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ORCIDICASSP 2008: Las Vegas, Nevada, USA
- Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2008, March 30 - April 4, 2008, Caesars Palace, Las Vegas, Nevada, USA. IEEE 2008, ISBN 1-4244-1484-9
Environmental Sound Processing and Audio Event Detection
- Selina Chu, Shrikanth S. Narayanan, C.-C. Jay Kuo
:
Environmental sound recognition using MP-based features. 1-4 - Jiachen Xue, Gordon Wichern, Harvey D. Thornburg, Andreas Spanias:
Fast query by example of environmental sounds via robust and efficient cluster-based indexing. 5-8 - Keansub Lee, Daniel P. W. Ellis:
Detecting music in ambient audio by long-window autocorrelation. 9-12 - Tomonori Izumitani, Ryo Mukai, Kunio Kashino:
A background music detection method based on robust feature extraction. 13-16 - Xiaodan Zhuang, Xi Zhou, Thomas S. Huang, Mark Hasegawa-Johnson:
Feature analysis and selection for acoustic event detection. 17-20 - Aggelos Pikrakis, Theodoros Giannakopoulos, Sergios Theodoridis:
Gunshot detection in audio streams from movies by means of dynamic programming and Bayesian networks. 21-24
Source Separation I
- Lingyun Gu, Richard M. Stern:
Single-channel speech separation based on modulation frequency. 25-28 - Hirokazu Kameoka, Nobutaka Ono, Shigeki Sagayama:
Auxiliary function approach to parameter estimation of constrained sinusoidal model for monaural speech separation. 29-32 - B. Vikrham Gowreesunker, Ahmed H. Tewfik:
Blind source separation using monochannel overcomplete dictionaries. 33-36 - Raphaël Blouet, Guy Rapaport, Cédric Févotte:
Evaluation of several strategies for single sensor speech/music separation. 37-40 - Andrew Nesbit, Mark D. Plumbley:
Oracle estimation of adaptive cosine packet transforms for underdetermined audio source separation. 41-44 - Nilesh Madhu, Colin Breithaupt, Rainer Martin:
Temporal smoothing of spectral masks in the cepstral domain for speech separation. 45-48
Content-based Audio Processing and Music Information Retrieval
- Shiva Sundaram, Shrikanth S. Narayanan:
Audio retrieval by latent perceptual indexing. 49-52 - Lie Lu, Alan Hanjalic:
Unsupervised anchor space generation for similarity measurement of general audio. 53-56 - Daniel P. W. Ellis, Courtenay V. Cotton, Michael I. Mandel:
Cross-correlation of beat-synchronous representations for music similarity. 57-60 - Joan Serrà, Emilia Gómez:
Audio cover song identification based on tonal sequence alignment. 61-64 - Meinard Müller, Daniel Appelt:
Path-constrained partial music synchronization. 65-68 - Hiromasa Fujihara, Masataka Goto:
Three techniques for improving automatic synchronization between music and lyrics: Fricative detection, filler model, and novel feature vectors for vocal activity detection. 69-72
Microphone Arrays for Speech Enhancement
- Ernst Warsitz, Alexander Krueger, Reinhold Haeb-Umbach:
Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller. 73-76 - Mehrez Souden, Jacob Benesty, Sofiène Affes:
Microphone arrays for noise reduction with low signal distortion in room acoustics. 77-80 - Malay Gupta, Scott C. Douglas:
An iterative spatio-temporal speech enhancement algorithm for microphone arrays. 81-84 - Tomohiro Nakatani, Takuya Yoshioka, Keisuke Kinoshita, Masato Miyoshi, Biing-Hwang Juang:
Blind speech dereverberation with multi-channel linear prediction based on short time fourier transform representation. 85-88 - Jacek P. Dmochowski, Zicheng Liu, Philip A. Chou:
Blind source separation in a distributed microphone meeting environment for improved teleconferencing. 89-92 - Shoko Araki, Masakiyo Fujimoto, Kentaro Ishizuka, Hiroshi Sawada, Shoji Makino:
Speaker indexing and speech enhancement in real meetings / conversations. 93-96
Audio Analysis and Synthesis
- Bob L. Sturm, John J. Shynk, Steffen Gauglitz:
Agglomerative clustering in sparse atomic decompositions of audio signals. 97-100 - Mads Græsbøll Christensen, Pedro Vera-Candeas, Samuel Dilshan Somasundaram, Andreas Jakobsson:
Robust subspace-based fundamental frequency estimation. 101-104 - Antonio Pertusa, José Manuel Iñesta Quereda:
Multiple fundamental frequency estimation using Gaussian smoothness. 105-108 - Emmanuel Vincent, Nancy Bertin, Roland Badeau:
Harmonic and inharmonic Nonnegative Matrix Factorization for Polyphonic Pitch transcription. 109-112 - Kenichi Miyamoto, Hirokazu Kameoka, Takuya Nishimoto, Nobutaka Ono, Shigeki Sagayama:
Harmonic-Temporal-Timbral Clustering (HTTC) for the analysis of multi-instrument polyphonic music signals. 113-116 - Olaf Schleusing, Bingjun Zhang, Ye Wang:
Onset detection in pitched non-percussive music using warping-compensated correlation. 117-120 - Hélène Papadopoulos, Geoffroy Peeters:
Simultaneous estimation of chord progression and downbeats from an audio file. 121-124 - Sylvain Le Groux, Paul F. M. J. Verschure:
Perceptsynth: mapping perceptual musical features to sound synthesis parameters. 125-128 - Mark Sterling, Xiaoxiao Dong, Mark Bocko:
Representation of solo clarinet music by physical modeling synthesis. 129-132 - Jussi Pekonen, Vesa Välimäki:
Filter-based alias reduction for digital classical waveform synthesis. 133-136 - Markus S. Schlosser:
Confidence measures for acoustic detection of film slates based on time-domain features. 137-140 - Cong Li, Zhijian Ou, Wei Hu, Tao Wang, Yimin Zhang:
Caption-aided speech detection in videos. 141-144
Source Separation II
- Intae Lee, Jiucang Hao, Te-Won Lee:
Adaptive independent vector analysis for the separation of convoluted mixtures using EM algorithm. 145-148 - Hirofumi Nakajima, Kazuhiro Nakadai, Yuji Hasegawa, Hiroshi Tsujino:
Adaptive step-size parameter control for real-world blind source separation. 149-152 - Kleanthis N. Mokios, Alexandros Potamianos, Nicholas D. Sidiropoulos:
On the effectiveness of PARAFAC-based estimation for blind speech separation. 153-156 - Jani Even, Hiroshi Saruwatari, Kiyohiro Shikano:
Frequency domain semi-blind signal separation: application to the rejection of internal noises. 157-160 - Pei Zhao, Zhiping Zhang, Xihong Wu:
Monaural speech separation based on multi-scale Fan-Chirp Transform. 161-164 - Mathieu Lagrange, Luis Gustavo Martins, George Tzanetakis:
A Computationally Efficient Scheme for Dominant Harmonic Source Separation. 165-168 - Jean-Louis Durrieu, Gaël Richard, Bertrand David:
Singer melody extraction in polyphonic signals using source separation methods. 169-172 - Yipeng Li, DeLiang Wang:
Musical Sound Separation Using Pitch-Based Labeling and Binary Time-Frequency Masking. 173-176 - Yuuki Haraguchi, Shigeki Miyabe, Hiroshi Saruwatari, Kiyohiro Shikano, Toshiyuki Nomura:
Source-oriented localization control of stereo audio signals based on blind source separation. 177-180 - Kenta Niwa, Takanori Nishino, Kazuya Takeda:
Encoding large array signals into a 3D sound field representation for selective listening point audio based on blind source separation. 181-184
Audio Coding
- Sang-Wook Shin, Chang-Heon Lee, Hyen-O Oh, Hong-Goo Kang:
Designing a unified speech/audio codec by adopting a single channel harmonic source separation module. 185-188 - Morten Holm Larsen, Mads Græsbøll Christensen, Søren Holdt Jensen:
Multiple description quantization of sinusoidal parameters. 189-192 - Florin Ghido, Ioan Tabus:
Optimization-quantization for least squares estimates and its application for lossless audio compression. 193-196 - Thomas R. Fischer, Hosang Sung, Jie Zhan, Eunmi Oh:
High-quality audio transform coded excitation using trellis codes. 197-200 - Vinay Melkote, Kenneth Rose:
A two-layered trellis approach to audio encoding. 201-204 - Te Li, Susanto Rahardja, Soo Ngee Koh:
A fully scalable audio coding structure with embedded psychoacoustic model. 205-208 - Martin Holters, Udo Zölzer:
Delay-free lossy audio coding using shelving pre- and post-filters. 209-212 - Ravi K. Chivukula, Yuriy A. Reznik:
Efficient implementation of a class of MDCT/IMDCT filterbanks for speech and audio coding applications. 213-216 - Tobias Friedrich, Matthias Gruhne, Gerald Schuller:
Subband conversion for feature extraction from compressed audio. 217-220 - Srivatsan Kandadai, Joseph C. Hardin, Charles D. Creusere:
Audio quality assessment using the mean structural similarity measure. 221-224 - David P. Varodayan, Yao-Chung Lin, Bernd Girod:
Audio authentication based on distributed source coding. 225-228
Echo Cancellation
- Andy W. H. Khong, Xiang Lin, Milos Doroslovacki, Patrick A. Naylor:
Frequency domain selective tap adaptive algorithms for sparse system identification. 229-232 - James D. Gordy, Tyseer Aboulnasr, Martin Bouchard:
Reduced-complexity proportionate nlms employing block-based selective coefficient updates. 233-236 - Andy W. H. Khong, Woon-Seng Gan, Patrick A. Naylor, Mike Brookes:
A lowcomplexity fast converging partial update adaptive algorithm employing variable step-size for acoustic echo cancellation. 237-240 - Asif Iqbal Mohammad, Steven L. Grant:
Novel variable step size nlms algorithms for echo cancellation. 241-244 - Constantin Paleologu, Silviu Ciochina, Jacob Benesty:
Double-talk robust VSS-NLMS algorithm for under-modeling acoustic echo cancellation. 245-248 - James D. Gordy, Franck Beaucoup, Rafik A. Goubran:
Delayed adaptation for improved doubletalk resilience in adaptive echo cancellers. 249-252 - Ted S. Wada, Biing-Hwang Juang:
Towards robust acoustic echo cancellation during double-talk and near-end background noise via enhancement of residual echo. 253-256 - Kun Shi, Xiaoli Ma, G. Tong Zhou:
A residual echo suppression technique for systems with nonlinear acoustic echo paths. 257-260 - Diego A. Bendersky, Jack W. Stokes, Henrique S. Malvar:
Nonlinear residual acoustic echo suppression for high levels of harmonic distortion. 261-264 - Nilesh Madhu, Ivan Tashev, Alex Acero:
AN EM-based probabilistic approach for Acoustic Echo Suppression. 265-268 - Toon van Waterschoot, Marc Moonen:
Adaptive feedback cancellation for audio signals using a warped all-pole near-end signal model. 269-272 - Tim Fingscheidt, Suhadi Suhadi, Kai Steinert:
Towards objective quality assessment of speech enhancement systems in a black box approach. 273-276
Microphone Arrays
- Dmitry N. Zotkin, Ramani Duraiswami, Nail A. Gumerov:
Sound field decomposition using spherical microphone arrays. 277-280 - Boaz Rafaely:
Spherical microphone array with multiple nulls for analysis of directional room impulse responses. 281-284 - Albenzio Cirillo, Raffaele Parisi, Aurelio Uncini:
Sound mapping in reverberant rooms by a robust direct method. 285-288 - Jacek P. Dmochowski, Jacob Benesty, Sofiène Affes:
Fast steered response power source localization using inverse mapping of relative delays. 289-292 - Xionghu Zhong, James R. Hopgood:
Nonconcurrent multiple speakers tracking based on extended Kalman particle filter. 293-296 - Angela Quinlan, Futoshi Asano:
Tracking a varying number of speakers using particle filtering. 297-300 - Hoang Do, Harvey F. Silverman:
A method for locating multiple sources from a frame of a large-aperture microphone array data without tracking. 301-304 - Ville Myllylä, Matti Hämäläinen:
Adaptive beamforming methods for dynamically steered microphone array systems. 305-308 - Benny Sällberg, Nedelko Grbic, Ingvar Claesson:
An adaptive blind beamformer with an integrated single-channel noise reduction method for robust realtime blind speech extraction. 309-312 - Shintaro Takada, Tetsuji Ogawa, Kenzo Akagiri, Tetsunori Kobayashi:
Speech enhancement using square microphone array for mobile devices. 313-316 - Nobutaka Ito, Nobutaka Ono, Shigeki Sagayama:
A blind noise decorrelation approach with crystal arrays on designing post-filters for diffuse noise suppression. 317-320 - Jacob Benesty, Jingdong Chen, Yiteng Huang:
A minimum speech distortion multichannel algorithm for noise reduction. 321-324
Acoustics, Active Noise Control, and Hearing Aids
- Shigekatsu Irie, Shigeki Hirobayashi:
Simulation of an acoustic system using Power Envelope Inverse Filtering. 325-328 - Jimi Yung-Chuan Wen, Emanuël A. P. Habets, Patrick A. Naylor:
Blind estimation of reverberation time based on the distribution of signal decay rates. 329-332 - Vincent Grulier, Sébastien Debert, Jérôme I. Mars, Marc Pachebat:
Acoustic and turbulent wavenumbers separation in wall pressure array signals using EMD in spatial domain. 333-336 - Yoko Yamakata, Michiaki Katsumoto, Toshiyuki Kimura:
Directional sound radiation system using a large planar diaphragm incorporating multiple vibrators. 337-340 - Alberto Carini, Silvia Malatini:
Auxiliary noise power scheduling for feedforward active noise control. 341-344 - Flavio P. Ribeiro, Vítor H. Nascimento:
A robust and computationally efficient method for tonal active noise control using a simplified secondary path model. 345-348 - Miguel Ferrer, Alberto González, Maria de Diego, Gema Piñero:
Mean square analysis of a fast filtered-x affine projection algorithm. 349-352 - Masaaki Nagahara, Yutaka Yamamoto:
Hybrid design of filtered-x adaptive algorithm via sampled-data control theory. 353-356 - Ashutosh Pandey, V. John Mathews, Michael Nilsson:
Adaptive gain processing to improve feedback cancellation in digital hearing aids. 357-360 - Harish Krishnamoorthi, Visar Berisha, Andreas Spanias:
A low-complexity loudness estimation algorithm. 361-364
Spatial and Multichannel Audio
- Francisco Pinto, Martin Vetterli:
Wave Field coding in the spacetime frequency domain. 365-368 - Bin Cheng, Christian H. Ritz, Ian S. Burnett:
A Spatial Squeezing approach to Ambisonic audio compression. 369-372 - Jens Ahrens, Sascha Spors:
Analytical driving functions for higher order Ambisonics. 373-376 - Yan Jennifer Wu, Thushara D. Abhayapala:
Soundfield reproduction using theoretical continuous loudspeaker. 377-380 - Ivan Tashev, Jasha Droppo, Michael L. Seltzer, Alex Acero:
Robust design of wideband loudspeaker arrays. 381-384 - Lars-Johan Brännmark, Anders Ahlén:
Robust loudspeaker equalization based on position-independent excess phase modeling. 385-388 - Andy W. H. Khong, Xiang Lin, Patrick A. Naylor:
Algorithms for identifying clusters of near-common zeros in multichannel blind system identification and equalization. 389-392 - Gerald Enzner:
Analysis and optimal control of LMS-type adaptive filtering for continuous-azimuth acquisition of head related impulse responses. 393-396 - Jens Hannemann, Christopher A. Leedy, Kevin D. Donohue, Sascha Spors, Alexander Raake:
A comparative study of perceptional quality between wavefield synthesis and Multipole-Matched Rendering for spatial audio. 397-400 - Yuuta Yuyama, Shigeki Miyabe, Hiroshi Saruwatari, Kiyohiro Shikano:
Hybrid structure of inverse filtering and DOA-parameterized wavefront synthesis. 401-404 - Yiteng Huang, Jacob Benesty, Jingdong Clien:
Generalized crosstalk cancellation and equalization using multiple loudspeakers for 3D sound reproduction at the ears of multiple listeners. 405-408 - Michael M. Goodwin:
Geometric signal decompositions for spatial audio enhancement. 409-412
EEG Signal Processing
- Shiliang Sun, Man Lan, Yue Lu:
Adaptive EEG signal classification using stochastic approximation methods. 413-416 - Xinyi Yong, Rabab K. Ward, Gary E. Birch:
Sparse spatial filter optimization for EEG channel reduction in brain-computer interface. 417-420 - Lisa Wong, Waleed H. Abdulla:
Automatic detection of preterm neonatal EEG background states. 421-424 - Toshihisa Tanaka, Yuki Saito:
Rhythmic component extraction for multi-channel EEG data analysis. 425-428 - Yonghong Huang, Deniz Erdogmus, Santosh Mathan, Misha Pavel:
Large-scale image database triage via EEG evoked responses. 429-432 - Kok-Kiong Poh, Pina Marziliano:
Compression of neonatal EEG seizure signalswith finite rate of innovation. 433-436
Functional MRI
- David M. Afonso, João M. Sanches, Martin H. Lauterbach:
Joint Bayesian detection of brain activated regions and local HRF estimation in functional MRI. 437-440 - Xiaomu Song, Tongyou Ji, Alice M. Wyrwicz:
Baseline drift and physiological noise removal in high field FMRI data using kernel PCA. 441-444 - Sarah Lee, Fernando O. Zelaya, Stephanie A. Amiel, Michael J. Brammer:
A completely data-driven method for detecting neuronal activation in FMRI. 445-448 - Jingyu Liu, Lai Xu, Arvind Caprihan, Vince D. Calhoun:
Extracting principle components for discriminant analysis of FMRI images. 449-452 - Joakim Rydell, Magnus Borga, Hans Knutsson:
Robust correlation analysis with an application to functional MRI. 453-456 - Zheng Ma, Z. Jane Wang, Martin J. McKeown:
Probabilistic Boolean Network for inferring brain connectivity using FMRI data. 457-460
Processing of Physiological Signals
- Thato Tsalaile, Syed M. Naqvi, Kianoush Nazarpour, Saeid Sanei, Jonathon A. Chambers:
Blind source extraction of heart sound signals from lung sound recordings exploiting periodicity of the heart sound. 461-464 - Hamid Reza Mohseni, Edward L. Wilding, Saeid Sanei:
Single trial estimation of event-related potentials using particle filtering. 465-468 - Jithendra Vepa, Paresh Tolay, Abhishek Jain:
Segmentation of heart sounds using simplicity features and timing information. 469-472