Detailed Program

MONDAY, AUGUST 30, 2010
19:00-22:00
Welcome Cocktail

TUESDAY, AUGUST 31, 2010
08:30-09:00
Registration

09:00-09:30
Opening Session:
Diamond Hall

Welcome Address and Greetings
09:30-10:30
Plenary Lecture 1
Diamond Hall


*This session is in memory of Frieda Gannot
 
09:30
Combatting Acoustic Echoes and Noise in Future Natural Immersive Human/Machine Interfaces
Walter Kellermann 1
1 Multimedia Communications and Signal Processing, University Erlangen-Nuremberg, Germany

10:30-11:00
Coffee Break

11:00-12:30
Poster Session A: Blind Source Separation, Speech Modeling, Filterbanks and Denoising
 Diamond Hall

P-1
Improving the Robustness of the Correlation Approach for Solving the Permutation Problem in theConvolutive Blind Source Separation
Radoslaw Mazur 1 Alfred Mertins 1
1 Institute for Signal Processing, University of Luebeck, Luebeck, Germany

P-2
Single-Channel Source Separation of Speech and Music using Short-Time Spectral Kurtosis
Yevgeni Litvin 1 Israel Cohen 1 Jacob Benesty 2
1 Electrical Engineering, Technion - Israel Institute of Technology, Haifa, Israel
2 INRS-EMT, Universite du Quebec, Montreal, QC H5A, 1K6, Canada

P-3
De-Noising of Acoustic Breathing Signals
Rene Derkx 1 Harm Belt 2
1 Digital Signal Processing group, Philips Research, Eindhoven, Netherlands
2 Video and Image Processing group, Philips Research, Eindhoven, Netherlands

P-4
Separation of Speech and Music Sources from a Single-Channel Mixture Using Discrete Energy Separation Algorithm
Yevgeni Litvin 1 Israel Cohen 1 Dan Chazan 2
1 Electrical Engineering, Technion - Israel Institute of Technology, Haifa, Israel
2 IBM Research Laboratory, Haifa, Israel

P-5
Complementary N-Band IIR Filterbank Based on 2-Band Complementary Filters
Alexis Favrot 1 Christof Faller 1
1 Illusonic LLC, Lausanne, Switzerland

P-6
Empirical Distributions of DFT-Domain Speech Coefficients Based on Estimated Speech Variances
Timo Gerkmann 1 Rainer Martin 1
1 Institute of Communication Acoustics, Ruhr-Universität Bochum, Bochum, Germany

P-7
Classification of Unvoiced Fricative Phonemes Using Geometric Methods
Michal Genussov 1 Yizhar Lavner 2 Israel Cohen 1
1 Department of Electrical Engineering, Technion- Israel Institute of Technology, Haifa, Israel
2 Department of Computer Science, Tel-Hai Academic College, Upper Galilee, Israel

P-8
On the Statistical Properties of Reverberant Speech Feature Vector Sequences
Armin Sehr 1 Walter Kellermann 1
1 Multimedia Communications and Signal Processing, University of Erlangen-Nuremberg, Erlangen, Germany

P-9
Towards a Better Understanding of the Effect of Reverberation on Speech Recognition Performance
Armin Sehr 1 Emanuel Habets 2 Roland Maas 1 Walter Kellermann 1
1 Multimedia Communications and Signal Processing, University of Erlangen-Nuremberg, Erlangen, Germany
2 Department of Electrical and Electronic Engineering, Imperial College London, London, United Kingdom

P-10
Single Microphone Blind Audio Source Separation Using Short+Long Term AR Modeling
Siouar Bensaid 1 Dirk Slock 2
1 Mobile Communication, Phd Student, Sophia Antipolis, Alpes Maritimes, France
2 Mobile Communication, Sophia Antipolis, Alpes Maritimes, France

P-11
Statistical Modeling of the Speech Signal
Ivan Tashev 1 Alex Acero 1
1 Speech Technology Group, Microsoft Research, Redmond, Washington 98034, United States

P-12
Blind Audio Source Separation using Short+Long Term AR Source Models and Iterative Itakura-Saito Distance Minimization
Antony Schutz 1 Dirk Slock 1
1 Mobile Department, EURECOM, SOPHIA ANTIPOLIS, France

12:30-13:30
Lunch Break

13:30-15:00
Poster Session B: Beamforming and Post-filtering Methods
 Diamond Hall

P-1
The Optimal Widely Linear MVDR Beamformer in Room Acoustics
Jacob Benesty 1 Jingdong Chen 2 Yiteng (Arden) Huang 2
1 INRS-EMT, University of Quebec, Montreal, Quebec, Canada
2 WeVoice, Inc., Bridgewater, NJ, United States

P-2
Overdetermined Blind Source Extraction exploiting a Generalized Sidelobe Canceller structure
Brian Bloemendal 1 Jakob van de Laar 2 Piet Sommen 1
1 Electrical Engineering, Eindhoven University of Technology, Eindhoven, Netherlands
2 Digital Signal Processing Group, Philips Research Laboratories, Eindhoven, Netherlands

P-3
Design of Robust Steerable Broadband Beamformers with Spiral Arrays and the Farrow Filter Structure
Chiong Ching Lai 1 Sven Nordholm 1 Yee Hong Leung 1
1 Department of Electrical and Computing Engineering, Curtin University, Bentley, Western Australia, Australia

P-4
Adaptive Azimuthal Null-Steeringfor a First-order Microphone Response
Rene Derkx 1
1 Digital Signal Processing group, Philips Research, Eindhoven, Netherlands

P-5
A Multi-Microphone Speech Enhancement Algorithm Tested Using Acoustic Vector Sensors
Ping-Kun Tony Wu 1 Craig Jin 1 Alan Kan 1 Andre Van Schaik 1
1 School of Electrical and Information Engineering, University of Sydney, Sydney, New South Wales, Australia

P-6
A Generalized View on Microphone Array Postfilters
Tobias Wolff 1 Markus Buck 1
1 Nuance Communications Aachen GmbH, Germany

P-7
Influence of Blocking Matrix Design on Microphone Array Postfilters
Tobias Wolff 1 Markus Buck 1
1 Nuance Communications Aachen GmbH, Germany

P-8
Computationally Efficient and Robust Frequency-Domain GSC
Ludovick LEPAULOUX 1 Pascal SCALART 2 Claude MARRO 1
1 TECH/OPERA, Orange Labs, Lannion, France
2 IRISA/CAIRN, ENSSAT, Lannion, France

P-9
A Distortionless Subband Beamformer for Noise Reduction in Reverberant Environments
Emanuel Habets 1
1 Department of Electrical and Electronic Engineering, Imperial College London, London, United Kingdom

P-10
Performance Characterization of Linear Arrays with Respect to Robust MVDR Beamforming
Sebastian Gergen 1 Nilesh Madhu 2 Rainer Martin 1
1 Institute of Communication Acoustics, Ruhr-Universität Bochum, Bochum, Germany
2 ExpORL, Dept. Neurosciences, Katholieke Universiteit Leuven, Leuven, Belgium

P-11
Dereverberation and Noise Reduction Using a Spherical Microphone Array: an Experimental Investigation
Yotam Peled 1 Boaz Rafaely 1
1 Electrical and Computer Engineering, Ben-Gurion University of the Negev, Beer-Sheva, Israel

15:00-15:30
Coffee Break

15:30-17:00
Poster Session C: Speech Enhancement and Hearing Aids Applications
 Diamond Hall

P-1
A Distortionless Noise-Reduction Filter Based on the Widely Linear Estimation Theory
Jingdong Chen Jacob Benesty Yiteng (Arden) Huang
1 WeVoice, Inc., Bridgewater, New Jersey, United States
2 INRS-EMT, University of Quebec, Montreal, Quebec, Canada

P-2
Speech Enhancement Using a Multidimensional Mixture-Maximum Model
Yochay Yeminy 1 Sharon Gannot 1 Yosi Keller 1
1 School of Engineering, Bar-Ilan University, Ramat-Gan, Israel

P-3
Dual-Microphone Speech Enhancement with Robustness to Variationsin Microphone Gain Characteristics
Shinya Matsui 1 Yoji Ishikawa 1
1 Information Technology Laboratory, Asahi Kasei Corporation, Kanagawa, Japan

P-4
Binaural Cue Preservation in Binaural Hearing Aids with Reduced-Bandwidth Multichannel Wiener Filter Based Noise Reduction
Bram Cornelis 1 Marc Moonen 1 Jan Wouters 2
1 Electrical engineering, ESAT/SCD - Katholieke Universiteit Leuven, Heverlee - Leuven, Belgium
2 Neurosciences, Exp ORL - Katholieke Universiteit leuven, Leuven, Belgium

P-5
The Benefit of Speech Enhancement to the Hearing Impaired
Nir Fink 1 Chava Muchnik 2 Miriam Furst 3
1 Department of Bio-Medical Engineering, Faculty of Engineering, Tel-Aviv University, Tel-Aviv, Israel
2 Department of Communications Disorders, Faculty of Medicine, Tel-Aviv University, Tel-Aviv, Israel
3 School of Electrical Engineering, Faculty of Engineering, Tel-Aviv University, Tel-Aviv, Israel

P-6
Multi-Channel Algorithms for Wind Noise Reduction and Signal Compensation in Binaural Hearing Aids
Sven Franz 1 Jörg Bitzer 1
1 Institute for Hearing Technology and Audiology, Jade University of Applied Sciences, Oldenburg, Germany

P-7
Noise Codebook Adaptation for Codebook-Based Noise Reduction
Tobias Rosenkranz 1
1 Siemens Audiologische Technik GmbH, Germany

P-8
On Predicting the Difference in Intelligibility Before and After Single-Channel Noise Reduction
Cees Taal 1 Richard Hendriks 1 Richard Heusdens 1 Jesper Jensen 2
1 Delft University of Technology, SIPlab, Netherlands
2 Oticon A/S, Denmark

P-9
Binaural Hearing Aid Using Sound-Localization-Preserved MMSE STSA Estimator with ICA-Based Noise Estimation
Hiroshi Saruwatari 1 Masanobu Go 1 Ryoi Okamoto 1 Kiyohiro Shikano 1 Hiroshi Hosoi 2
1 Graduate School of Information Science, Nara Institute of Science and Technology, Ikoma, Nara, Japan
2 Nara Medical University, Kashihara, Nara, Japan

P-10
Two-Sided Model Based Packet Loss Concealments
Nadav Linenberg 1 Yishai Gil 1 Ilan D Shallom 1
1 Department of Electrical and Computer Engineering, Ben-Gurion University of the Negev, Beer Sheva, Israel

P-11
Decoder State-Copying for Bluetooth CVSD Packet Loss Concealment
Xuejing Sun 1 Kuan-Chieh Yen 1
1 Cambridge Silicon Radio, Auburn Hills, MI, United States

P-12
Detection of Hum in Audio Signals
Matthias Brandt 1 Joerg Bitzer 1
1 Institute for Hearing Technology and Audiology, Jade University of Applied Sciences, Oldenburg, Germany

17:15-17:30
Group Photo

18:30-20:30
Optional Tours in Tel Aviv

WEDNESDAY, SEPTEMBER 01, 2010
08:00-08:15
Registration

08:15-09:15
Plenary Lecture 2
 Diamond Hall

08:15
Front-end Audio Processing: Reflections on Issues, Requirements, and Solutions
Tomas Gaensler 1
1 mh acoustics, Summit, NJ, United States

09:15-10:45
Poster Session D: Speaker Localization
 Diamond Hall

P-1
A Study of ICA Based DOA Estimation with Respect to Permutation Ambiguity, Scaling Ambiguity and Sensor Gain Mismatch
Benedikt Loesch 1 Bin Yang 1
1 System Theory and Signal Processing, University of Stuttgart, Stuttgart, Germany

P-2
Robust Neuro-Fuzzy Speaker Localization Using a Circular Microphone Array
Axel Plinge 1 Marius H. Hennecke 1 Gernot A. Fink 1
1 Intelligent Systems Group, Robotics Research Institute, TU Dortmund, Dortmund, Germany

P-3
Impact of Source Signal Coloration on Intensity Vector Based DOA Estimation
Dovid Levin 1 Emanuel Habets 2 Sharon Gannot 1
1 School of Engineering, Bar-Ilan University, Ramat-Gan, Israel
2 Department of Electrical and Electronic Engineering, Imperial College London, London, United Kingdom

P-4
A Vectorized Method for Computationally Efficient Srp-Phat Sound Source Localization
Bowon Lee 1 Ton Kalker 1
1 Multimedia Communications and Networking Lab, Hewlett-Packard Laboratories, Palo Alto, California, United States

P-5
Interactive Controller for Audio Object Localization Based on Spatial Representative Vector Operation
Noriyoshi Kamado 1 Hiroyuki Nawata 1 Hiroshi Saruwatari 1 Kiyohiro Shikano 1 Toshiyuki Nomura 2
1 Graduate school of Information Science, Nara Institute Science and Technology, Ikoma, Nara, Japan
2 NEC Corporation, Kanagawa, Japan

P-6
Advanced Disambiguation of TDOA Estimates in The Presence of Reverberation
Cecilia Maria Zannini 1 Albenzio Cirillo 1 Raffaele Parisi 1 Aurelio Uncini 1
1 Cecilia Maria Zannini, Albenzio Cirillo, Raffaele Parisi, Aurelio Uncini, INFOCOM Dept., University of Rome “La Sapienza”, Italy

P-7
Error Analysis on Source Localization in Ad-Hoc Wireless Microphone Networks
Wouter van Herpen 1 Sriram Srinivasan 2 Piet Sommen 1
1 Department of Electrical Engineering, Eindhoven University of Technology, Eindhoven, Netherlands
2 Digital Signal Processing Group, Philips Research, Eindhoven, Netherlands

P-8
Evaluation of Different Microphone Arrays and Localization Algorithms in the Context of Ambient Assisted Living
Christian Bartsch 1 Andreas Volgenandt 2 Thomas Rohdenburg 2 Joerg Bitzer 1,2
1 Institute for Hearing Technology and Audiology (IHA), Jade University Of Applied Sciences, Oldenburg, Germany
2 Department Hearing, Speech and Audio Technology (HSA), Fraunhofer Institute for Digital Media Technology (IDMT), Oldenburg, Germany

P-9
A GPU Implementation of the Srp-Phat Sound Source Localization Algorithm
Luiz Gonzaga da Silveira Jr. 1 Vicente Peruffo Minotto 1 Claudio Rosito Jung 2 Bowon Lee 3
1 Department of Computer Science, UNISINOS, So Leopoldo, RS, Brazil
2 Institute of Informatics, UFRGS, Porto Alegre, RS, Brazil
3 Multimedia Communications and Networking Lab, Hewlett-Packard Laboratories, Palo Alto, CA, United States

P-10
Robust Time Delay Estimation in the Presence of Reverberation Using Median Group Delay
Pratik Shah 1 Steven Grant 1
1 Electrical and Computer Engineering, Missouri University of Science and Technology, Rolla, Missouri, United States

P-11
A Wavelet-Based GCC Prefiltering Algorithm for Speech DOA Estimation
Di Liu 1 Andy Wai Hoong Khong 1
1 School of Electrical and Electronic Engineering, Nanyang Technological University, Singapore

10:45-11:00
Coffee Break

11:00-13:00
Best Student Papers Session - Finalists
 Diamond Hall


The Winners will be declared in the award session at the last day of the workshop
  
 
11:00
On the Use of Channel Shortening in Multichannel Acoustic System Equalization
Wancheng Zhang 1 Emanuel Habets 1 Patrick Naylor 1
1 Electrical and Electronic Engineering, Imperial College London, London, United Kingdom

11:20
An EM Approach to Integrated Multichannel Speech Separation and Noise Suppression
Dang Hai Tran Vu 1 Reinhold Haeb-Umbach 1
1 Department of Communications Engineering, University of Paderborn, Paderborn, NRW, Germany

11:40
A Reduced Bandwidth Binaural MVDR Beamformer
Shmulik Markovich Golan 1 Sharon Gannot 1 Israel Cohen 2
1 School of Engineering, Bar Ilan University, Ramat Gan, Israel
2 Department of Electrical Engineering, Technion, Haifa, Israel

12:00
Design of Robust Polynomial Beamformers as a Convex Optimization Problem
Edwin Mabande 1 Walter Kellermann 1
1 Multimedia Communications and Signal Processing, University of Erlangen-Nuremberg, Erlangen, Bavaria, Germany

12:20
Adaptive Distributed Noise Reduction for Speech Enhancementin Wireless Acoustic Sensor Networks
Alexander Bertrand 1 Jef Callebaut 1 Marc Moonen 1
1 ESAT/SCD-SISTA, Katholieke Universiteit Leuven, Leuven, Belgium

12:40
Blind Source Separation with Distributed Microphone Pairs Using Permutation Correction by Intra-Pair TDOA Clustering
Takuma Ono 1 Shigeki Miyabe 1 Nobutaka Ono 1 Shigeki Sagayama 1
1 Graduate School of Information Science and Technology, The University of Tokyo, Tokyo, Japan

13:00-14:00
Lunch Break

14:00-20:00
Excursion to the Old City of Jerusalem

20:00-23:59
Gala Dinner in Jerusalem

THURSDAY, SEPTEMBER 02, 2010
08:00-08:30
Registration

08:30-09:30
Plenary Lecture 3
 Diamond Hall

08:30
Distributed Microphone Array Signal Processingfor Hearing Aids
Doclo Simon 1
1 Institute of Physics - Signal Processing Group, University of Oldenburg, Germany

