| Acoustic Echo Reduction Systems (Single & Multi-Channel) |
| Acoustic Echo Cancellation on the Adaptive Multi-Rate Speech Codec Parameters |
Daniele Giacobello, Aalborg University; Danilo Neri, Nokia Siemens Networks; Luca Prati, Nokia Siemens Networks; Sergio Brofferio, Politecnico di Milano |
| Acoustic-Coupling Level Estimation for Performance Improvement of Echo Reduction |
Masahio Fukui, NTT Cyber Space Laboratories, NTT Corporation; Suehiro Shimauchi, NTT Cyber Space Laboratories, NTT Corporation; Akira Nakagawa, NTT Cyber Space Laboratories, NTT Corporation; Yoichi Haneda, NTT Cyber Space Laboratories, NTT Corporation; Akitoshi Kataoka, Ryukoku University |
| NLMS-Type System Identification of Miso Systems with Shifted Perfect Sequences |
Christiane Antweiler, RWTH Aachen University; Aulis Telle, RWTH Aachen University; Peter Vary, RWTH Aachen University |
| Multi-Channel Echo Control by Model Learning |
Majid Fozunbal, Hewlett-Packard Laboratories; Ton Kalker, Hewlett-Packard Laboratories; Ronald Schafer, Hewlett-Packard Laboratories |
| An Echo Canceller Using Smoothed-Coefficient Filter with Adaptive Time Constant Controlled by High-Pass Errors |
Osamu Hoshuyama, NEC Corporation |
| Speech Recognizer with Improved Detection Performance in an Automotive Environment |
Ashtosh Sapru, Aricent Communications Pvt Ltd; Ravi Lakkundi, Aricent Communications Pvt Ltd; Nisar Ahmed, Aricent Communications Pvt Ltd |
| Coefficient Pruning for Higher-Order Diagonals of Volterra Filters Representing Wiener-Hammerstein Models |
Marcus Zeller, University of Erlangen-Nuremberg; Walter Kellermann, University of Erlangen-Nuremberg |
| Model-Based Vs. Traditional Frequency-Domain Adaptive Filtering in the Presence of Continuous Double-Talk and Acoustic Echo Path Variability |
Sarmad Malik, Ruhr-University Bochum; Gerald Enzner, Ruhr-University Bochum |
| Robust Early Echo Cancellation and Late Echo Suppression in the STFT Domain |
Emanuël Habets, Technion - Israel Institute of Technology; Sharon Gannot, Bar-Ilan University; Israel Cohen, Technion - Israel Institute of Technology |
| Nonlinear Acoustic Echo Cancellation Based on a Multiplicative Transfer Function Approximation |
Yekutiel Avargel, Technion - Israel Institute of Technology; Israel Cohen, Technion - Israel Institute of Technology |
| Acoustic Echo Suppression Based on Separation of Stationary and Non-Stationary Echo Components |
Fabian Kuech, Fraunhofer IIS; Markus Kallinger, Fraunhofer IIS; Markus Schmidt, Fraunhofer IIS; Christof Faller, Illusonic; Alexis Favrot, Illusonic |
| Challenges and Solutions for Designing Software AEC on Personal Computers |
Qin Li, Microsoft Corporation; Chao He, Microsoft Corporation; Wei-Ge Chen, Microsoft Corporation |
| Acoustic Echo Control Based on Temporal Fluctuations of Short-Time Spectra |
Alexis Favrot, Illusonic; Christof Faller, Illusonic; Markus Kallinger, Fraunhofer IIS; Fabian Küch, Fraunhofer IIS; Markus Schmidt, Fraunhofer IIS |
| Multichannel Acoustic Echo Cancelation in Multiparty Spatial Audio Conferening with Constrained Kalman Filtering |
Zhengyou Zhang, Microsoft Corporation; Qin Cai, Microsoft Corporation; Jack Stokes, Microsoft Corporation |
| Wideband Echo Perception |
Silvia Poschen, HEAD Acoustics; Frank Kettler, HEAD Acoustics; Alexander Raake, Deutsche Telekom Laboratories; Sascha Spors, Deutsche Telekom Laboratories |
| Multichannel Transform Domain Adaptive Filtering: a Two Stage Approach and Illustration for Acoustic Echo Cancelation |
Sascha Spors, Deutsche Telekom Laboratories; Herbert Buchner, Deutsche Telekom Laboratories |
| A Study on Blind Source Separation for Preprocessing of an Acoustic Echo Canceller |
Yoshihiro Sakai, University of Electro-Communications; Wataru Mitsuhashi, University of Electro-Communications |
| Use of Decorrelation Procedure for Source and Echo Suppression |
Ted Wada, Georgia Institute of Technology; Shigeki Miyabe, Nara Institute of Science and Technology; Biing-Hwang Juang, Georgia Institute of Technology |
| Active Noise Control Systems |
| Combined Active Noise Control and Noise Reduction in Hearing Aids |
Romain Serizel, Katholieke Universiteit Leuven; Marc Moonen, Katholieke Universiteit Leuven; Jan Wouters, Katholieke Universiteit Leuven; Soren Holdt Jensen, Aalborg University |
| Nonlinear Active Noise Control Using VOLTERRA Filtering with a Variable Step-Size Gs-Pap Algorithm |
J B Seo, Hanyang University; K J Kim, Hanyang University; D W Kim, Dasan Cousultants Co, Ltd; M O Oh, Dasan Cousultants Co, Ltd; W H Cho, Dasan Cousultants Co, Ltd; H K Lee, Dasan Cousultants Co, Ltd; W Y Kim, Hanyang University; Sang Won Nam, Hanynag University |
| Adaptive Filtering Algorithms & Structures |
| An Improved Proportionate Affine Projection Algorithm for Network Echo Cancellation |
Kirill Sakhnov, Czech Technical University in Prague |
| A Weighted Overlap-Add Based Wave Domain Adaptive Filtering Algorithm |
Paolo Peretti, Universita Politecnica delle Marche; Laura Romoli, Universita Politecnica delle Marche; Stefania Cecchi, Universita Politecnica delle Marche; Lorenzo Palestini, Universita Politecnica delle Marche; Francesco Piazza, Universita Politecnica delle Marche |
| Low Distortion Decoupled Crosstalk Resistant Adaptive Noise Canceller |
Ludovick Lepauloux, Orange Labs; Pascal Scalart, IRISA/CAIRN; Claude Marro, Orange Labs |
| Microphone Arrays & Array Signal Processing |
| Optimal Azimuthal Steering of a First-Order Superdirectional Microphone Response |
Rene Derkx, Philips Research Laboratories |
| A New Approach to Digital Directivity Control of Loudspeakers Line Arrays Using Wave Field Synthesis Theory |
Laura Romoli, Universita Politecnica delle Marche; Paolo Peretti, Universita Politecnica delle Marche; Lorenzo Palestini, Universita Politecnica delle Marche; Stefania Cecchi, Universita Politecnica delle Marche; Francesco Piazza, Universita Politecnica delle Marche |
| A New Cost Function for Direction-of-Arrival Estimation of Multiple Sound Sources Using Two Microphones |
Minh The Dang, Paichai University; Seung-Hyon Nam, Paichai University |
| Interpolation Methods for the SRP-Phat Algorithm |
Sakari Tervo, Helsinki University of Technology; Tapio Lokki, Helsinki University of Technology |
| Acoustic Localization Using Reverberation with Virtual Microphones |
Teemu Korhonen, Tampere University of Technology |
| Score: a Low Complexity, Robust Algorithm for the Detection of Corrupt Sensors and Self-Calibration of Microphone Arrays |
Nilesh Madhu, Ruhr-University Bochum; Rainer Martin, Ruhr-University Bochum |
| A Scalable Framework for Multiple Speaker Localization and Tracking |
Nilesh Madhu, Ruhr-University Bochum; Rainer Martin, Ruhr-University Bochum |
| Maximum-Likelihood Sound Source Localization with a Multivariate Complex LAPLACIAN Distribution |
Bowon Lee, Hewlett-Packard Laboratories; Ton Kalker, Hewlett-Packard Laboratories; Ronald Schafer, Hewlett-Packard Laboratories |
| Non-Spherical Microphone Array Structures for 3D Beamforming and Spherical Harmonic Analysis |
Thushara Abhayapala, Australian National University; Aastha Gupta, Australian National University |
