgan:
  Deep and Wide: Multiple Layers in Automatic Speech Recognition. 7-13
  
• Abdel-rahman Mohamed, George E. Dahl, Geoffrey E. Hinton:
  Acoustic Modeling Using Deep Belief Networks. 14-22
  
• Garimella S. V. S. Sivaram, Hynek Hermansky:
  Sparse Multilayer Perceptron for Phoneme Recognition. 23-29
  
• George E. Dahl, Dong Yu, Li Deng, Alex Acero:
  Context-Dependent Pre-Trained Deep Neural Networks for Large-Vocabulary Speech Recognition. 30-42
  
• George Saon, Jen-Tzung Chien:
  Bayesian Sensing Hidden Markov Models. 43-54
  
• Jen-Tzung Chien, Chuang-Hua Chueh:
  Topic-Based Hierarchical Segmentation. 55-66
  
• I. Yücel Özbek, Mark Hasegawa-Johnson, Mübeccel Demirekler:
  On Improving Dynamic State Space Approaches to Articulatory Inversion With MAP-Based Parameter Estimation. 67-81
  
• Mark R. P. Thomas, Jon Gudnason, Patrick A. Naylor:
  Estimation of Glottal Closing and Opening Instants in Voiced Speech Using the YAGA Algorithm. 82-91
  
• Jesper Jensen, Richard C. Hendriks:
  Spectral Magnitude Minimum Mean-Square Error Estimation Using Binary and Continuous Gain Functions. 92-102
  
• Hen-Geul Yeh, Carlos Rangel Ruiz:
  Fixed-Point Implementation of Cascaded Forward-Backward Adaptive Predictors. 103-107
  
• Tobias May, Steven van de Par, Armin Kohlrausch:
  Noise-Robust Speaker Recognition Combining Missing Data Techniques and Universal Background Modeling. 108-121
  
• Alberto Carini, Stefania Cecchi, Francesco Piazza, Ivan Omiciuolo, Giovanni L. Sicuranza:
  Multiple Position Room Response Equalization in Frequency Domain. 122-135
  
• Iman S. Mossavat, Petko N. Petkov, W. Bastiaan Kleijn, Oliver Amft:
  A Hierarchical Bayesian Approach to Modeling Heterogeneity in Speech Quality Assessment. 136-146
  
• Thomas Ulrich Christiansen, Steven Greenberg:
  Perceptual Confusions Among Consonants, Revisited - Cross-Spectral Integration of Phonetic-Feature Information and Consonant Recognition. 147-161
  
• Enzo De Sena, Hüseyin Hacihabiboglu, Zoran Cvetkovic:
  On the Design and Implementation of Higher Order Differential Microphones. 162-174
  
• Ted S. Wada, Biing-Hwang Juang:
  Enhancement of Residual Echo for Robust Acoustic Echo Cancellation. 175-189
  
• Adam M. Stark, Mark D. Plumbley:
  Performance Following: Real-Time Prediction of Musical Sequences Without a Score. 190-199
  
• Matthias Mauch, Hiromasa Fujihara, Masataka Goto:
  Integrating Additional Chord Information Into HMM-Based Lyrics-to-Audio Alignment. 200-210
  
• Berlin Chen, Shih-Hsiang Lin:
  A Risk-Aware Modeling Framework for Speech Summarization. 211-222
  
• Richard C. Hendriks, Timo Gerkmann:
  Noise Correlation Matrix Estimation for Multi-Microphone Speech Enhancement. 223-233
  
• Giovanni L. Sicuranza, Alberto Carini:
  On the BIBO Stability Condition of Adaptive Recursive FLANN Filters With Application to Nonlinear Active Noise Control. 234-245
  
• Francesco Nesta, Maurizio Omologo:
  Generalized State Coherence Transform for Multidimensional TDOA Estimation of Multiple Sources. 246-260
  
• Yasmín Montenegro M., José Carlos M. Bermudez:
  Transient Mean-Square Analysis of Prediction Error Method-Based Adaptive Feedback Cancellation in Hearing Aids. 261-275
  
• Lei Xie, Lilei Zheng, Zihan Liu, Yanning Zhang:
  Laplacian Eigenmaps for Automatic Story Segmentation of Broadcast News. 276-289
  
• Norberto Degara, Enrique Argones-Rúa, Antonio Pena, Soledad Torres-Guijarro, Matthew E. P. Davies, Mark D. Plumbley:
  Reliability-Informed Beat Tracking of Musical Signals. 290-301
  
• Jen-Tzung Chien, Hsin-Lung Hsieh:
  Convex Divergence ICA for Blind Source Separation. 302-313
  
• Han-Gil Moon:
  A Low-Complexity Design for an MP3 Multi-Channel Audio Decoding System. 314-321
  
• Celia Shahnaz, Wei-Ping Zhu, M. Omair Ahmad:
  Pitch Estimation Based on a Harmonic Sinusoidal Autocorrelation Model and a Time-Domain Matching Scheme. 322-335
  
• Claudio Garretón, Néstor Becerra Yoma:
  Telephone Channel Compensation in Speaker Verification Using a Polynomial Approximation in the Log-Filter-Bank Energy Domain. 336-341
  
• Vishweshwara Rao, Pradeep Gaddipati, Preeti Rao:
  Signal-Driven Window-Length Adaptation for Sinusoid Detection in Polyphonic Music. 342-348
  

Volume 20, Number 2, February 2012

• Xavier Anguera Miró, Simon Bozonnet, Nicholas W. D. Evans, Corinne Fredouille, Gerald Friedland, Oriol Vinyals:
  Speaker Diarization: A Review of Recent Research. 356-370
  
