Stop the war!Остановите войну!
for scientists:
default search action
Google
Google Scholar
Semantic Scholar
Internet Archive Scholar
CiteSeerX
ORCIDICASSP 2010: Dallas, Texas, USA
- Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2010, 14-19 March 2010, Sheraton Dallas Hotel, Dallas, Texas, USA. IEEE 2010, ISBN 978-1-4244-4296-6
- Scott C. Douglas:
General chair's message.
Source Separation
- Francesco Nesta, Maurizio Omologo
:
Cooperative Wiener-ICA for source localization and Separation by distributed microphone arrays. 1-4 - Shoko Araki, Tomohiro Nakatani, Hiroshi Sawada:
Simultaneous clustering of mixing and spectral model parameters for blind sparse source separation. 5-8 - Ngoc Q. K. Duong, Emmanuel Vincent, Rémi Gribonval:
Under-determined convolutive blind source separation using spatial covariance models. 9-12 - Tomohiro Nakatani, Shoko Araki:
Single channel source separation based on sparse source observation model with harmonic constraint. 13-16 - Thom-Thi Ngo, Seung-Hyon Nam:
An expectation-maximization method for the permutation problem in frequency-domain blind source separation. 17-20 - Pejman Mowlaee, Mads Græsbøll Christensen, Søren Holdt Jensen:
Improved single-channel speech separation using sinusoidal modeling. 21-24
Audio Analysis and Synthesis
- Joonas Nikunen, Tuomas Virtanen:
Noise-to-mask ratio minimization by weighted non-negative matrix factorization. 25-28 - Damián Marelli, Mitsuko Aramaki, Richard Kronland-Martinet, Charles Verron:
A unified time-frequency method for synthesizing noisy sounds with short transients and narrow spectral components. ICASSP 2010: 29-32 - Mahdi Triki, Dirk T. M. Slock:
First-order global AM-FM decomposition and application to music analysis and transformation. 33-36 - Alex Fink, Harvey D. Thornburg, Andreas Spanias:
An individually tunable perfect reconstruction filterbank for banded waveguide synthesis. 37-40 - Jörg-Hendrik Bach, Birger Kollmeier, Jörn Anemüller:
Modulation-based detection of speech in real background noise: Generalization to novel background classes. 41-44 - Mickael Rouvier, Georges Linarès, Driss Matrouf:
On-the-fly video genre classification by combination of audio features. 45-48
Music Signal Processing
- Keisuke Kinoshita, Tomohiro Nakatani, Masato Miyoshi:
Blind upmix of stereo music signals using multi-step linear prediction based reverberation extraction. 49-52 - Naoki Yasuraoka, Takuya Yoshioka, Tomohiro Nakatani, Atsushi Nakamura, Hiroshi G. Okuno:
Music dereverberation using harmonic structure source model and Wiener filter. 53-56 - Zhiyao Duan, Jinyu Han, Bryan Pardo:
Song-level multi-pitch tracking by heavily constrained clustering. 57-60 - Dan Tidhar, Matthias Mauch, Simon Dixon:
High precision frequency estimation for harpsichord tuning classification. 61-64 - Suman V. Ravuri, Daniel P. W. Ellis:
Cover song detection: From high scores to general classification. 65-68 - Qian Lin, Lie Lu, Christopher Weare, Frank Seide:
Music rhythm characterization with application to workout-mix generation. 69-72
Loudspeaker and Microphone Array Signal Processing
- Mihailo Kolundzija, Christof Faller, Martin Vetterli:
Baffled circular loudspeaker array with broadband high directivity. 73-76 - Aastha Gupta, Thushara D. Abhayapala:
Double sided cone array for spherical harmonic analysis of wavefields. 77-80 - Mehrez Souden, Jacob Benesty, Sofiène Affes:
Eigenanalysis-based broadband source localization. 81-84 - Alexander Bertrand, Marc Moonen:
Energy-based multi-speaker voice activity detection with an ad hoc microphone array. 85-88 - Mehrez Souden, Jacob Benesty, Sofiène Affes:
Linear filtering for noise reduction and interference rejection. 89-92 - Yu Takahashi, Hiroshi Saruwatari, Hiroshi Shikano, Kazunobu Kondo:
Theoretical musical-noise analysis and its generalization for methods of integrating beamforming and spectral subtraction based on higher-order statistics. 93-96
Loudspeaker and Microphone Array Signal Processing
- Joshua Atkins:
Optimal spatial sampling for spherical loudspeaker arrays. 97-100 - Paul D. Teal, Terence Betlehem, Mark A. Poletti:
An algorithm for power constrained holographic reproduction of sound. 101-104 - Jens Ahrens, Sascha Spors:
An analytical approach to 2.5D sound field reproduction employing linear distributions of non-omnidirectional loudspeakers. 105-108 - Wancheng Zhang, Emanuël A. P. Habets, Patrick A. Naylor:
A System-Identification-Error-Robust Method for equalization of multichannel acoustic systems. 109-112 - Yotam Peled, Boaz Rafaely:
Method for dereverberation and noise reduction using spherical microphone arrays. 113-116 - Haohai Sun, Shefeng Yan, U. Peter Svensson:
Space domain optimal beamforming for spherical microphone arrays. 117-120 - Lae-Hoon Kim, Mark Hasegawa-Johnson, Gerasimos Potamianos, Vit Libal:
Joint estimation of DOA and speech based on EM beamforming. 121-124 - Hoang Do, Harvey F. Silverman:
SRP-PHAT methods of locating simultaneous multiple talkers using a frame of microphone array data. 125-128 - Kenta Niwa, Yusuke Hioka, Sumitaka Sakauchi, Ken'ichi Furuya, Youichi Haneda:
Estimation of sound source orientation using eigenspace of spatial correlation matrix. 129-132 - Masahito Togami, Yohei Kawaguchi:
Head orientation estimation of a speaker by utilizing kurtosis of a DOA histogram with restoration of distance effect. 133-136 - Muawiyath Shujau, Christian H. Ritz, Ian S. Burnett:
Speech enhancement via separation of sources from co-located microphone recordings. 137-140 - Deep Sen, Shuai Wang, Andy Deffrasnes:
Psychoacoustically motivated, frequency dependent Tikhonov regularization for soundfield parametrization. 141-144 - Pengxiao Teng, Anthony Lombard, Walter Kellermann:
Disambiguation in multidimensional tracking of multiple acoustic sources using a Gaussian likelihood criterion. 145-148
Room Acoustics and Hearing Aids
- Yusuke Hioka, Kenta Niwa, Sumitaka Sakauchi, Ken'ichi Furuya, Youichi Haneda:
Estimating direct-to-reverberant energy ratio based on spatial correlation model segregating direct sound and reverberation. 149-152 - Sakari Tervo, Teemu Korhonen:
Estimation of reflective surfaces from continuous signals. 153-156 - Demba E. Ba, Flavio P. Ribeiro, Cha Zhang, Dinei A. F. Florêncio:
L1 regularized room modeling with compact microphone arrays. 157-160 - Ren Gang, Mark F. Bocko, Dave Headlam:
Reverberation features identification from music recordings using the discrete wavelet transform. 161-164 - Rebecca Stewart, Mark B. Sandler:
Database of omnidirectional and B-format room impulse responses. 165-168 - Navarun Gupta, Armando Barreto, Manan Joshi, Juan Carlos Agudelo:
HRTF database at FIU DSP Lab. 169-172 - Ashutosh Pandey, V. John Mathews:
Improving adaptive feedback cancellation in digital hearing aids through offending frequency suppression. 173-176 - Randall S. Plate, Yongchao Wang, Zhi-Quan Luo, Chris Gao:
Adaptive feedback cancellation in hearing aids using the IPLS algorithm. 177-180 - Kim Ngo, Toon van Waterschoot, Mads Græsbøll Christensen, Marc Moonen, Søren Holdt Jensen, Jan Wouters:
Adaptive feedback cancellation in hearing aids using a sinusoidal near-end signal model. 181-184 - Juan-Juan Xiang, Martin F. McKinney, Kelly Fitz, Tao Zhang:
Evaluation of sound classification algorithms for hearing aid applications. 185-188 - Janet C. Rutledge, Peggy B. Nelson, Juan Carlos Tejero-Calado, Jonathan K. Chang, Russell R. Williams II:
Performance of sinusoidal model based amplitude compression in fluctuating noise. 189-192 - Qing Yang, John G. Harris:
A higher-order spectro-temporal integration model for predicting signal audibility. 193-196 - Konrad Kowalczyk, Maarten van Walstijn:
A comparison of nonstaggered compact FDTD schemes for the 3D wave equation. 197-200
Signal Enhancement and Source Separation
- Shmulik Markovich Golan, Sharon Gannot, Israel Cohen:
Subspace tracking of multiple sources and its application to speakers extraction. 201-204 - Jacob Benesty, Jingdong Chen, Yiteng Huang:
Study of the widely linear Wiener filter for noise reduction. 205-208 - Jingdong Chen, Jacob Benesty, Yiteng Huang:
Analysis of the frequency-domain Wiener filter with the prediction gain. 209-212 - Jani Even, Hiroshi Saruwatari, Kiyohiro Shikano, Tomoya Takatani:
Complex Newton algorithm for blind signal extraction of speech in diffuse noise. 213-216 - Kyuhong Kim, So-Young Jeong, Jae-Hoon Jeong, Kwang-Cheol Oh, Jeongsu Kim:
Dual channel noise reduction method using phase difference-based spectral amplitude estimation. 217-220 - Manuel Moussallam, Pierre Leveau, Si-Mohamed Aziz Sbai:
Sound enhancement using sparse approximation with speclets. 221-224 - Benedikt Loesch, Francesco Nesta, Bin Yang:
On the robustness of the multidimensional state coherence transform for solving the permutation problem of frequency-domain ICA. 225-228 - Motoki Ogasawara, Takanori Nishino, Kazuya Takeda:
A small dodecahedral microphone array for blind source separation. 229-232 - Stratis Sofianos, Aladdin M. Ariyaeeinia, Richard Polfreman:
Towards effective singing voice extraction from stereophonic recordings. 233-236 - Michael Stark, Franz Pernkopf:
On optimizing the computational complexity for VQ-based single channel source separation. 237-240 - Dang Hai Tran Vu, Reinhold Haeb-Umbach:
Blind speech separation employing directional statistics in an Expectation Maximization framework. 241-244 - Mathieu Parvaix, Laurent Girin:
Informed source separation of underdetermined instantaneous stereo mixtures using source index embedding. 245-248
Audio Analysis, Synthesis, and Processing
- Yannis Panagakis, Constantine Kotropoulos:
Music genre classification via Topology Preserving Non-Negative Tensor Factorization and sparse representations. 249-252 - Gordon Wichern, Makoto Yamada, Harvey D. Thornburg, Masashi Sugiyama, Andreas Spanias:
Automatic audio tagging using covariate shift adaptation. 253-256 - Dirk von Zeddelmann, Frank Kurth, Meinard Müller:
Perceptual audio features for unsupervised key-phrase detection. 257-260 - Vishwa Gupta, Gilles Boulianne, Patrick Cardinal:
Content-based audio copy detection using nearest-neighbor mapping. 261-264 - Emiru Tsunoo, Taichi Akase, Nobutaka Ono, Shigeki Sagayama:
Music mood classification by rhythm and bass-line unit pattern analysis. 265-268 - Danil Korchagin, Philip N. Garner, John Dines:
Automatic temporal alignment of AV data with confidence estimation. 269-272 - Toni Hirvonen, Athanasios Mouchtaris:
Top-down strategies in parameter selection of sinusoidal modeling of audio. 273-276 - Kazuhiro Kondo:
Evaluation of a stereo audio data hiding method using inter-channel decorrelator polarity. 277-280 - Elsa Dupraz, Gaël Richard:
Robust frequency-based Audio Fingerprinting. 281-284 - Dimitar Kostadinov, Joshua D. Reiss, Valeri M. Mladenov:
Evaluation of Distance Based Amplitude panning for spatial audio. ICASSP 2010: 285-288 - Bo Hang, Ruimin Hu:
Spatial audio cues based surveillance audio attention model. 289-292 - Martin Graciarena, Michelle Delplanche, Elizabeth Shriberg, Andreas Stolcke, Luciana Ferrer:
Acoustic front-end optimization for bird species recognition. 293-296 - Marius Vasile Ghiurcau, Corneliu Rusu, Radu Ciprian Bilcu:
Wildlife intruder detection using sounds captured by acoustic sensors. 297-300 - Nay Oo, Woon-Seng Gan, Wee-Tong Lim:
Generalized harmonic analysis of Arc-Tangent Square Root (ATSR) nonlinear device for virtual bass system. 301-304
Echo Cancellation and Active Noise Control
- Christian Schüldt, Fredric Lindström, Ingvar Claesson:
An improved deviation measure for two-path echo cancellation. 305-308 - Constantin Paleologu, Jacob Benesty, Silviu Ciochina:
An improved proportionate NLMS algorithm based on the l0 norm. 309-312 - Moctar Mossi Idrissa, Nicholas W. D. Evans, Christophe Beaugeant:
An assessment of linear adaptive filter performance with nonlinear distortions. 313-316 - Pradeep Loganathan, Emanuël A. P. Habets, Patrick A. Naylor:
Performance analysis of IPNLMS for identification of time-varying systems. 317-320 - Karim Helwani, Herbert Buchner, Sascha Spors:
Source-domain adaptive filtering for MIMO systems with application to acoustic echo cancellation. 321-324 - Akihiko Sugiyama, Yuusuke Mizuno, Akihiro Hirano, Kenji Nakayama:
A stereo echo canceller with simultaneous input-sliding and sliding-period control. 325-328 - Laura Romoli, Stefania Cecchi, Lorenzo Palestini, Paolo Peretti, Francesco Piazza:
A novel approach to channel decorrelation for stereo Acoustic Echo Cancellation based on missing fundamental theory. 329-332 - Tiange Shao, Yahong Rosa Zheng, Jacob Benesty:
A variable step-size normalized sign algorithm for acoustic echo cancelation. 333-336 - Mehdi Bekrani, Andy W. H. Khong, Mojtaba Lotfizad:
A clipping-based adaptive filtering approach for stereophonic acoustic echo cancellation. 337-340 - Hua Bao, Issa M. S. Panahi:
Psychoacoustic active noise control based on delayless subband adaptive filtering. 341-344 - Hyeon-Jin Jeon, Shin-Wook Kim, Min-Woo Park, Woo-Geun Lee, Tae-Gyu Chang, Sen M. Kuo:
Frequency mismatch in narrowband active noise control. 345-348 - Ali A. Milani, Govind Kannan, Issa M. S. Panahi:
On maximum achievable noise reduction in ANC systems. 349-352
Audio Coding
- Michael Werner, Gerald Schuller:
An SBR tool for very lowdelay applications with flexible crossover frequency. 353-356 - Frederik Nagel, Sascha Disch, Stephan Wilde:
A continuous modulated single sideband bandwidth extension. 357-360 - Janusz Klejsa, Minyue Li, W. Bastiaan Kleijn:
Flexcode - flexible audio coding. 361-364 - Emmanuel Ravelli, Vinay Melkote, Tejaswi Nanjundaswamy, Kenneth Rose:
Joint optimization of the perceptual core and lossless compression layers in scalable audio coding. 365-368 - Shuhua Zhang, Weibei Dou, Ping Chi, Huazhong Yang:
MDCT spectrum separation: Catching the fine spectral structures for stereo coding. 369-372 - Wisarn Patchoo, Thomas R. Fischer:
Gaussian-mixture modeling of lattice-based spherical vector quantization performance in transform audio coding. 373-376 - Haiyan Shu, Haibin Huang, Te Li, Susanto Rahardja:
Enhanced scalable to lossless audio coding scheme. 377-380 - Yong-tao Sha, Changchun Bao, Mao-shen Jia, Xin Liu:
High frequency reconstruction of audio signal based on chaotic prediction theory. 381-384 - Se-Woon Jeon, Dongil Hyun, Jeongil Seo, Young-Cheol Park, Dae Hee Youn:
Enhancement of principal to ambient energy ratio for PCA-based parametric audio coding. 385-388 - Toni Hirvonen, Athanasios Mouchtaris:
Sinusoidal spatial audio coding for low-bitrate binaural reproduction. 389-392 - Pasi Ojala, Mikko Tammi, Miikka Vilermo:
Parametric binaural audio coding. 393-396 - Ruijing Hu, Xucen Huang, Michel Kieffer, Olivier Derrien, Pierre Duhamel:
Robust critical data recovery for MPEG-4 AAC encoded bitstreams. 397-400
Music Signal Processing
- Jayme G. A. Barbedo, George Tzanetakis:
Instrument identification in polyphonic music signals based on individual partials. 401-404 - Mathieu Lagrange, Roland Badeau, Gaël Richard:
Robust similarity metrics between audio signals based on asymmetrical spectral envelope matching. 405-408 - Cyril Joder, Slim Essid, Gaël Richard:
A comparative study of tonal acoustic features for a symbolic level music-to-score alignment. 409-412 - Steven K. Tjoa, Matthew C. Stamm, W. Sabrina Lin, K. J. Ray Liu:
Harmonic variable-size dictionary learning for music source separation. 413-416 - Thomas A. Baran, Nicolas Malyska, Thomas F. Quatieri:
Preserving the character of perturbations in scaled pitch contours. 417-420 - Aida Austin, Elliot Moore II, Udit Gupta, Parag Chordia:
Characterization of movie genre based on music score. 421-424 - Hideyuki Tachibana, Takuma Ono, Nobutaka Ono, Shigeki Sagayama:
Melody line estimation in homophonic music audio signals based on temporal-variability of melodic source. 425-428 - Aaron B. Greenblatt, Jonathan S. Abel, David P. Berners:
A hybrid reverberation crossfading technique. 429-432 - Mark Sterling, Mark Bocko:
Empirical physical modeling for bowed string instruments. 433-436 - Garth Griffin, Youngmoo E. Kim, Douglas Turnbull:
Beat-Sync-Mash-Coder: A web application for real-time creation of beat-synchronous music mashups. 437-440 - Lise Regnier, Geoffroy Peeters:
Partial clustering using a time-varying frequency model for singing voice detection. 441-444 - Romain Hennequin, Roland Badeau, Bertrand David:
NMF with time-frequency activations to model non stationary audio events. 445-448 - Steven K. Tjoa, K. J. Ray Liu:
Multiplicative update rules for nonnegative matrix factorization with co-occurrence constraints. 449-452
Biomedical Imaging
- Doru-Cristian Balcan, Gowri Srinivasa, Matthew C. Fickus, Jelena Kovacevic:
Convergence behavior of the Active Mask segmentation algorithm. 453-456 - Jun Kong, Lee A. D. Cooper, Ashish Sharma, Tahsin M. Kurç, Daniel J. Brat, Joel H. Saltz:
Texture based image recognition in microscopy images of diffuse gliomas with multi-class gentle boosting mechanism. 457-460