WASPAA 2009 - 2009 IEEE Workshop on Applications of 
             Signal Processing to Audio and Acoustics - October 18 - 21 - 
             Mohonk Mountain House, New Paltz, New York, U.S.A.

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

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

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

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

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

 

 

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

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

PM-03

Fast Bayesian NMF Algorithms Enforcing Harmonicity And Temporal Continuity In Polyphonic Music Transcription

Nancy Bertin, Roland Badeau, and Emmanuel Vincent

PM-04

Computing Predominant Local Periodicity Information In Music Recordings

Peter Grosche and Meinard Mueller

PM-05

Acoustic Topic Model For Audio Information Retrieval

Samuel Kim, Shrikanth Narayanan, and Shiva Sundaram

PM-06

Guided Harmonic Sinusoid Estimation In A Multi-Pitch Environment

Christine Smit and Daniel P.W. Ellis

PM-07

Improving MIDI-Audio Alignment With Acoustic Features

Johanna Devaney, Michael Mandel, and Daniel Ellis

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

PM-09

Multi-Voice Polyphonic Music Transcription Using Eigeninstruments

Graham Grindlay and Daniel Ellis

PM-10

Polyphonic Music Transcription Employing Max-Margin Classification Of Spectrograhic Features

Gang Ren, Mark F. Bocko, Dave Headlam, and Justin Lundberg

PM-11

Towards A Musical Beat Emphasis Function

Matthew Davies, Mark Plumbley, and Douglas Eck

PM-12

Towards Co-Channel Speaker Separation By 2-D Demodulation Of Spectrograms

Tianyu T. Wang and Thomas F. Quatieri

PM-13

Separation By "Humming": User-Guided Sound Extraction From Monophonic Mixtures

Paris Smaragdis and Gautham Mysore

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

PM-15

Improving Separation Of Harmonic Sources With Iterative Estimation Of Spatial Cues

Jinyu Han and Bryan Pardo

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

PM-17

The Ideal Interaural Parameter Mask: A Bound On Binaural Separation Systems

Michael Mandel and Daniel Ellis

PM-18

Source Enumeration of Speech Mixtures Using Pitch Harmonics

Keith Gilbert and Karen Payton

PM-19

Unsupervised Single-Channel Source Separation Using Bayesian NMF

Onur Dikmen and A. Taylan Cemgil

PM-20

An Investigation Of Discrete-State Discriminant Approaches To Single-Sensor Source Separation

Valentin Emiya, Emmanuel Vincent, and Rémi Gribonval

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

PM-22

Coherent Spectral Estimation For A Robust Solution Of The Permutation Problem

Francesco Nesta, Ted S. Wada, and Biing-Hwang (Fred) Juang

 

 

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

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

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

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

LM2-05 5:20 pm - 5:40 pm

Finding Similar Acoustic Events Using Matching Pursuit and Locality-Sensitive Hashing

Courtenay Cotton and Daniel Ellis

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

 

 

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

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

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

LT1-04 9:45 am - 10:05 am

Statistical Models For Speech Dereverberation

Takuya Yoshioka, Hirokazu Kameoka, Tomohiro Nakatani, and Hiroshi Okuno

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

 

 

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

PT-02

On Robustness Of Multi-Channel Minimum Mean-Squared Error Estimators Under Super-Gaussian Priors

Richard Christian Hendriks, Richard Heusdens, and Jesper Jensen

PT-03

Blind Alignment Of Asynchronously Recorded Signals For Distributed Microphone Array

Nobutaka Ono, Hitoshi Kohno, Nobutaka Ito, and Shigeki Sagayama

PT-04

Artifacts In The Sound Field Of A Moving Sound Source Reconstructed From A Microphone Array Recording

Jens Ahrens and Sascha Spors

PT-05

Dolph-Chebyshev Radial Filter For The Near-Field Spherical Microphone Array

Etan Fisher and Boaz Rafaely

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

PT-07

Single-Microphone Wind Noise Reduction By Adaptive Postfiltering

Elias Nemer and Wilf Leblanc

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

PT-10

Improved A Priori Snr Estimation With Application In Log-Mmse Speech Estimation

Nils Höglund and Sven Nordholm

PT-11

Robust Audio Precompensation With Probabilistic Modeling Of Transfer Function Variability

Lars-Johan Brannmark

PT-12

Variable Control Of The Pre-Response Error In Mixed Phase Audio Precompensation

Lars-Johan Brannmark and Anders Ahlén

PT-13

Dynamic Impulse Response Model For Nonlinear Acoustic System And Its Application To Acoustic Echo Canceller