• Gerald Friedland, Adam Janin, David Imseng, Xavier Anguera Miró, Luke R. Gottlieb, Marijn Huijbregts, Mary Tai Knox, Oriol Vinyals:
  The ICSI RT-09 Speaker Diarization System. 371-381
  
• Nicholas W. D. Evans, Simon Bozonnet, Dong Wang, Corinne Fredouille, Raphaël Troncy:
  A Comparative Study of Bottom-Up and Top-Down Approaches to Speaker Diarization. 382-392
  
• Marijn Huijbregts, David A. van Leeuwen, Chuck Wooters:
  Speaker Diarization Error Analysis Using Oracle Components. 393-403
  
• Marijn Huijbregts, David A. van Leeuwen:
  Large-Scale Speaker Diarization for Long Recordings and Small Collections. 404-413
  
• Oshry Ben-Harush, Itshak Lapidot, Hugo Guterman:
  Initialization of Iterative-Based Speaker Diarization Systems for Telephone Conversations. 414-425
  
• José Manuel Pardo, Roberto Barra-Chicote, Rubén San Segundo, Ricardo de Córdoba, Beatriz Martínez-González:
  Speaker Diarization Features: The UPM Contribution to the RT09 Evaluation. 426-435
  
• Martin Zelenák, Carlos Segura, Jordi Luque, Javier Hernando:
  Simultaneous Speech Detection With Spatial Features for Speaker Diarization. 436-446
  
• Katsuhiko Ishiguro, Takeshi Yamada, Shoko Araki, Tomohiro Nakatani, Hiroshi Sawada:
  Probabilistic Speaker Diarization With Bag-of-Words Representations of Speaker Angle Information. 447-460
  
• Tin Lay Nwe, Hanwu Sun, Bin Ma, Haizhou Li:
  Speaker Clustering and Cluster Purification Methods for RT07 and RT09 Evaluation Meeting Data. 461-473
  
• Fernando Batista, Helena Moniz, Isabel Trancoso, Nuno J. Mamede:
  Bilingual Experiments on Automatic Recovery of Capitalization and Punctuation of Automatic Speech Transcripts. 474-485
  
• Thomas Hain, Lukás Burget, John Dines, Philip N. Garner, Frantisek Grézl, Asmaa El Hannani, Marijn Huijbregts, Martin Karafiát, Mike Lincoln, Vincent Wan:
  Transcribing Meetings With the AMIDA Systems. 486-498
  
• Takaaki Hori, Shoko Araki, Takuya Yoshioka, Masakiyo Fujimoto, Shinji Watanabe, Takanobu Oba, Atsunori Ogawa, Kazuhiro Otsuka, Dan Mikami, Keisuke Kinoshita, Tomohiro Nakatani, Atsushi Nakamura, Junji Yamato:
  Low-Latency Real-Time Meeting Recognition and Understanding Using Distant Microphones and Omni-Directional Camera. 499-513
  
• Joan Serrà, Holger Kantz, Xavier Serra, Ralph G. Andrzejak:
  Predictability of Music Descriptor Time Series and its Application to Cover Song Detection. 514-525
  
• Marco Dinarelli, Alessandro Moschitti, Giuseppe Riccardi:
  Discriminative Reranking for Spoken Language Understanding. 526-539
  
• Ebru Arisoy, Murat Saraclar, Brian Roark, Izhak Shafran:
  Discriminative Language Modeling With Linguistic and Statistically Derived Features. 540-550
  
• Björn Hoffmeister, Georg Heigold, David Rybach, Ralf Schlüter, Hermann Ney:
  WFST Enabled Solutions to ASR Problems: Beyond HMM Decoding. 551-564
  
• Alberto Sanchís, Alfons Juan, Enrique Vidal:
  A Word-Based Naïve Bayes Classifier for Confidence Estimation in Speech Recognition. 565-574
  
• Wen Zhang, Mengqiu Zhang, Rodney A. Kennedy, Thushara D. Abhayapala:
  On High-Resolution Head-Related Transfer Function Measurements: An Efficient Sampling Scheme. 575-584
  
• Sungrack Yun, Chang D. Yoo:
  Loss-Scaled Large-Margin Gaussian Mixture Models for Speech Emotion Classification. 585-598
  
• Nima Yousefian, Philipos C. Loizou:
  A Dual-Microphone Speech Enhancement Algorithm Based on the Coherence Function. 599-609
  
• Laura E. Boucheron, Phillip L. De Leon, Steven Sandoval:
  Low Bit-Rate Speech Coding Through Quantization of Mel-Frequency Cepstral Coefficients. 610-619
  
• Nam Soo Kim, Tae Gyoon Kang, Shin Jae Kang, Chang Woo Han, Doo Hwa Hong:
  Speech Feature Mapping Based on Switching Linear Dynamic System. 620-631
  
• Yi-Cheng Pan, Hung-yi Lee, Lin-Shan Lee:
  Interactive Spoken Document Retrieval With Suggested Key Terms Ranked by a Markov Decision Process. 632-645
  
• Jake Gunther:
  Learning Echo Paths During Continuous Double-Talk Using Semi-Blind Source Separation. 646-660
  
• Meng Yu, Wenye Ma, Jack Xin, Stanley Osher:
  Multi-Channel l₁ Regularized Convex Speech Enhancement Model and Fast Computation by the Split Bregman Method. 661-675
  
• Hüseyin Hacihabiboglu, Zoran Cvetkovic:
  Multichannel Dereverberation Theorems and Robustness Issues. 676-689
  
• Laura Romoli, Stefania Cecchi, Paolo Peretti, Francesco Piazza:
  A Mixed Decorrelation Approach for Stereo Acoustic Echo Cancellation Based on the Estimation of the Fundamental Frequency. 690-698
  
• Jacob Benesty, Mehrez Souden, Yiteng Huang:
  A Perspective on Differential Microphone Arrays in the Context of Noise Reduction. 699-704
  
• Frédéric Mustière, Martin Bouchard, Miodrag Bolic:
  All-Pole Modeling of Discrete Spectral Powers: A Unified Approach. 705-708
  
• Takayuki Arai, Nao Hodoshima, Keiichi Yasu:
  Errata to "Using Steady-State Suppression to Improve Speech Intelligibility in Reverberant Environments for Elderly Listeners". 709
  

Volume 20, Number 3, March 2012

• Kazuyoshi Yoshii, Masataka Goto:
  A Nonparametric Bayesian Multipitch Analyzer Based on Infinite Latent Harmonic Allocation. 717-730
  
• Siddika Parlak, Murat Saraclar:
  Performance Analysis and Improvement of Turkish Broadcast News Retrieval. 731-741
  
• Haohai Sun, Shefeng Yan, U. Peter Svensson:
  Optimal Higher Order Ambisonics Encoding With Predefined Constraints. 742-754
  
• Mitchell McLaren, David A. van Leeuwen:
  Source-Normalized LDA for Robust Speaker Recognition Using i-Vectors From Multiple Speech Sources. 755-766
  
• Elias K. Kokkinis, Joshua D. Reiss, John Mourjopoulos:
  A Wiener Filter Approach to Microphone Leakage Reduction in Close-Microphone Applications. 767-779
  
• Qiang Fu, Yong Zhao, Biing-Hwang Juang:
  Automatic Speech Recognition Based on Non-Uniform Error Criteria. 780-793
  
• Heiga Zen, Mark J. F. Gales, Yoshihiko Nankaku, Keiichi Tokuda:
  Product of Experts for Statistical Parametric Speech Synthesis. 794-805
  
• Elina Helander, Hanna Silén, Tuomas Virtanen, Moncef Gabbouj:
  Voice Conversion Using Dynamic Kernel Partial Least Squares Regression. 806-817
  
• Ning Ma, Jon Barker, Heidi Christensen, Phil Green:
  Combining Speech Fragment Decoding and Adaptive Noise Floor Modeling. 818-827
  
• Liang-Che Sun, Lin-Shan Lee:
  Modulation Spectrum Equalization for Improved Robust Speech Recognition. 828-843
  
• Matija Marolt:
  Automatic Transcription of Bell Chiming Recordings. 844-853
  
• Emanuël Anco Peter Habets, Jacob Benesty, Patrick A. Naylor:
  A Speech Distortion and Interference Rejection Constraint Beamformer. 854-867
  
• Yousheng Chen, Qin Gong:
  A Normalized Beamforming Algorithm for Broadband Speech Using a Continuous Interleaved Sampling Strategy. 868-874
  
• Sabato Marco Siniscalchi, Dau-Cheng Lyu, Torbjørn Svendsen, Chin-Hui Lee:
  Experiments on Cross-Language Attribute Detection and Phone Recognition With Minimal Target-Specific Training Data. 875-887
  
• Xiang Lin, Andy W. H. Khong, Patrick A. Naylor:
  A Forced Spectral Diversity Algorithm for Speech Dereverberation in the Presence of Near-Common Zeros. 888-899
  
• Yu-Hsiang Bosco Chiu, Bhiksha Raj, Richard M. Stern:
  Learning-Based Auditory Encoding for Robust Speech Recognition. 900-914
  
• Amir Adler, Valentin Emiya, Maria G. Jafari, Michael Elad, Rémi Gribonval, Mark D. Plumbley:
  Audio Inpainting. 922-932
  
• Ana M. Barbancho, Anssi Klapuri, Lorenzo J. Tardón, Isabel Barbancho:
  Automatic Transcription of Guitar Chords and Fingering From Audio. 915-921
  
• Wei Chu, Abeer Alwan:
  SAFE: A Statistical Approach to F0 Estimation Under Clean and Noisy Conditions. 933-944
  
• Ashish Panda, Thambipillai Srikanthan:
  Psychoacoustic Model Compensation for Robust Speaker Verification in Environmental Noise. 945-953
  
• Emanuel A. P. Habets, Jacob Benesty:
  A Perspective on Frequency-Domain Beamformers in Room Acoustics. 947-960
  
• Thomas Drugman, Thierry Dutoit:
  The Deterministic Plus Stochastic Model of the Residual Signal and Its Applications. 968-981
  
• S. C. Chan, Y. Chu:
  Performance Analysis and Design of FxLMS Algorithm in Broadband ANC System With Online Secondary-Path Modeling. 982-993
  
• Thomas Drugman, Mark R. P. Thomas, Jon Gudnason, Patrick A. Naylor, Thierry Dutoit:
  Detection of Glottal Closure Instants From Speech Signals: A Quantitative Review. 994-1006
  
• Alfonso Perez Carrillo, Jordi Bonada, Esteban Maestre, Enric Guaus, Merlijn Blaauw:
  Performance Control Driven Violin Timbre Model Based on Neural Networks. 1007-1021
  
• Ravi K. Chivukula, Yuriy A. Reznik, Venkat Devarajan, Mythreya Jayendra-Lakshman:
  Fast Algorithms for Low-Delay SBR Filterbanks in MPEG-4 AAC-ELD. 1022-1031
  
