Table of Contents

• Multizone sound reproduction in reverberant environments using an iterative least-squares filter design method with a spatiotemporal weighting functionMichael Buerger, Christian Hofmann, Cornelius Frankenbach, Walter Kellermann. 1-5 [doi]
• Keynotes: Parametric time-frequency-domain spatial audio - Delivering sound according to human spatial resolutionVille Pulkki. 1-6 [doi]
• Metric learning based data augmentation for environmental sound classificationRui Lu, Zhiyao Duan, Changshui Zhang. 1-5 [doi]
• Transfer learning of weakly labelled audioAleksandr Diment, Tuomas Virtanen. 6-10 [doi]
• Sound event detection in synthetic audio: Analysis of the dcase 2016 task resultsGrégoire Lafay, Emmanouil Benetos, Mathieu Lagrange. 11-15 [doi]
• Learning vocal mode classifiers from heterogeneous data sourcesShuyang Zhao, Toni Heittola, Tuomas Virtanen. 16-20 [doi]
• Multi-Scale multi-band densenets for audio source separationNaoya Takahashi, Yuki Mitsufuji. 21-25 [doi]
• Underdetermined methods for multichannel audio enhancement with partial preservation of background sourcesRyan M. Corey, Andrew C. Singer. 26-30 [doi]
• A convex optimization approach for time-frequency mask estimationFeng Bao 0003, Waleed H. Abdulla. 31-35 [doi]
• Music/Voice separation using the 2D fourier transformPrem Seetharaman, Fatemeh Pishdadian, Bryan Pardo. 36-40 [doi]
• Exploiting the intermittency of speech for joint separation and diarizationDionyssos Kounades-Bastian, Laurent Girin, Xavier Alameda-Pineda, Radu Horaud, Sharon Gannot. 41-45 [doi]
• A novel target speaker dependent postfiltering approach for multichannel speech enhancementRitwik Giri, Karim Helwani, Tao Zhang. 46-50 [doi]
• Explaining the parameterized wiener filter with alpha-stable processesMathieu Fontaine, Antoine Liutkus, Laurent Girin, Roland Badeau. 51-55 [doi]
• An em algorithm for audio source separation based on the convolutive transfer functionXiaofei Li, Laurent Girin, Radu Horaud. 56-60 [doi]
• Guiding audio source separation by video object informationSanjeel Parekh, Slim Essid, Alexey Ozerov, Ngoc Q. K. Duong, Patrick Pérez, Gaël Richard. 61-65 [doi]
• Low-Latency approximation of bidirectional recurrent networks for speech denoisingGordon Wichern, Alexey Lukin. 66-70 [doi]
• Low latency sound source separation using convolutional recurrent neural networksGaurav Naithani, Tom Barker, Giambattista Parascandolo, Lars Bramslow, Niels Henrik Pontoppidan, Tuomas Virtanen. 71-75 [doi]
• PSD estimation of multiple sound sources in a reverberant room using a spherical microphone arrayAbdullah Fahim, Prasanga N. Samarasinghe, Thushara D. Abhayapala. 76-80 [doi]
• Joint wideband source localization and acquisition based on a grid-shift approachChristos Tzagkarakis, W. Bastiaan Kleijn, Jan Skoglund. 81-85 [doi]
• Experimental study of robust beamforming techniques for acoustic applicationsYingke Zhao, Jesper Rindom Jensen, Mads Græsbøll Christensen, Simon Doclo, Jingdong Chen. 86-90 [doi]
• Modulation spectrum based beamforming for speech enhancementSam Karimian-Azari, Tiago H. Falk. 91-95 [doi]
• A conformal, helmet-mounted microphone array for auditory situational awareness and hearing protectionPaul Calamia, Shakti Davis, Christopher Smalt, Christine Weston. 96-100 [doi]
• Multi-Channel late reverberation power spectral density estimation based on nuclear norm minimizationIna Kodrasi, Simon Doclo. 101-105 [doi]
• Design of robust two-dimensional polynomial beamformers as a convex optimization problem with application to robot auditionHendrik Barfuss, Markus Bachmann, Michael Buerger, Martin Schneider 0009, Walter Kellermann. 106-110 [doi]
• Amplitude engineering for beamformers with self-bending directivity based on convex optimizationJens Ahrens. 116-120 [doi]
• Directional source modeling in wave-based room acoustics simulationStefan Bilbao, Brian Hamilton. 121-125 [doi]
• Frequency domain singular value decomposition for efficient spatial audio codingSina Zamani, Tejaswi Nanjundaswamy, Kenneth Rose. 126-130 [doi]
• Blind microphone geometry calibration using one reverberant speech eventOfer Schwartz, Axel Plinge, Emanuel A. P. Habets, Sharon Gannot. 131-135 [doi]
• Broadband doa estimation using convolutional neural networks trained with noise signalsSoumitro Chakrabarty, Emanuel A. P. Habets. 136-140 [doi]
• Antiderivative antialiasing, lagrange interpolation and spectral flatnessStefan Bilbao, Fabian Esqueda, Vesa Välimäki. 141-145 [doi]
• An augmented lagrangian method for piano transcription using equal loudness thresholding and lstm-based decodingSebastian Ewert, Mark B. Sandler. 146-150 [doi]
• Towards end-to-end polyphonic music transcription: Transforming music audio directly to a scoreRalf Gunter Correa Carvalho, Paris Smaragdis. 151-155 [doi]
• A note on the implementation of audio processing by short-term fourier transformJames A. Moorer. 156-159 [doi]
• Colouration in 2.5D local wave field synthesis using spatial bandwidth-limitationFiete Winter, Christoph Hold, Hagen Wierstorf, Alexander Raake, Sascha Spors. 160-165 [doi]
• Differential microphone arrays for the underwater acoustic channelRotem Mulayoff, Yaakov Buchris, Israel Cohen. 165-169 [doi]
• Localization of acoustic sources in the ray space for distributed microphone sensorsFederico Borra, Fabio Antonacci, Augusto Sarti, Stefano Tubaro. 170-174 [doi]
• A penalized inequality-constrained minimum variance beamformer with applications in hearing aidsWenqiang Pu, Jinjun Xiao, Tao Zhang, Zhi-Quan Luo. 175-179 [doi]
• Angular spectrum decomposition-based 2.5D higher-order spherical harmonic sound field synthesis with a linear loudspeaker arrayTakuma Okamoto. 180-184 [doi]
• Extended sound field recording using position information of directional sound sourcesKeigo Wakayama, Jorge Trevino, Hideaki Takada, Shuichi Sakamoto, Yôiti Suzuki. 185-189 [doi]
• Asymmetric beampatterns with circular differential microphone arraysYaakov Buchris, Israel Cohen, Jacob Benesty. 190-194 [doi]
• Robust phase replication method for spatial aliasing problem in multiple sound sources localizationKainan Chen, Jürgen T. Geiger, Wenyu Jin, Mohammad Javad Taghizadeh, Walter Kellermann. 195-199 [doi]
• Speech enhancement using extreme learning machinesBabafemi O. Odelowo, David V. Anderson. 200-204 [doi]
• Continuous measurement of spatial room impulse responses using a non-uniformly moving microphoneNara Hahn, Sascha Spors. 205-208 [doi]
• Incoherent idempotent ambisonics renderingW. Bastiaan Kleijn, Andrew Allen, Jan Skoglund, Felicia Lim. 209-213 [doi]
• Comparison of reverberation models for sparse sound field decompositionShoichi Koyama, Laurent Daudet. 214-218 [doi]
• A DNN regression approach to speech enhancement by artificial bandwidth extensionJohannes Abel, Tim Fingscheidt. 219-223 [doi]
• Comparing modeled and measurement-based spherical harmonic encoding filters for spherical microphone arraysArchontis Politis, Hannes Gamper. 224-228 [doi]
• Multi-Microphone acoustic echo cancellation using relative echo transfer functionsMaria Luis Valero, Emanuel A. P. Habets. 229-233 [doi]
• Noise power spectral density estimation for binaural noise reduction exploiting direction of arrival estimatesDaniel Marquardt, Simon Doclo. 234-238 [doi]
• Ray space analysis with sparse recoveryCraig Jin, Fabio Antonacci, Augusto Sarti. 239-243 [doi]
• Distributed lcmv beamforming: Considerations of spatial topology and local preprocessingDovid Y. Levin, Shmulik Markovich Golan, Sharon Gannot. 244-248 [doi]
• Performance analysis of a planar microphone array for three dimensional soundfield analysisPrasanga N. Samarasinghe, Hanchi Chen, Abdullah Fahim, Thushara D. Abhayapala. 249-253 [doi]