Shoichiro Saito, Akira Nakagawa, and Yoichi Haneda

PT-14

Acoustic Echo Cancellation Based On Independent Component Analysis And Integrated Residual Echo Enhancement

Ted S. Wada and Biing-Hwang (Fred) Juang

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

PT-16

Stochastic Particle Filtering: A Fast SRP-PHAT Single Source Localization Algorithm

Hoang Do and Harvey Silverman

PT-17

Multiple Sound Source Tracking Method Based On Subspace Tracking

Noboru Ohwada and Kenji Suyama

PT-18

Coherent Signals Direction-Of-Arrival Estimation Using A Spherical Microphone Array: Frequency Smoothing Approach

Dima Khaykin and Boaz Rafaely

PT-19

Multichannel Voice Activity Detection With Spherically Invariant Sparse Distributions

Bowon Lee and Ton Kalker

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

PT-21

A Wiener-Based Implementation Of Equalization-Cancellation Pre-Processing For Binaural Speech Intelligibility Prediction

Nicolas Ellaham, Christian Giguère, and Wail Gueaieb

 

 

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

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

LT2-03 4:40 pm - 5:00 pm

Acoustic Reflection Path Tracing Using A Highly Directional Loudspeaker

Sakari Tervo, Jukka Pätynen, and Tapio Lokki

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

LT2-05 5:20 pm - 5:40 pm

Parsimonious Sound Field Synthesis Using Compressive Sampling

Georgios N. Lilis, Daniele Angelosante, and Georgios B. Giannakis

LT2-06 5:40 pm - 6:00 pm

Regularized Hrtf Fitting Using Spherical Harmonics

Dmitry N. Zotkin, Ramani Duraiswami, and Nail A. Gumerov

 

 

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

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

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

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

LW-05 10:05 am - 10:25 am

Coding Of Spatio-Temporal Audio Spectra Using Tree-Structured Directional Filterbanks

Francisco Pinto and Martin Vetterli

 

 

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

PW-02

Diffuseness Estimation Using Temporal Variation Of Intensity Vectors

Jukka Ahonen and Ville Pulkki

PW-03

Estimating Pressure At Eardrum With Pressure-Velocity Measurement From Ear Canal Entrance

Marko Hiipakka, Matti Karjalainen, and Ville Pulkki

PW-04

HRTF Interpolation In The Wavelet Transform Domain

Julio Cesar B. Torres and Mariane R. Petraglia

PW-05

Sound Texture Synthesis Via Filter Statistics

Josh McDermott, Andrew Oxenham, and Eero Simoncelli

PW-06

An Overall Optimization Method For Arbitrary Sample Rate Converters Based On Integer Rate SRC And Lagrange Interpolation

Andreas Franck and Karlheinz Brandenburg

PW-07

Spectral HRTF Enhancement For Improved Vertical-Polar Auditory Localization

Douglas Brungart and Griffin Romigh

PW-08

Multizone 2D Soundfield Reproduction Via Spatial Band Stop Filters

Yan Jennifer WU and Thushara D Abhayapala

PW-09

Soundfield Rendering With Loudspeaker Arrays Through Multiple Beam Shaping

Fabio Antonacci, Alberto Calatroni, Antonio Canclini, Andrea Galbiati, Augusto Sarti, and Stefano Tubaro

PW-10

Wave Field Analysis Using Multiple Radii Measurements

Achim Kuntz and Rudolf Rabenstein

PW-11

Controlling A Spatialized Environmental Sound Synthesizer

Charles Verron, Mitsuko Aramaki, Richard Kronland-Martinet, and Pallone Grégory

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

PW-14

Parametric AM/FM Decomposition For Speech And Audio Coding

Tom Bäckström and Sascha Disch

PW-15

Binaural Reproduction For Directional Audio Coding

Mikko-Ville Laitinen and Ville Pulkki

PW-16

Applications Of Signal Analysis Using Autoregressive Models For Amplitude Modulation

Sriram Ganapathy, Samuel Thomas, Petr Motlicek, and Hynek Hermansky

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

PW-18

Sinewave Parameter Estimation Using The Fast Fan-Chirp Transform

Robert Dunn, Thomas Quatieri, and Nicolas Malyska

PW-19

Realization Of Arbitrary Filters In The STFT Domain

Michael Goodwin