• Xianyu Zhao, Yuan Dong:
  Variational Bayesian Joint Factor Analysis Models for Speaker Verification. 1032-1042
  
• Ashutosh Pandey, V. John Mathews:
  Adaptive Gain Processing With Offending Frequency Suppression for Digital Hearing Aids. 1043-1055
  
• Tamar Shoham, David Malah, Slava Shechtman:
  Quality Preserving Compression of a Concatenative Text-To-Speech Acoustic Database. 1056-1068
  
• Vladimir Despotovic, Norbert Goertz, Zoran Peric:
  Nonlinear Long-Term Prediction of Speech Based on Truncated Volterra Series. 1069-1073
  
• Siow Yong Low, Svetha Venkatesh, Sven Nordholm:
  A Spectral Slit Approach to Doubletalk Detection. 1074-1080
  

Volume 20, Number 4, May 2012

• S. Nakagawa, L. Wang, S. Ohtsuka:
  Speaker Identification and Verification by Combining MFCC and Phase Information. 1085-1095
  
• Riccardo Miotto, Gert R. G. Lanckriet:
  A Generative Context Model for Semantic Music Annotation and Retrieval. 1096-1108
  
• C.-C. Lin, R. T.-H. Tsai:
  A Generative Data Augmentation Model for Enhancing Chinese Dialect Pronunciation Prediction. 1109-1117
  
• Alexey Ozerov, Emmanuel Vincent, Frédéric Bimbot:
  A General Flexible Framework for the Handling of Prior Information in Audio Source Separation. 1118-1133
  
• Jia-Min Ren, Jyh-Shing Roger Jang:
  Discovering Time-Constrained Sequential Patterns for Music Genre Classification. 1134-1144
  
• Virginia Estellers, Mihai Gurban, Jean-Philippe Thiran:
  On Dynamic Stream Weighting for Audio-Visual Speech Recognition. 1145-1157
  
• Navin Chatlani, John J. Soraghan:
  EMD-Based Filtering (EMDF) of Low-Frequency Noise for Speech Enhancement. 1158-1166
  
• Haiyan Shu, Haibin Huang, Susanto Rahardja:
  Analysis of Bit-Plane Probability for Generalized Gaussian Distribution and its Application in Audio Coding. 1167-1176
  
• Tobias Rosenkranz, Henning Puder:
  Improving Robustness of Codebook-Based Noise Estimation Approaches With Delta Codebooks. 1177-1188
  
• Ines Hafizovic, Carl-Inge Colombo Nilsen, Sverre Holm:
  Transformation Between Uniform Linear and Spherical Microphone Arrays With Symmetric Responses. 1189-1195
  
• Xiaohong Yang, Yufang Yang:
  Prosodic Realization of Rhetorical Structure in Chinese Discourse. 1196-1206
  
• David T. Yeh:
  Automated Physical Modeling of Nonlinear Audio Circuits for Real-Time Audio Effects - Part II: BJT and Vacuum Tube Examples. 1207-1216
  
• Manish Narwaria, Weisi Lin, Ian Vince McLoughlin, Sabu Emmanuel, Liang-Tien Chia:
  Nonintrusive Quality Assessment of Noise Suppressed Speech With Mel-Filtered Energies and Support Vector Regression. 1217-1232
  
• Wei-Ho Tsai, Hsin-Chieh Lee:
  Automatic Evaluation of Karaoke Singing Based on Pitch, Volume, and Rhythm Features. 1233-1243
  
• Takanobu Oba, Takaaki Hori, Atsushi Nakamura, Akinori Ito:
  Round-Robin Duel Discriminative Language Models. 1244-1255
  
• Yiteng Arden Huang, Jacob Benesty:
  A Multi-Frame Approach to the Frequency-Domain Single-Channel Noise Reduction Problem. 1256-1269
  
• Miroslav Zivanovic, Johan Schoukens:
  Single and Piecewise Polynomials for Modeling of Pitched Sounds. 1270-1281
  
• Yaakov Bucris, Israel Cohen, Miriam A. Doron:
  Bayesian Focusing for Coherent Wideband Beamforming. 1282-1296
  
• Hélène Papadopoulos, Geoffroy Peeters:
  Local Key Estimation From an Audio Signal Relying on Harmonic and Metrical Structures. 1297-1312
  
• Elizabeth Godoy, Olivier Rosec, Thierry Chonavel:
  Voice Conversion Using Dynamic Frequency Warping With Amplitude Scaling, for Parallel or Nonparallel Corpora. 1313-1323
  
• Ruofei Chen, Cheung-Fat Chan, Hing-Cheung So:
  Model-Based Speech Enhancement With Improved Spectral Envelope Estimation via Dynamics Tracking. 1324-1336
  
• Qun Feng Tan, Shrikanth S. Narayanan:
  Novel Variations of Group Sparse Regularization Techniques With Applications to Noise Robust Automatic Speech Recognition. 1337-1346
  
• Rubén Solera-Ureña, Ana I. García-Moral, Carmen Peláez-Moreno, Manel Martínez-Ramón, Fernando Díaz-de-María:
  Real-Time Robust Automatic Speech Recognition Using Compact Support Vector Machines. 1347-1361
  
• Amin Fazel, Shantanu Chakrabartty:
  Sparse Auditory Reproducing Kernel (SPARK) Features for Noise-Robust Speech Recognition. 1362-1371
  
• Jorge I. Marin-Hurtado, Devangi N. Parikh, David V. Anderson:
  Perceptually Inspired Noise-Reduction Method for Binaural Hearing Aids. 1372-1382
  
