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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
- 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
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
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 G. 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 S. 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
Improving MIDI-audio alignment with acoustic features. 45-48 - Stanislaw Andrzej Raczynski, Nobutaka Ono
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
Polyphonic music transcription employing max-margin classification of spectrograhic features. 57-60 - Matthew E. P. Davies
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
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
Unsupervised single-channel source separation using bayesian NMF. 93-96 - Valentin Emiya
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
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
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 - Emanuël A. P. Habets
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
Adaptive fir filters with automatic length optimization by monitoring a normalized combination scheme. 149-152 - Morag Agmon, Boaz Rafaely
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
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
A zone of quiet based approach to integrated active noise control and noise reduction in hearing AIDS. 229-232 - Nicolas Ellaham, Christian Giguère, 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
A probabilistic speaker clustering for DOA-based diarization. 241-244 - Sakari Tervo, Jukka Pätynen
Acoustic reflection path tracing using a highly directional loudspeaker. 245-248 - Noam R. Shabtai, Yaniv Zigel
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
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
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
HRTF interpolation in the wavelet transform domain. 293-296 - Josh H. McDermott, Andrew J. Oxenham
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
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 - Tom Bäckström
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
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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