Table of Contents

• Ad-hoc microphone array calibration from partial distance measurementsMohammad Javad Taghizadeh, Afsaneh Asaei, Philip N. Garner, Hervé Bourlard. 1-5 [doi]
• Kernel spectrogram models for source separationAntoine Liutkus, Zafar Rafii, Bryan Pardo, Derry Fitzgerald, Laurent Daudet. 6-10 [doi]
• Supervised non-euclidean sparse NMF via bilevel optimization with applications to speech enhancementPablo Sprechmann, Alexander M. Bronstein, Guillermo Sapiro. 11-15 [doi]
• Content-adaptive speech enhancement by a sparsely-activated dictionary plus low rank decompositionZhuo Chen, Hélène Papadopoulos, Daniel P. W. Ellis. 16-20 [doi]
• The self-taught vocal interfaceJort F. Gemmeke. 21-22 [doi]
• Speech dereverberation with multi-channel linear prediction and sparse priors for the desired signalAnte Jukic, Toon van Waterschoot, Timo Gerkmann, Simon Doclo. 23-26 [doi]
• Transient noise reduction using nonnegative matrix factorizationNasser Mohammadiha, Simon Dodo. 27-31 [doi]
• A fast phoneme recognition system based on sparse representation of test utterancesArmin Saeb, Farbod Razzazi, Massoud Babaie-Zadeh. 32-36 [doi]
• Discriminativetensor dictionaries and sparsity for speaker identificationS. Zubair, W. Wang, J. A. Chambers. 37-41 [doi]
• Far-field criterion for spherical microphone arrays and directional sourcesHai Morgenstern, Boaz Rafaely. 42-46 [doi]
• Adaptive interference rejection using generalized sidelobe canceller in spherical harmonics domainJounghoon Beh, Dmitry N. Zotkin, Ramani Duraiswami. 47-51 [doi]
• Adaptive beamformer for spherical eigenbeamforming microphone arraysGary W. Elko, Jens Meyer. 52-56 [doi]
• Methods to learn bank of filters steering nulls toward potential positions of a target sourceJirí Málek, David Botha, Zbynek Koldovský, Sharon Gannot. 57-61 [doi]
• Tuning methodology for speech enhancement algorithms using a simulated conversational database and perceptual objective measuresDaniele Giacobello, Jason Wung, Ramin Pichevar, Joshua Atkins. 62-66 [doi]
• Improved hands-free automatic speech recognition in reverberant environment conditionRandy Gomez, Keisuke Nakamura, Takeshi Mizumoto, Kazuhiro Nakadai. 67-71 [doi]
• Multiple acoustic sources localization using distributed expectation-maximization algorithmYuval Dorfan, Gershon Hazan, Sharon Gannot. 72-76 [doi]
• Study of a generalized spherical array beamformer with adjustable binaural reproductionMichael Jeffet, Boaz Rafaely. 77-81 [doi]
• Near-field source localization using spherical microphone arrayLalan Kumar, Kushagra Singhal, Rajesh M. Hegde. 82-86 [doi]
• Short-time multichannel noise correlation matrix estimators for acoustic signalsJonathan Blanchette, Martin Bouchard. 87-91 [doi]
• Divergence optimization in nonnegative matrix factorization with spectrogram restoration for multichannel signal separationDaichi Kitamura, Hiroshi Saruwatari, Satoshi Nakamura, Yu Takahashi, Kazunobu Kondo, Hirokazu Kameoka. 92-96 [doi]
• Circular microphone array with tangential pressure gradient sensorsFalk-Martin Hoffmann, Filippo Maria Fazi. 97-101 [doi]
• A GPU-accelerated real-time implementation of TRINICON-BSS for multiple separation unitsCraig A. Anderson, Stefan Meier, Walter Kellermann, Paul D. Teal, Mark A. Poletti. 102-106 [doi]
• An auxiliary-function approach to online independent vector analysis for real-time blind source separationToru Taniguchi, Nobutaka Ono, Akinori Kawamura, Shigeki Sagayama. 107-111 [doi]
• A comparison of different loudspeaker models to empirically estimated non-linearitiesLeela K. Gudupudi, Christophe Beaugeant, Nicholas W. D. Evans, Moctar Mossi Mossi, Ludovick Lepauloux. 112-116 [doi]
• Spectrogram patch based acoustic event detection and classification in speech overlapping conditionsMiquel Espi, Masakiyo Fujimoto, Yotaro Kubo, Tomohiro Nakatani. 117-121 [doi]
• Theoretical analysis of biased MMSE short-time spectral amplitude estimator and its extension to musical-noise-free speech enhancementShunsuke Nakai, Hiroshi Saruwatari, Ryoichi Miyazaki, Satoshi Nakamura, Kazunobu Kondo. 122-126 [doi]
• A Minimum variance beamformer for spatially distributed microphones using a soft reference selectionSebastian Stenzel, Jürgen Freudenberger, Gerhard Schmidt. 127-131 [doi]
• A hierarchical approach for the online, on-board detection and localisation of brake squeal using microphone arraysN. Madhu, R. Martin, H.-W. Rehn, S. Gergen, A. Fischer. 132-136 [doi]
• Spatial aliasing-cancellation for circular microphone arraysDavid L. Alon, Boaz Rafaely. 137-141 [doi]
• Extraction of pinna spectral notches in the median plane of a virtual spherical microphone arrayAnkit Sohni, Chaitanya Ahuja, Rajesh M. Hegde. 142-146 [doi]
• Utilizing motion in humanoid robots to enhance spatial information recordedby microphone arraysVladimir Tourbabin, Boaz Rafaely. 147-151 [doi]
• Self-localization of wireless acoustic sensors in meeting roomsMikko Parviainen, Pasi Pertilä, Matti S. Hämäläinen. 152-156 [doi]
• A speech event detection and localization task for multiroom environmentsAlessio Brutti, Mirco Ravanelli, Piergiorgio Svaizer, Maurizio Omologo. 157-161 [doi]
• Ensemble integration of calibrated speaker localization and statistical speech detection in domestic environmentsYuuki Tachioka, Tomohiro Narita, Shinji Watanabe, Jonathan Le Roux. 162-166 [doi]
• The Athena-RC system for speech activity detection and speaker localization in the DIRHA smart homePanagiotis Giannoulis, Antigoni Tsiami, Isidoros Rodomagoulakis, Athanasios Katsamanis, Gerasimos Potamianos, Petros Maragos. 167-171 [doi]
• Neural networks for distant speech recognitionSteve Renals, Pawel Swietojanski. 172-176 [doi]
• Efficient training of acoustic models for reverberation-robust medium-vocabulary automatic speech recognitionArmin Sehr, Hendrik Barfuss, Christian Hofmann, Roland Maas, Walter Kellermann. 177-181 [doi]
• Optimized joint noise suppression and dereverberation based on blind signal extraction for hands-free speech recognition systemFine Dwinita Aprilyanti, Hiroshi Saruwatari, Satoshi Nakamura, Tomoya Takatani. 182-186 [doi]
• Word boundary agreementto combine multi-microphone hypotheses in distant speech recognitionCristina Guerrero, Maurizio Omologo. 187-191 [doi]
• Investigating stranded GMM for improving automatic speech recognitionArseniy Gorin, Denis Jouvet, Emmanuel Vincent, Dung T. Tran. 192-196 [doi]
• Exploring deep neural networks and deep autoencoders in reverberant speech recognitionMasato Mimura, Shinsuke Sakai, Tatsuya Kawahara. 197-201 [doi]