09:30-10:30
Research and Development in Sponsoring Companies
 Diamond Hall

09:30
MT103 - OEM Audio DSP Development Platform
Joseph Marash 1
1 Phoenix Audio Technologies, USA

09:50
Signal Processing Technologies in Voice over IP Applications
Eli Shoval 1 Oren Klimker 1 Guy Shterlich 1
1 AudioCodes Ltd, Israel

10:10
S-Cube™ :Generic controller for active acoustics
Yoel Naor 1
1 Silentium, Israel

10:30-11:00
Coffee Break

11:00-12:30
Poster Session E: Dereverberation and Room Modeling
 Diamond Hall

P-1
Room Volume Classification from Reverberant Speech
Noam Shabtai 1 Boaz Rafaely 1 Yaniv Zigel 2
1 Electrical and Computer Engineering, Ben Gurion University of the Negev, Beer Sheva, Israel
2 Biomedical Engineering, Ben Gurion University of the Negev, Beer Sheva, Israel

P-2
On the Robustness of Room Impulse Response Reshaping
Tiemin Mei 2 Alfred Mertins 1
1 Institute for Signal Processing, University of Lübeck, Lübeck, Germany
2 School of Information Science and Engineering, Shenyang Ligong University of Technology, Shenyang, China

P-3
A Parametric Least-Squares Approximation for Multichannel Equalization of Room Acoustics
Dominic Schmid 1 Gerald Enzner 1
1 Institute of Communication Acoustics, Ruhr-Universität Bochum, Bochum, Germany

P-4
An Improved Algorithm for Blind Reverberation Time Estimation
Heinrich Loellmann 1 Emre Yilmaz 1 Marco Jeub 1 Peter Vary 1
1 Institute of Communication Systems and Data Processing, RWTH Aachen University, Aachen, Germany

P-5
A Blind Multichannel Dereverberation Algorithm Based on the Natural Gradient
Massimiliano Tonelli 1 Mike Davies 1
1 IDCOM & Joint Research Institute for Signal and Image Processing, Edinburgh University, Edinburgh, United Kingdom

P-6
Analysis of Reflected Wavefronts by Means of a Line Microphone Array
Piergiorgio Svaizer 1 Alessio Brutti 1 Maurizio Omologo 1
1 Fondazione Bruno Kessler - IRST, Povo - Trento, Italy

P-7
Multipoint Room Response Equalization with Group Delay Compensation
Alberto Carini 1 Stefania Cecchi 2 Laura Romoli 2
1 MFI, Università di Urbino "Carlo Bo", Urbino, (PU), Italy
2 A3Lab - DIBET, Università Politecnica delle Marche, Ancona, (AN), Italy

P-8
Estimation of Acoustic Resonances for Room Ttransfer Function Equalization
Pepe Gil-Cacho 1 Toon van Waterschoot 1 Marc Moonen 1 Soren Holdt Jensen 2
1 ESAT-SCD, Katholieke Univeristeit Leuven, Leuven, Belgium
2 Dept. Electronic Systems, Aalborg University, Aalborg, Denmark

P-9
A Two-Step Approach to Blindly Infer Room Geometries
Jason Filos 1 Emanuel Habets 1 Patrick A Naylor 1
1 Electrical and Electronic Engineering, Imperial College London, London, United Kingdom

P-10
Subband Scale Factor Ambiguity Correction Using Multiple Filterbanks
Boaz Castro 1 Nikolay Gaubitch 2 Emanuel Habets 2 Sharon Gannot 1 Patrick Naylor 2 Steven Grant 3
1 School of Engineering, Bar-Ilan University, Ramat-Gan, Israel
2 Department of Electrical and Electronic Engineering, Imperial College, London, United Kingdom
3 Electrical and Computer Engineering, Missouri University of Science and Technology, Rolla, Missouri, United States

P-11
Supervised Identification and Removal of Common Filter Components in Adaptive Blind SIMO System Identification
Mark Thomas 1 Nikolay Gaubitch 1 Emanuel Habets 1 Patrick Naylor 1
1 Electrical and Electronic Engineering, Imperial College London, London, United Kingdom