• Timo Gerkmann, Richard C. Hendriks:
  Unbiased MMSE-Based Noise Power Estimation With Low Complexity and Low Tracking Delay. 1383-1393
  
• Haiquan Zhao, Xiangping Zeng, Xiaoqiang Zhang, Zhengyou He, Tian-rui Li, Weidong Jin:
  Adaptive Extended Pipelined Second-Order Volterra Filter for Nonlinear Active Noise Controller. 1394-1399
  
• Damián Marelli, Mitsuko Aramaki, Richard Kronland-Martinet, Charles Verron:
  An Efficient Time-Frequency Method for Synthesizing Noisy Sounds With Short Transients and Narrow Spectral Components. 1400-1408
  
• Maurice F. Fallon, Simon J. Godsill:
  Acoustic Source Localization and Tracking of a Time-Varying Number of Speakers. 1409-1415
  

Volume 20, Number 5, July 2012

• Vesa Välimäki, Julian D. Parker, Lauri Savioja, Julius O. Smith, Jonathan S. Abel:
  Fifty Years of Artificial Reverberation. 1421-1448
  
• Flavio P. Ribeiro, Dinei A. F. Florêncio, Demba E. Ba, Cha Zhang:
  Geometrically Constrained Room Modeling With Compact Microphone Arrays. 1449-1460
  
• Wenliang Chen, Jun'ichi Kazama, Min Zhang, Yoshimasa Tsuruoka, Yujie Zhang, Yiou Wang, Kentaro Torisawa, Haizhou Li:
  Bitext Dependency Parsing With Auto-Generated Bilingual Treebank. 1461-1472
  
• K. Lakhdhar, R. Lefebvre:
  Context-Based Adaptive Arithmetic Encoding of EAVQ Indices. 1473-1481
  
• Chao-Ling Hsu, DeLiang Wang, Jyh-Shing Roger Jang, Ke Hu:
  A Tandem Algorithm for Singing Pitch Extraction and Voice Separation From Music Accompaniment. 1482-1491
  
• Zhen-Hua Ling, Li-Rong Dai:
  Minimum Kullback-Leibler Divergence Parameter Generation for HMM-Based Speech Synthesis. 1492-1502
  
• John Woodruff, DeLiang Wang:
  Binaural Localization of Multiple Sources in Reverberant and Noisy Environments. 1503-1512
  
• Welly Naptali, Masatoshi Tsuchiya, Seiichi Nakagawa:
  Topic-Dependent-Class-Based $n$-Gram Language Model. 1513-1525
  
• Jesper Rindom Jensen, Jacob Benesty, Mads Græsbøll Christensen, Søren Holdt Jensen:
  Non-Causal Time-Domain Filters for Single-Channel Noise Reduction. 1526-1541
  
• Kamil Adilolu, Robert Anniés, Elio Wahlen, Hendrik Purwins, Klaus Obermayer:
  A Graphical Representation and Dissimilarity Measure for Basic Everyday Sound Events. 1542-1552
  
• Cees H. Taal, Richard C. Hendriks, Richard Heusdens:
  A Low-Complexity Spectro-Temporal Distortion Measure for Audio Processing Applications. 1553-1564
  
• Huawei Chen, Wee Ser, Jianjiang Zhou:
  Robust Nearfield Wideband Beamformer Design Using Worst Case Mean Performance Optimization With Passband Response Variance Constraint. 1565-1572
  
• D. Rama Sanand, Srinivasan Umesh:
  VTLN Using Analytically Determined Linear-Transformation on Conventional MFCC. 1573-1584
  
• Sandro Cumani, Pietro Laface:
  Analysis of Large-Scale SVM Training Algorithms for Language and Speaker Recognition. 1585-1596
  
• Xiaoyan Cai, Wenjie Li:
  Mutually Reinforced Manifold-Ranking Based Relevance Propagation Model for Query-Focused Multi-Document Summarization. 1597-1607
  
• Xiaojia Zhao, Yang Shao, DeLiang Wang:
  CASA-Based Robust Speaker Identification. 1608-1616
  
• Saeed Mosayyebpour, Hamid Sheikhzadeh, T. Aaron Gulliver, Morteza Esmaeili:
  Single-Microphone LP Residual Skewness-Based Inverse Filtering of the Room Impulse Response. 1617-1632
  
• Upendra V. Chaudhari, Michael Picheny:
  Matching Criteria for Vocabulary-Independent Search. 1633-1643
  
• Daniele Giacobello, Mads Græsbøll Christensen, Manohar N. Murthi, Søren Holdt Jensen, Marc Moonen:
  Sparse Linear Prediction and Its Applications to Speech Processing. 1644-1657
  
• Stefan Bilbao:
  Optimized FDTD Schemes for 3-D Acoustic Wave Propagation. 1658-1663
  

Volume 20, Number 6, August 2012

• Sin-Horng Chen, Jyh-Her Yang, Chen-Yu Chiang, Ming-Chieh Liu, Yih-Ru Wang:
  A New Prosody-Assisted Mandarin ASR System. 1669-1684
  
• Romain Serizel, Marc Moonen, Jan Wouters, Søren Holdt Jensen:
  A Zone-of-Quiet Based Approach to Integrated Active Noise Control and Noise Reduction for Speech Enhancement in Hearing Aids. 1685-1697
  
• Christian D. Sigg, Tomas Dikk, Joachim M. Buhmann:
  Speech Enhancement Using Generative Dictionary Learning. 1698-1712
  
• Heiga Zen, Norbert Braunschweiler, Sabine Buchholz, Mark J. F. Gales, Kate Knill, Sacha Krstulovic, Javier Latorre:
  Statistical Parametric Speech Synthesis Based on Speaker and Language Factorization. 1713-1724
  
• Christian Schüldt, Fredric Lindström, Ingvar Claesson:
  A Delay-Based Double-Talk Detector. 1725-1733
  
• Alastair J. Manders, David M. Simpson, Steven L. Bell:
  Objective Prediction of the Sound Quality of Music Processed by an Adaptive Feedback Canceller. 1734-1745
  
• Shoichi Koyama, Ken'ichi Furuya, Yusuke Hiwasaki, Yoichi Haneda:
  Reproducing Virtual Sound Sources in Front of a Loudspeaker Array Using Inverse Wave Propagator. 1746-1758
  
• Justin Salamon, Emilia Gómez:
  Melody Extraction From Polyphonic Music Signals Using Pitch Contour Characteristics. 1759-1770
  
• Yizhao Ni, Matt McVicar, Raúl Santos-Rodriguez, Tijl De Bie:
  An End-to-End Machine Learning System for Harmonic Analysis of Music. 1771-1783
  
• Daisuke Saito, Shinji Watanabe, Atsushi Nakamura, Nobuaki Minematsu:
  Statistical Voice Conversion Based on Noisy Channel Model. 1784-1794
  
• Daniel Angus, Andrew E. Smith, Janet Wiles:
  Human Communication as Coupled Time Series: Quantifying Multi-Participant Recurrence. 1795-1807
  
• Claire Masterson, Gavin Kearney, Marcin Gorzel, Francis M. Boland:
  HRIR Order Reduction Using Approximate Factorization. 1808-1817
  
• Jan Vanek, Jan Trmal, Josef V. Psutka, Josef Psutka:
  Optimized Acoustic Likelihoods Computation for NVIDIA and ATI/AMD Graphics Processors. 1818-1828
  
• Jan Ole Jungmann, Radoslaw Mazur, Markus Kallinger, Tiemin Mei, Alfred Mertins:
  Combined Acoustic MIMO Channel Crosstalk Cancellation and Room Impulse Response Reshaping. 1829-1842
  
• Tianyu T. Wang, Thomas F. Quatieri:
  Two-Dimensional Speech-Signal Modeling. 1843-1856
  
• Isabel Barbancho, Lorenzo J. Tardón, Simone Sammartino, Ana M. Barbancho:
  Inharmonicity-Based Method for the Automatic Generation of Guitar Tablature. 1857-1868
  
• Amit Das, John H. L. Hansen:
  Constrained Iterative Speech Enhancement Using Phonetic Classes. 1869-1883
  
• Abbas Keshavarz, Saeed Mosayyebpour, Mehrzad Biguesh, T. Aaron Gulliver, Morteza Esmaeili:
  Speech-Model Based Accurate Blind Reverberation Time Estimation Using an LPC Filter. 1884-1893
  
• Anil Kumar Vuppala, Jainath Yadav, Saswat Chakrabarti, K. Sreenivasa Rao:
  Vowel Onset Point Detection for Low Bit Rate Coded Speech. 1894-1903
  

Volume 20, Number 7, 2012

• Theodoros Giannakopoulos, Sergios Petridis:
  Fisher Linear Semi-Discriminant Analysis for Speaker Diarization. 1913-1922
  
• Xiaodong Cui, Jing Huang, Jen-Tzung Chien:
  Multi-View and Multi-Objective Semi-Supervised Learning for HMM-Based Automatic Speech Recognition. 1923-1935
  
• Jacob L. Newman, Stephen J. Cox:
  Language Identification Using Visual Features. 1936-1947
  
• Jesper Rindom Jensen, Jacob Benesty, Mads Græsbøll Christensen, Søren Holdt Jensen:
  Enhancement of Single-Channel Periodic Signals in the Time-Domain. 1948-1963
  
• Marco Compagnoni, Paolo Bestagini, Fabio Antonacci, Augusto Sarti, Stefano Tubaro:
  Localization of Acoustic Sources Through the Fitting of Propagation Cones Using Multiple Independent Arrays. 1964-1975
  
• Jung-Woo Choi, Yang-Hann Kim:
  Integral Approach for Reproduction of Virtual Sound Source Surrounded by Loudspeaker Array. 1976-1989
  
• Tomi Kinnunen, Rahim Saeidi, Filip Sedlak, Kong-Aik Lee, Johan Sandberg, Maria Hansson-Sandsten, Haizhou Li:
  Low-Variance Multitaper MFCC Features: A Case Study in Robust Speaker Verification. 1990-2001
  
• Wen-Lin Zhang, Weiqiang Zhang, Bi-Cheng Li, Dan Qu, Michael T. Johnson:
  Bayesian Speaker Adaptation Based on a New Hierarchical Probabilistic Model. 2002-2015
  
• Tobias May, Steven van de Par, Armin Kohlrausch:
  A Binaural Scene Analyzer for Joint Localization and Recognition of Speakers in the Presence of Interfering Noise Sources and Reverberation. 2016-2030
  
• Armando Muscariello, Guillaume Gravier, Frédéric Bimbot:
  Unsupervised Motif Acquisition in Speech via Seeded Discovery and Template Matching Combination. 2031-2044
  
• César González Ferreras, David Escudero Mancebo, Carlos Vivaracho-Pascual, Valentín Cardeñoso-Payo:
  Improving Automatic Classification of Prosodic Events by Pairwise Coupling. 2045-2058
  
