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ORCIDIWAENC 2014: Juan-les-Pins, France
- 14th International Workshop on Acoustic Signal Enhancement, IWAENC 2014, Juan-les-Pins, France, September 8-11, 2014. IEEE 2014, ISBN 978-1-4799-6808-4
- Patrick Bauer, Johannes Abel
, Tim Fingscheidt:
HMM-based artificial bandwidth extension supported by neural networks. 1-5 - Andreas Schwarz, Walter Kellermann:
Unbiased coherent-to-diffuse ratio estimation for dereverberation. 6-10 - Balázs Fodor, Timo Gerkmann:
A posteriori speech presence probability estimation based on averaged observations and a super-Gaussian speech model. 11-15 - Hendrik Barfuss, Walter Kellermann:
An adaptive microphone array topology for target signal extraction with humanoid robots. 16-20 - Gerald Enzner, Philipp Thüne:
On the statistics and the detection of multichannel common zeros. 21-25 - Kamil Adiloglu, Tobias Herzke, Volker Hohmann, Matthieu Recugnat, Martin Besnard, Teng Huang, Bradford Backus:
The ABCIT research platform. 26-29 - Boaz Schwartz, Sharon Gannot, Emanuël A. P. Habets:
LPC-based speech dereverberation using Kalman-EM algorithm. 30-34 - Kenta Niwa, Yusuke Hioka, Kazunori Kobayashi:
Post-filter design for speech enhancement in various noisy environments. 35-39 - Jesper Kjær Nielsen, Jesper Rindom Jensen, Søren Holdt Jensen, Mads Græsbøll Christensen:
The single- and multichannel audio recordings database (SMARD). 40-44 - Henning F. Schepker, Simon Doclo:
Estimation of the common part of acoustic feedback paths in hearing aids using iterative quadratic programming. 45-49 - Ante Jukic, Toon van Waterschoot, Timo Gerkmann, Simon Doclo:
Speech dereverberation with convolutive transfer function approximation using map and variational deconvolution approaches. 50-54 - João Felipe Santos, Mohammed Senoussaoui, Tiago H. Falk:
An improved non-intrusive intelligibility metric for noisy and reverberant speech. 55-59 - Satoru Emura, Hitoshi Ohmuro:
Wave-domain canceling of residual echo with subspace tracking. 60-64 - Joachim Flocon-Cholet, Julien Faure, Alexandre Guérin, Pascal Scalart:
A robust howling detection algorithm based on a statistical approach. 65-69 - W. Bastiaan Kleijn, Turaj Zakizadeh Shabestary, Jan Skoglund:
Sinusoidal interpolation across missing data. 70-74 - Mark R. P. Thomas, Ivan J. Tashev, Felicia Lim, Patrick A. Naylor:
Optimal beamforming as a time domain equalization problem with application to room acoustics. 75-79 - Pejman Mowlaee, Christian Nachbar:
Speaker dependent speech enhancement using sinusoidal model. 80-84 - Yusuke Hioka, Kenta Niwa:
PSD estimation in beamspace for source separation in a diffuse noise field. 85-88 - Joao B. Crespo, Richard C. Hendriks:
Speech reinforcement with a globally optimized perceptual distortion measure for noisy reverberant channels. 89-93 - Nikolay D. Gaubitch, Jorge Martínez, W. Bastiaan Kleijn, Richard Heusdens:
On near-field beamforming with smartphone-based ad-hoc microphone arrays. 94-98 - Hendrik Kayser, Jörn Anemüller:
A discriminative learning approach to probabilistic acoustic source localization. 99-103 - Jochen Withopf, Gerhard Schmidt:
Estimation of time-variant acoustic feedback paths in in-car communication systems. 104-108 - Oliver Thiergart, Konrad Kowalczyk, Emanuël A. P. Habets:
An acoustical zoom based on informed spatial filtering. 109-113 - Niccolò Antonello, Toon van Waterschoot, Marc Moonen, Patrick A. Naylor:
Identification of surface acoustic impedances in a reverberant room using the FDTD method. 114-118 - Felicia Lim, Patrick A. Naylor:
Statistical modelling of multichannel blind system identification errors. 119-123 - Sebastian Braun, Matteo Torcoli, Daniel Marquardt, Emanuël A. P. Habets, Simon Doclo:
Multichannel dereverberation for hearing aids with interaural coherence preservation. 124-128 - Konrad Kowalczyk, Alexandra Craciun, Christian Dachmann, Emanuël A. P. Habets:
Spatial perception of virtual X-Y recordings. 129-133 - Leela K. Gudupudi, Christophe Beaugeant, Nicholas W. D. Evans:
Characterisation and modelling of non-linear loudspeakers. 134-138 - Ina Kodrasi, Simon Doclo:
Joint dereverberation and noise reduction based on acoustic multichannel equalization. 139-143 - Masanori Kato, Akihiko Sugiyama:
A wind-noise suppressor based on wind-onset detection and spectral gain modification. 144-148 - Hiroki Katahira, Nobutaka Ono, Shigeki Miyabe, Takeshi Yamada, Shoji Makino:
Generalized amplitude interpolation by β-divergence for virtual microphone array. 149-153 - Tobias May, Timo Gerkmann:
Generalization of supervised learning for binary mask estimation. 154-158 - Daniele Giacobello, Jason Wung, Ramin Pichevar, Joshua Atkins:
A computationally constrained optimization framework for implementation and tuning of speech enhancement systems. 159-163 - Emma Jokinen, Ulpu Remes, Marko Takanen, Kalle J. Palomäki, Mikko Kurimo, Paavo Alku:
Spectral tilt modelling with extrapolated GMMs for intelligibility enhancement of narrowband telephone speech. 164-168 - Maja Taseska, Shmulik Markovich Golan, Emanuël A. P. Habets, Sharon Gannot:
Near-field source extraction using speech presence probabilities for ad hoc microphone arrays. 169-173 - Karim Helwani, Herbert Buchner:
Multichannel adaptive filtering in compressive domains. 174-177 - Trung-Kien Le, Nobutaka Ono:
Numerical formulae for TOA-based microphone and source localization. 178-182 - Dani Cherkassky, Sharon Gannot:
Blind synchronization in wireless sensor networks with application to speech enhancement. 183-187 - Chris Oreinos, Jörg M. Buchholz:
Validation of realistic acoustic environments for listening tests using directional hearing aids. 188-192 - Alexis Favrot:
Noise coloration filter design by pole-zero placement. 193-197 - Zbynek Koldovský, Jirí Málek, Michael Muller, Petr Tichavský:
On semi-blind estimation of echo paths during double-talk based on nonstationarity. 198-202 - Hironobu Chiba, Nobutaka Ono, Shigeki Miyabe, Yu Takahashi, Takeshi Yamada, Shoji Makino:
Amplitude-based speech enhancement with nonnegative matrix factorization for asynchronous distributed recording. 203-207 - Toby Christian Lawin-Ore, Sebastian Stenzel, Jürgen Freudenberger, Simon Doclo:
Alternative formulation and robustness analysis of the multichannel wiener filter for spatially distributed microphones. 208-212 - Lukas Drude, Aleksej Chinaev, Dang Hai Tran Vu, Reinhold Haeb-Umbach:
Towards online source counting in speech mixtures applying a variational EM for complex Watson mixture models. 213-217 - Christin Baasch, Vasudev Kandade Rajan, Mohamed Krini, Gerhard Schmidt:
Low-complexity noise power spectral density estimation for harsh automobile environments. 218-222 - Masoumeh Azarpour, Gerald Enzner:
Fast noise PSD estimation based on blind channel identification. 223-227 - Youssef Oualil, Rahil Mahdian Toroghi, Dietrich Klakow:
Online unsupervised overlapping speaker detection using enhanced classification history-based features. 228-232 - Stefan Goetze, Anna Warzybok, Ina Kodrasi, Jan Ole Jungmann, Benjamin Cauchi, Jan Rennies, Emanuël A. P. Habets, Alfred Mertins, Timo Gerkmann, Simon Doclo, Birger Kollmeier:
A study on speech quality and speech intelligibility measures for quality assessment of single-channel dereverberation algorithms. 233-237 - Nurit Spingarn, Saman Mousazadeh, Israel Cohen:
Voice activity detection in transient noise environment using Laplacian pyramid algorithm. 238-242 - Axel Plinge, Gernot A. Fink:
Geometry calibration of multiple microphone arrays in highly reverberant environments. 243-247 - Giacomo Vairetti, Toon van Waterschoot, Marc Moonen, Michael Catrysse, Søren Holdt Jensen:
An automatic model-building algorithm for sparse approximation of room impulse responses with Orthonormal Basis Functions. 248-252 - Mahfoud Hamidia, Abderrahmane Amrouche:
A new structure for acoustic echo cancellation in double-talk scenario using auxiliary filter. 253-257 - Christine Evers, Alastair H. Moore, Patrick A. Naylor:
Multiple source localisation in the spherical harmonic domain. 258-262 - Marco Matassoni, Alessio Brutti, Piergiorgio Svaizer:
Acoustic modeling based on early-to-late reverberation ratio for robust ASR. 263-267 - Nobutaka Ito, Shoko Araki, Takuya Yoshioka, Tomohiro Nakatani:
Relaxed disjointness based clustering for joint blind source separation and dereverberation. 268-272 - Mahmoud Fakhry, Piergiorgio Svaizer, Maurizio Omologo:
Reverberant audio source separation using partially pre-trained nonnegative matrix factorization. 273-277 - Anne Theiß, Gerhard Schmidt:
Investigation of self-masking effects for the evaluation of in-car communication systems. 278-282 - Steffen Kortlang, Stephan Dieter Ewert, Timo Gerkmann:
Single channel noise reduction based on an auditory filterbank. 283-287 - Daniel Marquardt, Elior Hadad, Sharon Gannot, Simon Doclo:
Optimal binaural LCMV beamformers for combined noise reduction and binaural cue preservation. 288-292 - Wei Zhang, Jingdong Chen, Jacob Benesty:
A reduced-rank approach to single-channel noise reduction. 293-297 - Pablo Peso Parada, Dushyant Sharma, Jose Lainez, Daniel Barreda, Patrick A. Naylor, Toon van Waterschoot:
A quantitative comparison of blind C50 estimators. 298-302 - Jiuwen Cao, Andy W. H. Khong, Sharon Gannot:
On the performance of widely linear quaternion based MVDR beamformer for an acoustic vector sensor. 303-307 - Shoichi Koyama, Prakhar Srivastava, Ken'ichi Furuya, Suehiro Shimauchi, Hitoshi Ohmuro:
STSP: Space-time stretched pulse for measuring spatio-temporal impulse response. 308-312 - Elior Hadad, Florian Heese, Peter Vary, Sharon Gannot:
Multichannel audio database in various acoustic environments. 313-317 - Takuya Toyoda, Nobutaka Ono, Shigeki Miyabe, Takeshi Yamada, Shoji Makino:
Traffic monitoring with ad-hoc microphone array. 318-322 - Yoshihiro Sakai, Muhammad Tahir Akhtar:
The acoustic echo cancelation using blind source separation to reduce double-talk interference. 323-326 - Christoph Matthias Nelke, Peter Vary:
Measurement, analysis and simulation of wind noise signals for mobile communication devices. 327-331 - Anna Warzybok, Ina Kodrasi, Jan Ole Jungmann, Emanuël A. P. Habets, Timo Gerkmann, Alfred Mertins, Simon Doclo, Birger Kollmeier, Stefan Goetze:
Subjective speech quality and speech intelligibility evaluation of single-channel dereverberation algorithms. 332-336 - Pejman Mowlaee, Rahim Saeidi:
Time-frequency constraints for phase estimation in single-channel speech enhancement. 337-341 - Wen Zhang, Thushara D. Abhayapala:
2.5D sound field reproduction in higher order Ambisonics. 342-346
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