Table of Contents

• HMM-based artificial bandwidth extension supported by neural networksPatrick Bauer, Johannes Abel, Tim Fingscheidt. 1-5 [doi]
• Unbiased coherent-to-diffuse ratio estimation for dereverberationAndreas Schwarz, Walter Kellermann. 6-10 [doi]
• A posteriori speech presence probability estimation based on averaged observations and a super-Gaussian speech modelBalázs Fodor, Timo Gerkmann. 11-15 [doi]
• An adaptive microphone array topology for target signal extraction with humanoid robotsHendrik Barfuss, Walter Kellermann. 16-20 [doi]
• On the statistics and the detection of multichannel common zerosGerald Enzner, Philipp Thüne. 21-25 [doi]
• The ABCIT research platformKamil Adiloglu, Tobias Herzke, Volker Hohmann, Matthieu Recugnat, Martin Besnard, Teng Huang, Bradford Backus. 26-29 [doi]
• LPC-based speech dereverberation using Kalman-EM algorithmBoaz Schwartz, Sharon Gannot, Emanuel A. P. Habets. 30-34 [doi]
• Post-filter design for speech enhancement in various noisy environmentsKenta Niwa, Yusuke Hioka, Kazunori Kobayashi. 35-39 [doi]
• The single- and multichannel audio recordings database (SMARD)Jesper Kjaer Nielsen, Jesper Rindom Jensen, Søren Holdt Jensen, Mads Græsbøll Christensen. 40-44 [doi]
• Estimation of the common part of acoustic feedback paths in hearing aids using iterative quadratic programmingHenning F. Schepker, Simon Doclo. 45-49 [doi]
• Speech dereverberation with convolutive transfer function approximation using map and variational deconvolution approachesAnte Jukic, Toon van Waterschoot, Timo Gerkmann, Simon Doclo. 50-54 [doi]
• An improved non-intrusive intelligibility metric for noisy and reverberant speechJoão Felipe Santos, Mohammed Senoussaoui, Tiago H. Falk. 55-59 [doi]
• Wave-domain canceling of residual echo with subspace trackingSatoru Emura, Hitoshi Ohmuro. 60-64 [doi]
• A robust howling detection algorithm based on a statistical approachJoachim Flocon-Cholet, Julien Faure, Alexandre Guérin, Pascal Scalart. 65-69 [doi]
• Sinusoidal interpolation across missing dataW. Bastiaan Kleijn, Turaj Zakizadeh Shabestary, Jan Skoglund. 70-74 [doi]
• Optimal beamforming as a time domain equalization problem with application to room acousticsMark R. P. Thomas, Ivan J. Tashev, Felicia Lim, Patrick A. Naylor. 75-79 [doi]
• Speaker dependent speech enhancement using sinusoidal modelPejman Mowlaee, Christian Nachbar. 80-84 [doi]
• PSD estimation in beamspace for source separation in a diffuse noise fieldYusuke Hioka, Kenta Niwa. 85-88 [doi]
• Speech reinforcement with a globally optimized perceptual distortion measure for noisy reverberant channelsJoao B. Crespo, Richard C. Hendriks. 89-93 [doi]
• On near-field beamforming with smartphone-based ad-hoc microphone arraysNikolay D. Gaubitch, Jorge Martinez, W. Bastiaan Kleijn, Richard Heusdens. 94-98 [doi]
• A discriminative learning approach to probabilistic acoustic source localizationHendrik Kayser, Jörn Anemüller. 99-103 [doi]
• Estimation of time-variant acoustic feedback paths in in-car communication systemsJochen Withopf, Gerhard Schmidt. 104-108 [doi]
• An acoustical zoom based on informed spatial filteringOliver Thiergart, Konrad Kowalczyk, Emanuel A. P. Habets. 109-113 [doi]
• Identification of surface acoustic impedances in a reverberant room using the FDTD methodNiccolo Antonello, Toon van Waterschoot, Marc Moonen, Patrick A. Naylor. 114-118 [doi]
• Statistical modelling of multichannel blind system identification errorsFelicia Lim, Patrick A. Naylor. 119-123 [doi]
• Multichannel dereverberation for hearing aids with interaural coherence preservationSebastian Braun, Matteo Torcoli, Daniel Marquardt, Emanuel A. P. Habets, Simon Doclo. 124-128 [doi]
• Spatial perception of virtual X-Y recordingsKonrad Kowalczyk, Alexandra Craciun, Christian Dachmann, Emanuel A. P. Habets. 129-133 [doi]
• Characterisation and modelling of non-linear loudspeakersLeela K. Gudupudi, Christophe Beaugeant, Nicholas W. D. Evans. 134-138 [doi]
• Joint dereverberation and noise reduction based on acoustic multichannel equalizationIna Kodrasi, Simon Doclo. 139-143 [doi]
• A wind-noise suppressor based on wind-onset detection and spectral gain modificationMasanori Kato, Akihiko Sugiyama. 144-148 [doi]
• Generalized amplitude interpolation by β-divergence for virtual microphone arrayHiroki Katahira, Nobutaka Ono, Shigeki Miyabe, Takeshi Yamada, Shoji Makino. 149-153 [doi]
• Generalization of supervised learning for binary mask estimationTobias May, Timo Gerkmann. 154-158 [doi]
• A computationally constrained optimization framework for implementation and tuning of speech enhancement systemsDaniele Giacobello, Jason Wung, Ramin Pichevar, Joshua Atkins. 159-163 [doi]
• Spectral tilt modelling with extrapolated GMMs for intelligibility enhancement of narrowband telephone speechEmma Jokinen, Ulpu Remes, Marko Takanen, Kalle Palomäki, Mikko Kurimo, Paavo Alku. 164-168 [doi]
• Near-field source extraction using speech presence probabilities for ad hoc microphone arraysMaja Taseska, Shmulik Markovich Golan, Emanuel A. P. Habets, Sharon Gannot. 169-173 [doi]
• Multichannel adaptive filtering in compressive domainsKarim Helwani, Herbert Buchner. 174-177 [doi]
• Numerical formulae for TOA-based microphone and source localizationL. E. Trung-Kien, Nobutaka Ono. 178-182 [doi]
• Blind synchronization in wireless sensor networks with application to speech enhancementDani Cherkassky, Sharon Gannot. 183-187 [doi]
• Validation of realistic acoustic environments for listening tests using directional hearing aidsChris Oreinos, Jörg M. Buchholz. 188-192 [doi]
• Noise coloration filter design by pole-zero placementAlexis Favrot. 193-197 [doi]
• On semi-blind estimation of echo paths during double-talk based on nonstationarityZbynek Koldovský, Jirí Málek, Michael Muller, Petr Tichavskjy. 198-202 [doi]
• Amplitude-based speech enhancement with nonnegative matrix factorization for asynchronous distributed recordingHironobu Chiba, Nobutaka Ono, Shigeki Miyabe, Yu Takahashi, Takeshi Yamada, Shoji Makino. 203-207 [doi]
• Alternative formulation and robustness analysis of the multichannel wiener filter for spatially distributed microphonesToby Christian Lawin-Ore, Sebastian Stenzel, Jürgen Freudenberger, Simon Doclo. 208-212 [doi]
• Towards online source counting in speech mixtures applying a variational EM for complex Watson mixture modelsLukas Drude, Aleksej Chinaev, Dang Hai Tran Vu, Reinhold Häb-Umbach. 213-217 [doi]
• Low-complexity noise power spectral density estimation for harsh automobile environmentsChristin Baasch, Vasudev Kandade Rajan, Mohamed Krini, Gerhard Schmidt. 218-222 [doi]
• Fast noise PSD estimation based on blind channel identificationMasoumeh Azarpour, Gerald Enzner. 223-227 [doi]
• Online unsupervised overlapping speaker detection using enhanced classification history-based featuresYoussef Oualil, Rahil Mahdian Toroghi, Dietrich Klakow. 228-232 [doi]
