WASPAA 2009: New Paltz, NY, USA
IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, WASPAA '09, New Paltz, NY, USA, October 18-21, 2009. IEEE 2009 ISBN 978-1-4244-3678-1- 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
- 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, Wilfried 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
- 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
- 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
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last updated on 2013-06-18T211851 by the dblp team