default search action
HSCMA 2014: Villers-les-Nancy, France
- 4th Joint Workshop on Hands-free Speech Communication and Microphone Arrays, HSCMA 2014, Villers-les-Nancy, France, May 12-14, 2014. IEEE 2014
- Mohammad Javad Taghizadeh, Afsaneh Asaei, Philip N. Garner
, Hervé Bourlard:
Ad-hoc microphone array calibration from partial distance measurements. 1-5 - Antoine Liutkus, Zafar Rafii, Bryan Pardo, Derry Fitzgerald, Laurent Daudet:
Kernel spectrogram models for source separation. 6-10 - Pablo Sprechmann, Alexander M. Bronstein, Guillermo Sapiro:
Supervised non-euclidean sparse NMF via bilevel optimization with applications to speech enhancement. 11-15 - Zhuo Chen, Hélène Papadopoulos, Daniel P. W. Ellis:
Content-adaptive speech enhancement by a sparsely-activated dictionary plus low rank decomposition. 16-20 - Jort F. Gemmeke:
The self-taught vocal interface. 21-22 - Ante Jukic, Toon van Waterschoot
Speech dereverberation with multi-channel linear prediction and sparse priors for the desired signal. 23-26 - Nasser Mohammadiha, Simon Dodo:
Transient noise reduction using nonnegative matrix factorization. 27-31 - Armin Saeb, Farbod Razzazi
A fast phoneme recognition system based on sparse representation of test utterances. 32-36 - Syed Zubair, Wenwu Wang, Jonathon A. Chambers:
Discriminativetensor dictionaries and sparsity for speaker identification. 37-41 - Hai Morgenstern, Boaz Rafaely
Far-field criterion for spherical microphone arrays and directional sources. 42-46 - Jounghoon Beh, Dmitry N. Zotkin, Ramani Duraiswami
Adaptive interference rejection using generalized sidelobe canceller in spherical harmonics domain. 47-51 - Gary W. Elko, Jens Meyer:
Adaptive beamformer for spherical eigenbeamforming microphone arrays. 52-56 - Jirí Málek, David Botha, Zbynek Koldovský
Methods to learn bank of filters steering nulls toward potential positions of a target source. 57-61 - Daniele Giacobello, Jason Wung, Ramin Pichevar, Joshua Atkins:
Tuning methodology for speech enhancement algorithms using a simulated conversational database and perceptual objective measures. 62-66 - Randy Gomez, Keisuke Nakamura, Takeshi Mizumoto, Kazuhiro Nakadai:
Improved hands-free automatic speech recognition in reverberant environment condition. 67-71 - Yuval Dorfan, Gershon Hazan, Sharon Gannot
Multiple acoustic sources localization using distributed expectation-maximization algorithm. 72-76 - Michael Jeffet, Boaz Rafaely
Study of a generalized spherical array beamformer with adjustable binaural reproduction. 77-81 - Lalan Kumar
Near-field source localization using spherical microphone array. 82-86 - Jonathan Blanchette, Martin Bouchard
Short-time multichannel noise correlation matrix estimators for acoustic signals. 87-91 - Daichi Kitamura, Hiroshi Saruwatari, Satoshi Nakamura, Yu Takahashi, Kazunobu Kondo, Hirokazu Kameoka:
Divergence optimization in nonnegative matrix factorization with spectrogram restoration for multichannel signal separation. 92-96 - Falk-Martin Hoffmann
Circular microphone array with tangential pressure gradient sensors. 97-101 - Craig A. Anderson, Stefan Meier, Walter Kellermann, Paul D. Teal
A GPU-accelerated real-time implementation of TRINICON-BSS for multiple separation units. 102-106 - Toru Taniguchi, Nobutaka Ono
An auxiliary-function approach to online independent vector analysis for real-time blind source separation. 107-111 - Leela K. Gudupudi, Christophe Beaugeant, Nicholas W. D. Evans, Moctar Mossi Idrissa, Ludovick Lepauloux:
A comparison of different loudspeaker models to empirically estimated non-linearities. 112-116 - Miquel Espi, Masakiyo Fujimoto, Yotaro Kubo, Tomohiro Nakatani:
Spectrogram patch based acoustic event detection and classification in speech overlapping conditions. 117-121 - Shunsuke Nakai, Hiroshi Saruwatari, Ryoichi Miyazaki, Satoshi Nakamura, Kazunobu Kondo:
Theoretical analysis of biased MMSE short-time spectral amplitude estimator and its extension to musical-noise-free speech enhancement. 122-126 - Sebastian Stenzel, Jürgen Freudenberger
A Minimum variance beamformer for spatially distributed microphones using a soft reference selection. 127-131 - Nilesh Madhu
A hierarchical approach for the online, on-board detection and localisation of brake squeal using microphone arrays. 132-136 - David L. Alon, Boaz Rafaely
Spatial aliasing-cancellation for circular microphone arrays. 137-141 - Ankit Sohni, Chaitanya Ahuja, Rajesh M. Hegde:
Extraction of pinna spectral notches in the median plane of a virtual spherical microphone array. 142-146 - Vladimir Tourbabin, Boaz Rafaely
Utilizing motion in humanoid robots to enhance spatial information recordedby microphone arrays. 147-151 - Mikko Parviainen, Pasi Pertilä, Matti S. Hämäläinen:
Self-localization of wireless acoustic sensors in meeting rooms. 152-156 - Alessio Brutti, Mirco Ravanelli
A speech event detection and localization task for multiroom environments. 157-161 - Yuuki Tachioka, Tomohiro Narita, Shinji Watanabe
Ensemble integration of calibrated speaker localization and statistical speech detection in domestic environments. 162-166 - Panagiotis Giannoulis, Antigoni Tsiami, Isidoros Rodomagoulakis, Athanasios Katsamanis
The Athena-RC system for speech activity detection and speaker localization in the DIRHA smart home. 167-171 - Steve Renals
Neural networks for distant speech recognition. 172-176 - Armin Sehr, Hendrik Barfuss, Christian Hofmann
Efficient training of acoustic models for reverberation-robust medium-vocabulary automatic speech recognition. 177-181 - Fine Dwinita Aprilyanti, Hiroshi Saruwatari, Satoshi Nakamura, Tomoya Takatani:
Optimized joint noise suppression and dereverberation based on blind signal extraction for hands-free speech recognition system. 182-186 - Cristina Guerrero, Maurizio Omologo
Word boundary agreementto combine multi-microphone hypotheses in distant speech recognition. 187-191 - Arseniy Gorin, Denis Jouvet, Emmanuel Vincent, Dung T. Tran:
Investigating stranded GMM for improving automatic speech recognition. 192-196 - Masato Mimura, Shinsuke Sakai, Tatsuya Kawahara
Exploring deep neural networks and deep autoencoders in reverberant speech recognition. 197-201
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 2024-12-31 21:31 CET 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: