WASPAA 2009: New Paltz, NY, USA

• Enrique Perez Gonzalez, Joshua D. Reiss:
  Automatic gain and fader control for live mixing. 1-4
  
• Matt Speed, Damian T. Murphy, David M. Howard:
  Acoustic coupling in multi-dimensional finite difference schemes for physically modeled voice synthesis. 5-8
  
• Laurent Oudre, Yves Grenier, Cédric Févotte:
  Chord recognition using measures of fit, chord templates and filtering methods. 9-12
  
• Gordon Wichern, Harvey D. Thornburg, Andreas Spanias:
  Unifying semantic and content-based approaches for retrieval of environmental sounds. 13-16
  
• Hiromasa Fujihara, Masataka Goto, Hiroshi G. Okuno:
  A novel framework for recognizing phonemes of singing voice in polyphonic music. 17-20
  
• Moonseok Kim, Gary P. Scavone:
  Domain decomposition method for the digital waveguide mesh. 21-24
  
• François Germain, Gianpaolo Evangelista:
  Synthesis of guitar by digital waveguides: Modeling the plectrum in the physical interaction of the player with the instrument. 25-28
  
• Nancy Bertin, Roland Badeau, Emmanuel Vincent:
  Fast bayesian nmf algorithms enforcing harmonicity and temporal continuity in polyphonic music transcription. 29-32
  
• Peter Grosche, Meinard Müller:
  Computing predominant local periodicity information in music recordings. 33-36
  
• Samuel Kim, Shrikanth Narayanan, Shiva Sundaram:
  Acoustic topic model for audio information retrieval. 37-40
  
• Christine Smit, Daniel P. W. Ellis:
  Guided harmonic sinusoid estimation in a multi-pitch environment. 41-44
  
• Johanna Devaney, Michael I. Mandel, Daniel P. W. Ellis:
  Improving MIDI-audio alignment with acoustic features. 45-48
  
• Stanislaw Andrzej Raczynski, Nobutaka Ono, Shigeki Sagayama:
  Note detection with dynamic bayesian networks as a postanalysis step for NMF-based multiple pitch estimation techniques. 49-52
  
• Graham Grindlay, Daniel P. W. Ellis:
  Multi-voice polyphonic music transcription using eigeninstruments. 53-56
  
• Ren Gang, Mark F. Bocko, Dave Headlam, Justin Lundberg:
  Polyphonic music transcription employing max-margin classification of spectrograhic features. 57-60
  
• Matthew E. P. Davies, Mark D. Plumbley, Douglas Eck:
  Towards a musical beat emphasis function. 61-64
  
• Tao T. Wang, Thomas F. Quatieri:
  Towards co-channel speaker separation BY 2-D demodulation of spectrograms. 65-68
  
• Paris Smaragdis, Gautham J. Mysore:
  Separation by "humming": User-guided sound extraction from monophonic mixtures. 69-72
  
• So-Young Jeong, Kyuhong Kim, Jae-Hoon Jeong, Kwang-Cheol Oh:
  Semi-blind disjoint non-negative matrix factorization for extracting target source from single channel noisy mixture. 73-76
  
• Jinyu Han, Bryan Pardo:
  Improving separation of harmonic sources with iterative estimation of spatial cues. 77-80
  
• Jack Xin, Meng Yu, Yingyong Qi, Hsin-I. Yang, Fan-Gang Zeng:
  A nonlocally weighted soft-constrained natural gradient algorithm for blind separation of reverberant speech. 81-84
  
• Michael I. Mandel, Daniel P. W. Ellis:
  The Ideal Interaural Parameter Mask: A bound on binaural separation systems. 85-88
  
• Keith D. Gilbert, Karen L. Payton:
  Source enumeration of speech mixtures using pitch harmonics. 89-92
  
• Onur Dikmen, Ali Taylan Cemgil:
  Unsupervised single-channel source separation using bayesian NMF. 93-96
  
• Valentin Emiya, Emmanuel Vincent, Rémi Gribonval:
  An investigation of discrete-state discriminant approaches to single-sensor source separation. 97-100
  
• Francesco Nesta, Ted S. Wada, Shigeki Miyabe, Biing-Hwang Juang:
  On the non-uniqueness problem and the semi-blind source separation. 101-104
  
• Francesco Nesta, Ted S. Wada, Biing-Hwang Juang:
  Coherent spectral estimation for a robust solution of the permutation problem. 105-108
  
• Mehrez Souden, Jacob Benesty, Sofiène Affes:
  On optimal beamforming for noise reduction and interference rejection. 109-112
  
• Haohai Sun, Shefeng Yan, U. Peter Svensson:
  Robust spherical microphone array beamforming with multi-beam-multi-null steering, and sidelobe control. 113-116
  
• Hüseyin Hacihabiboglu, Zoran Cvetkovic:
  Panoramic recording and reproduction of multichannel audio using a circular microphone array. 117-120
  
• Alexey Ozerov, Cédric Févotte, Maurice Charbit:
  Factorial Scaled Hidden Markov Model for polyphonic audio representation and source separation. 121-124
  
• Courtenay V. Cotton, Daniel P. W. Ellis:
  Finding similar acoustic events using matching pursuit and locality-sensitive hashing. 125-128
  
• Ngoc Q. K. Duong, Emmanuel Vincent, Rémi Gribonval:
  Spatial covariance models for under-determined reverberant audio source separation. 129-132
  
• Junfeng Li, Shuichi Sakamoto, Satoshi Hongo, Masato Akagi, Yôiti Suzuki:
  Two-stage binaural speech enhancement with wiener filter based on equalization-cancellation model. 133-136
  
• Malay Gupta, Sylvain Angrignon, Chris Forrester, Sean Simmons, Scott C. Douglas:
  A spatio-temporal power method for time-domain multi-channel speech enhancement. 137-140
  
• Emanuel A. P. Habets, Jacob Benesty, Sharon Gannot, Patrick A. Naylor, Israel Cohen:
  On the application of the LCMV beamformer to speech enhancement. 141-144
  
• Takuya Yoshioka, Hirokazu Kameoka, Tomohiro Nakatani, Hiroshi G. Okuno:
  Statistical models for speech dereverberation. 145-148
  
• Marcus Zeller, Luis Antonio Azpicueta-Ruiz, Walter Kellermann:
  Adaptive fir filters with automatic length optimization by monitoring a normalized combination scheme. 149-152
  
• Morag Agmon, Boaz Rafaely, Joseph Tabrikian:
  Maximum directivity beamformer for spherical-aperture microphones. 153-156
  
• Richard C. Hendriks, Richard Heusdens, Jesper Jensen:
  On robustness of multi-channel minimum mean-squared error estimators under super-Gaussian priors. 157-160
  
• Nobutaka Ono, Hitoshi Kohno, Nobutaka Ito, Shigeki Sagayama:
  Blind alignment of asynchronously recorded signals for distributed microphone array. 161-164
  
• Jens Ahrens, Sascha Spors:
  Artifacts in the sound field of a moving sound source reconstructed from a microphone array recording. 165-168
  
• Etan Fisher, Boaz Rafaely:
  Dolph-Chebyshev radial filter for the near-field spherical microphone array. 169-172
  
• Daniel M. Rasetshwane, J. Robert Boston, Ching-Chung Li, John D. Durrant, Gregory Genna:
  Enhancement of speech intelligibility using transients extracted by wavelet packets. 173-176
  
• Elias Nemer, Wilfrid Leblanc:
  Single-microphone wind noise reduction by adaptive postfiltering. 177-180
  
• Qi Li:
  An auditory-based transfrom for audio signal processing. 181-184
  
• Devangi N. Parikh, Sourabh Ravindran, David V. Anderson:
  Gain adaptation based on signal-to-noise ratio for noise suppression. 185-188
  
• Nils Höglund, Sven Nordholm:
  Improved a priori SNR estimation with application in Log-MMSE speech estimation. 189-192
  
• Lars-Johan Brännmark:
  Robust audio precompensation with probabilistic modeling of transfer function variability. 193-196
  
• Lars-Johan Brännmark, Anders Ahlén:
  Variable control of the pre-response error in mixed phase audio precompensation. 197-200
  
• Shoichiro Saito, Akira Nakagawa, Yoichi Haneda:
  Dynamic impulse response model for nonlinear acoustic system and its application to acoustic echo canceller. 201-204
  
• Ted S. Wada, Biing-Hwang Juang:
  Acoustic echo cancellation based on independent component analysis and integrated residual echo enhancement. 205-208
  
• Zaher El-Chami, Alexandre Guérin, Antoine Dinh-Tuan Pham, Christine Servière:
  A phase-based dual microphone method to count and locate audio sources in reverberant rooms. 209-212
  
• Hoang Do, Harvey F. Silverman:
  Stochastic particle filtering: A fast SRP-PHAT single source localization algorithm. 213-216
  
• Noboru Ohwada, Kenji Suyama:
  Multiple sound sources tracking method based on Subspace Tracking. 217-220
  
• Dima Khaykin, Boaz Rafaely:
  Coherent signals direction-of-arrival estimation using a spherical microphone array: Frequency smoothing approach. 221-224
  
• Bowon Lee, Ton Kalker:
  Multichannel voice activity detection with spherically invariant sparse distributions. 225-228
  
• Romain Serizel, Marc Moonen, Jan Wouters, Søren Holdt Jensen:
  A zone of quiet based approach to integrated active noise control and noise reduction in hearing AIDS. 229-232
  
• Nicolas Ellaham, Christian Giguere, Wail Gueaieb:
  A Wiener-based implementation of equalization-cancellation pre-processing for binaural speech intelligibility prediction. 233-236
  
• Francesco Nesta, Maurizio Omologo:
  Generalized State Coherence Transform for multidimensional localization of multiple sources. 237-240
  