• Deep recurrent NMF for speech separation by unfolding iterative thresholdingScott Wisdom, Thomas Powers, James W. Pitton, Les Atlas. 254-258 [doi]
• Lévy NMF for robust nonnegative source separationPaul Magron, Roland Badeau, Antoine Liutkus. 259-263 [doi]
• Separating time-frequency sources from time-domain convolutive mixtures using non-negative matrix factorizationSimon Leglaive, Roland Badeau, Gaël Richard. 264-268 [doi]
• Consistent anisotropic Wiener filtering for audio source separationPaul Magron, Jonathan Le Roux, Tuomas Virtanen. 269-273 [doi]
• Predicting algorithm efficacy for adaptive multi-cue source separationEthan Manilow, Prem Seetharaman, Fatemeh Pishdadian, Bryan Pardo. 274-278 [doi]
• The selection of spectral magnitude exponents for separating two sources is dominated by phase distribution not magnitude distributionStephen Voran. 279-283 [doi]
• Low-Complexity Kalman filter for multi-channel linear-prediction-based blind speech dereverberationThomas Dietzen, Simon Doclo, Ann Spriet, Wouter Tirry, Marc Moonen, Toon van Waterschoot. 284-288 [doi]
• Dynamic range compression for noisy mixtures using source separation and beamformingRyan M. Corey, Andrew C. Singer. 289-293 [doi]
• Amplitude and phase dereverberation of harmonic signalsArthur Belhomme, Roland Badeau, Yves Grenier, Eric Humbert. 294-298 [doi]
• Audio soft declipping based on weighted L1-normFlávio R. Avila, Luiz W. P. Biscainho. 299-303 [doi]
• IMINET: Convolutional semi-siamese networks for sound search by vocal imitationYichi Zhang, Zhiyao Duan. 304-308 [doi]
• Leveraging repetition to do audio imputationEthan Manilow, Bryan Pardo. 309-313 [doi]
• A Kalman-based fundamental frequency estimation algorithmLiming Shi, Jesper Kjær Nielsen, Jesper Rindom Jensen, Max A. Little, Mads Græsbøll Christensen. 314-318 [doi]
• Assessment of human and machine performance in acoustic scene classification: Dcase 2016 case studyAnnamaria Mesaros, Toni Heittola, Tuomas Virtanen. 319-323 [doi]
• Speech coding with transform domain predictionLars F. Villemoes, Janusz Klejsa, Per Hedelin. 324-328 [doi]
• Voice conversion based on a mixture density networkMohsen Ahangar, Mostafa Ghorbandoost, Sudhendu R. Sharma, Mark J. T. Smith. 329-333 [doi]
• Source rendering on dynamic audio displaysMichael C. Heilemann, David Anderson, Mark F. Bocko. 334-338 [doi]
• Zero-Delay large signal convolution using multiple processor architecturesNicholas Jillings, Joshua D. Reiss, Ryan Stables. 339-343 [doi]
• Scaper: A library for soundscape synthesis and augmentationJustin Salamon, Duncan MacConnell, Mark Cartwright, Peter Li, Juan Pablo Bello. 344-348 [doi]
• Transient-to-noise ratio restoration of coded applause-like signalsAlexander Adami, Adrian Herzog, Sascha Disch, Jürgen Herre. 349-353 [doi]
• Diagonal rnns in symbolic music modelingY. Cem Sübakan, Paris Smaragdis. 354-358 [doi]
• Deep recurrent mixture of experts for speech enhancementShlomo E. Chazan, Jacob Goldberger, Sharon Gannot. 359-363 [doi]
• Fast reconstruction of sparse relative impulse responses via second-order cone programmingPavel Rajmic, Zbynek Koldovský, Marie Danková. 364-368 [doi]
• QRD based MVDR beamforming for fast tracking of speech and noise dynamicsAnna Barnov, Vered Bar Bracha, Shmulik Markovich Golan. 369-373 [doi]
• Automated audio captioning with recurrent neural networksKonstantinos Drossos, Sharath Adavanne, Tuomas Virtanen. 374-378 [doi]
• Enhancement of ambisonic binaural reproduction using directional audio coding with optimal adaptive mixingArchontis Politis, Leo McCormack, Ville Pulkki. 379-383 [doi]
• Optimizing differentiated discretization for audio circuits beyond driving point transfer functionsFrançois G. Germain, Kurt James Werner. 384-388 [doi]