default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCIDWASPAA 2017: New Paltz, NY, USA
- 2017 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, WASPAA 2017, New Paltz, NY, USA, October 15-18, 2017. IEEE 2017, ISBN 978-1-5386-1632-1
- Ville Pulkki:
Keynotes: Parametric time-frequency-domain spatial audio - Delivering sound according to human spatial resolution. 1-6 - Rui Lu, Zhiyao Duan, Changshui Zhang:
Metric learning based data augmentation for environmental sound classification. 1-5 - Aleksandr Diment, Tuomas Virtanen
:
Transfer learning of weakly labelled audio. 6-10 - Grégoire Lafay, Emmanouil Benetos, Mathieu Lagrange:
Sound event detection in synthetic audio: Analysis of the dcase 2016 task results. 11-15 - Shuyang Zhao, Toni Heittola, Tuomas Virtanen:
Learning vocal mode classifiers from heterogeneous data sources. 16-20 - Naoya Takahashi, Yuki Mitsufuji:
Multi-Scale multi-band densenets for audio source separation. 21-25 - Ryan M. Corey, Andrew C. Singer:
Underdetermined methods for multichannel audio enhancement with partial preservation of background sources. 26-30 - Feng Bao, Waleed H. Abdulla:
A convex optimization approach for time-frequency mask estimation. 31-35 - Prem Seetharaman, Fatemeh Pishdadian, Bryan Pardo:
Music/Voice separation using the 2D fourier transform. 36-40 - Dionyssos Kounades-Bastian, Laurent Girin, Xavier Alameda-Pineda, Radu Horaud, Sharon Gannot:
Exploiting the intermittency of speech for joint separation and diarization. 41-45 - Ritwik Giri, Karim Helwani, Tao Zhang:
A novel target speaker dependent postfiltering approach for multichannel speech enhancement. 46-50 - Mathieu Fontaine, Antoine Liutkus, Laurent Girin, Roland Badeau:
Explaining the parameterized wiener filter with alpha-stable processes. 51-55 - Xiaofei Li, Laurent Girin, Radu Horaud:
An em algorithm for audio source separation based on the convolutive transfer function. 56-60 - Sanjeel Parekh, Slim Essid, Alexey Ozerov, Ngoc Q. K. Duong, Patrick Pérez, Gaël Richard:
Guiding audio source separation by video object information. 61-65 - Gordon Wichern, Alexey Lukin:
Low-Latency approximation of bidirectional recurrent networks for speech denoising. 66-70 - Gaurav Naithani, Tom Barker, Giambattista Parascandolo, Lars Bramslow, Niels Henrik Pontoppidan, Tuomas Virtanen:
Low latency sound source separation using convolutional recurrent neural networks. 71-75 - Abdullah Fahim, Prasanga N. Samarasinghe, Thushara D. Abhayapala:
PSD estimation of multiple sound sources in a reverberant room using a spherical microphone array. 76-80 - Christos Tzagkarakis, W. Bastiaan Kleijn, Jan Skoglund:
Joint wideband source localization and acquisition based on a grid-shift approach. 81-85 - Yingke Zhao, Jesper Rindom Jensen, Mads Græsbøll Christensen, Simon Doclo, Jingdong Chen:
Experimental study of robust beamforming techniques for acoustic applications. 86-90 - Sam Karimian-Azari, Tiago H. Falk:
Modulation spectrum based beamforming for speech enhancement. 91-95 - Paul Calamia, Shakti Davis, Christopher Smalt, Christine Weston:
A conformal, helmet-mounted microphone array for auditory situational awareness and hearing protection. 96-100 - Ina Kodrasi, Simon Doclo:
Multi-Channel late reverberation power spectral density estimation based on nuclear norm minimization. 101-105 - Hendrik Barfuss, Markus Bachmann, Michael Buerger, Martin Schneider, Walter Kellermann:
Design of robust two-dimensional polynomial beamformers as a convex optimization problem with application to robot audition. 106-110 - Michael Buerger, Christian Hofmann, Cornelius Frankenbach, Walter Kellermann:
Multizone sound reproduction in reverberant environments using an iterative least-squares filter design method with a spatiotemporal weighting function. 1-5 - Jens Ahrens:
Amplitude engineering for beamformers with self-bending directivity based on convex optimization. 116-120 - Stefan Bilbao, Brian Hamilton:
Directional source modeling in wave-based room acoustics simulation. 121-125 - Sina Zamani, Tejaswi Nanjundaswamy, Kenneth Rose:
Frequency domain singular value decomposition for efficient spatial audio coding. 126-130 - Ofer Schwartz, Axel Plinge, Emanuël A. P. Habets, Sharon Gannot:
Blind microphone geometry calibration using one reverberant speech event. 131-135 - Soumitro Chakrabarty, Emanuël A. P. Habets:
Broadband doa estimation using convolutional neural networks trained with noise signals. 136-140 - Stefan Bilbao, Fabian Esqueda, Vesa Välimäki:
Antiderivative antialiasing, lagrange interpolation and spectral flatness. 141-145 - Sebastian Ewert, Mark B. Sandler:
An augmented lagrangian method for piano transcription using equal loudness thresholding and lstm-based decoding. 146-150 - Ralf Gunter Correa Carvalho, Paris Smaragdis:
Towards end-to-end polyphonic music transcription: Transforming music audio directly to a score. 151-155 - James Anderson Moorer:
A note on the implementation of audio processing by short-term fourier transform. 156-159 - Fiete Winter, Christoph Hold, Hagen Wierstorf, Alexander Raake, Sascha Spors:
Colouration in 2.5D local wave field synthesis using spatial bandwidth-limitation. 160-165 - Rotem Mulayoff, Yaakov Buchris, Israel Cohen:
Differential microphone arrays for the underwater acoustic channel. 165-169 - Federico Borra, Fabio Antonacci, Augusto Sarti, Stefano Tubaro:
Localization of acoustic sources in the ray space for distributed microphone sensors. 170-174 - Wenqiang Pu, Jinjun Xiao, Tao Zhang, Zhi-Quan Luo:
A penalized inequality-constrained minimum variance beamformer with applications in hearing aids. 175-179 - Takuma Okamoto:
Angular spectrum decomposition-based 2.5D higher-order spherical harmonic sound field synthesis with a linear loudspeaker array. 180-184 - Keigo Wakayama, Jorge Treviño, Hideaki Takada, Shuichi Sakamoto, Yôiti Suzuki:
Extended sound field recording using position information of directional sound sources. 185-189 - Yaakov Buchris, Israel Cohen, Jacob Benesty:
Asymmetric beampatterns with circular differential microphone arrays. 190-194 - Kainan Chen, Jürgen T. Geiger, Wenyu Jin, Mohammad Javad Taghizadeh, Walter Kellermann:
Robust phase replication method for spatial aliasing problem in multiple sound sources localization. 195-199 - Babafemi O. Odelowo, David V. Anderson:
Speech enhancement using extreme learning machines. 200-204 - Nara Hahn, Sascha Spors:
Continuous measurement of spatial room impulse responses using a non-uniformly moving microphone. 205-208 - W. Bastiaan Kleijn, Andrew Allen, Jan Skoglund, Felicia Lim:
Incoherent idempotent ambisonics rendering. 209-213 - Shoichi Koyama, Laurent Daudet:
Comparison of reverberation models for sparse sound field decomposition. 214-218 - Johannes Abel, Tim Fingscheidt:
A DNN regression approach to speech enhancement by artificial bandwidth extension. 219-223 - Archontis Politis, Hannes Gamper:
Comparing modeled and measurement-based spherical harmonic encoding filters for spherical microphone arrays. 224-228 - Maria Luis Valero, Emanuël A. P. Habets:
Multi-Microphone acoustic echo cancellation using relative echo transfer functions. 229-233 - Daniel Marquardt, Simon Doclo:
Noise power spectral density estimation for binaural noise reduction exploiting direction of arrival estimates. 234-238 - Craig T. Jin, Fabio Antonacci, Augusto Sarti:
Ray space analysis with sparse recovery. 239-243 - Dovid Y. Levin, Shmulik Markovich Golan, Sharon Gannot:
Distributed lcmv beamforming: Considerations of spatial topology and local preprocessing. 244-248 - Prasanga N. Samarasinghe, Hanchi Chen, Abdullah Fahim, Thushara D. Abhayapala:
Performance analysis of a planar microphone array for three dimensional soundfield analysis. 249-253 - Scott Wisdom, Thomas Powers, James W. Pitton, Les Atlas:
Deep recurrent NMF for speech separation by unfolding iterative thresholding. 254-258 - Paul Magron, Roland Badeau, Antoine Liutkus:
Lévy NMF for robust nonnegative source separation. 259-263 - Simon Leglaive, Roland Badeau, Gaël Richard:
Separating time-frequency sources from time-domain convolutive mixtures using non-negative matrix factorization. 264-268 - Paul Magron, Jonathan Le Roux, Tuomas Virtanen:
Consistent anisotropic Wiener filtering for audio source separation. 269-273 - Ethan Manilow, Prem Seetharaman, Fatemeh Pishdadian, Bryan Pardo:
Predicting algorithm efficacy for adaptive multi-cue source separation. 274-278 - Stephen Voran:
The selection of spectral magnitude exponents for separating two sources is dominated by phase distribution not magnitude distribution. 279-283 - Thomas Dietzen, Simon Doclo, Ann Spriet, Wouter Tirry, Marc Moonen, Toon van Waterschoot:
Low-Complexity Kalman filter for multi-channel linear-prediction-based blind speech dereverberation. 284-288 - Ryan M. Corey, Andrew C. Singer:
Dynamic range compression for noisy mixtures using source separation and beamforming. 289-293 - Arthur Belhomme, Roland Badeau, Yves Grenier, Eric Humbert:
Amplitude and phase dereverberation of harmonic signals. 294-298 - Flávio R. Avila, Luiz W. P. Biscainho:
Audio soft declipping based on weighted L1-norm. 299-303 - Yichi Zhang, Zhiyao Duan:
IMINET: Convolutional semi-siamese networks for sound search by vocal imitation. 304-308 - Ethan Manilow, Bryan Pardo:
Leveraging repetition to do audio imputation. 309-313 - Liming Shi, Jesper Kjær Nielsen, Jesper Rindom Jensen, Max A. Little, Mads Græsbøll Christensen:
A Kalman-based fundamental frequency estimation algorithm. 314-318 - Annamaria Mesaros, Toni Heittola, Tuomas Virtanen:
Assessment of human and machine performance in acoustic scene classification: Dcase 2016 case study. 319-323 - Lars F. Villemoes, Janusz Klejsa, Per Hedelin:
Speech coding with transform domain prediction. 324-328 - Mohsen Ahangar, Mostafa Ghorbandoost, Sudhendu R. Sharma, Mark J. T. Smith:
Voice conversion based on a mixture density network. 329-333 - Michael C. Heilemann, David Anderson, Mark F. Bocko:
Source rendering on dynamic audio displays. 334-338 - Nicholas Jillings, Joshua D. Reiss, Ryan Stables:
Zero-Delay large signal convolution using multiple processor architectures. 339-343 - Justin Salamon, Duncan MacConnell, Mark Cartwright, Peter Li, Juan Pablo Bello:
Scaper: A library for soundscape synthesis and augmentation. 344-348 - Alexander Adami, Adrian Herzog, Sascha Disch, Jürgen Herre:
Transient-to-noise ratio restoration of coded applause-like signals. 349-353 - Y. Cem Sübakan, Paris Smaragdis:
Diagonal rnns in symbolic music modeling. 354-358 - Shlomo E. Chazan, Jacob Goldberger, Sharon Gannot:
Deep recurrent mixture of experts for speech enhancement. 359-363 - Pavel Rajmic, Zbynek Koldovský, Marie Danková:
Fast reconstruction of sparse relative impulse responses via second-order cone programming. 364-368 - Anna Barnov, Vered Bar Bracha, Shmulik Markovich Golan:
QRD based MVDR beamforming for fast tracking of speech and noise dynamics. 369-373 - Konstantinos Drossos, Sharath Adavanne, Tuomas Virtanen:
Automated audio captioning with recurrent neural networks. 374-378 - Archontis Politis, Leo McCormack, Ville Pulkki:
Enhancement of ambisonic binaural reproduction using directional audio coding with optimal adaptive mixing. 379-383 - François G. Germain, Kurt James Werner:
Optimizing differentiated discretization for audio circuits beyond driving point transfer functions. 384-388
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 2025-12-04 01:32 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:
