default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCIDIEEE Transactions on Audio, Speech and Language Processing, Volume 21
Volume 21, Number 1, January 2013
- Stanislaw Gorlow, Sylvain Marchand:
Informed Audio Source Separation Using Linearly Constrained Spatial Filters. 1-11 - Florin Ghido, Ioan Tabus
:
Sparse Modeling for Lossless Audio Compression. 12-26 - Xiguang Zheng, Christian H. Ritz, Jiangtao Xi:
Encoding Navigable Speech Sources: A Psychoacoustic-Based Analysis-by-Synthesis Approach. 27-36 - Jwu-Sheng Hu, Ming-Tang Lee, Chia-Hsing Yang:
Robust Adaptive Beamformer for Speech Enhancement Using the Second-Order Extended H∞ Filter. 37-48 - Yi-Chin Huang, Chung-Hsien Wu, Yu-Ting Chao:
Personalized Spectral and Prosody Conversion Using Frame-Based Codeword Distribution and Adaptive CRF. 49-60 - Nilesh Madhu, Ann Spriet, Sofie Jansen, Raphael Koning, Jan Wouters:
The Potential for Speech Intelligibility Improvement Using the Ideal Binary Mask and the Ideal Wiener Filter in Single Channel Noise Reduction Systems: Application to Auditory Prostheses. 61-70 - Zafar Rafii, Bryan Pardo:
REpeating Pattern Extraction Technique (REPET): A Simple Method for Music/Voice Separation. 71-82 - Sree Hari Krishnan Parthasarathi, Hervé Bourlard, Daniel Gatica-Perez:
Wordless Sounds: Robust Speaker Diarization Using Privacy-Preserving Audio Representations. 83-96 - Feng Huang, Tan Lee:
Pitch Estimation in Noisy Speech Using Accumulated Peak Spectrum and Sparse Estimation Technique. 97-107 - Néstor Becerra Yoma, Claudio Garretón, Fernando Huenupán, Ignacio Catalan, Jorge Wuth Sepúlveda:
On Reducing Harmonic and Sampling Distortion in Vocal Tract Length Normalization. 108-119 - Ke Hu, DeLiang Wang:
An Unsupervised Approach to Cochannel Speech Separation. 120-129 - Ronen Talmon, Israel Cohen, Sharon Gannot:
Single-Channel Transient Interference Suppression With Diffusion Maps. 130-142 - Nima Yousefian, Philipos C. Loizou:
A Dual-Microphone Algorithm That Can Cope With Competing-Talker Scenarios. 143-153 - Miguel Ferrer, Alberto González, Maria de Diego, Gema Pinero:
Convex Combination Filtered-X Algorithms for Active Noise Control Systems. 154-165 - Kun Han, DeLiang Wang:
Towards Generalizing Classification Based Speech Separation. 166-175 - Nicolas Sturmel, Laurent Daudet:
Informed Source Separation Using Iterative Reconstruction. 176-183 - Ziqiang Shi, Jiqing Han, Tieran Zheng, Shiwen Deng:
Audio Segment Classification Using Online Learning Based Tensor Representation Feature Discrimination. 184-194 - Hai Son Le, Ilya Oparin, Alexandre Allauzen, Jean-Luc Gauvain, François Yvon:
Structured Output Layer Neural Network Language Models for Speech Recognition. 195-204 - Zhen-Hua Ling, Korin Richmond, Junichi Yamagishi:
Articulatory Control of HMM-Based Parametric Speech Synthesis Using Feature-Space-Switched Multiple Regression. 205-217
Volume 21, Number 2, February 2013
- Alexandre Trilla, Francesc Alías:
Sentence-Based Sentiment Analysis for Expressive Text-to-Speech. 223-233 - Paolo Annibale, Jason Filos, Patrick A. Naylor, Rudolf Rabenstein:
TDOA-Based Speed of Sound Estimation for Air Temperature and Room Geometry Inference. 234-246 - Jung-Woo Choi, Yang-Hann Kim:
Sound Field Reproduction of a Virtual Source Inside a Loudspeaker Array With Minimal External Radiation. 247-259 - Wei Wu, Mari Ostendorf:
Graph-Based Query Strategies for Active Learning. 260-269 - Yuxuan Wang, Kun Han, DeLiang Wang:
Exploring Monaural Features for Classification-Based Speech Segregation. 270-279 - Yao Qian, Frank K. Soong, Zhi-Jie Yan:
A Unified Trajectory Tiling Approach to High Quality Speech Rendering. 280-290 - Erinç Dikici, Murat Semerci, Murat Saraclar, Ethem Alpaydin:
Classification and Ranking Approaches to Discriminative Language Modeling for ASR. 291-300 - Boyan Huang, Yegui Xiao, Jinwei Sun, Guo Wei:
A Variable Step-Size FXLMS Algorithm for Narrowband Active Noise Control. 301-312 - Stefano D'Angelo, Jyri Pakarinen, Vesa Välimäki:
New Family of Wave-Digital Triode Models. 313-321 - Saeed Mosayyebpour, Morteza Esmaeili, T. Aaron Gulliver:
Single-Microphone Early and Late Reverberation Suppression in Noisy Speech. 322-335 - Chengshi Zheng, Hao Liu, Renhua Peng, Xiaodong Li:
A Statistical Analysis of Two-Channel Post-Filter Estimators in Isotropic Noise Fields. 336-342 - Shmulik Markovich Golan, Sharon Gannot, Israel Cohen:
Distributed Multiple Constraints Generalized Sidelobe Canceler for Fully Connected Wireless Acoustic Sensor Networks. 343-356 - Ian McGraw, Ibrahim Badr, James R. Glass:
Learning Lexicons From Speech Using a Pronunciation Mixture Model. 357-366 - Jonathan William Dennis, Tran Huy Dat, Engsiong Chng:
Image Feature Representation of the Subband Power Distribution for Robust Sound Event Classification. 367-377 - Nasim Radmanesh, Ian S. Burnett:
Generation of Isolated Wideband Sound Fields Using a Combined Two-stage Lasso-LS Algorithm. 378-387 - Dong Yu, Li Deng, Frank Seide:
The Deep Tensor Neural Network With Applications to Large Vocabulary Speech Recognition. 388-396 - Manas A. Pathak, Bhiksha Raj:
Privacy-Preserving Speaker Verification and Identification Using Gaussian Mixture Models. 397-406 - Abigail A. Kressner, David V. Anderson, Christopher J. Rozell:
Evaluating the Generalization of the Hearing Aid Speech Quality Index (HASQI). 407-415 - Sridhar Krishna Nemala, Kailash Patil, Mounya Elhilali:
A Multistream Feature Framework Based on Bandpass Modulation Filtering for Robust Speech Recognition. 416-426 - Chiong-Ching Lai, Sven Nordholm, Yee-Hong Leung:
Design of Steerable Spherical Broadband Beamformers With Flexible Sensor Configurations. 427-438 - Antonio Canclini, Fabio Antonacci, Augusto Sarti, Stefano Tubaro:
Acoustic Source Localization With Distributed Asynchronous Microphone Networks. 439-443 - Stefano Gaiotto:
A Tuning-Less Approach in Secondary Path Modeling in Active Noise Control Systems. 444-448 - Matt Speed, Damian T. Murphy, David M. Howard:
Three-Dimensional Digital Waveguide Mesh Simulation of Cylindrical Vocal Tract Analogs. 449-455
Volume 21, Number 3, March 2013
- Hongsen He, Lifu Wu, Jing Lu, Xiaojun Qiu, Jingdong Chen:
Time Difference of Arrival Estimation Exploiting Multichannel Spatio-Temporal Prediction. 463-475 - Shan Liang, Wenju Liu, Wei Jiang:
A New Bayesian Method Incorporating With Local Correlation for IBM Estimation. 476-487 - Stephan Tassart:
Band-Limited Impulse Train Generation Using Sampled Infinite Impulse Responses of Analog Filters. 488-497 - Xin Chen, Yunxin Zhao:
Building Acoustic Model Ensembles by Data Sampling With Enhanced Trainings and Features. 498-507 - Simone Spagnol, Michele Geronazzo, Federico Avanzini:
On the Relation Between Pinna Reflection Patterns and Head-Related Transfer Function Features. 508-519 - Vipul Arora, Laxmidhar Behera:
On-Line Melody Extraction From Polyphonic Audio Using Harmonic Cluster Tracking. 520-530 - Meinard Müller, Nanzhu Jiang, Peter Grosche:
A Robust Fitness Measure for Capturing Repetitions in Music Recordings With Applications to Audio Thumbnailing. 531-543 - Shi-Xiong Zhang, Mark J. F. Gales:
Structured SVMs for Automatic Speech Recognition. 544-555 - Daniel Erro, Eva Navas, Inma Hernáez:
Parametric Voice Conversion Based on Bilinear Frequency Warping Plus Amplitude Scaling. 556-566 - Shuhua Zhang, Laurent Girin:
Fast and Accurate Direct MDCT to DFT Conversion With Arbitrary Window Functions. 567-578 - Ladan Baghai-Ravary:
The Inherent Temporal Precision of Phoneme Transitions. 579-586 - Matt Shannon, Heiga Zen, William Byrne:
Autoregressive Models for Statistical Parametric Speech Synthesis. 587-597 - Jesper Kjær Nielsen, Mads Græsbøll Christensen, Søren Holdt Jensen:
Default Bayesian Estimation of the Fundamental Frequency. 598-610 - Ziqiang Shi, Jiqing Han, Tieran Zheng, Ji Li:
Identification of Objectionable Audio Segments Based on Pseudo and Heterogeneous Mixture Models. 611-623 - José A. González, Antonio M. Peinado, Ning Ma, Angel M. Gomez, Jon Barker:
MMSE-Based Missing-Feature Reconstruction With Temporal Modeling for Robust Speech Recognition. 624-635 - Bassam Jabaian, Laurent Besacier, Fabrice Lefèvre:
Comparison and Combination of Lightly Supervised Approaches for Language Portability of a Spoken Language Understanding System. 636-648 - Renxian Zhang, Wenjie Li, Dehong Gao, Ouyang You:
Automatic Twitter Topic Summarization With Speech Acts. 649-658 - Weibin Zhang, Pascale Fung:
Sparse Inverse Covariance Matrices for Low Resource Speech Recognition. 659-668 - Jonathan Botts, José Escolano, Ning Xiang:
Design of IIR Filters With Bayesian Model Selection and Parameter Estimation. 669-674 - Kais Khaldi, Abdel-Ouahab Boudraa:
Audio Watermarking Via EMD. 675-680
Volume 21, Number 4, April 2013
- Shoichi Koyama, Ken'ichi Furuya, Yusuke Hiwasaki, Yoichi Haneda:
Analytical Approach to Wave Field Reconstruction Filtering in Spatio-Temporal Frequency Domain. 685-696 - Xiao-Lei Zhang, Ji Wu:
Deep Belief Networks Based Voice Activity Detection. 697-710 - Emmanuel Ravelli, Vinay Melkote, Tejaswi Nanjundaswamy, Kenneth Rose:
Joint Optimization of Base and Enhancement Layers in Scalable Audio Coding. 711-724 - Cemil Demir, Murat Saraclar, Ali Taylan Cemgil:
Single-Channel Speech-Music Separation for Robust ASR With Mixture Models. 725-736 - Athanasia Zlatintsi, Petros Maragos:
Multiscale Fractal Analysis of Musical Instrument Signals With Application to Recognition. 737-748 - Shakeel Ahmed, Muhammad Tahir Akhtar, Xi Zhang:
Robust Auxiliary-Noise-Power Scheduling in Active Noise Control Systems With Online Secondary Path Modeling. 749-761 - Yakun Hu, Dapeng Wu, Antonio Nucci:
Fuzzy-Clustering-Based Decision Tree Approach for Large Population Speaker Identification. 762-774 - Nicki Holighaus, Monika Dörfler, Gino Angelo M. Velasco, Thomas Grill:
A Framework for Invertible, Real-Time Constant-Q Transforms. 775-785 - Sabato Marco Siniscalchi, Torbjørn Svendsen, Chin-Hui Lee:
A Bottom-Up Modular Search Approach to Large Vocabulary Continuous Speech Recognition. 786-797 - Jihoon Park, Kwang-Ki Kim, Minsoo Hahn:
Vocal Removal From Multiobject Audio Using Harmonic Information for Karaoke Service. 798-805 - John Woodruff, DeLiang Wang:
Binaural Detection, Localization, and Segregation in Reverberant Environments Based on Joint Pitch and Azimuth Cues. 806-815 - Carlos Vaquero, Alfonso Ortega, Antonio Miguel, Eduardo Lleida:
Quality Assessment for Speaker Diarization and Its Application in Speaker Characterization. 816-827 - Alireza Masnadi-Shirazi, Bhaskar D. Rao:
An ICA-SCT-PHD Filter Approach for Tracking and Separation of Unknown Time-Varying Number of Sources. 828-841 - Taufiq Hasan, John H. L. Hansen:
Acoustic Factor Analysis for Robust Speaker Verification. 842-853 - Gayadhar Pradhan, S. R. Mahadeva Prasanna:
Speaker Verification by Vowel and Nonvowel Like Segmentation. 854-867 - Shing-Chow Chan, Y. J. Chu, Z. G. Zhang:
A New Variable Regularized Transform Domain NLMS Adaptive Filtering Algorithm - Acoustic Applications and Performance Analysis. 868-878 - Nicolas Ellaham, Christian Giguère, Wail Gueaieb:
Evaluation of the Phase-Inversion Signal Separation Method When Using Nonlinear Hearing Aids. 879-888
Volume 21, Number 5, May 2013
- Guangzhao Bao, Zhongfu Ye, Xu Xu, Yingyue Zhou:
A Compressed Sensing Approach to Blind Separation of Speech Mixture Based on a Two-Layer Sparsity Model. 899-906 - Shing-Chow Chan, Y. J. Chu, Z. G. Zhang, Kai Man Tsui:
A New Variable Regularized QR Decomposition-Based Recursive Least M-Estimate Algorithm - Performance Analysis and Acoustic Applications. 907-922 - Jesper Rindom Jensen, Mads Græsbøll Christensen, Søren Holdt Jensen:
Nonlinear Least Squares Methods for Joint DOA and Pitch Estimation. 923-933 - Sandro Cumani, Pietro Laface:
Memory and Computation Trade-Offs for Efficient I-Vector Extraction. 934-944 - Emanuël A. P. Habets, Jacob Benesty:
A Two-Stage Beamforming Approach for Noise Reduction and Dereverberation. 945-958 - Marco Liuni, Axel Röbel, Ewa Matusiak, Marco Romito, Xavier Rodet:
Automatic Adaptation of the Time-Frequency Resolution for Sound Analysis and Re-Synthesis. 959-970 - Hiroshi Sawada, Hirokazu Kameoka, Shoko Araki, Naonori Ueda:
Multichannel Extensions of Non-Negative Matrix Factorization With Complex-Valued Data. 971-982 - Robert M. Nickel, Ramón Fernandez Astudillo, Dorothea Kolossa, Rainer Martin:
Corpus-Based Speech Enhancement With Uncertainty Modeling and Cepstral Smoothing. 983-997 - Nasser Mohammadiha, Arne Leijon:
Nonnegative HMM for Babble Noise Derived From Speech HMM: Application to Speech Enhancement. 998-1011 - Wei Rao, Man-Wai Mak:
Boosting the Performance of I-Vector Based Speaker Verification via Utterance Partitioning. 1012-1022 - Ramón Fernandez Astudillo, Reinhold Orglmeister:
Computing MMSE Estimates and Residual Uncertainty Directly in the Feature Domain of ASR using STFT Domain Speech Distortion Models. 1023-1034 - Petko Nikolov Petkov, Gustav Eje Henter, W. Bastiaan Kleijn:
Maximizing Phoneme Recognition Accuracy for Enhanced Speech Intelligibility in Noise. 1035-1045 - Maciej Niedzwiecki, Marcin Ciolek:
Elimination of Impulsive Disturbances From Archive Audio Signals Using Bidirectional Processing. 1046-1059 - Li Deng, Xiao Li:
Machine Learning Paradigms for Speech Recognition: An Overview. 1060-1089 - Coskun Mermer, Murat Saraclar, Ruhi Sarikaya:
Improving Statistical Machine Translation Using Bayesian Word Alignment and Gibbs Sampling. 1090-1101 - Xiaodan Zhu, Colin Cherry, Gerald Penn:
A Graph-Partitioning Framework for Aligning Hierarchical Topic Structures to Presentations. 1102-1112 - Romain Serizel, Marc Moonen, Jan Wouters, Søren Holdt Jensen:
Binaural Integrated Active Noise Control and Noise Reduction in Hearing Aids. 1113-1118
Volume 21, Number 6, June 2013
- Emanuël A. P. Habets, Jacob Benesty:
Multi-Microphone Noise Reduction Based on Orthogonal Noise Signal Decompositions. 1123-1133 - Ping Xu, Pascale Fung:
Cross-Lingual Language Modeling for Low-Resource Speech Recognition. 1134-1144 - Dongwen Ying, Yonghong Yan:
Noise Estimation Using a Constrained Sequential Hidden Markov Model in the Log-Spectral Domain. 1145-1157 - Ciprian Chelba, Peng Xu, Fernando Pereira, Thomas Richardson:
Large Scale Distributed Acoustic Modeling With Back-Off ℕ-Grams. 1158-1169 - John Kane, Christer Gobl:
Wavelet Maxima Dispersion for Breathy to Tense Voice Discrimination. 1170-1179 - Bin Zhang, Alex Marin, Brian Hutchinson, Mari Ostendorf:
Learning Phrase Patterns for Text Classification. 1180-1189 - Ilker Bayram, Mustafa E. Kamasak:
A Simple Prior for Audio Signals. 1190-1200 - Lars-Johan Brännmark, Adrian Bahne, Anders Ahlén:
Compensation of Loudspeaker-Room Responses in a Robust MIMO Control Framework. 1201-1216 - Sandro Cumani, Niko Brummer, Lukás Burget, Pietro Laface, Oldrich Plchot, Vasileios Vasilakakis:
Pairwise Discriminative Speaker Verification in the 𝕀-Vector Space. 1217-1227 - Ki-Seung Lee:
Position-Dependent Crosstalk Cancellation Using Space Partitioning. 1228-1239 - Yusuke Hioka, Ken'ichi Furuya, Kazunori Kobayashi, Kenta Niwa, Yoichi Haneda:
Underdetermined Sound Source Separation Using Power Spectrum Density Estimated by Combination of Directivity Gain. 1240-1250 - Benjamin Lecouteux, Georges Linarès, Yannick Estève, Guillaume Gravier:
Dynamic Combination of Automatic Speech Recognition Systems by Driven Decoding. 1251-1260 - Saman Mousazadeh, Israel Cohen:
Voice Activity Detection in Presence of Transient Noise Using Spectral Clustering. 1261-1271 - Hung-yi Lee, Lin-Shan Lee:
Enhanced Spoken Term Detection Using Support Vector Machines and Weighted Pseudo Examples. 1272-1284 - Tom Ko, Brian Mak:
Eigentriphones for Context-Dependent Acoustic Modeling. 1285-1294 - David Rybach, Hermann Ney, Ralf Schlüter:
Lexical Prefix Tree and WFST: A Comparison of Two Dynamic Search Concepts for LVCSR. 1295-1307
Volume 21, Number 7, 2013
- Charles Verron, Philippe-Aubert Gauthier, Jennifer Langlois, Catherine Guastavino:
Spectral and Spatial Multichannel Analysis/Synthesis of Interior Aircraft Sounds. 1317-1329 - Chun-an Chan, Lin-Shan Lee:
Model-Based Unsupervised Spoken Term Detection with Spoken Queries. 1330-1342 - Jingdong Chen, Jacob Benesty:
On the Time-Domain Widely Linear LCMV Filter for Noise Reduction With a Stereo System. 1343-1354 - Ji Ming, Ramji Srinivasan, Danny Crookes, Ayeh Jafari:
CLOSE - A Data-Driven Approach to Speech Separation. 1355-1368 - Masahito Togami, Yohei Kawaguchi, Ryu Takeda, Yasunari Obuchi, Nobuo Nukaga:
Optimized Speech Dereverberation From Probabilistic Perspective for Time Varying Acoustic Transfer Function. 1369-1380 - Yuxuan Wang, DeLiang Wang:
Towards Scaling Up Classification-Based Speech Separation. 1381-1390 - Simon Arberet, Pierre Vandergheynst, Rafael E. Carrillo, Jean-Philippe Thiran, Yves Wiaux:
Sparse Reverberant Audio Source Separation via Reweighted Analysis. 1391-1402 - Shuhua Zhang, Weibei Dou, Huazhong Yang:
MDCT Sinusoidal Analysis for Audio Signals Analysis and Processing. 1403-1414 - Balaji Vasan Srinivasan, Yuancheng Luo, Daniel Garcia-Romero, Dmitry N. Zotkin, Ramani Duraiswami:
A Symmetric Kernel Partial Least Squares Framework for Speaker Recognition. 1415-1423 - Xiaoyan Cai, Wenjie Li:
Ranking Through Clustering: An Integrated Approach to Multi-Document Summarization. 1424-1433 - Stanislaw Gorlow, Joshua D. Reiss:
Model-Based Inversion of Dynamic Range Compression. 1434-1444 - Matthew C. McCallum, Bernard J. Guillemin:
Stochastic-Deterministic MMSE STFT Speech Enhancement With General A Priori Information. 1445-1457 - Ali Hassan, Robert I. Damper, Mahesan Niranjan:
On Acoustic Emotion Recognition: Compensating for Covariate Shift. 1458-1468 - Fei Liu, Yang Liu:
Towards Abstractive Speech Summarization: Exploring Unsupervised and Supervised Approaches for Spoken Utterance Compression. 1469-1480 - Vesa Välimäki, Heidi-Maria Lehtonen, Marko Takanen:
A Perceptual Study on Velvet Noise and Its Variants at Different Pulse Densities. 1481-1488 - Olivier Derrien, Roland Badeau, Gaël Richard:
Parametric Audio Coding With Exponentially Damped Sinusoids. 1489-1501 - Danilo Comminiello, Michele Scarpiniti, Luis Antonio Azpicueta-Ruiz, Jerónimo Arenas-García, Aurelio Uncini:
Functional Link Adaptive Filters for Nonlinear Acoustic Echo Cancellation. 1502-1512 - Shmulik Markovich Golan, Sharon Gannot, Israel Cohen:
Performance of the SDW-MWF With Randomly Located Microphones in a Reverberant Enclosure. 1513-1523 - Stefan Bilbao:
Modeling of Complex Geometries and Boundary Conditions in Finite Difference/Finite Volume Time Domain Room Acoustics Simulation. 1524-1533
Volume 21, Number 8, August 2013
- Constantin Paleologu, Jacob Benesty, Silviu Ciochina:
Study of the General Kalman Filter for Echo Cancellation. 1539-1549 - Anaïk Olivero, Bruno Torrésani, Richard Kronland-Martinet:
A Class of Algorithms for Time-Frequency Multiplier Estimation. 1550-1559 - Olaf Schleusing, Tomi Kinnunen, Brad H. Story, Jean-Marc Vesin:
Joint Source-Filter Optimization for Accurate Vocal Tract Estimation Using Differential Evolution. 1560-1572 - Dumidu S. Talagala, Wen Zhang, Thushara D. Abhayapala:
Broadband DOA Estimation Using Sensor Arrays on Complex-Shaped Rigid Bodies. 1573-1585 - Jiajun Zhang, Feifei Zhai, Chengqing Zong:
Syntax-Based Translation With Bilingually Lexicalized Synchronous Tree Substitution Grammars. 1586-1597 - Chang Woo Han, Shin Jae Kang, Nam Soo Kim:
Reverberation and Noise Robust Feature Compensation Based on IMM. 1598-1611 - Anoop Deoras, Gökhan Tür, Ruhi Sarikaya, Dilek Hakkani-Tür:
Joint Discriminative Decoding of Words and Semantic Tags for Spoken Language Understanding. 1612-1621 - Ville Hautamäki, Tomi Kinnunen, Filip Sedlak, Kong-Aik Lee, Bin Ma, Haizhou Li:
Sparse Classifier Fusion for Speaker Verification. 1622-1631 - Sarthak Khanal, Harvey F. Silverman, Rahul R. Shakya:
A Free-Source Method (FrSM) for Calibrating a Large-Aperture Microphone Array. 1632-1639 - Volker Leutnant, Alexander Krueger, Reinhold Haeb-Umbach:
Bayesian Feature Enhancement for Reverberation and Noise Robust Speech Recognition. 1640-1652 - Enzo De Sena, Hüseyin Hacihabiboglu, Zoran Cvetkovic:
Analysis and Design of Multichannel Systems for Perceptual Sound Field Reconstruction. 1653-1665 - Marcelo F. Caetano, Xavier Rodet:
Musical Instrument Sound Morphing Guided by Perceptually Motivated Features. 1666-1675 - Bin Cheng, Christian H. Ritz, Ian S. Burnett, Xiguang Zheng:
A General Compression Approach to Multi-Channel Three-Dimensional Audio. 1676-1688 - Weifeng Li, Longbiao Wang, Yicong Zhou, Hervé Bourlard, Qingmin Liao:
Robust Log-Energy Estimation and its Dynamic Change Enhancement for In-car Speech Recognition. 1689-1698 - Alexey Ozerov, Antoine Liutkus, Roland Badeau, Gaël Richard:
Coding-Based Informed Source Separation: Nonnegative Tensor Factorization Approach. 1699-1712 - David Imseng, Hervé Bourlard, John Dines, Philip N. Garner, Mathew Magimai-Doss:
Applying Multi- and Cross-Lingual Stochastic Phone Space Transformations to Non-Native Speech Recognition. 1713-1726 - Eleftheria Georganti, Tobias May, Steven van de Par, John Mourjopoulos:
Sound Source Distance Estimation in Rooms based on Statistical Properties of Binaural Signals. 1727-1741 - Michael Wohlmayr, Franz Pernkopf:
Model-Based Multiple Pitch Tracking Using Factorial HMMs: Model Adaptation and Inference. 1742-1754 - Ron M. Hecht, Elad Noor, Gil Dobry, Yaniv Zigel, Aharon Bar-Hillel, Naftali Tishby:
Effective Model Representation by Information Bottleneck Principle. 1755-1759 - Luis Weruaga, Leonid Dimitrov:
The Spectral Nature of Maximum Likelihood Noise Compensated Linear Prediction. 1760-1765
Volume 21, Number 9, September 2013
- Zhanyu Ma, Arne Leijon, W. Bastiaan Kleijn:
Vector quantization of LSF parameters with a mixture of dirichlet distributions. 1777-1790 - Liang Lu, K. K. Chin, Arnab Ghoshal, Stephen Renals:
Joint Uncertainty Decoding for Noise Robust Subspace Gaussian Mixture Models. 1791-1804 - Dimitrios Giannoulis, Anssi Klapuri:
Musical Instrument Recognition in Polyphonic Audio Using Missing Feature Approach. 1805-1817 - Freddy William, Abhijeet Sangwan, John H. L. Hansen:
Automatic Accent Assessment Using Phonetic Mismatch and Human Perception. 1818-1829 - Stanislaw Andrzej Raczynski, Emmanuel Vincent, Shigeki Sagayama:
Dynamic Bayesian Networks for Symbolic Polyphonic Pitch Modeling. 1830-1840 - Raymond W. M. Ng, Tan Lee, Cheung-Chi Leung, Bin Ma, Haizhou Li:
Spoken Language Recognition With Prosodic Features. 1841-1853 - Benoit Fuentes, Roland Badeau, Gaël Richard:
Harmonic Adaptive Latent Component Analysis of Audio and Application to Music Transcription. 1854-1866 - Jose Manuel Gil-Cacho, Marco Signoretto, Toon van Waterschoot, Marc Moonen, Søren Holdt Jensen:
Nonlinear Acoustic Echo Cancellation Based on a Sliding-Window Leaky Kernel Affine Projection Algorithm. 1867-1878 - Ina Kodrasi, Stefan Goetze, Simon Doclo:
Regularization for Partial Multichannel Equalization for Speech Dereverberation. 1879-1890 - Roman Scharrer, Michael Vorländer:
Sound Field Classification in Small Microphone Arrays Using Spatial Coherences. 1891-1899 - Muhammad Salman Khan, Syed M. Naqvi, Ata ur-Rehman, Wenwu Wang, Jonathon A. Chambers:
Video-Aided Model-Based Source Separation in Real Reverberant Rooms. 1900-1912 - Mehrez Souden, Shoko Araki, Keisuke Kinoshita, Tomohiro Nakatani, Hiroshi Sawada:
A Multichannel MMSE-Based Framework for Speech Source Separation and Noise Reduction. 1913-1928 - Matías Zanartu, Julio C. Ho, Daryush D. Mehta, Robert E. Hillman, George R. Wodicka:
Subglottal Impedance-Based Inverse Filtering of Voiced Sounds Using Neck Surface Acceleration. 1929-1939 - Alex Southern, Samuel Siltanen, Damian T. Murphy, Lauri Savioja:
Room Impulse Response Synthesis and Validation Using a Hybrid Acoustic Model. 1940-1952 - Futoshi Asano, Hideki Asoh, Kazuhiro Nakadai:
Sound Source Localization Using Joint Bayesian Estimation With a Hierarchical Noise Model. 1953-1965 - Neil Wachowski, Mahmood R. Azimi-Sadjadi:
Characterization of Multiple Transient Acoustical Sources From Time-Transform Representations. 1966-1978 - Cheng-Yuan Chang, Sen M. Kuo:
Complete Parallel Narrowband Active Noise Control Systems. 1979-1986
Volume 21, Number 10, October 2013
- William Hartmann, Arun Narayanan, Eric Fosler-Lussier, DeLiang Wang:
A Direct Masking Approach to Robust ASR. 1993-2005 - Yow-Bang Wang, Shang-wen Li, Lin-Shan Lee:
An Experimental Analysis on Integrating Multi-Stream Spectro-Temporal, Cepstral and Pitch Information for Mandarin Speech Recognition. 2006-2014 - Stephen Shum, Najim Dehak, Réda Dehak, James R. Glass:
Unsupervised Methods for Speaker Diarization: An Integrated and Iterative Approach. 2015-2028 - Zbynek Koldovský, Jirí Málek, Petr Tichavský, Francesco Nesta:
Semi-Blind Noise Extraction Using Partially Known Position of the Target Source. 2029-2041 - Mads Græsbøll Christensen:
Accurate Estimation of Low Fundamental Frequencies From Real-Valued Measurements. 2042-2056 - Philippe Esling, Carlos Agón:
Multiobjective Time Series Matching for Audio Classification and Retrieval. 2057-2072 - Chao Zhang, Yi Liu, Yunqing Xia, Xuan Wang, Chin-Hui Lee:
Reliable Accent-Specific Unit Generation With Discriminative Dynamic Gaussian Mixture Selection for Multi-Accent Chinese Speech Recognition. 2073-2084 - Gilles Degottex, Yannis Stylianou:
Analysis and Synthesis of Speech Using an Adaptive Full-Band Harmonic Model. 2085-2095 - Bilei Zhu, Wei Li, Ruijiang Li, Xiangyang Xue:
Multi-Stage Non-Negative Matrix Factorization for Monaural Singing Voice Separation. 2096-2107 - Sadao Hiroya:
Non-Negative Temporal Decomposition of Speech Parameters by Multiplicative Update Rules. 2108-2117 - Cyril Joder, Slim Essid, Gaël Richard:
Learning Optimal Features for Polyphonic Audio-to-Score Alignment. 2118-2128 - Zhen-Hua Ling, Li Deng, Dong Yu:
Modeling Spectral Envelopes Using Restricted Boltzmann Machines and Deep Belief Networks for Statistical Parametric Speech Synthesis. 2129-2139 - Nasser Mohammadiha, Paris Smaragdis, Arne Leijon:
Supervised and Unsupervised Speech Enhancement Using Nonnegative Matrix Factorization. 2140-2151 - Sabato Marco Siniscalchi, Jinyu Li, Chin-Hui Lee:
Hermitian Polynomial for Speaker Adaptation of Connectionist Speech Recognition Systems. 2152-2161 - Nikolay D. Gaubitch, Mike Brookes, Patrick A. Naylor:
Blind Channel Magnitude Response Estimation in Speech Using Spectrum Classification. 2162-2171 - Masayuki Suzuki, Takuya Yoshioka, Shinji Watanabe, Nobuaki Minematsu, Keikichi Hirose:
Feature Enhancement With Joint Use of Consecutive Corrupted and Noise Feature Vectors With Discriminative Region Weighting. 2172-2181 - Takuya Yoshioka, Tomohiro Nakatani:
Noise Model Transfer: Novel Approach to Robustness Against Nonstationary Noise. 2182-2192 - Despoina Pavlidi, Anthony Griffin, Matthieu Puigt, Athanasios Mouchtaris:
Real-Time Multiple Sound Source Localization and Counting Using a Circular Microphone Array. 2193-2206 - Sefki Kolozali, Mathieu Barthet, György Fazekas, Mark B. Sandler:
Automatic Ontology Generation for Musical Instruments Based on Audio Analysis. 2207-2220
Volume 21, Number 11, November 2013
- Stephen J. Wright, Dimitri Kanevsky, Li Deng, Xiaodong He, Georg Heigold, Haizhou Li:
Optimization Algorithms and Applications for Speech and Language Processing. 2231-2243 - Gillian M. Chin, Jorge Nocedal, Peder A. Olsen, Steven J. Rennie:
Second Order Methods for Optimizing Convex Matrix Functions and Sparse Covariance Clustering. 2244-2254 - Theodoros Tsiligkaridis, Etienne Marcheret, Vaibhava Goel:
A Difference of Convex Functions Approach to Large-Scale Log-Linear Model Estimation. 2255-2266 - Tara N. Sainath, Brian Kingsbury, Hagen Soltau, Bhuvana Ramabhadran:
Optimization Techniques to Improve Training Speed of Deep Neural Networks for Large Speech Tasks. 2267-2276 - Tuomas Virtanen, Jort Florent Gemmeke, Bhiksha Raj:
Active-Set Newton Algorithm for Overcomplete Non-Negative Representations of Audio. 2277-2289 - Patrick Cardinal, Pierre Dumouchel, Gilles Boulianne:
Large Vocabulary Speech Recognition on Parallel Architectures. 2290-2300 - Rémi Mignot, Laurent Daudet, François Ollivier:
Room Reverberation Reconstruction: Interpolation of the Early Part Using Compressed Sensing. 2301-2312 - Min Zhang, Wenliang Chen, Xiangyu Duan, Rong Zhang:
Improving Graph-Based Dependency Parsing Models With Dependency Language Models. 2313-2323 - Florian Pflug, Tim Fingscheidt:
Robust Ultra-Low Latency Soft-Decision Decoding of Linear PCM Audio. 2324-2336 - Koji Seto, Tokunbo Ogunfunmi:
Scalable Speech Coding for IP Networks: Beyond iLBC. 2337-2345 - Kenta Niwa, Yusuke Hioka, Ken'ichi Furuya, Yoichi Haneda:
Diffused Sensing for Sharp Directive Beamforming. 2346-2355 - Symeon Delikaris-Manias, Ville Pulkki:
Cross Pattern Coherence Algorithm for Spatial Filtering Applications Utilizing Microphone Arrays. 2356-2367 - Bai Ying Lei, Ing Yann Soon, Ee-Leng Tan:
Robust SVD-Based Audio Watermarking Scheme With Differential Evolution Optimization. 2368-2378 - Nikos Malandrakis, Alexandros Potamianos, Elias Iosif, Shrikanth S. Narayanan:
Distributional Semantic Models for Affective Text Analysis. 2379-2392 - Pasi Pertilä, Matti S. Hämäläinen, Mikael Mieskolainen:
Passive Temporal Offset Estimation of Multichannel Recordings of an Ad-Hoc Microphone Array. 2393-2402 - Liang Vincent Wang, Woon-Seng Gan, Andy W. H. Khong, Sen M. Kuo:
Convergence Analysis of Narrowband Feedback Active Noise Control System With Imperfect Secondary Path Estimation. 2403-2411 - Chi-Man Pun, Xiaochen Yuan:
Robust Segments Detector for De-Synchronization Resilient Audio Watermarking. 2412-2424 - Miranti Indar Mandasari, Rahim Saeidi, Mitchell McLaren, David A. van Leeuwen:
Quality Measure Functions for Calibration of Speaker Recognition Systems in Various Duration Conditions. 2425-2438 - Fabian Triefenbach, Azarakhsh Jalalvand, Kris Demuynck, Jean-Pierre Martens:
Acoustic Modeling With Hierarchical Reservoirs. 2439-2450 - Donghyeon Lee, Minwoo Jeong, Kyungduk Kim, Seonghan Ryu, Gary Geunbae Lee:
Unsupervised Spoken Language Understanding for a Multi-Domain Dialog System. 2451-2464
Volume 21, Number 12, December 2013
- A. P. Prathosh, T. V. Ananthapadmanabha, A. G. Ramakrishnan:
Epoch Extraction Based on Integrated Linear Prediction Residual Using Plosion Index. 2471-2480 - Tomas Dekens, Werner Verhelst:
Body Conducted Speech Enhancement by Equalization and Signal Fusion. 2481-2492 - Dejan Markovic, Fabio Antonacci, Augusto Sarti, Stefano Tubaro:
Soundfield Imaging in the Ray Space. 2493-2505 - Partha Lal, Simon King:
Cross-Lingual Automatic Speech Recognition Using Tandem Features. 2506-2515 - Tomohiro Nakatani, Shoko Araki, Takuya Yoshioka, Marc Delcroix, Masakiyo Fujimoto:
Dominance Based Integration of Spatial and Spectral Features for Speech Enhancement. 2516-2531 - Yotam Peled, Boaz Rafaely:
Linearly-Constrained Minimum-Variance Method for Spherical Microphone Arrays Based on Plane-Wave Decomposition of the Sound Field. 2532-2540 - Jean-Louis Durrieu, Jean-Philippe Thiran:
Source/Filter Factorial Hidden Markov Model, With Application to Pitch and Formant Tracking. 2541-2553 - Katherine Ellis, Emanuele Coviello, Antoni B. Chan, Gert R. G. Lanckriet:
A Bag of Systems Representation for Music Auto-Tagging. 2554-2569 - Aroor Dinesh Dileep, C. Chandra Sekhar:
HMM Based Intermediate Matching Kernel for Classification of Sequential Patterns of Speech Using Support Vector Machines. 2570-2582 - Oliver Thiergart, Giovanni Del Galdo, Maja Taseska, Emanuël A. P. Habets:
Geometry-Based Spatial Sound Acquisition Using Distributed Microphone Arrays. 2583-2594 - Jesper Rindom Jensen, Jacob Benesty, Mads Græsbøll Christensen, Jingdong Chen:
A Class of Optimal Rectangular Filtering Matrices for Single-Channel Signal Enhancement in the Time Domain. 2595-2606 - Yizhao Ni, Matt McVicar, Raúl Santos-Rodríguez, Tijl De Bie:
Understanding Effects of Subjectivity in Measuring Chord Estimation Accuracy. 2607-2615 - Georg Heigold, Hermann Ney, Ralf Schlüter:
Investigations on an EM-Style Optimization Algorithm for Discriminative Training of HMMs. 2616-2626 - Bruno Defraene, Naim Mansour, Steven De Hertogh, Toon van Waterschoot, Moritz Diehl, Marc Moonen:
Declipping of Audio Signals Using Perceptual Compressed Sensing. 2627-2637
manage site settings
To protect your privacy, all features that rely on external API calls from your browser are turned off by default. You need to opt-in for them to become active. All settings here will be stored as cookies with your web browser. For more information see our F.A.Q.
last updated on 2025-08-07 23:35 CEST by the dblp team
all metadata released as open data under CC0 1.0 license
see also: Terms of Use | Privacy Policy | Imprint
dblp was originally created in 1993 at:
since 2018, dblp has been operated and maintained by:
the dblp computer science bibliography is funded and supported by:
