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HSCMA 2017: San Francisco, CA, USA
- Hands-free Speech Communications and Microphone Arrays, HSCMA 2017, San Francisco, CA, USA, March 1-3, 2017. IEEE 2017, ISBN 978-1-5090-5925-6
- Nico Gößling, Daniel Marquardt, Simon Doclo:
Performance analysis of the extended binaural MVDR beamformer with partial noise estimation in a homogeneous noise field. 1-5 - JeeSok Lee, Soo-Whan Chung, Min-Seok Choi, Hong-Goo Kang:
A study on search grid points for data-driven 3-D beamsteering. 6-10 - Reza Varzandeh, Maja Taseska, Emanuël A. P. Habets:
An iterative multichannel subspace-based covariance subtraction method for relative transfer function estimation. 11-15 - Shoko Araki, Nobutaka Ito, Marc Delcroix, Atsunori Ogawa, Keisuke Kinoshita, Takuya Higuchi, Takuya Yoshioka, Dung T. Tran, Shigeki Karita, Tomohiro Nakatani:
Online meeting recognition in noisy environments with time-frequency mask based MVDR beamforming. 16-20 - Vladimir Tourbabin, Ilan Malka, Eli Tzirkel-Hancock:
Performance of fixed in-car microphone array beamformer under variations in car noise. 21-25 - Karan Nathwani, Juan Andres Morales-Cordovilla, Sunit Sivasankaran, Irina Illina, Emmanuel Vincent:
An extended experimental investigation of DNN uncertainty propagation for noise robust ASR. 26-30 - Christian Huemmer, Ramón Fernandez Astudillo, Walter Kellermann:
An improved uncertainty decoding scheme with weighted samples for multi-channel DNN-HMM hybrid systems. 31-35 - Bo Wu, Kehuang Li, Zhen Huang, Sabato Marco Siniscalchi, Minglei Yang, Chin-Hui Lee:
A unified deep modeling approach to simultaneous speech dereverberation and recognition for the reverb challenge. 36-40 - Jinkyu Lee, Keulbit Kim, Turaj Shabestary, Hong-Goo Kang:
Deep bi-directional long short-term memory based speech enhancement for wind noise reduction. 41-45 - Daniel Gerber, Stefan Meier, Walter Kellermann:
Efficient target activity detection based on recurrent neural networks. 46-50 - Heinrich W. Löllmann, Alastair H. Moore, Patrick A. Naylor, Boaz Rafaely, Radu Horaud, Alexandre Mazel, Walter Kellermann:
Microphone array signal processing for robot audition. 51-55 - Hendrik Barfuss, Michael Buerger, Jasper Podschus, Walter Kellermann:
HRTF-based two-dimensional robust least-squares frequency-invariant beamformer design for robot audition. 56-60 - Quan V. Nguyen, Francis Colas, Emmanuel Vincent, François Charpillet:
Long-term robot motion planning for active sound source localization with Monte Carlo tree search. 61-65 - Lior Madmoni, Hendrik Barfuss, Boaz Rafaely, Walter Kellermann:
A unified framework for multiple arrays on a robot and application to sound localization. 66-70 - Israel D. Gebru, Christine Evers, Patrick A. Naylor, Radu Horaud:
Audio-visual tracking by density approximation in a sequential Bayesian filtering framework. 71-75 - Ivan J. Tashev, Long Le, Vani Gopalakrishna, Andrew Lovitt:
Cost function for sound source localization with arbitrary microphone arrays. 76-80 - Sina Hafezi, Alastair H. Moore, Patrick A. Naylor:
Multi-source estimation consistency for improved multiple direction-of-arrival estimation. 81-85 - Ofer Schwartz, Yuval Dorfan, Maja Taseska, Emanuël A. P. Habets, Sharon Gannot:
DOA estimation in noisy environment with unknown noise power using the EM algorithm. 86-90 - Christine Evers, Boaz Rafaely, Patrick A. Naylor:
Speaker tracking in reverberant environments using multiple directions of arrival. 91-95 - Boaz Rafaely, Dorothea Kolossa, Yanir Maymon:
Towards acoustically robust localization of speakers in a reverberant environment. 96-100 - Qing Wang, Jun Du, Li-Rong Dai, Chin-Hui Lee:
Joint noise and mask aware training for DNN-based speech enhancement with SUB-band features. 101-105 - Samy Elshamy, Nilesh Madhu, Wouter Tirry, Tim Fingscheidt:
Two-stage speech enhancement with manipulation of the cepstral excitation. 106-110 - Nikolaos Dionelis, Mike Brookes:
Modulation-domain speech enhancement using a Kalman filter with a Bayesian update of speech and noise in the log-spectral domain. 111-115 - Ina Kodrasi, Simon Doclo:
EVD-based multi-channel dereverberation of a moving speaker using different RETF estimation methods. 116-120 - Radoslaw Mazur, Fabrice Katzberg, Huy Phan, Alfred Mertins:
Room equalization based on measurements with moving microphones. 121-125 - Feifei Xiong, Bernd T. Meyer, Benjamin Cauchi, Ante Jukic, Simon Doclo, Stefan Goetze:
Performance comparison of real-time single-channel speech dereverberation algorithms. 126-130 - James Eaton, Hamza A. Javed, Patrick A. Naylor:
Estimation of the perceived level of reverberation using non-intrusive single-channel variance of decay rates. 131-135 - Lei Sun, Jun Du, Li-Rong Dai, Chin-Hui Lee:
Multiple-target deep learning for LSTM-RNN based speech enhancement. 136-140 - Li Li, Hirokazu Kameoka, Shoji Makino:
Discriminative non-negative matrix factorization with majorization-minimization. 141-145 - Federico Borra, Fabio Antonacci, Augusto Sarti, Stefano Tubaro:
Extraction of acoustic sources for multiple arrays based on the ray space transform. 146-150 - Abdullah Fahim, Prasanga N. Samarasinghe, Thushara D. Abhayapala:
Sound field separation in a mixed acoustic environment using a sparse array of higher order spherical microphones. 151-155 - W. Bastiaan Kleijn, Felicia Lim:
Robust and low-complexity blind source separation for meeting rooms. 156-160 - Shahab Pasha, Jacob Donley, Christian H. Ritz, Yue Xian Zou:
Towards real-time source counting by estimation of coherent-to-diffuse ratios from ad-hoc microphone array recordings. 161-165 - Pierre Narvor, Bertrand Rivet, Christian Jutten:
Single sensor audiovisual speech source separation. 166-170 - Camelia Elisei-Iliescu, Cristian Lucian Stanciu, Constantin Paleologu, Jacob Benesty, Cristian Anghel, Silviu Ciochina:
Robust variable-regularized RLS algorithms. 171-175 - Maria Luis Valero, Ilkay Yildiz, Edwin Mabande, Emanuël A. P. Habets:
Coherence-aware stereophonic residual echo estimation. 176-180 - Stefan Kühl, Christiane Antweiler, Tobias Hübschen, Peter Jax:
Kalman filter based stereo system identification with auto- and cross-decorrelation. 181-185 - Hannes Gamper:
Clock drift estimation and compensation for asynchronous impulse response measurements. 186-190 - Fabrice Katzberg, Radoslaw Mazur, Marco Maaß, Philipp Koch, Alfred Mertins:
Multigrid reconstruction of sound fields using moving microphones. 191-195 - Natsuki Ueno, Shoichi Koyama, Hiroshi Saruwatari:
Listening-area-informed sound field reproduction with Gaussian prior based on circular harmonic expansion. 196-200 - Mark R. P. Thomas:
Fast computation of cubature formulae for the sphere. 201-205
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