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ORCIDICASSP 2013: Vancouver, BC, Canada
- IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2013, Vancouver, BC, Canada, May 26-31, 2013. IEEE 2013
AASP-L1: MUSIC - TRANSCRIPTION
- Hélène Papadopoulos, George Tzanetakis
:
Exploiting structural relationships in audio music signals using Markov Logic Networks. 1-5 - Felix Weninger, Christian Kirst, Björn W. Schuller, Hans-Joachim Bungartz:
A discriminative approach to polyphonic piano note transcription using supervised non-negative matrix factorization. 6-10 - François Rigaud, Antoine Falaize, Bertrand David, Laurent Daudet:
Does inharmonicity improve an NMF-based piano transcription model? 11-15 - Ken O'Hanlon, Mark D. Plumbley:
Automatic Music Transcription using row weighted decompositions. 16-20 - Tiago Fernandes Tavares, Jayme Garcia Arnal Barbedo, Romis Attux, Amauri Lopes:
Unsupervised training of detection threshold for polyphonic musical note tracking based on event periodicity. 21-25 - Holger Kirchhoff, Simon Dixon, Anssi Klapuri:
Missing template estimation for user-assisted music transcription. 26-30
AASP-L2: MUSIC - CONTENT
- Chuang Shi, Hao Mu, Woon-Seng Gan:
A psychoacoustical preprocessing technique for virtual bass enhancement of the parametric loudspeaker. 31-35 - Hao Mu, Woon-Seng Gan, Ee-Leng Tan:
A timbre matching approach to enhance audio quality of psychoacoustic bass enhancement system. 36-40 - Andreas Franck, Vesa Välimäki:
An ideal integrator for higher-order integrated wavetable synthesis. 41-45 - Sebastian Ewert, Meinard Müller, Mark B. Sandler:
Efficient data adaption for musical source separation methods based on parametric models. 46-50 - José Ricardo Zapata, Emilia Gómez:
Using voice suppression algorithms to improve beat tracking in the presence of highly predominant vocals. 51-55 - Antonis Theodoridis, Constantine Kotropoulos, Yannis Panagakis:
Music recommendation using hypergraphs and group sparsity. 56-60
AASP-L3: MULTICHANNEL SOURCE SEPARATION
- Shuhua Zhang, Laurent Girin, Antoine Liutkus:
Informed Source Separation from compressed mixtures using spatial wiener filter and quantization noise estimation. 61-65 - Antoine Liutkus, Roland Badeau, Gaël Richard:
Low bitrate informed source separation of realistic mixtures. 66-70 - Yuki Mitsufuji, Axel Roebel:
Sound source separation based on non-negative tensor factorization incorporating spatial cue as prior knowledge. 71-75 - Antoine Deleforge, Florence Forbes, Radu Horaud:
Variational EM for binaural sound-source separation and localization. 76-80 - Benxu Liu, V. G. Reju, Andy W. H. Khong:
Underdetermined instantaneous blind source separation of sparse signals with temporal structure using the state-space model. 81-85 - Francesco Nesta, Mahmoud Fakhry:
Unsupervised spatial dictionary learning for sparse underdetermined multichannel source separation. 86-90
AASP-L4: MICROPHONE-ARRAY BEAMFORMING AND SOURCE LOCALIZATION
- Dovid Levin, Emanuël A. P. Habets, Sharon Gannot:
Robust beamforming using sensors with nonidentical directivity patterns. 91-95 - Florian Heese, Magnus Schaefer, Jona Wernerus, Peter Vary:
Numerical near field optimization of a non-uniform sub-band filter-and-sum beamformer. 96-100 - Noam R. Shabtai, Boaz Rafaely:
Binaural sound reproduction beamforming using spherical microphone arrays. 101-105 - Nikolay D. Gaubitch, W. Bastiaan Kleijn, Richard Heusdens:
Auto-localization in ad-hoc microphone arrays. 106-110 - Mehrez Souden, Keisuke Kinoshita, Tomohiro Nakatani:
An integration of source location cues for speech clustering in distributed microphone arrays. 111-115 - Florian Jacob, Joerg Schmalenstroeer, Reinhold Haeb-Umbach:
DOA-based microphone array postion self-calibration using circular statistics. 116-120 - Ken'ichi Kumatani, Rita Singh, Friedrich Faubel, John W. McDonough, Youssef Oualil:
Joint constrained maximum likelihood regression for overlapping speech recognition. 121-125 - Emmanuel Vincent, Jon Barker, Shinji Watanabe, Jonathan Le Roux, Francesco Nesta, Marco Matassoni:
The second 'chime' speech separation and recognition challenge: Datasets, tasks and baselines. 126-130 - Cyril Joder, Felix Weninger, David Virette, Björn W. Schuller:
Integrating noise estimation and factorization-based speech separation: A novel hybrid approach. 131-135 - Pablo Sprechmann, Alexander M. Bronstein, Michael M. Bronstein, Guillermo Sapiro:
Learnable low rank sparse models for speech denoising. 136-140 - Dennis L. Sun, Gautham J. Mysore:
Universal speech models for speaker independent single channel source separation. 141-145 - Yu Ting Yeung, Tan Lee, Cheung-Chi Leung:
Using dynamic conditional random field on single-microphone speech separation. 146-150
AASP-L6: ROOM ACOUSTICS AND ACOUSTIC SYSTEM MODELING
- Ina Kodrasi, Stefan Goetze, Simon Doclo:
A perceptually constrained channel shortening technique for speech dereverberation. 151-155 - Ronen Talmon, Emanuël A. P. Habets:
Blind reverberation time estimation by intrinsic modeling of reverberant speech. 156-160 - James Eaton, Nikolay D. Gaubitch, Patrick A. Naylor:
Noise-robust reverberation time estimation using spectral decay distributions with reduced computational cost. 161-165 - Samuel Siltanen, Alex Southern, Lauri Savioja:
Finite-difference time domain method source calibration for hybrid acoustics modeling. 166-170 - Jonathan Botts, Lauri Savioja:
Integrating finite difference schemes for scalar and vector wave equations. 171-175 - Joshua Atkins, Adam Strauss, Chen Zhang:
Approximate convolution using partitioned truncated singular value decomposition filtering. 176-180
AASP-P1: MUSIC TRANSCRIPTION AND ANALYSIS
- Akshay Anantapadmanabhan, Ashwin Bellur, Hema A. Murthy:
Modal analysis and transcription of strokes of the mridangam using non-negative matrix factorization. 181-185 - Rajeev Rajan, Hema A. Murthy:
Group delay based melody monopitch extraction from music. 186-190 - Ryuichi Yamamoto, Shinji Sako, Tadashi Kitamura:
Robust on-line algorithm for real-time audio-to-score alignment based on a delayed decision and anticipation framework. 191-195 - Kazuki Yazawa, Daichi Sakaue, Kohei Nagira, Katsutoshi Itoyama, Hiroshi G. Okuno:
Audio-based guitar tablature transcription using multipitch analysis and playability constraints. 196-200 - Dogac Basaran, A. Taylan Cemgil, Emin Anarim:
SMC samplers for multiresolution audio sequence alignment. 201-205 - Cyril Joder, Björn W. Schuller:
Off-line refinement of audio-to-score alignment by observation template adaptation. 206-210 - José Manuel Iñesta Quereda, Carlos Pérez-Sancho:
Interactive multimodal music transcription. 211-215 - Sundar Harshavardhan, H. G. Ranjani, T. V. Sreenivas:
Student's-t mixture model based multi-instrument recognition in polyphonic music. 216-220 - Marius Miron, Matthew E. P. Davies, Fabien Gouyon:
An open-source drum transcription system for Pure Data and Max MSP. 221-225 - Daichi Sakaue, Takuma Otsuka, Katsutoshi Itoyama, Hiroshi G. Okuno:
Initialization-robust Bayesian multipitch analyzer based on psychoacoustical and musical criteria. 226-230 - Florian Kaiser, Geoffroy Peeters:
Multiple hypotheses at multiple scales for audio novelty computation within music. 231-235 - Oriol Nieto, Tristan Jehan:
Convex non-negative matrix factorization for automatic music structure identification. 236-240 - George Tzanetakis, Graham Percival:
An effective, simple tempo estimation method based on self-similarity and regularity. 241-245 - Chin-Chia Michael Yeh, Li Su, Yi-Hsuan Yang:
Dual-layer bag-of-frames model for music genre classification. 246-250
AASP-P2: SPATIAL AND MULTICHANNEL AUDIO
- Gerald Enzner, Michael W. Weinert, Stefan Abeling, Jan-Mark Batke, Peter Jax:
Advanced system options for binaural rendering of Ambisonic format. 251-255 - Yuancheng Luo, Dmitry N. Zotkin, Hal Daumé III, Ramani Duraiswami:
Kernel regression for Head-Related Transfer Function interpolation and spectral extrema extraction. 256-260 - Umamahesh Srinivas, Nasser M. Nasrabadi, Vishal Monga:
Graph-based multi-sensor fusion for acoustic signal classification. 261-265 - Jianjun He, Ee-Leng Tan, Woon-Seng Gan:
Time-shifted principal component analysis based cue extraction for stereo audio signals. 266-270 - Shoichi Koyama, Ken'ichi Furuya, Yusuke Hiwasaki, Yoichi Haneda, Yôiti Suzuki:
Sound field reproduction using multiple linear arrays based on wave field reconstruction filtering in helicalwave spectrum domain. 271-275 - Shoichi Koyama, Timothy Lee, Ken'ichi Furuya, Yusuke Hiwasaki, Yoichi Haneda:
Improvement using circular harmonics beamforming on reverberation problem of wave field reconstruction filtering. 276-280 - Xiguang Zheng, Christian H. Ritz, Jiangtao Xi:
A psychoacoustic-based analysis-by-synthesis scheme for jointly encoding multiple audio objects into independent mixtures. 281-285 - Jing Wang, Chundong Xu, Xiang Xie, Jingming Kuang:
Multichannel audio signal compression based on tensor decomposition. 286-290 - Yesenia Lacouture-Parodi, Emanuël A. P. Habets:
Application of particle filtering to an interaural time difference based head tracker for crosstalk cancellation. 291-295 - Anastasios Alexandridis, Anthony Griffin, Athanasios Mouchtaris:
Directional coding of audio using a circular microphone array. 296-300 - Tatsuya Komatsu, Takanori Nishino, Gareth W. Peters, Tomoko Matsui, Kazuya Takeda:
Modeling head-related transfer functions via spatial-temporal Gaussian process. 301-305 - Prasanga N. Samarasinghe, Mark A. Poletti, S. M. Akramus Salehin, Thushara D. Abhayapala, Filippo Maria Fazi:
3D soundfield reproduction using higher order loudspeakers. 306-310 - Wenyu Jin, W. Bastiaan Kleijn, David Virette:
Multizone soundfield reproduction using orthogonal basis expansion. 311-315 - Nasim Radmanesh, Ian S. Burnett:
Effectiveness of horizontal personal sound systems for listeners of variable heights. 316-320
AASP-P3: LOUDSPEAKER/MICROPHONE ARRAYS AND ACTIVE NOISE CONTROL
- Hanieh Khalilian, Ivan V. Bajic, Rodney G. Vaughan:
Towards optimal loudspeaker placement for sound field reproduction. 321-325 - Lucio Bianchi, Fabio Antonacci, Antonio Canclini, Augusto Sarti, Stefano Tubaro:
Localization of virtual acoustic sources based on the Hough transform for sound field rendering applications. 326-330 - Yu Bao, Huawei Chen:
Some insights into farfield wideband beamformers in the presence of microphone mismatches. 331-335 - Joseph Szurley, Alexander Bertrand, Marc Moonen:
Improved tracking performance for distributed node-specific signal enhancement inwireless acoustic sensor networks. 336-340 - Yefeng Cai, Ming Wu, Jun Yang:
Design of a time-domain acoustic contrast control for broadband input signals in personal audio systems. 341-345 - Tahereh Noohi, Nicolas Epain, Craig T. Jin:
Direction of arrival estimation for spherical microphone arrays by combination of independent component analysis and sparse recovery. 346-349 - Nicolas Epain, Craig T. Jin:
Super-resolution sound field imaging with sub-space pre-processing. 350-354 - Martin Kreißig, Bin Yang:
Fast and reliable TDOA assignment in multi-source reverberant environments. 355-359 - Stefanie Brown, Deep Sen:
Error analysis of spherical harmonic soundfield representations in terms of truncation and aliasing errors. 360-364 - Kai Wu, Shu Ting Goh, Andy W. H. Khong:
Speaker localization and tracking in the presence of sound interference by exploiting speech harmonicity. 365-369 - Hai Morgenstern, Boaz Rafaely:
Spherical loudspeaker array beamforming in enclosed sound fields by MIMO optimization. 370-374 - Lichuan Liu, Sen M. Kuo:
Wireless communication integrated active noise control system for infant incubators. 375-378 - Iman Tabatabaei Ardekani, Waleed H. Abdulla:
Stability of Residual Acoustic Noise Variance in active control of stochastic noise. 379-382 - Nobuhiro Miyazaki, Yoshinobu Kajikawa:
Adaptive feedback ANC system using virtual microphones. 383-387
AASP-P4: AUDITORY MODELING, HEARING AIDS, ROOM ACOUSTICS, AND SYSTEM MODELING
- Enzo De Sena, Zoran Cvetkovic:
A computational model for the estimation of localisation uncertainty. 388-392 - Shu-Hsien Chu, Mingyi Hong, Zhi-Quan Luo, Kelly Fitz, Martin F. McKinney, Tao Zhang:
Derivative-free optimization of hearing aid parameters. 393-397 - Jens Brehm Nielsen, Jakob Nielsen:
Efficient individualization of hearing aid processed sound. 398-402 - Mengyao Zhu, Jia Zheng, Craig T. Jin, Wanggen Wan:
Structural Similarity Analysis of Modulation for audio quality assessment. 403-407 - Khan Baykaner, Christopher Hummersone, Russell Mason, Soren Bech:
Selection of temporal windows for the computational prediction of masking thresholds. 408-412 - Philipp Thuene, Gerald Enzner:
Improved online identification of acoustic MISO systems based on separated input signal components. 413-417 - Rajan S. Rashobh, Andy W. H. Khong:
A multichannel time-domain subspace approach exploiting multiple time-delays for acoustic channel equalization. 418-422 - Masato Nakayama, Kazuhiro Suzuki, Noboru Nakasako:
Acoustic distance measurement based on phase interference using the cross-spectral method with adjacent microphones. 423-427 - Adam A. Hersbach, Stefan J. Mauger, David B. Grayden, James B. Fallon, Hugh J. McDermott:
Algorithms to improve listening in noise for cochlear implant users. 428-432 - Jan Ole Jungmann, Radoslaw Mazur, Alfred Mertins:
Perturbation of room impulse responses and its application in robust listening room compensation. 433-437 - Ivo Merks, Gerald Enzner, Tao Zhang:
Sound source localization with binaural hearing aids using adaptive blind channel identification. 438-442 - Feifei Xiong, Stefan Goetze, Bernd T. Meyer:
Blind estimation of reverberation time based on spectro-temporal modulation filtering. 443-447 - Albino Nogueiras Rodríguez, Jordi Colom Olivares:
A statistical approach to reverberation in non-diffusive rectangular rooms based on the image source model. 448-452
AASP-P5: AUDIO ANALYSIS AND SYNTHESIS
- Björn W. Schuller, Florian B. Pokorny, Stefan Ladstaetter, Maria Fellner, Franz Graf, Lucas Paletta:
Acoustic Geo-Sensing: Recognising cyclists' route, route direction, and route progress from cell-phone audio. 453-457 - Jürgen T. Geiger, Martin Hofmann, Björn W. Schuller, Gerhard Rigoll:
Gait-based person identification by spectral, cepstral and energy-related audio features. 458-462 - Kazuyoshi Yoshii, Masataka Goto:
Infinite kernel linear prediction for joint estimation of spectral envelope and fundamental frequency. 463-467 - Dan Stowell, Saso Musevic, Jordi Bonada, Mark D. Plumbley:
Improved multiple birdsong tracking with distribution derivative method and Markov renewal process clustering. 468-472 - TaeJin Park, Kyeong Ok Kang:
Position estimation using a microphone and stereo loudspeaker with an audio-embedded hidden time synchronization signal. 473-477 - Vikas Joshi, Nithya Rajamani, Naveen Prathapaneni, L. Venkata Subramaniam:
Traffic density state estimation based on acoustic fusion. 478-482 - Florian Eyben, Felix Weninger, Stefano Squartini, Björn W. Schuller:
Real-life voice activity detection with LSTM Recurrent Neural Networks and an application to Hollywood movies. 483-487 - Saso Musevic, Jordi Bonada:
Derivative analysis of complex polynomial amplitude, complex exponential with exponential damping. 488-492 - Jens Schröder, Stefan Goetze, Volker Grutzmacher, Jörn Anemüller:
Automatic acoustic siren detection in traffic noise by part-based models. 493-497 - Thibaud Necciari, Péter Balázs, Nicki Holighaus, Peter L. Søndergaard:
The ERBlet transform: An auditory-based time-frequency representation with perfect reconstruction. 498-502 - Wai-tian Tan, Ramin Samadani, Bowon Lee, Mary Baker:
Determining co-location using a sequential hypothesis test on patterns of silence. 503-507 - Marcela Charfuelan, Geert-Jan M. Kruijff:
Classification of speech under stress and cognitive load in USAR operations. 508-512 - Talal Ahmed, Momin Uppal, Abubakr Muhammad:
Improving efficiency and reliability of gunshot detection systems. 513-517
AASP-P6: AUDIO CODING
- Sascha Disch, Benjamin Schubert, Bernd Edler:
Cheap beeps - Efficient synthesis of sinusoids and sweeps in the MDCT domain. 518-522 - Yuki Yamamoto, Toru Chinen, Masayuki Nishiguchi:
A new bandwidth extension technology for MPEG Unified Speech and Audio Coding. 523-527 - David Virette, Yue Lang, Lei Miao, Wenhai Wu, Balázs Kövesi, Claude Lamblin, Stéphane Ragot:
G.722 annex D and G.711.1 Annex F - New ITU-T stereo codecs. 528-532 - Huan Zhou, Haiyan Shu, Rongshan Yu, Haibin Huang, Susanto Rahardja:
A novel scalable audio coding scheme. 533-537 - Stefan Bayer, Bernd Edler:
Improved arithmetic coding for Time-Warped MDCT based audio coding. 538-542 - Haojie Liu, Changchun Bao, Xin Liu:
Spectral envelope estimation used for audio bandwidth extension based on RBF neural network. 543-547 - Shi Dong, Ruimin Hu, Xiaochen Wang, Weiping Tu, Xiang Zheng:
An expanded Mid/Side coding for 3D audio signal compression. 548-551 - Wenxin He, Tianshu Qu:
Audio lossless coding/decoding method using basis pursuit algorithm. 552-555 - Wenhai Wu, Lei Miao, Yue Lang, David Virette:
Parametric stereo coding scheme with a new downmix method and whole band inter channel time/phase differences. 556-560 - Stanislaw Gorlow, Emanuël A. P. Habets, Sylvain Marchand:
Multichannel object-based audio coding with controllable quality. 561-565 - Christian Neukam, Frederik Nagel, Gerald Schuller, Michael Schnabel:
A MDCT based harmonic spectral bandwidth extension method. 566-570 - Anderson Fraiha Machado, Antonio Bonafonte, Marcelo Queiroz:
Parametric decomposition of the spectral envelope. 571-574 - Harish Krishnamoorthi, Andreas Spanias:
Sinusoidal component selection based on partial loudness criteria. 575-579