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ICASSP 2011: Prague, Czech Republic
- Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2011, May 22-27, 2011, Prague Congress Center, Prague, Czech Republic. IEEE 2011, ISBN 978-1-4577-0539-7
- Petr Tichavský, Honza Cernocký, Ales Procházka:
General chair's message. - Alle-Jan van der Veen, Jonathon A. Chambers:
Technical chair's overview.
Acoustic Source Separation I
- Yun Wang, Zhijian Ou:
Combining HMM-based melody extraction and NMF-based soft masking for separating voice and accompaniment from monaural audio. 1-4 - Xabier Jaureguiberry, Pierre Leveau, Simon Maller, Juan José Burred:
Adaptation of source-specific dictionaries in Non-Negative Matrix Factorization for source separation. 5-8 - Ngoc Q. K. Duong, Emmanuel Vincent, Rémi Gribonval:
An acoustically-motivated spatial prior for under-determined reverberant source separation. 9-12 - Jani Even, Norihiro Hagita:
Resolving FD-BSS permutation for arbitrary array in presence of spatial aliasing. 13-16 - Gautham J. Mysore, Paris Smaragdis:
A non-negative approach to semi-supervised separation of speech from noise with the use of temporal dynamics. 17-20 - Augustin Lefèvre, Francis R. Bach, Cédric Févotte:
Itakura-Saito nonnegative matrix factorization with group sparsity. 21-24
Music Signal Processing I
- Jun Wu, Emmanuel Vincent, Stanislaw Andrzej Raczynski, Takuya Nishimoto, Nobutaka Ono
, Shigeki Sagayama:
Multipitch estimation by joint modeling of harmonic and transient sounds. 25-28 - Sascha Disch, Bernd Edler:
Frequency selective pitch transposition of audio signals. 29-32 - Jinyu Han, Ching-Wei Chen:
Improving melody extraction using Probabilistic Latent Component Analysis. 33-36 - Emmanouil Benetos
, Simon Dixon:
Polyphonic music transcription using note onset and offset detection. 37-40 - Hiroyuki Nawata, Noriyoshi Kamado, Hiroshi Saruwatari, Kiyohiro Shikano:
Automatic musical thumbnailing based on audio object localization and its evaluation. 41-44 - Romain Hennequin, Bertrand David, Roland Badeau:
Score informed audio source separation using a parametric model of non-negative spectrogram. 45-48
Spatial and Multichannel Signal Processing
- Sascha Spors, Jens Ahrens
:
Efficient range extrapolation of head-related impulse responses by wave field synthesis techniques. 49-52 - Mengqiu Zhang, Rodney A. Kennedy
, Thushara D. Abhayapala
:
Efficiency evaluation and orthogonal basis determination in functional HRTF modeling. 53-56 - Mark A. Poletti
, Thushara D. Abhayapala
:
Spatial sound reproduction systems using higher order loudspeakers. 57-60 - Mikko-Ville Laitinen, Ville Pulkki
:
Converting 5.1 audio recordings to B-format for directional audio coding reproduction. 61-64 - Jens Ahrens
, Sascha Spors:
An analytical approach to local sound field synthesis using linear arrays of loudspeakers. 65-68 - Antonio Canclini, Paolo Annibale, Fabio Antonacci
, Augusto Sarti, Rudolf Rabenstein, Stefano Tubaro:
A methodology for evaluating the accuracy of wave field rendering techniques. 69-72
Echo Cancellation
- Pradeep Loganathan, Emanuël A. P. Habets
, Patrick A. Naylor
:
A proportionate adaptive algorithm with variable partitioned block length for acoustic echo cancellation. 73-76 - Constantin Paleologu, Jacob Benesty
, Felix Albu
, Silviu Ciochina:
An efficient variable step-size proportionate affine projection algorithm. 77-80 - Tao Yu, John H. L. Hansen:
Relative proportionate NLMS: Improving convergence for acoustic channel identification. 81-84 - Sarmad Malik, Gerald Enzner
:
Fourier expansion of hammerstein models for nonlinear acoustic system identification. 85-88 - Moctar Mossi Idrissa, Christelle Yemdji, Nicholas W. D. Evans, Christophe Beaugeant, Philippe Degry:
Robust and low-cost cascaded non-linear acoustic echo cancellation. 89-92 - Karim Helwani, Sascha Spors, Herbert Buchner:
Spatio-temporal signal preprocessing for multichannel acoustic echo cancellation. 93-96
Microphone Array Signal Processing
- Jukka Ahonen, Ville Pulkki
:
Broadband direction estimation method utilizing combined pressure and energy gradients from optimized microphone array. 97-100 - Chiong-Ching Lai, Sven Nordholm
, Yee-Hong Leung:
Design of robust steerable broadband beamformers incorporating microphone gain and phase error characteristics. 101-104 - Dovid Levin, Sharon Gannot
, Emanuël A. P. Habets
:
Direction-of-arrival estimation using acoustic vector sensors in the presence of noise. 105-108 - Ina Kodrasi
, Thomas Rohdenburg, Simon Doclo
:
Microphone position optimization for planar superdirective beamforming. 109-112 - Haohai Sun, Edwin Mabande, Konrad Kowalczyk
, Walter Kellermann:
Joint DOA and TDOA estimation for 3D localization of reflective surfaces using eigenbeam MVDR and spherical microphone arrays. 113-116 - Haohai Sun, Heinz Teutsch, Edwin Mabande, Walter Kellermann:
Robust localization of multiple sources in reverberant environments using EB-ESPRIT with spherical microphone arrays. 117-120
Loudspeaker and Microphone Array Signal Processing
- Shmulik Markovich Golan, Sharon Gannot
, Israel Cohen:
Performance analysis of a randomly spaced wireless microphone array. 121-124 - Enzo De Sena
, Hüseyin Hacihabiboglu
, Zoran Cvetkovic:
A generalized design method for directivity patterns of spherical microphone arrays. 125-128 - Daniel P. Jarrett, Emanuël A. P. Habets
, Mark R. P. Thomas, Patrick A. Naylor
:
Simulating room impulse responses for spherical microphone arrays. 129-132 - Markus Kallinger, Michael Buerger, Oliver Thiergart, Fabian Kuech, Dirk Mahne:
Resolving spatial sampling effects in parametric directional filtering. 133-136 - Huajun Yu, Tim Fingscheidt
:
A data-driven post-filter design based on spatially and temporally smoothed a priori SNR. 137-140 - Yoichi Haneda, Ken'ichi Furuya
, Hiroaki Itou:
Design of multipole loudspeaker array based on spherical harmonic expansion. 141-144 - Benxu Liu, Bremananth Ramachandran, Andy W. H. Khong:
A wavenumber-fitting extrapolation method for FFT-based near-field acoustic holography using microphone array. 145-148 - Francesco Nesta, Maurizio Omologo
:
Approximated kernel density estimation for multiple TDOA detection. 149-152 - Edwin Mabande, Haohai Sun, Konrad Kowalczyk
, Walter Kellermann:
On 2D localization of reflectors using robust beamforming techniques. 153-156 - Anthony Lombard, Yuanhang Zheng, Walter Kellermann:
Synthesis of ICA-based methods for localization of multiple broadband sound sources. 157-160
Music Signal Processing II
- Marcelo F. Caetano
, Xavier Rodet:
Sound morphing by feature interpolation. 161-164 - Alexandros Nanopoulos, Ioannis Karydis
:
Know Thy Neighbor: Combining audio features and social tags for effective music similarity. 165-168 - Joan Serrà, Carlos A. de los Santos, Ralph G. Andrzejak
:
Nonlinear audio recurrence analysis with application to genre classification. 169-172 - Jia-Min Ren, Jyh-Shing Roger Jang
:
Time-constrained sequential pattern discovery for music genre classification. 173-176 - Thierry Bertin-Mahieux, Graham Grindlay, Ron J. Weiss, Daniel P. W. Ellis:
Evaluating music sequence models through missing data. 177-180 - Maksim Khadkevich, Maurizio Omologo
:
Time-frequency reassigned features for automatic chord recognition. 181-184 - Akira Maezawa, Hiroshi G. Okuno
, Tetsuya Ogata
, Masataka Goto
:
Polyphonic audio-to-score alignment based on Bayesian Latent Harmonic Allocation Hidden Markov Model. 185-188 - Jakob Abeßer, Olivier Lartillot, Christian Dittmar, Tuomas Eerola
, Gerald Schuller:
Modeling musical attributes to characterize ensemble recordings using rhythmic audio features. 189-192 - Nicola Montecchio, Arshia Cont:
A unified approach to real time audio-to-score and audio-to-audio alignment using sequential Montecarlo inference techniques. 193-196 - Zhiyao Duan, Bryan Pardo:
A state space model for online polyphonic audio-score alignment. 197-200
Acoustic Source Separation II
- Juan José Burred, Pierre Leveau:
Geometric multichannel common signal separation with application to music and effects extraction from film soundtracks. 201-204 - Ngoc Q. K. Duong, Hideyuki Tachibana
, Emmanuel Vincent, Nobutaka Ono
, Rémi Gribonval, Shigeki Sagayama:
Multichannel harmonic and percussive component separation by joint modeling of spatial and spectral continuity. 205-208 - Atiyeh Alinaghi, Wenwu Wang, Philip J. B. Jackson
:
Integrating binaural cues and blind source separation method for separating reverberant speech mixtures. 209-212 - Masahito Togami:
Online speech source separation based on maximum likelihood of local Gaussian modeling. 213-216 - Zafar Rafii, Bryan Pardo:
Degenerate Unmixing Estimation Technique using the Constant Q Transform. 217-220 - Zafar Rafii, Bryan Pardo:
A simple music/voice separation method based on the extraction of the repeating musical structure. 221-224 - Shoko Araki
, Tomohiro Nakatani:
Hybrid approach for multichannel source separation combining time-frequency mask with multi-channel Wiener filter. 225-228 - Hiroshi Sawada, Hirokazu Kameoka, Shoko Araki
, Naonori Ueda:
Formulations and algorithms for multichannel complex NMF. 229-232 - Yusuke Hioka
, W. Bastiaan Kleijn
:
Distributed blind source separation with an application to audio signals. 233-236 - Tomohiro Nakatani, Shoko Araki
, Takuya Yoshioka, Masakiyo Fujimoto:
Joint unsupervised learning of hidden Markov source models and source location models for multichannel source separation. 237-240
Acoustic Source Separation and Noise Reduction
- Syed Mohsen Naqvi
, Miao Yu, Jonathon A. Chambers:
Multimodal blind source separation for moving sources based on robust beamforming. 241-244 - Rajesh Jaiswal, Derry Fitzgerald, Dan Barry
, Eugene Coyle, Scott Rickard:
Clustering NMF basis functions using Shifted NMF for monaural sound source separation. 245-248 - Jinyu Han, Bryan Pardo:
Reconstructing completely overlapped notes from musical mixtures. 249-252 - Serap Kirbiz, Paris Smaragdis:
An adaptive time-frequency resolution approach for Non-negative Matrix Factorization based single channel sound source separation. 253-256 - Alexey Ozerov, Cédric Févotte, Raphaël Blouet, Jean-Louis Durrieu:
Multichannel nonnegative tensor factorization with structured constraints for user-guided audio source separation. 257-260 - James K. Murphy, Simon J. Godsill:
Joint Bayesian removal of impulse and background noise. 261-264 - Akihiko Sugiyama, Ryoji Miyahara, Masanori Kato:
An adaptive noise canceller with adaptive delay compensation for a distant noise source. 265-268 - Toby Christian Lawin-Ore, Simon Doclo
:
Analysis of rate constraints for MWF-based noise reduction in acoustic sensor networks. 269-272 - Jacob Benesty
, Yiteng Huang:
A single-channel noise reduction MVDR filter. 273-276 - Jingdong Chen, Jacob Benesty
, Yiteng Huang, Tomas Gänsler:
On single-channel noise reduction in the time domain. 277-280 - Bram Cornelis, Marc Moonen, Jan Wouters
:
A VAD-robust Multichannel Wiener Filter algorithm for noise reduction in hearing aids. 281-284
Acoustic System Modelling and Hearing Aids
- Phyu P. Khing, Eliathamby Ambikairajah
, Brett A. Swanson
:
Effect of fast AGC on cochlear implant speech intelligibility. 285-288 - Marko Hiipakka:
Estimating pressure and volume velocity in the ear canal for insert headphones. 289-292 - Nicolas Ellaham, Christian Giguère, Wail Gueaieb:
Evaluation of two speech and noise estimation methods for the assessment of nonlinear hearing aids. 293-296 - John Woodruff, DeLiang Wang:
Directionality-based speech enhancement for hearing aids. 297-300 - Ashutosh Pandey, V. John Mathews:
Offending frequency suppression with a reset algorithm to improve feedback cancellation in digital hearing aids. 301-304 - Takanori Nishino, Kazuya Takeda:
Improving head-related impulse response measured in noisy environments with spatio-temporal frequency analysis. 305-308 - Heinrich W. Löllmann, Peter Vary:
Estimation of the frequency dependent reverberation time by means of warped filter-banks. 309-312 - Lei Liao, Andy W. H. Khong:
Equalization of multichannel acoustic system using sub-systems for speech dereverberation. 313-316 - Craig J. Webb, Stefan Bilbao:
Computing room acoustics with CUDA - 3D FDTD schemes with boundary losses and viscosity. 317-320 - Ivan Dokmanic
, Yue M. Lu, Martin Vetterli
:
Can one hear the shape of a room: The 2-D polygonal case. 321-324
Audio Signal Processing
- Leung Kin Chiu, Nathan V. Parrish, David V. Anderson:
A perceptually transparent audio power reduction algorithm for loudspeaker power management. 325-328 - Amir Adler
, Valentin Emiya
, Maria G. Jafari, Michael Elad, Rémi Gribonval, Mark D. Plumbley
:
A constrained matching pursuit approach to audio declipping. 329-332 - Bruno Defraene, Toon van Waterschoot
, Moritz Diehl, Marc Moonen:
A fast projected gradient optimization method for real-time perception-based clipping of audio signals. 333-336 - Felix Weninger, Björn W. Schuller
:
Audio recognition in the wild: Static and dynamic classification on a real-world database of animal vocalizations. 337-340 - Martin Graciarena, Michelle Delplanche, Elizabeth Shriberg, Andreas Stolcke:
Bird species recognition combining acoustic and sequence modeling. 341-344 - Wei Chu, Daniel T. Blumstein
:
Noise robust bird song detection using syllable pattern-based hidden Markov models. 345-348 - Po-Sen Huang, Xiaodan Zhuang, Mark Hasegawa-Johnson:
Improving acoustic event detection using generalizable visual features and multi-modality modeling. 349-352 - Taoufik Majoul, Fathi Raouafi, Meriem Jaïdane
:
An improved scheme of audio watermarking based on turbo codes and channel effect modeling. 353-356 - Taras Butko, Climent Nadeu:
Audio segmentation of broadcast news: A hierarchical system with feature selection for the Albayzin-2010 evaluation. 357-360 - Christos Vezyrtzis, Aaron E. Klein, Dan Ellis, Yannis P. Tsividis:
Direct processing of mpeg audio using companding and BFP techniques. 361-364
Music Signal Processing III
- Hiromasa Fujihara, Masataka Goto
:
Concurrent estimation of singing voice F0 and phonemes by using spectral envelopes estimated from polyphonic music. 365-368 - Naoki Yasuraoka, Hirokazu Kameoka, Takuya Yoshioka, Hiroshi G. Okuno
:
I-Divergence-based dereverberation method with auxiliary function approach. 369-372 - Xander Fiss, Andres Kwasinski:
Automatic real-time electric guitar audio transcription. 373-376 - Ana M. Barbancho, Isabel Barbancho
, Beatriz Soto, Lorenzo J. Tardón
:
SIC receiver for polyphonic piano music. 377-380 - François Rigaud, Mathieu Lagrange, Axel Röbel, Geoffroy Peeters:
Drum extraction from polyphonic music based on a spectro-temporal model of percussive sounds. 381-384 - Sebastian Ewert
, Meinard Müller
:
Estimating note intensities in music recordings. 385-388 - Mathieu Lagrange, George Tzanetakis
:
Adaptive N-normalization for enhancing music similarity. 389-392 - Chao-Ling Hsu, DeLiang Wang, Jyh-Shing Roger Jang
:
A trend estimation algorithm for singing pitch detection in musical recordings. 393-396 - Cyril Joder, Slim Essid, Gaël Richard:
Hidden Discrete Tempo Model: A tempo-aware timing model for audio-to-score alignment. 397-400 - Benoit Fuentes, Roland Badeau, Gaël Richard:
Adaptive harmonic time-frequency decomposition of audio using shift-invariant PLCA. 401-404 - Zhi Zeng, Shuwu Zhang:
A hierarchical generative model for Generic Audio Document Categorization. 405-408
Acoustic Signal Processing
- Yoshinobu Kajikawa
:
Linearization ability evaluation of nonlinear filters employing dynamic distortion measurement. 409-412 - Hua Bao, Issa M. S. Panahi:
A new structure with spectrum-tuning of residual noise for active noise control. 413-416 - Thomas Schumacher, Hauke Krüger, Marco Jeub, Peter Vary, Christophe Beaugeant:
Active noise control in headsets: A new approach for broadband feedback ANC. 417-420 - Satoru Emura
, Yoichi Haneda:
A method for posterior frequency-domain multi-channel residual echo canceling. 421-424 - Constantin Paleologu, Jacob Benesty
, Tomas Gänsler, Silviu Ciochina:
Class of double-talk detectors based on the holder inequality. 425-428 - Felix Albu
, Constantin Paleologu, Jacob Benesty
:
A variable step size evolutionary affine projection algorithm. 429-432 - Meng Guo
, Thomas Bo Elmedyb, Søren Holdt Jensen, Jesper Jensen:
Analysis of adaptive feedback and echo cancelation algorithms in a general multiple-microphone and single-loudspeaker system. 433-436 - Terence Betlehem, Paul D. Teal
:
A constrained optimization approach for multi-zone surround sound. 437-440 - Noriyoshi Kamado, Hiroshi Saruwatari, Kiyohiro Shikano:
Robust sound field reproduction integrating multi-point sound field control and wave field synthesis. 441-444 - Jason Wung, Ted S. Wada, Biing-Hwang Juang, Bowon Lee, Ton Kalker, Ronald W. Schafer:
A system approach to residual echo suppression in robust hands-free teleconferencing. 445-448
Sound Reproduction, Synthesis, and Classification
- Zhixin Wang, Cheung-Fat Chan
:
Efficient implementation of virtual 3D sound synthesis based on combining grouped PCA and BMT. 449-452 - Tomoyasu Nakano, Masataka Goto
:
Vocalistener2: A singing synthesis system able to mimic a user's singing in terms of voice timbre changes as well as pitch and dynamics. 453-456 - Aki Härmä
:
Stereo audio classification for audio enhancement. 457-460 - Nasim Radmanesh, Ian S. Burnett
:
Reproduction of independent narrowband soundfields in a multizone surround system and its extension to speech signal sources. 461-464 - Andrew Wabnitz, Nicolas Epain, André van Schaik
, Craig T. Jin
:
Time domain reconstruction of spatial sound fields using compressed sensing. 465-468 - M. Umair Bin Altaf, Biing-Hwang Juang:
Audio signal classification with temporal envelopes. 469-472 - Courtenay V. Cotton, Daniel P. W. Ellis, Alexander C. Loui:
Soundtrack classification by transient events. 473-476