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IEEE Transactions on Audio, Speech & Language Processing, Volume 18
Volume 18, Number 1, January 2010
- Ali H. Sayed:
Free Electronic Access to SP Publications. 1 - Dmitry N. Zotkin, Ramani Duraiswami
, Nail A. Gumerov:
Plane-Wave Decomposition of Acoustical Scenes Via Spherical and Cylindrical Microphone Arrays. 2-16 - Ramdas Kumaresan, Nitesh Panchal:
Encoding Bandpass Signals Using Zero/Level Crossings: A Model-Based Approach. 17-33 - Péter Balázs, Bernhard Laback
Time-Frequency Sparsity by Removing Perceptually Irrelevant Components Using a Simple Model of Simultaneous Masking. 34-49 - Antti J. Eronen, Anssi Klapuri:
Music Tempo Estimation With k -NN Regression. 50-57 - Jean-Marc Valin, Timothy B. Terriberry, Christopher Montgomery, Gregory Maxwell:
A High-Quality Speech and Audio Codec With Less Than 10-ms Delay. 58-67 - Martin Raspaud
Binaural Source Localization by Joint Estimation of ILD and ITD. 68-77 - Konrad Kowalczyk
Wideband and Isotropic Room Acoustics Simulation Using 2-D Interpolated FDTD Schemes. 78-89 - Tiago H. Falk
Modulation Spectral Features for Robust Far-Field Speaker Identification. 90-100 - Vaninirappuputhenpurayil Gopalan Reju
Underdetermined Convolutive Blind Source Separation via Time-Frequency Masking. 101-116 - Ian McLoughlin
Vowel Intelligibility in Chinese. 117-125 - Bernd Matschkal, Johannes B. Huber:
Spherical Logarithmic Quantization. 126-140 - Shih-Sian Cheng, Hsin-Min Wang
BIC-Based Speaker Segmentation Using Divide-and-Conquer Strategies With Application to Speaker Diarization. 141-157 - Emanuël Anco Peter Habets
New Insights Into the MVDR Beamformer in Room Acoustics. 158-170 - Tianyu T. Wang, Thomas F. Quatieri:
High-Pitch Formant Estimation by Exploiting Temporal Change of Pitch. 171-186 - Feifan Liu, Yang Liu:
Exploring Correlation Between ROUGE and Human Evaluation on Meeting Summaries. 187-196 - Mehryar Mohri, Pedro J. Moreno, Eugene Weinstein:
Efficient and Robust Music Identification With Weighted Finite-State Transducers. 197-207
Volume 18, Number 2, February 2010
- Hüseyin Hacihabiboglu
Simulation of Directional Microphones in Digital Waveguide Mesh-Based Models of Room Acoustics. 213-223 - Claudius Gläser, Martin Heckmann
Combining Auditory Preprocessing and Bayesian Estimation for Robust Formant Tracking. 224-236 - Damián Marelli, Péter Balázs:
On Pole-Zero Model Estimation Methods Minimizing a Logarithmic Criterion for Speech Analysis. 237-248 - Alfred Mertins, Tiemin Mei
Room Impulse Response Shortening/Reshaping With Infinity- and p -Norm Optimization. 249-259 - Mehrez Souden, Jacob Benesty
On Optimal Frequency-Domain Multichannel Linear Filtering for Noise Reduction. 260-276 - Avram Levi, Harvey F. Silverman:
A Robust Method to Extract Talker Azimuth Orientation Using a Large-Aperture Microphone Array. 277-285 - Roberto Napoli, Luigi Piroddi
Nonlinear Active Noise Control With NARX Models. 286-295 - Luis Buera, Antonio Miguel, Oscar Saz, Alfonso Ortega
Unsupervised Data-Driven Feature Vector Normalization With Acoustic Model Adaptation for Robust Speech Recognition. 296-309 - Chao-Ling Hsu, Jyh-Shing Roger Jang:
On the Improvement of Singing Voice Separation for Monaural Recordings Using the MIR-1K Dataset. 310-319 - Ümit Güz, Sébastien Cuendet, Dilek Hakkani-Tür, Gökhan Tür:
Multi-View Semi-Supervised Learning for Dialog Act Segmentation of Speech. 320-329 - Vinay Melkote, Kenneth Rose:
Trellis-Based Approaches to Rate-Distortion Optimized Audio Encoding. 330-341 - Bram Cornelis, Simon Doclo