• Maximo Cobos, José J. López:
  Maximum a Posteriori Binary Mask Estimation for Underdetermined Source Separation Using Smoothed Posteriors. 2059-2064
  
• Sarmad Malik, Gerald Enzner:
  State-Space Frequency-Domain Adaptive Filtering for Nonlinear Acoustic Echo Cancellation. 2065-2079
  
• Ryoichi Miyazaki, Hiroshi Saruwatari, Takayuki Inoue, Yu Takahashi, Kiyohiro Shikano, Kazunobu Kondo:
  Musical-Noise-Free Speech Enhancement Based on Optimized Iterative Spectral Subtraction. 2080-2094
  
• Hung-yi Lee, Chia-Ping Chen, Lin-Shan Lee:
  Integrating Recognition and Retrieval With Relevance Feedback for Spoken Term Detection. 2095-2110
  
• Mohamed I. Alkanhal, Mohammed A. Al-Badrashiny, Mansour M. Alghamdi, Abdulaziz O. Al-Qabbany:
  Automatic Stochastic Arabic Spelling Correction With Emphasis on Space Insertions and Deletions. 2111-2122
  
• Stephen J. Elliott, Jordan Cheer, Jung-Woo Choi, Youngtae Kim:
  Robustness and Regularization of Personal Audio Systems. 2123-2133
  
• Lakshmi Saheer, John Dines, Philip N. Garner:
  Vocal Tract Length Normalization for Statistical Parametric Speech Synthesis. 2134-2148
  
• Yongqiang Wang, M. J. F. Gales:
  Speaker and Noise Factorization for Robust Speech Recognition. 2149-2158
  

Volume 20, Number 8, October 2012

• Leonardo O. Nunes, Flávio R. Avila, Alan Freihof Tygel, Luiz W. P. Biscainho, Bowon Lee, Amir Said, Ronald W. Schafer:
  A Parametric Objective Quality Assessment Tool for Speech Signals Degraded by Acoustic Echo. 2181-2190
  
• Yong Zhao, Biing-Hwang Juang:
  Nonlinear Compensation Using the Gauss-Newton Method for Noise-Robust Speech Recognition. 2191-2206
  
• Brian McFee, Luke Barrington, Gert R. G. Lanckriet:
  Learning Content Similarity for Music Recommendation. 2207-2218
  
• Hannu Pulakka, Ulpu Remes, Santeri Yrttiaho, Kalle J. Palomäki, Mikko Kurimo, Paavo Alku:
  Bandwidth Extension of Telephone Speech to Low Frequencies Using Sinusoidal Synthesis and a Gaussian Mixture Model. 2219-2231
  
• Iynkaran Natgunanathan, Yong Xiang, Yue Rong, Wanlei Zhou, Song Guo:
  Robust Patchwork-Based Embedding and Decoding Scheme for Digital Audio Watermarking. 2232-2239
  
• Yotaro Kubo, Shinji Watanabe, Takaaki Hori, Atsushi Nakamura:
  Structural Classification Methods Based on Weighted Finite-State Transducers for Automatic Speech Recognition. 2240-2251
  
• Xiaodong Cui, Jian Xue, Xin Chen, Peder A. Olsen, Pierre L. Dognin, Upendra V. Chaudhari, John R. Hershey, Bowen Zhou:
  Hidden Markov Acoustic Modeling With Bootstrap and Restructuring for Low-Resourced Languages. 2252-2264
  
• Amit Das, John H. L. Hansen:
  Phoneme Selective Speech Enhancement Using Parametric Estimators and the Mixture Maximum Model: A Unifying Approach. 2265-2279
  
• Phillip L. De Leon, Michael Pucher, Junichi Yamagishi, Inma Hernáez, Ibon Saratxaga:
  Evaluation of Speaker Verification Security and Detection of HMM-Based Synthetic Speech. 2280-2290
  
• Wei-Ho Tsai, Hsin-Chieh Lee:
  Singer Identification Based on Spoken Data in Voice Characterization. 2291-2300
  
• Daniel Felps, Christian Geng, Ricardo Gutierrez-Osuna:
  Foreign Accent Conversion Through Concatenative Synthesis in the Articulatory Domain. 2301-2312
  
• Gustavo Reis, Francisco Fernández de Vega, Aníbal Ferreira:
  Automatic Transcription of Polyphonic Piano Music Using Genetic Algorithms, Adaptive Spectral Envelope Modeling, and Dynamic Noise Level Estimation. 2313-2328
  
• Soroosh Mariooryad, Carlos Busso:
  Generating Human-Like Behaviors Using Joint, Speech-Driven Models for Conversational Agents. 2329-2340
  
• Hasim Sak, Murat Saraclar, Tunga Gungor:
  Morpholexical and Discriminative Language Models for Turkish Automatic Speech Recognition. 2341-2351
  
• Yongwon Jeong:
  Adaptation of Hidden Markov Models Using Model-as-Matrix Representation. 2352-2364
  
• Seyedmahdad Mirsamadi, Shabnam Ghaffarzadegan, Hamid Sheikhzadeh, Seyed Mohammad Ahadi, Amir Hossein Rezaie:
  Efficient Frequency Domain Implementation of Noncausal Multichannel Blind Deconvolution for Convolutive Mixtures of Speech. 2365-2377
  
• Barry-John Theobald, Iain Matthews:
  Relating Objective and Subjective Performance Measures for AAM-Based Visual Speech Synthesis. 2378-2387
  
• Terence Betlehem, Christopher Withers:
  Sound Field Reproduction With Energy Constraint on Loudspeaker Weights. 2388-2392
  

Volume 20, Number 9, November 2012

• Nikolaos Mitianoudis:
  A Generalized Directional Laplacian Distribution : Estimation, Mixture Models and Audio Source Separation. 2397-2408
  
• Janne Pylkkönen, Mikko Kurimo:
  Analysis of Extended Baum-Welch and Constrained Optimization for Discriminative Training of HMMs. 2409-2419
  
• Alex Southern, Damian T. Murphy, Lauri Savioja:
  Spatial Encoding of Finite Difference Time Domain Acoustic Models for Auralization. 2420-2432
  
• Marco Crocco, Andrea Trucco:
  Stochastic and Analytic Optimization of Sparse Aperiodic Arrays and Broadband Beamformers With Robust Superdirective Patterns. 2433-2447
  
• Masashi Okada, Takao Onoye, Wataru Kobayashi:
  A Ray Tracing Simulation of Sound Diffraction Based on the Analytic Secondary Source Model. 2448-2460
  
• Ryouichi Nishimura:
  Audio Watermarking Using Spatial Masking and Ambisonics. 2461-2469
  
• Flávio R. Avila, Luiz W. P. Biscainho:
  Bayesian Restoration of Audio Signals Degraded by Impulsive Noise Modeled as Individual Pulses. 2470-2481
  
• Woojay Jeon, Changxue Ma, Dusan Macho:
  Statistical Utterance Comparison for Speaker Clustering Using Factor Analysis. 2482-2491
  
• Justin Jian Zhang, Pascale Fung:
  Automatic Parliamentary Meeting Minute Generation Using Rhetorical Structure Modeling. 2492-2504
  
• Tomoki Toda, Mikihiro Nakagiri, Kiyohiro Shikano:
  Statistical Voice Conversion Techniques for Body-Conducted Unvoiced Speech Enhancement. 2505-2517
  
• Arun Narayanan, DeLiang Wang:
  A CASA-Based System for Long-Term SNR Estimation. 2518-2527
  
• Ronen Talmon, Israel Cohen, Sharon Gannot, Ronald R. Coifman:
  Supervised Graph-Based Processing for Sequential Transient Interference Suppression. 2528-2538
  
• Andre Holzapfel, Matthew E. P. Davies, José R. Zapata, João Lobato Oliveira, Fabien Gouyon:
  Selective Sampling for Beat Tracking Evaluation. 2539-2548
  
• Meng Guo, Søren Holdt Jensen, Jesper Jensen:
  Novel Acoustic Feedback Cancellation Approaches in Hearing Aid Applications Using Probe Noise and Probe Noise Enhancement. 2549-2563
  
• Jens Ahrens, Sascha Spors:
  A Modal Analysis of Spatial Discretization of Spherical Loudspeaker Distributions Used for Sound Field Synthesis. 2564-2574
  
• V. Tourbabin, Morag Agmon, Boaz Rafaely, Joseph Tabrikian:
  Optimal Real-Weighted Beamforming With Application to Linear and Spherical Arrays. 2575-2585
  
• Pejman Mowlaee, Rahim Saeidi, Mads Græsbøll Christensen, Zheng-Hua Tan, Tomi Kinnunen, Pasi Fränti, Søren Holdt Jensen:
  A Joint Approach for Single-Channel Speaker Identification and Speech Separation. 2586-2601
  
• Berlin Chen, Kuan-Yu Chen, Pei-Ning Chen, Yi-Wen Chen:
  Spoken Document Retrieval With Unsupervised Query Modeling Techniques. 2602-2612
  
• Kruthiventi S. S. Srinivas, Kishore Prahallad:
  An FIR Implementation of Zero Frequency Filtering of Speech Signals. 2613-2617
  

Volume 20, Number 10, December 2012

• Mari Ostendorf:
  A Message from the Vice President of Publications on New Developments in Signal Processing Society Publications. 2625
  
• Sundar Harshavardhan, Chandra Sekhar Seelamantula, Thippur V. Sreenivas:
  A Mixture Model Approach for Formant Tracking and the Robustness of Student's-t Distribution. 2626-2636
  
• Steven Hargreaves, Anssi Klapuri, Mark Sandler:
  Structural Segmentation of Multitrack Audio. 2637-2647
  
• Matthew Gibson, Thomas Hain:
  Correctness-Adjusted Unsupervised Discriminative Acoustic Model Adaptation. 2648-2656
  
• Bruno Defraene, Toon van Waterschoot, Hans Joachim Ferreau, Moritz Diehl, Marc Moonen:
  Real-Time Perception-Based Clipping of Audio Signals Using Convex Optimization. 2657-2671
  
• G. Ananthakrishnan, Olov Engwall, Daniel Neiberg:
  Exploring the Predictability of Non-Unique Acoustic-to-Articulatory Mappings. 2672-2682
  
• Fabio Antonacci, Jason Filos, Mark R. P. Thomas, Emanuël Anco Peter Habets, Augusto Sarti, Patrick A. Naylor, Stefano Tubaro:
  Inference of Room Geometry From Acoustic Impulse Responses. 2683-2695
  
• João Lobato Oliveira, Matthew E. P. Davies, Fabien Gouyon, Luís Paulo Reis:
  Beat Tracking for Multiple Applications: A Multi-Agent System Architecture With State Recovery. 2696-2706
  
• Takuya Yoshioka, Tomohiro Nakatani:
  Generalization of Multi-Channel Linear Prediction Methods for Blind MIMO Impulse Response Shortening. 2707-2720
  

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