P-12
A Noise Robust Multichannel Algorithm for Blind Estimation of Room Impulse Responses
Liao Lei 1 Khong Andy W. H. 2
1 Electrical and Electronic Engineering, Nanyang Technological University, Singapore
2 Electrical and Electronic Engineering, Nanyang Technological University, Singapore

12:30-13:30
Lunch Break

13:30-15:00
Poster Session F: Echo cancellation and Adaptive Filtering
 Diamond Hall

P-1
Multifrequency Self-Optimizing Narrowband Interference Canceller
Maciej Niedzwiecki 1 Michal Meller 1
1 Department of Electronics, Telecommunications and Computer Science, Gdansk University of Technology, Gdansk, Poland

P-2
Multi-channel Acoustic Echo Cancellation Based On Residual Echo Enhancement with Effective Channel Decorrelation via Resampling
Ted S. Wada 1 Biing-Hwang Juang 1
1 Center for Signal and Image Processing, Georgia Institute of Technology, Atlanta, Georgia, United States

P-3
Investigation and Development of Digital Active Noise Control Headsets
Hauke Krüger 1 Marco Jeub 1 Thomas Schumacher 1 Peter Vary 1 Christophe Beaugeant 2
1 Institut for Communication Systems and Data Processing, RWTH Aachen University, Aachen, NRW, Germany
2 Business Group Communication Solutions, Infineon Technologies France, Sophia-Antipolis, France

P-4
Low Delay Filtering for Joint Noise Reduction and Residual Echo Suppression
Christelle YEMDJI 1 Moctar Mossi I. 1 Nicholas Evans W. D. 1 Christophe Beaugeant 2
1 EURECOM Institute, Sophia-Antipolis, France
2 Infineon Technologies, Sophia-Antipolis, France

P-5
Self-Configuring System Identification via Evolutionary Frequency-Domain Adaptive Filters
Marcus Zeller 1 Walter Kellermann 1
1 Multimedia Communications and Signal Processing, University of Erlangen-Nuremberg, Erlangen, Germany

P-6
Improved Approach to Stereophonic Channel Decorrelation Based on Missing Fundamental Theory
Laura Romoli 1 Stefania Cecchi 1 Paolo Peretti 1 Francesco Piazza 1
1 Dipartimento di Ingegneria Biomedica, Elettronica e Telecomunicazioni, Università Politecnica delle Marche, Ancona, Italy

P-7
Fixed-point Implementation of a Coherence-based Double-talk Detector for Line Echo Cancelers
Radu Pralea 1
1 Packet Telephony Applications, Freescale Semiconductor Romania, Bucharest, Romania

P-8
A Functional Link Based Nonlinear Echo Canceller Exploiting Sparsity
Danilo Comminiello 1 Michele Scarpiniti 1 Raffaele Parisi 1 Aurelio Uncini 1
1 INFOCOM Dpt., "Sapienza" University of Rome, Rome, Rome, Italy

P-9
A Phase Robust Spectral Magnitude Estimator for Acoustic Echo Suppression
Oystein Birkenes 1
1 TANDBERG, now a part of Cisco, Oslo, Norway

P-10
New Models for Characterizing Non-Linear Distortions in Mobile Terminal Loudspeakers
Moctar Mossi Idrissa 1 Christelle Yemdji 1 Nicholas Evans 1 Christian Herglotz 2 Christophe Beaugeant 2 Philippe Degry 2
1 EURECOM, 06560 Sophia-Antipols, France
2 Infineon Technologies, 06560 Sophia-Antipols, France

P-11
Evaluation of an Improved Deviation Measure for Two-Path Echo Cancellation
Christian Schüldt 1 Fredric Lindstrom 2 Ingvar Claesson 1
1 Department of Signal Processing, Blekinge Institute of Technology, Ronneby, Sweden
2 Limes Audio AB, Umeå, Sweden

P-12
A Closed-Form Solution to the Delay Constraint in Noise Cancellation and Feedback Signal Separation
Akihiko Sugiyama 1 Michael Matejko 2
1 Information and Media Processing Research Laboratories, NEC Corporation, Kawasaki, Kanagawa, Japan
2 EECE, University of Erlangen-Nuernberg, Nuernberg, Germany

15:00-15:30
Coffee Break

15:30-16:30
Plenary Lecture 4
 Diamond Hall

15:30
Noise and Echo Control for Immersive Voice Communication in Spacesuits
Yiteng (Arden) Huang 1
1 Wevoice, Inc., New Jersey, United States

16:30-17:30
Closing Session and Awards Ceremony
Diamond Hall