Theoretical Analysis of Binaural Multimicrophone Noise Reduction Techniques. 342-355 - Wen Jin, Xin Liu, Michael S. Scordilis, Lu Han:
Speech Enhancement Using Harmonic Emphasis and Adaptive Comb Filtering. 356-368 - Nathalie Camelin
Detection and Interpretation of Opinion Expressions in Spoken Surveys. 369-381 - Michael I. Mandel, Ron J. Weiss, Daniel P. W. Ellis:
Model-Based Expectation-Maximization Source Separation and Localization. 382-394 - Shinji Watanabe
Predictor-Corrector Adaptation by Using Time Evolution System With Macroscopic Time Scale. 395-406 - Alexandros Nanopoulos, Dimitrios Rafailidis
MusicBox: Personalized Music Recommendation Based on Cubic Analysis of Social Tags. 407-412
Volume 18, Number 3, March 2010
- Bertrand David, Masataka Goto
Editorial for the Special Issue on Signal Models and Representations of Musical and Environmental Sounds. 417-419 - Vittoria Bruni, Silvia Marconi
Time-Scale Atoms Chains for Transients Detection in Audio Signals. 420-433 - Emmanuel Ravelli, Gaël Richard, Laurent Daudet:
Audio Signal Representations for Indexing in the Transform Domain. 434-446 - Nicolás Ruiz-Reyes
Adaptive Signal Modeling Based on Sparse Approximations for Scalable Parametric Audio Coding. 447-460 - Bob L. Sturm, John J. Shynk:
Sparse Approximation and the Pursuit of Meaningful Signal Models With Interference Adaptation. 461-472 - Julio J. Carabias-Orti
Music Scene-Adaptive Harmonic Dictionary for Unsupervised Note-Event Detection. 473-486 - Johan Xi Zhang, Mads Græsbøll Christensen
A Robust and Computationally Efficient Subspace-Based Fundamental Frequency Estimator. 487-497 - Jeremy Wells, Damian T. Murphy:
A Comparative Evaluation of Techniques for Single-Frame Discrimination of Nonstationary Sinusoids. 498-508 - Mathieu Lagrange, Gary P. Scavone, Philippe Depalle:
Analysis/Synthesis of Sounds Generated by Sustained Contact Between Rigid Objects. 509-518 - Paul H. Peeling, Ali Taylan Cemgil
Generative Spectrogram Factorization Models for Polyphonic Piano Transcription. 519-527 - Emmanuel Vincent, Nancy Bertin, Roland Badeau:
Adaptive Harmonic Spectral Decomposition for Multiple Pitch Estimation. 528-537 - Nancy Bertin, Roland Badeau, Emmanuel Vincent:
Enforcing Harmonicity and Smoothness in Bayesian Non-Negative Matrix Factorization Applied to Polyphonic Music Transcription. 538-549 - Alexey Ozerov, Cédric Févotte:
Multichannel Nonnegative Matrix Factorization in Convolutive Mixtures for Audio Source Separation. 550-563 - Jean-Louis Durrieu, Gaël Richard, Bertrand David, Cédric Févotte:
Source/Filter Model for Unsupervised Main Melody Extraction From Polyphonic Audio Signals. 564-575 - Yannis Panagakis
Non-Negative Multilinear Principal Component Analysis of Auditory Temporal Modulations for Music Genre Classification. 576-588 - Onur Dikmen
Gamma Markov Random Fields for Audio Source Modeling. 589-601 - Luke Barrington, Antoni B. Chan
Modeling Music as a Dynamic Texture. 602-612 - Anssi Klapuri, Tuomas Virtanen
Representing Musical Sounds With an Interpolating State Model. 613-624 - Kris West, Stephen Cox:
Incorporating Cultural Representations of Features Into Audio Music Similarity Estimation. 625-637 - Hiromasa Fujihara, Masataka Goto
A Modeling of Singing Voice Robust to Accompaniment Sounds and Its Application to Singer Identification and Vocal-Timbre-Similarity-Based Music Information Retrieval. 638-648 - Meinard Müller
Towards Timbre-Invariant Audio Features for Harmony-Based Music. 649-662 - Juan José Burred, Axel Röbel, Thomas Sikora:
Dynamic Spectral Envelope Modeling for Timbre Analysis of Musical Instrument Sounds. 663-674 - Geoffroy Peeters, Emmanuel Deruty:
Sound Indexing Using Morphological Description. 675-687 - Gordon Wichern, Jiachen Xue, Harvey D. Thornburg, Brandon Mechtley, Andreas Spanias:
Segmentation, Indexing, and Retrieval for Environmental and Natural Sounds. 688-707
Volume 18, Number 4, May 2010
- Vesa Välimäki
Introduction to the Special Issue on Virtual Analog Audio Effects and Musical Instruments. 713-714 - Giovanni De Sanctis, Augusto Sarti:
Virtual Analog Modeling in the Wave-Digital Domain. 715-727 - David T. Yeh, Jonathan S. Abel, Julius O. Smith III
Automated Physical Modeling of Nonlinear Audio Circuits For Real-Time Audio Effects - Part I: Theoretical Development. 728-737 - Jyri Pakarinen, Matti Karjalainen:
Enhanced Wave Digital Triode Model for Real-Time Tube Amplifier Emulation. 738-746 - Thomas Hélie:
Volterra Series and State Transformation for Real-Time Simulations of Audio Circuits Including Saturations: Application to the Moog Ladder Filter. 747-759 - Federico Fontana
Modeling of the EMS VCS3 Voltage-Controlled Filter as a Nonlinear Filter Network. 760-772 - Juhan Nam
Efficient Antialiasing Oscillator Algorithms Using Low-Order Fractional Delay Filters. 773-785 - Vesa Välimäki
Alias-Suppressed Oscillators Based on Differentiated Polynomial Waveforms. 786-798 - Stefan Bilbao, Julian Parker:
A Virtual Model of Spring Reverberation. 799-808 - Balázs Bank
A Modal-Based Real-Time Piano Synthesizer. 809-821 - Gianpaolo Evangelista
Player-Instrument Interaction Models for Digital Waveguide Synthesis of Guitar: Touch and Collisions. 822-832 - Nelson Lee, Julius O. Smith III
Analysis and Synthesis of Coupled Vibrating Strings Using a Hybrid Modal-Waveguide Synthesis Model. 833-842 - Rémi Mignot, Thomas Hélie, Denis Matignon:
Digital Waveguide Modeling for Wind Instruments: Building a State-Space Representation Based on the Webster-Lokshin Model. 843-854 - Esteban Maestre
Statistical Modeling of Bowing Control Applied to Violin Sound Synthesis. 855-871 - Stefan Bilbao:
Percussion Synthesis Based on Models of Nonlinear Shell Vibration. 872-880 - Rudolf Rabenstein, Tilman Koch, Christian Popp:
Tubular Bells: A Physical and Algorithmic Model. 881-890 - Federico Avanzini
A Modular Physically Based Approach to the Sound Synthesis of Membrane Percussion Instruments. 891-902
Volume 18, Number 5, July 2010
- Yannis Stylianou, Tomoki Toda
Introduction to the Special Section on Voice Transformation. 909-911 - Elina Helander
Voice Conversion Using Partial Least Squares Regression. 912-921 - Daniel Erro, Asunción Moreno, Antonio Bonafonte
Voice Conversion Based on Weighted Frequency Warping. 922-931 - Jianhua Tao, Meng Zhang, Jani Nurminen, Jilei Tian, Xia Wang:
Supervisory Data Alignment for Text-Independent Voice Conversion. 932-943 - Daniel Erro, Asunción Moreno, Antonio Bonafonte
INCA Algorithm for Training Voice Conversion Systems From Nonparallel Corpora. 944-953 - Srinivas Desai, Alan W. Black, B. Yegnanarayana, Kishore Prahallad:
Spectral Mapping Using Artificial Neural Networks for Voice Conversion. 954-964 - Oytun Türk, Marc Schröder:
Evaluation of Expressive Speech Synthesis With Voice Conversion and Copy Resynthesis Techniques. 965-973 - Daniel Erro, Eva Navas
Emotion Conversion Based on Prosodic Unit Selection. 974-983 - Junichi Yamagishi, Bela Usabaev, Simon King
Thousands of Voices for HMM-Based Speech Synthesis-Analysis and Application of TTS Systems Built on Various ASR Corpora. 984-1004 - Oliver Watts, Junichi Yamagishi, Simon King
Synthesis of Child Speech With HMM Adaptation and Voice Conversion. 1005-1016 - Purvis Bedenbaugh, Diana K. Sarko, Heidi L. Roth, Eugene M. Martin:
Prosody-Preserving Voice Transformation to Evaluate Brain Representations of Speech Sounds. 1017-1029 - Daniel Felps, Ricardo Gutierrez-Osuna
Developing Objective Measures of Foreign-Accent Conversion. 1030-1040 - Carlos Molina, Néstor Becerra Yoma, Fernando Huenupán
Maximum Entropy-Based Reinforcement Learning Using a Confidence Measure in Speech Recognition for Telephone Speech. 1041-1052 - Xugang Lu, Jianwu Dang:
Vowel Production Manifold: Intrinsic Factor Analysis of Vowel Articulation. 1053-1062 - S. Abdallah:
Comment on "Unified View of Prediction and Repetition Structure in Audio Signals With Application to Interest Point Detection". 1063-1065 - Cong-Thanh Do, Dominique Pastor
On the Recognition of Cochlear Implant-Like Spectrally Reduced Speech With MFCC and HMM-Based ASR. 1065-1068 - Parham Mokhtari, Hironori Takemoto, Ryouichi Nishimura, Hiroaki Kato:
Optimum Loss Factor for a Perfectly Matched Layer in Finite-Difference Time-Domain Acoustic Simulation. 1068-1071 - Mehrez Souden, Jingdong Chen, Jacob Benesty
Gaussian Model-Based Multichannel Speech Presence Probability. 1072-1077 - Stas Tiomkin, David Malah
Statistical Text-to-Speech Synthesis Based on Segment-Wise Representation With a Norm Constraint. 1077-1082 - Claudio Garretón, Néstor Becerra Yoma, Matias Torres:
Channel Robust Feature Transformation Based on Filter-Bank Energy Filtering. 1082-1086
Volume 18, Number 6, August 2010
- Hamed Ketabdar, Hervé Bourlard:
Enhanced Phone Posteriors for Improving Speech Recognition Systems. 1094-1106 - Sergio Canazza, Giovanni De Poli
Restoration of Audio Documents by Means of Extended Kalman Filter. 1107-1115 - Chunghsin Yeh, Axel Röbel, Xavier Rodet:
Multiple Fundamental Frequency Estimation and Polyphony Inference of Polyphonic Music Signals. 1116-1126 - Jiucang Hao, Te-Won Lee, Terrence J. Sejnowski:
Speech Enhancement Using Gaussian Scale Mixture Models. 1127-1136 - Romain Serizel, Marc Moonen, Jan Wouters
Integrated Active Noise Control and Noise Reduction in Hearing Aids. 1137-1146 - Justin Jian Zhang, Ricky Ho Yin Chan, Pascale Fung:
Extractive Speech Summarization Using Shallow Rhetorical Structure Modeling. 1147-1157 - Xiong Xiao, Jinyu Li
A Study on the Generalization Capability of Acoustic Models for Robust Speech Recognition. 1158-1169 - Chung-Hsien Wu
Sentence Correction Incorporating Relative Position and Parse Template Language Models. 1170-1181 - Robbie Vogt, Sridha Sridharan, Michael Mason
Making Confident Speaker Verification Decisions With Minimal Speech. 1182-1192 - Dimitri Nion, Kleanthis N. Mokios, Nicholas D. Sidiropoulos
Batch and Adaptive PARAFAC-Based Blind Separation of Convolutive Speech Mixtures. 1193-1207 - Vaclav Eksler, Milan Jelinek:
Glottal-Shape Codebook to Improve Robustness of CELP Codecs. 1208-1217 - Moo Young Kim, W. Bastiaan Kleijn
Reduction of the Impact of Distortion Outliers and Source Mismatch in Resolution-Constrained Quantization. 1218-1227 - Maurice F. Fallon, Simon J. Godsill:
Acoustic Source Localization and Tracking Using Track Before Detect. 1228-1242 - Xiaoqiang Xiao, Robert M. Nickel
Speech Enhancement With Inventory Style Speech Resynthesis. 1243-1257 - Angel M. Gomez
A Multipulse-Based Forward Error Correction Technique for Robust CELP-Coded Speech Transmission Over Erasure Channels. 1258-1268 - Matt Gibson, Thomas Hain
Error Approximation and Minimum Phone Error Acoustic Model Estimation. 1269-1279 - Matthias Mauch, Simon Dixon:
Simultaneous Estimation of Chords and Musical Context From Audio. 1280-1289 - Jingen Ni, Feng Li:
A Variable Step-Size Matrix Normalized Subband Adaptive Filter. 1290-1299 - Chang Huai You, Kong-Aik Lee
GMM-SVM Kernel With a Bhattacharyya-Based Distance for Speaker Recognition. 1300-1312 - Ilknur Durgar El-Kahlout, Kemal Oflazer
Exploiting Morphology and Local Word Reordering in English-to-Turkish Phrase-Based Statistical Machine Translation. 1313-1322 - Jingbo Zhu, Huizhen Wang, Benjamin K. Tsou, Matthew Y. Ma:
Active Learning With Sampling by Uncertainty and Density for Data Annotations. 1323-1331 - Chi-Sang Jung, Moo Young Kim, Hong-Goo Kang:
Selecting Feature Frames for Automatic Speaker Recognition Using Mutual Information. 1332-1340 - José L. Carmona, Antonio M. Peinado
MMSE-Based Packet Loss Concealment for CELP-Coded Speech Recognition. 1341-1353 - Kentaro Ishizuka, Shoko Araki
Speech Activity Detection for Multi-Party Conversation Analyses Based on Likelihood Ratio Test on Spatial Magnitude. 1354-1365 - Marc Ferras, Cheung-Chi Leung, Claude Barras, Jean-Luc Gauvain:
Comparison of Speaker Adaptation Methods as Feature Extraction for SVM-Based Speaker Recognition. 1366-1378 - Hynek Boril, John H. L. Hansen:
Unsupervised Equalization of Lombard Effect for Speech Recognition in Noisy Adverse Environments. 1379-1393 - Chung-Hsien Wu
Hierarchical Prosody Conversion Using Regression-Based Clustering for Emotional Speech Synthesis. 1394-1405 - Keansub Lee, Daniel P. W. Ellis:
Audio-Based Semantic Concept Classification for Consumer Video. 1406-1416 - Charturong Tantibundhit, Franz Pernkopf
Joint Time-Frequency Segmentation Algorithm for Transient Speech Decomposition and Speech Enhancement. 1417-1428 - Eric A. Lehmann
Diffuse Reverberation Model for Efficient Image-Source Simulation of Room Impulse Responses. 1429-1439 - Øystein Birkenes, Tomoko Matsui, Kunio Tanabe, Sabato Marco Siniscalchi
Penalized Logistic Regression With HMM Log-Likelihood Regressors for Speech Recognition. 1440-1454 - Jerome R. Bellegarda:
A Dynamic Cost Weighting Framework for Unit Selection Text-to-Speech Synthesis. 1455-1463 - Mathieu Parvaix, Laurent Girin, Jean-Marc Brossier:
A Watermarking-Based Method for Informed Source Separation of Audio Signals With a Single Sensor. 1464-1475 - Hirofumi Nakajima, Kazuhiro Nakadai, Yuji Hasegawa, Hiroshi Tsujino
Blind Source Separation With Parameter-Free Adaptive Step-Size Method for Robot Audition. 1476-1485 - Murat Akbacak, John H. L. Hansen:
Spoken Proper Name Retrieval for Limited Resource Languages Using Multilingual Hybrid Representations. 1486-1495 - Mitchell McLaren, Robbie Vogt, Brendan Baker, Sridha Sridharan:
Data-Driven Background Dataset Selection for SVM-Based Speaker Verification. 1496-1506 - Hirokazu Kameoka, Nobutaka Ono
Speech Spectrum Modeling for Joint Estimation of Spectral Envelope and Fundamental Frequency. 1507-1516 - Andre Holzapfel, Yannis Stylianou, Ali Cenk Gedik
Three Dimensions of Pitched Instrument Onset Detection. 1517-1527 - Panikos Heracleous, V.-A. Tran, Takayuki Nagai, Kiyohiro Shikano:
Analysis and Recognition of NAM Speech Using HMM Distances and Visual Information. 1528-1538 - Yuya Akita, Tatsuya Kawahara
Statistical Transformation of Language and Pronunciation Models for Spontaneous Speech Recognition. 1539-1549 - Charles Verron, Mitsuko Aramaki, Richard Kronland-Martinet
A 3-D Immersive Synthesizer for Environmental Sounds. 1550-1561 - Yi-Cheng Pan, Lin-Shan Lee:
Performance Analysis for Lattice-Based Speech Indexing Approaches Using Words and Subword Units. 1562-1574 - Mehrez Souden, Jacob Benesty
Broadband Source Localization From an Eigenanalysis Perspective. 1575-1587 - Péter Mihajlik
Improved Recognition of Spontaneous Hungarian Speech - Morphological and Acoustic Modeling Techniques for a Less Resourced Task. 1588-1600 - Gökhan Tür, Andreas Stolcke, L. Lynn Voss, Stanley Peters, Dilek Hakkani-Tür, John Dowding, Benoît Favre, Raquel Fernández, Matthew Frampton, Michael W. Frandsen, Clint Frederickson, Martin Graciarena, Donald Kintzing, Kyle Leveque, Shane Mason, John Niekrasz, Matthew Purver
The CALO Meeting Assistant System. 1601-1611 - Bengt J. Borgstrom, Abeer Alwan:
HMM-Based Reconstruction of Unreliable Spectrographic Data for Noise Robust Speech Recognition. 1612-1623 - Sriram Ganapathy, Petr Motlícek
Autoregressive Models of Amplitude Modulations in Audio Compression. 1624-1631 - Hyeon-Jin Jeon, Tae-Gyu Chang, Sen M. Kuo:
Analysis of Frequency Mismatch in Narrowband Active Noise Control. 1632-1642 - Valentin Emiya
Multipitch Estimation of Piano Sounds Using a New Probabilistic Spectral Smoothness Principle. 1643-1654 - Thushara D. Abhayapala
Spherical Harmonic Analysis of Wavefields Using Multiple Circular Sensor Arrays. 1655-1666
Volume 18, Number 7, September 2010
- Tomohiro Nakatani, Walter Kellermann, Patrick A. Naylor
Introduction to the Special Issue on Processing Reverberant Speech: Methodologies and Applications. 1673-1675 - Armin Sehr, Roland Maas, Walter Kellermann:
Reverberation Model-Based Decoding in the Logmelspec Domain for Robust Distant-Talking Speech Recognition. 1676-1691 - Alexander Krueger, Reinhold Haeb-Umbach
Model-Based Feature Enhancement for Reverberant Speech Recognition. 1692-1707 - Randy Gomez, Tatsuya Kawahara
Robust Speech Recognition Based on Dereverberation Parameter Optimization Using Acoustic Model Likelihood. 1708-1716 - Tomohiro Nakatani
Speech Dereverberation Based on Variance-Normalized Delayed Linear Prediction. 1717-1731 - Marco Jeub, Magnus Schäfer, Thomas Esch, Peter Vary:
Model-Based Dereverberation Preserving Binaural Cues. 1732-1745 - Jan S. Erkelens, Richard Heusdens:
Correlation-Based and Model-Based Blind Single-Channel Late-Reverberation Suppression in Noisy Time-Varying Acoustical Environments. 1746-1765 - Tiago H. Falk
A Non-Intrusive Quality and Intelligibility Measure of Reverberant and Dereverberated Speech. 1766-1774 - Takayuki Arai, Nao Hodoshima
Using Steady-State Suppression to Improve Speech Intelligibility in Reverberant Environments for Elderly Listeners. 1775-1780 - Flavio P. Ribeiro, Cha Zhang, Dinei A. F. Florêncio, Demba E. Ba:
Using Reverberation to Improve Range and Elevation Discrimination for Small Array Sound Source Localization. 1781-1792 - Yan-Chen Lu, Martin Cooke:
Binaural Estimation of Sound Source Distance via the Direct-to-Reverberant Energy Ratio for Static and Moving Sources. 1793-1805 - Fotios Talantzis:
An Acoustic Source Localization and Tracking Framework Using Particle Filtering and Information Theory. 1806-1817 - Matthieu Kowalski, Emmanuel Vincent, Rémi Gribonval:
Beyond the Narrowband Approximation: Wideband Convex Methods for Under-Determined Reverberant Audio Source Separation. 1818-1829 - Ngoc Q. K. Duong, Emmanuel Vincent, Rémi Gribonval:
Under-Determined Reverberant Audio Source Separation Using a Full-Rank Spatial Covariance Model. 1830-1840 - Alireza Masnadi-Shirazi, Wenyi Zhang, Bhaskar D. Rao:
Glimpsing IVA: A Framework for Overcomplete/Complete/Undercomplete Convolutive Source Separation. 1841-1855 - John Woodruff, DeLiang Wang:
Sequential Organization of Speech in Reverberant Environments by Integrating Monaural Grouping and Binaural Localization. 1856-1866 - Chris Hummersone, Russell Mason
Dynamic Precedence Effect Modeling for Source Separation in Reverberant Environments. 1867-1871 - Michael I. Mandel, Scott Bressler, Barbara G. Shinn-Cunningham, Daniel P. W. Ellis:
Evaluating Source Separation Algorithms With Reverberant Speech. 1872-1883
Volume 18, Number 8, November 2010
- Ozlem Kalinli, Michael L. Seltzer, Jasha Droppo
Noise Adaptive Training for Robust Automatic Speech Recognition. 1889-1901 - Georgios N. Lilis, Daniele Angelosante, Georgios B. Giannakis
Sound Field Reproduction using the Lasso. 1902-1912 - Wenyi Zhang, Bhaskar D. Rao:
A Two Microphone-Based Approach for Source Localization of Multiple Speech Sources. 1913-1928 - Damián Marelli, Mitsuko Aramaki, Richard Kronland-Martinet
Time-Frequency Synthesis of Noisy Sounds With Narrow Spectral Components. 1929-1940 - Songfang Huang, Steve Renals
Hierarchical Bayesian Language Models for Conversational Speech Recognition. 1941-1954 - Emmanouil Benetos
Non-Negative Tensor Factorization Applied to Music Genre Classification. 1955-1967 - Emmanouil Benetos
Auditory Spectrum-Based Pitched Instrument Onset Detection. 1968-1977 - Jian Liu, Yegui Xiao, Jinwei Sun, Li Xu:
Analysis of Online Secondary-Path Modeling With Auxiliary Noise Scaled by Residual Noise Signal. 1978-1993 - Chi-Chun Hsia, Chung-Hsien Wu
Exploiting Prosody Hierarchy and Dynamic Features for Pitch Modeling and Generation in HMM-Based Speech Synthesis. 1994-2003 - Huijun Ding, Ing Yann Soon, Chai Kiat Yeo
Over-Attenuated Components Regeneration for Speech Enhancement. 2004-2014 - Aarthi M. Reddy, Richard C. Rose:
Integration of Statistical Models for Dictation of Document Translations in a Machine-Aided Human Translation Task. 2015-2027 - David Imseng, Gerald Friedland:
Tuning-Robust Initialization Methods for Speaker Diarization. 2028-2037 - Jens Ahrens
Sound Field Reproduction Using Planar and Linear Arrays of Loudspeakers. 2038-2050 - Gregory Sell, Malcolm Slaney
Solving Demodulation as an Optimization Problem. 2051-2066 - Guoning Hu, DeLiang L. Wang:
A Tandem Algorithm for Pitch Estimation and Voiced Speech Segregation. 2067-2079 - Gibak Kim, Philipos C. Loizou:
Improving Speech Intelligibility in Noise Using Environment-Optimized Algorithms. 2080-2090 - Maor Kleider, Boaz Rafaely
Golden-Ratio Sampling for Scanning Circular Microphone Arrays. 2091-2098 - Seokhwan Jo, Chang D. Yoo:
Psychoacoustically Constrained and Distortion Minimized Speech Enhancement. 2099-2110 - Wooil Kim, John H. L. Hansen:
Missing-Feature Reconstruction by Leveraging Temporal Spectral Correlation for Robust Speech Recognition in Background Noise Conditions. 2111-2120 - Zhiyao Duan
Multiple Fundamental Frequency Estimation by Modeling Spectral Peaks and Non-Peak Regions. 2121-2133 - Nikoletta Bassiou, Vassiliki Moschou, Constantine Kotropoulos
Speaker Diarization Exploiting the Eigengap Criterion and Cluster Ensembles. 2134-2144 - Vishweshwara Rao, Preeti Rao:
Vocal Melody Extraction in the Presence of Pitched Accompaniment in Polyphonic Music. 2145-2154 - Brady Laska
Particle Filter Enhancement of Speech Spectral Amplitudes. 2155-2167
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