• Katsuhiko Ishiguro, Takeshi Yamada, Shoko Araki, Tomohiro Nakatani:
  A probabilistic speaker clustering for DOA-based diarization. 241-244
  
• Sakari Tervo, Jukka Pätynen, Tapio Lokki:
  Acoustic reflection path tracing using a highly directional loudspeaker. 245-248
  
• N. R. Shabtai, Yaniv Zigel, Boaz Rafaely:
  Feature selection for room volume identification from room impulse response. 249-252
  
• Georgios N. Lilis, Daniele Angelosante, Georgios B. Giannakis:
  Parsimonious sound field synthesis using compressive sampling. 253-256
  
• Dmitry N. Zotkin, Ramani Duraiswami, Nail A. Gumerov:
  Regularized HRTF fitting using spherical harmonics. 257-260
  
• Shigeki Miyabe, Keisuke Masatoki, Hiroshi Saruwatari, Kiyohiro Shikano, Toshiyuki Nomura:
  Temporal quantization of spatial information using directional clustering for multichannel audio coding. 261-264
  
• Minjie Xie, Peter Chu, Anisse Taleb, Manuel Briand:
  ITU-T G.719: A new low-complexity full-band (20 kHZ) audio coding standard for high-quality conversational applications. 265-268
  
• Heinrich W. Löllmann, Matthias Hildenbrand, Bernd Geiser, Peter Vary:
  IIR QMF-bank design for speech and audio subband coding. 269-272
  
• Giovanni Del Galdo, Oliver Thiergart, Fabian Kuech:
  Nested microphone array processing for parameter estimation in Directional Audio Coding. 273-276
  
• Francisco Pinto, Martin Vetterli:
  Coding of spatio-temporal audio spectra using tree-structured directional filterbanks. 277-280
  
• Christiane Antweiler, Gerald Enzner:
  Perfect sequence lms for rapid acquisition of continuous-azimuth head related impulse responses. 281-284
  
• Jukka Ahonen, Ville Pulkki:
  Diffuseness estimation using temporal variation of intensity vectors. 285-288
  
• Marko Hiipakka, Matti Karjalainen, Ville Pulkki:
  Estimating ressure at eardrum with pressure-velocity measurement from ear canal entrance. 289-292
  
• Julio C. B. Torres, Mariane R. Petraglia:
  HRTF interpolation in the wavelet transform domain. 293-296
  
• Josh H. McDermott, Andrew J. Oxenham, Eero P. Simoncelli:
  Sound texture synthesis via filter statistics. 297-300
  
• Andreas Franck, Karlheinz Brandenburg:
  An overall optimization method for arbitrary sample rate converters based on integer rate SRC and lagrange interpolation. 301-304
  
• Douglas Brungart, Griffin D. Romigh:
  Spectral HRTF enhancement for improved vertical-polar auditory localization. 305-308
  
• Yan Jennifer Wu, Thushara D. Abhayapala:
  Multizone 2D soundfield reproduction via spatial band stop filters. 309-312
  
• Fabio Antonacci, Alberto Calatroni, Antonio Canclini, Andrea Galbiati, Augusto Sarti, Stefano Tubaro:
  Soundfield rendering with loudspeaker arrays through multiple beam shaping. 313-316
  
• Achim Kuntz, Rudolf Rabenstein:
  Wave field analysis using multiple radii measurements. 317-320
  
• Charles Verron, Grégory Pallone, Mitsuko Aramaki, Richard Kronland-Martinet:
  Controlling a spatialized environmental sound synthesizer. 321-324
  
• Gerald Enzner:
  3D-continuous-azimuth acquisition of head-related impulse responses using multi-channel adaptive filtering. 325-328
  
• Emmanuel Ravelli, Vinay Melkote, Kenneth Rose:
  A perceptually enhanced Scalable-to-Lossless audio coding scheme and a trellis-based approach for its optimization. 329-332
  
• Tomas Bäckström, Sascha Disch:
  Parametric AM/FM decomposition for speech and audio coding. 333-336
  
• Mikko-Ville Laitinen, Ville Pulkki:
  Binaural reproduction for Directional Audio Coding. 337-340
  
• Sriram Ganapathy, Samuel Thomas, Petr Motlícek, Hynek Hermansky:
  Applications of signal analysis using autoregressive models for amplitude modulation. 341-344
  
• Brian Hamilton, Philippe Depalle, Sylvain Marchand:
  Theoretical and practical comparisons of the reassignment method and the derivative method for the estimation of the frequency slope. 345-348
  
• Robert B. Dunn, Thomas F. Quatieri, Nicolas Malyska:
  Sinewave parameter estimation using the fast Fan-Chirp Transform. 349-352
  
• Michael M. Goodwin:
  Realization of arbitrary filters in the STFT domain. 353-356
  

Last update Fri May 25 08:47:44 2012 CET by the DBLP Team — Data released under the ODC-BY 1.0 license — See also our legal information page