• A study on speech quality and speech intelligibility measures for quality assessment of single-channel dereverberation algorithmsStefan Goetze, Anna Warzybok, Ina Kodrasi, Jan Ole Jungmann, Benjamin Cauchi, Jan Rennies, Emanuel A. P. Habets, Alfred Mertins, Timo Gerkmann, Simon Doclo, Birger Kollmeier. 233-237 [doi]
• Voice activity detection in transient noise environment using Laplacian pyramid algorithmNurit Spingarn, Saman Mousazadeh, Israel Cohen. 238-242 [doi]
• Geometry calibration of multiple microphone arrays in highly reverberant environmentsAxel Plinge, Gernot A. Fink. 243-247 [doi]
• An automatic model-building algorithm for sparse approximation of room impulse responses with Orthonormal Basis FunctionsGiacomo Vairetti, Toon van Waterschoot, Marc Moonen, Michael Catrysse, Søren Holdt Jensen. 248-252 [doi]
• A new structure for acoustic echo cancellation in double-talk scenario using auxiliary filterMahfoud Hamidia, Abderrahmane Amrouche. 253-257 [doi]
• Multiple source localisation in the spherical harmonic domainChristine Evers, Alastair H. Moore, Patrick A. Naylor. 258-262 [doi]
• Acoustic modeling based on early-to-late reverberation ratio for robust ASRMarco Matassoni, Alessio Brutti, Piergiorgio Svaizer. 263-267 [doi]
• Relaxed disjointness based clustering for joint blind source separation and dereverberationNobutaka Ito, Shoko Araki, Takuya Yoshioka, Tomohiro Nakatani. 268-272 [doi]
• Reverberant audio source separation using partially pre-trained nonnegative matrix factorizationMahmoud Fakhry, Piergiorgio Svaizer, Maurizio Omologo. 273-277 [doi]
• Investigation of self-masking effects for the evaluation of in-car communication systemsAnne Theib, Gerhard Schmidt. 278-282 [doi]
• Single channel noise reduction based on an auditory filterbankSteffen Kortlang, Stephan Dieter Ewert, Timo Gerkmann. 283-287 [doi]
• Optimal binaural LCMV beamformers for combined noise reduction and binaural cue preservationDaniel Marquardt, Elior Hadad, Sharon Gannot, Simon Doclo. 288-292 [doi]
• A reduced-rank approach to single-channel noise reductionWei Zhang, Jingdong Chen, Jacob Benesty. 293-297 [doi]
• A quantitative comparison of blind C50 estimatorsP. Peso Parada, D. Sharma, J. Lainez, D. Barreda, Patrick A. Naylor, Toon van Waterschoot. 298-302 [doi]
• On the performance of widely linear quaternion based MVDR beamformer for an acoustic vector sensorJiuwen Cao, Andy W. H. Khong, Sharon Gannot. 303-307 [doi]
• STSP: Space-time stretched pulse for measuring spatio-temporal impulse responseShoichi Koyama, Prakhar Srivastava, Ken'ichi Furuya, Suehiro Shimauchi, Hitoshi Ohmuro. 308-312 [doi]
• Multichannel audio database in various acoustic environmentsElior Hadad, Florian Heese, Peter Vary, Sharon Gannot. 313-317 [doi]
• Traffic monitoring with ad-hoc microphone arrayTakuya Toyoda, Nobutaka Ono, Shigeki Miyabe, Takeshi Yamada, Shoji Makino. 318-322 [doi]
• The acoustic echo cancelation using blind source separation to reduce double-talk interferenceYoshihiro Sakai, Muhammad Tahir Akhtar. 323-326 [doi]
• Measurement, analysis and simulation of wind noise signals for mobile communication devicesChristoph Matthias Nelke, Peter Vary. 327-331 [doi]
• Subjective speech quality and speech intelligibility evaluation of single-channel dereverberation algorithmsAnna Warzybok, Ina Kodrasi, Jan Ole Jungmann, Emanuel A. P. Habets, Timo Gerkmann, Alfred Mertins, Simon Doclo, Birger Kollmeier, Stefan Goetze. 332-336 [doi]
• Time-frequency constraints for phase estimation in single-channel speech enhancementPejman Mowlaee, Rahim Saeidi. 337-341 [doi]
• 2.5D sound field reproduction in higher order AmbisonicsWen Zhang, Thushara D. Abhayapala. 342-346 [doi]