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Program: Agenda
Lecture Session LM1:
Music Perception, Analysis, and Synthesis
8:45 am - 10:25 am, Monday, October 19, 2009
Chair: Dan Ellis, Columbia University, USA |
LM1-01 |
8:45 am - 9:05 am |
Automatic Gain and Fader Control For Live Mixing
Enrique Perez Gonzalez and Joshua Reiss
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LM1-02 |
9:05 am - 9:25am |
Acoustic Coupling in Multi-Dimensional Finite
Difference Schemes for Physically Modeled
Voice Synthesis
Matt Speed, Damian Murphy, and David Howard
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LM1-03 |
9:25 am - 9:45 am |
Chord Recognition Using Measures Of Fit,
Chord Templates And Filtering Methods.
Laurent Oudre, Yves Grenier, and
Cédric Févotte
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LM1-04 |
9:45 am - 10:05 am |
Unifying Semantic and Content-based Approaches
for Retrieval of Environmental Sounds
Gordon Wichern, Harvey Thornburg, and Andreas Spanias
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LM1-05 |
10:05 am - 10:25 am |
A Novel Framework For Recognizing Phonemes Of
Singing Voice In Polyphonic Music
Hiromasa Fujihara, Masataka Goto, and Hiroshi G. Okuno
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Poster Session PM:
Music Processing and Source Separation
10:40 am - 12:20 pm, Monday, October 19, 2009
Chair: Eric Diethorn,
MH Acoustics, USA |
PM-01 |
Domain Decomposition Method For The
Digital Waveguide Mesh
Moonseok Kim and Gary Scavone
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PM-02 |
Synthesis Of Guitar By Digital Waveguides:
Modeling The Plectrum In The Physical
Interaction Of The Player With The Instrument
François Germain and Gianpaolo Evangelista
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PM-03 |
Fast Bayesian NMF Algorithms Enforcing Harmonicity
And Temporal Continuity In Polyphonic Music
Transcription
Nancy Bertin, Roland Badeau, and Emmanuel Vincent
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PM-04 |
Computing Predominant Local Periodicity
Information In Music Recordings
Peter Grosche and Meinard Mueller
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PM-05 |
Acoustic Topic Model For Audio
Information Retrieval
Samuel Kim, Shrikanth Narayanan, and Shiva Sundaram
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PM-06 |
Guided Harmonic Sinusoid Estimation In
A Multi-Pitch Environment
Christine Smit and Daniel P.W. Ellis
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PM-07 |
Improving MIDI-Audio Alignment With
Acoustic Features
Johanna Devaney, Michael Mandel, and Daniel Ellis
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PM-08 |
Note Detection With Dynamic Bayesian
Networks As A Postanalysis Step For
NMF-Based Multiple Pitch Estimation Techniques
Stanislaw Andrzej Raczynski,
Nobutaka Ono, and Shigeki Sagayama
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PM-09 |
Multi-Voice Polyphonic Music Transcription
Using Eigeninstruments
Graham Grindlay and Daniel Ellis
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PM-10 |
Polyphonic Music Transcription Employing
Max-Margin Classification Of
Spectrograhic Features
Gang Ren, Mark F. Bocko, Dave Headlam, and
Justin Lundberg
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PM-11 |
Towards A Musical Beat Emphasis Function
Matthew Davies, Mark Plumbley, and Douglas Eck
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PM-12 |
Towards Co-Channel Speaker Separation
By 2-D Demodulation Of Spectrograms
Tianyu T. Wang and Thomas F. Quatieri
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PM-13 |
Separation By "Humming": User-Guided Sound
Extraction From Monophonic Mixtures
Paris Smaragdis and Gautham Mysore
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PM-14 |
Semi-Blind Disjoint Non-Negative Matrix
Factorization For Extracting Target Source
From Single Channel Noisy Mixture
So-Young Jeong, Kyuhong Kim, Jae-Hoon Jeong,
and Kwang-Cheol Oh
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PM-15 |
Improving Separation Of Harmonic Sources
With Iterative Estimation Of Spatial Cues
Jinyu Han and Bryan Pardo
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PM-16 |
A Nonlocally Weighted Soft-Constrained
Natural Gradient Algorithm For
Blind Separation Of Reverberant Speech
Jack Xin, Meng Yu, Yingyong Qi, Hsin-I Yang,
and Fan-Gang Zeng
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PM-17 |
The Ideal Interaural Parameter Mask:
A Bound On Binaural Separation Systems
Michael Mandel and Daniel Ellis
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PM-18 |
Source Enumeration of Speech Mixtures
Using Pitch Harmonics
Keith Gilbert and Karen Payton
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PM-19 |
Unsupervised Single-Channel
Source Separation Using Bayesian NMF
Onur Dikmen and A. Taylan Cemgil
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PM-20 |
An Investigation Of Discrete-State Discriminant
Approaches To Single-Sensor Source Separation
Valentin Emiya, Emmanuel Vincent,
and Rémi Gribonval
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PM-21 |
On The Non-Uniqueness Problem And The Semi-Blind
Source Separation
Francesco Nesta, Ted S. Wada, Shigeki Miyabe,
and Biing-Hwang (Fred) Juang
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PM-22 |
Coherent Spectral Estimation For A Robust Solution
Of The Permutation Problem
Francesco Nesta, Ted S. Wada, and
Biing-Hwang (Fred) Juang
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Lecture Session LM2:
Microphone Array and Source Separation
4:00 pm - 6:00 pm, Monday, October 19, 2009
Chair: Gary W. Elko, MH Acoustics, USA |
LM2-01 |
4:00 pm - 4:20 pm |
On Optimal Beamforming For Noise Reduction
And Interference Rejection
Mehrez Souden, Jacob Benesty, and Sofiène Affes
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LM2-02 |
4:20 pm - 4:40 pm |
Robust Spherical Microphone Array Beamforming
With Multi-Beam-Multi-Null Steering, And
Sidelobe Control
Haohai Sun, Shefeng Yan, and Peter Svensson
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LM2-03 |
4:40 pm - 5:00 pm |
Panoramic Recording And Reproduction Of
Multichannel Audio Using A Circular
Microphone Array
Huseyin Hacihabiboglu and Zoran Cvetkovic
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LM2-04 |
5:00 pm - 5:20 pm |
Factorial Scaled Hidden Markov Model For
Polyphonic Audio Representation And
Source Separation
Alexey Ozerov, Cedric Fevotte, and Maurice Charbit
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LM2-05 |
5:20 pm - 5:40 pm |
Finding Similar Acoustic Events Using
Matching Pursuit and Locality-Sensitive Hashing
Courtenay Cotton and Daniel Ellis
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LM2-06 |
5:40 pm - 6:00 pm |
Spatial Covariance Models For Under-Determined
Reverberant Audio Source Separation
Ngoc Duong, Emmanuel Vincent, and Rémi Gribonval
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Lecture Session LT1:
Signal Enhancement, Dereverberation, and Echo Control
8:45 am - 10:25 am, Tuesday, October 20, 2009
Chair: Rudolf Rabenstein, University of Erlangen-Nuremberg, Germany |
LT1-01 |
8:45 am - 9:05 am |
TS-BASE/WF: Two-Stage BinAural Speech Enhancement
with Wiener Filter Based On
Equalization-Cancellation Model
Junfeng Li, Shuichi Sakamoto, Satoshi Hongo,
Masato Akagi, and Yo-iti Suzuki
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LT1-02 |
9:05 am - 9:25am |
A Spatio-Temporal Power Method for
Time-Domain Multi-Channel Speech Enhancement
Malay Gupta, Sylvain Angrignon, Chris Forrester,
and Sean Simmons
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LT1-03 |
9:25 am - 9:45 am |
On the Application Of The LCMV Beamformer
To Speech Enhancement
Emanuel Habets, Jacob Benesty, Sharon Gannot,
Patrick Naylor, and Israel Cohen
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LT1-04 |
9:45 am - 10:05 am |
Statistical Models For Speech Dereverberation
Takuya Yoshioka, Hirokazu Kameoka, Tomohiro Nakatani,
and Hiroshi Okuno
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LT1-05 |
10:05 am - 10:25 am |
Adaptive FIR Filters With Automatic Length
Optimization By Monitoring A Normalized
Combination Scheme
Marcus Zeller, Luis Antonio Azpicueta-Ruiz,
and Walter Kellermann
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Poster Session PT:
Speech Enhancement, Multichannel Audio Processing,
and Hearing Aids
10:40 am - 12:20 pm, Tuesday, October 20, 2009
Chair: Jingdong Chen,
WeVoice, USA |
PT-01 |
Maximum Directivity Beamformer For
Spherical-Aperture Microphones
Morag Agmon, Boaz Rafaely, and Joseph Tabrikian
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PT-02 |
On Robustness Of Multi-Channel Minimum
Mean-Squared Error Estimators Under
Super-Gaussian Priors
Richard Christian Hendriks, Richard Heusdens,
and Jesper Jensen
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PT-03 |
Blind Alignment Of Asynchronously
Recorded Signals For Distributed
Microphone Array
Nobutaka Ono, Hitoshi Kohno, Nobutaka Ito,
and Shigeki Sagayama
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PT-04 |
Artifacts In The Sound Field Of A Moving
Sound Source Reconstructed From
A Microphone Array Recording
Jens Ahrens and Sascha Spors
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PT-05 |
Dolph-Chebyshev Radial Filter For The
Near-Field Spherical Microphone Array
Etan Fisher and Boaz Rafaely
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PT-06 |
Enhancement Of Speech Intelligibility
Using Transients Extracted By Wavelet Packets
Daniel M. Rasetshwane, J. R. Boston, Ching-Chung Li,
John Durrant, and Greg Genna
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PT-07 |
Single-Microphone Wind Noise Reduction
By Adaptive Postfiltering
Elias Nemer and Wilf Leblanc
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PT-08 |
An Auditory-Based Transform For
Audio Signal Processing
Qi Li |
PT-09 |
Gain Adaptation Based On
Signal-To-Noise Ratio for Noise Suppression
Devangi Parikh, Sourabh Ravindran,
and David Anderson
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PT-10 |
Improved A Priori Snr Estimation With
Application In Log-Mmse Speech Estimation
Nils Höglund and Sven Nordholm
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PT-11 |
Robust Audio Precompensation With
Probabilistic Modeling Of
Transfer Function Variability
Lars-Johan Brannmark
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PT-12 |
Variable Control Of The Pre-Response Error
In Mixed Phase Audio Precompensation
Lars-Johan Brannmark and Anders Ahlén
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PT-13 |
Dynamic Impulse Response Model For Nonlinear
Acoustic System And Its Application To
Acoustic Echo Canceller
Shoichiro Saito, Akira Nakagawa, and Yoichi Haneda
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PT-14 |
Acoustic Echo Cancellation Based On
Independent Component Analysis And
Integrated Residual Echo Enhancement
Ted S. Wada and Biing-Hwang (Fred) Juang
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PT-15 |
A Phase-Based Dual Microphone Method
To Count And Locate Audio Sources
In Reverberant Rooms
Zaher El chami, Antoine Pham, Christine Serviere,
and Alexandre Guerin
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PT-16 |
Stochastic Particle Filtering: A Fast SRP-PHAT
Single Source Localization Algorithm
Hoang Do and Harvey Silverman
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PT-17 |
Multiple Sound Source Tracking Method Based
On Subspace Tracking
Noboru Ohwada and Kenji Suyama
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PT-18 |
Coherent Signals Direction-Of-Arrival Estimation
Using A Spherical Microphone Array:
Frequency Smoothing Approach
Dima Khaykin and Boaz Rafaely
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PT-19 |
Multichannel Voice Activity Detection With
Spherically Invariant Sparse Distributions
Bowon Lee and Ton Kalker
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PT-20 |
A Zone Of Quiet Based Approach To
Integrated Active Noise Control And
Noise Reduction In Hearing Aids
Romain Serizel, Marc Moonen, Jan Wouters,
and Soren Holdt Jensen
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PT-21 |
A Wiener-Based Implementation Of
Equalization-Cancellation Pre-Processing
For Binaural Speech Intelligibility Prediction
Nicolas Ellaham, Christian Giguère,
and Wail Gueaieb
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Lecture Session LT2:
Spatial Sound Perception, Analysis, and Reproduction
4:00 pm - 6:00 pm, Tuesday, October 20, 2009
Chair: Mike Goodwin, Audience, USA |
LT2-01 |
4:00 pm - 4:20 pm |
Generalized State Coherence Transform For
Multidimensional Localization Of
Multiple Sources
Francesco Nesta and Maurizio Omologo
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LT2-02 |
4:20 pm - 4:40 pm |
A Probabilistic Speaker Clustering
For DOA-based Diarization
Katsuhiko Ishiguro, Takeshi Yamada, Shoko Araki,
and Tomohiro Nakatani
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LT2-03 |
4:40 pm - 5:00 pm |
Acoustic Reflection Path Tracing Using
A Highly Directional Loudspeaker
Sakari Tervo, Jukka Pätynen, and Tapio Lokki
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LT2-04 |
5:00 pm - 5:20 pm |
Feature Selection For Room Volume Identification
From Room Impulse Response
Noam R. Shabtai, Yaniv Zigel, and Boaz Rafaely
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LT2-05 |
5:20 pm - 5:40 pm |
Parsimonious Sound Field Synthesis Using
Compressive Sampling
Georgios N. Lilis, Daniele Angelosante,
and Georgios B. Giannakis
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LT2-06 |
5:40 pm - 6:00 pm |
Regularized Hrtf Fitting Using Spherical Harmonics
Dmitry N. Zotkin, Ramani Duraiswami,
and Nail A. Gumerov
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Lecture Session LW:
Speech and Audio Coding
8:45 am - 10:25 am, Wednesday, October 21, 2009
Chair: Peter Kroon, Infineon Technologies, USA |
LW-01 |
8:45 am - 9:05 am |
Temporal Quantization Of Spatial Information
Using Directional Clustering
For Multichannel Audio Coding
Shigeki Miyabe, Keisuke Masatoki, Hiroshi Saruwatari,
Kiyohiro Shikano, and Toshiyuki Nomura
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LW-02 |
9:05 am - 9:25am |
ITU-T G.719: A New Low-Complexity Full-Band
(20 Khz) Audio Coding Standard For
High-Quality Conversational Applications
Minjie Xie, Anisse Taleb, Manuel Briand, and Peter Chu
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LW-03 |
9:25 am - 9:45 am |
IIR QMF-Bank Design for Speech and
Audio Subband Coding
Heinrich Loellmann, Matthias Hildenbrand,
Bernd Geiser, and Peter Vary
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LW-04 |
9:45 am - 10:05 am |
Nested Microphone Array Processing For
Parameter Estimation In
Directional Audio Coding
Giovanni Del Galdo, Oliver Thiergart,
and Fabian Kuech
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LW-05 |
10:05 am - 10:25 am |
Coding Of Spatio-Temporal Audio Spectra
Using Tree-Structured Directional Filterbanks
Francisco Pinto and Martin Vetterli
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Poster Session PW:
Spatial Sound Perception and Reproduction, and
Speech and Audio Coding
10:40 am - 12:20 pm, Wednesday, October 21, 2009
Chair:
Shoji Makino,
Univ. of Tsukuba, Japan |
PW-01 |
Perfect Sequence LMS For Rapid Acquisition
Of Continuous-Azimuth Head Related
Impulse Responses
Christiane Antweiler and Gerald Enzner
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PW-02 |
Diffuseness Estimation Using Temporal Variation
Of Intensity Vectors
Jukka Ahonen and Ville Pulkki
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PW-03 |
Estimating Pressure At Eardrum With
Pressure-Velocity Measurement
From Ear Canal Entrance
Marko Hiipakka, Matti Karjalainen, and Ville Pulkki
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PW-04 |
HRTF Interpolation In The Wavelet Transform Domain
Julio Cesar B. Torres and Mariane R. Petraglia
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PW-05 |
Sound Texture Synthesis Via Filter Statistics
Josh McDermott, Andrew Oxenham, and Eero Simoncelli
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PW-06 |
An Overall Optimization Method For Arbitrary
Sample Rate Converters Based On
Integer Rate SRC And Lagrange Interpolation
Andreas Franck and Karlheinz Brandenburg
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PW-07 |
Spectral HRTF Enhancement For Improved
Vertical-Polar Auditory Localization
Douglas Brungart and Griffin Romigh
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PW-08 |
Multizone 2D Soundfield Reproduction Via
Spatial Band Stop Filters
Yan Jennifer WU and Thushara D Abhayapala
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PW-09 |
Soundfield Rendering With Loudspeaker
Arrays Through Multiple Beam Shaping
Fabio Antonacci, Alberto Calatroni,
Antonio Canclini, Andrea Galbiati,
Augusto Sarti, and Stefano Tubaro
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PW-10 |
Wave Field Analysis Using Multiple
Radii Measurements
Achim Kuntz and Rudolf Rabenstein
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PW-11 |
Controlling A Spatialized Environmental
Sound Synthesizer
Charles Verron, Mitsuko Aramaki,
Richard Kronland-Martinet,
and Pallone Grégory
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PW-12 |
3D-Continuous-Azimuth Acquisition Of
Head Related Impulse Responses
Using Multi-Channel Adaptive Filtering
Gerald Enzner |
PW-13 |
A Perceptually Enhanced Scalable-To-Lossless
Audio Coding Scheme And A Trellis-Based
Approach For Its Optimization
Emmanuel Ravelli, Vinay Melkote, and Kenneth Rose
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PW-14 |
Parametric AM/FM Decomposition For
Speech And Audio Coding
Tom Bäckström and Sascha Disch
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PW-15 |
Binaural Reproduction For Directional Audio Coding
Mikko-Ville Laitinen and Ville Pulkki
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PW-16 |
Applications Of Signal Analysis Using
Autoregressive Models For
Amplitude Modulation
Sriram Ganapathy, Samuel Thomas, Petr Motlicek,
and Hynek Hermansky
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PW-17 |
Theoretical And Practical Comparisons Of
The Reassignment Method And
The Derivative Method For The Estimation
Of The Frequency Slope
Brian Hamilton, Philippe Depalle, and
Sylvain Marchand
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PW-18 |
Sinewave Parameter Estimation Using The
Fast Fan-Chirp Transform
Robert Dunn, Thomas Quatieri,
and Nicolas Malyska
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PW-19 |
Realization Of Arbitrary Filters
In The STFT Domain
Michael Goodwin
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