| A Log-MMSE Adaptive Beamformer Using a Nonlinear Spatial Filter |
Michael Seltzer, Microsoft Corporation; Ivan Tashev, Microsoft Corporation |
| On Hidden Markov Model Maximum Negentropy Beamforming |
Barbara Rauch, Spoken Language Systems; Kenichi Kumatani, Saarland University; Friedrich Faubel, Saarland University; John McDonough, Saarland University; Dietrich Klakow, Saarland University |
| Analytical Solution of Nonlinear Microphone Array Based on Complementary Beamforming |
Shigeki Miyabe, Nara Institute of Science and Technology; Biing-Hwang Juang, Georgia Institute of Technology; Hiroshi Saruwatari, Nara Institute of Science and Technology; Kiyohiro Shikano, Nara Institute of Science and Technology |
| Data Driven Beamformer Design for Binaural Headset |
Ivan Tashev, Microsoft Corporation; Michael Seltzer, Microsoft Corporation |
| Multi-Rate Filter Banks & Sub-Band Systems |
| 9065Scale Factor Ambiguity Correction for Subband Blind Multichannel Identification |
Nikolay Gaubitch, Imperial College London; Xiang Lin, Imperial College London; Patrick Naylor, Imperial College London |
| Noise And Acoustic Environments & Characteristics |
| Binaural Distance Perception Based on Direct-to-Reverberant Energy Ratio |
Yan-Chen Lu, University of Sheffield; Martin Cooke, University of Sheffield |
| Variable Speech Distortion Weighted Multichannel Wiener Filter Based on Soft Output Voice Activity Detection for Noise Reduction in Hearing Aids |
Kim Ngo, Katholieke Universiteit Leuven; Ann Spriet, Katholieke Universiteit Leuven; Marc Moonen, Katholieke Universiteit Leuven; Jan Wouters, Katholieke Universiteit Leuven; Søren Holdt Jensen, Aalborg University |
| Noise Suppression with Adaptive Adjustment of the Maximum Attenuation |
Mohamed Krini, Harman/Becker Automotive Systems; Gerhard Schmidt, Harman/Becker Automotive Systems |
| Sound Capture, Processing & Reproduction Applications |
| Multiple Position Room Response Equalization with Frequency Domain Fuzzy C-Means Prototype Design |
Ivan Omiciuolo, University of Trieste; Alberto Carini, University of Urbino; Giovanni L Sicuranza, University of Trieste |
| Multichannel Cross-Talk Cancellation in a Call-Center Scenario Using Frequency-Domain Adaptive Filtering |
Anthony Lombard, University of Erlangen-Nuremberg; Walter Kellermann, University of Erlangen-Nuremberg |
| Sound Source Separation |
| Source Number Estimation and Clustering for Underdetermined Blind Source Separation |
Benedikt Loesch, University of Stuttgart; Bin Yang, University of Stuttgart |
| A Novel Robust Solution to the Permutation Problem Based on a Joint Multiple TDOA Estimation |
Francesco Nesta, Fondazione Bruno Kessler; Maurizio Omologo, Fondazione Bruno Kessler; Piergiorgio Svaizer, Fondazione Bruno Kessler |
| Generalized Eigenvector Blind Speech Separation Under Coherent Noise in a GSC Configuration |
Dang Hai Tran Vu, University of Paderborn; Alexander Krueger, University of Paderborn; Reinhold Haeb-Umbach, University of Paderborn |
| Sound Source Separation with Distributed Microphone Arrays in the Presence of Clock Synchronization Errors |
Zicheng Liu, Microsoft Corporation |
| Using Phase Linearity to Tackle the Permutation Problem in Audio Source Separation |
Keisuke Toyama, Queen Mary University of London; Andrew Nesbit, Queen Mary University of London; Maria Jafari, Queen Mary University of London; Mark Plumbley, Queen Mary University of London |
| Instantaneous Blind Signal Extraction Using Second Order Statistics |
Brian Bloemendal, Eindhoven University of Technology; Jakob van de Laar, Philips Research Laboratories; Piet Sommen, Eindhoven University of Technology |
| A Source Reassignment Technique for Time-Frequency Masking Audio Separation |
Maximo Cobos, Technical University of Valencia; Jose Javier Lopez, Technical University of Valencia; Jan Oliver Hinz, Technical University of Valencia |
| A New Model-Based Underdetermined Speech Separation |
Zaher El Chami, Orange Labs; Antoine Dinh-Tuan Pham, Laboratory of Modeling and Computation; Christine Servière, GIPSA Lab; Alexandre Guerin, Orange Labs |
| Estimation of the Ideal Binary Mask Using Directional Systems |
Jesper Boldt, Oticon A/S; Ulrik Kjems, Oticon A/S; Michael Pedersen, Oticon A/S; Thomas Lunner, Oticon Research Centre Eriksholm; Deliang Wang, Ohio State University |
| New Architecture Combining Blind Signal Extraction and Modified Spectral Subtraction for Suppression of Background Noise |
Jani Even, Nara Institute of Science and Technology; Hiroshi Saruwatari, Nara Institute of Science and Technology; Kiyohiro Shikano, Nara Institute of Science and Technology |
| Performance Improvement of Higher-Order ICA Using Learning Period Detection Based on Closed-Form Second-Order ICA and Kurtosis |
Yuuki Fujihara, Nara Institute of Science and Technology; Yu Takahashi, Nara Institute of Science and Technology; Shigeki Miyabe, Nara Institute of Science and Technology; Hiroshi Saruwatari, Nara Institute of Science and Technology; Kiyohiro Shikano, Nara Institute of Science and Technology; Akira Tanaka, Hokkaido University |
| Frequency-Domain Blind Source Separation with Permutation Control |
Peng Xie, Missouri University of Science and Technology; Steven Grant, Missouri University of Science and Technology |
| Speech-Databases & Software Tools |
| Instrumental Speech Distortion Assessment of Black Box Speech Enhancement Systems |
Kai Steinert, Siemens AG, Corporate Technology; Suhadi Suhadi, Braunschweig Technical University, Institute for Communications Technology; Tim Fingscheidt, Braunschweig Technical University; Martin Schönle, Siemens AG, Corporate Technology |
| Speech Enhancement & Dereverberation Techniques |
| A Non-Intrusive Quality Measure of Dereverberated Speech |
Tiago Falk, Queen's University; Wai-Yip Chan, Queen's University |
| Robust Hands-Free Voice Control for Medical Applications |
Bahaa Eddine Sarroukh, Philips Research Laboratories; L C A van Stuivenberg, Philips Research Laboratories; C P Janse, Philips Research Laboratories |
| A Simplified Decoding Method for a Robust Distant-Talking ASR Concept Based on Feature-Domain Dereverberation |
Armin Sehr, University of Erlangen-Nuremberg; Walter Kellermann, University of Erlangen-Nuremberg |
| Techniques for Estimating the Ideal Binary Mask |
Yi Hu, University of Texas at Dallas; Philipos C Loizou, University of Texas at Dallas |
| A Decoupled Filtered-X LMS Algorithm for Listening-Room Compensation |
Stefan Goetze, University of Bremen; Markus Kallinger, University of Oldenburg; Alfred Mertins, University of Luebeck; Karl-Dirk Kammeyer, University of Bremen |
| Estimation of the Reverberation Time in Noisy Environments |
Heinrich Löllmann, RWTH Aachen University; Peter Vary, RWTH Aachen University |
| Modified Kalman Filter Exploiting Interframe Correlation of Speech and Noise Magnitudes |
Thomas Esch, RWTH Aachen University; Peter Vary, RWTH Aachen University |
| An Analysis of Quefrency Selective Temporal Smoothing of the Cepstrum in Speech Enhancement |
Dirk Mauler, Ruhr-University Bochum; Timo Gerkmann, Ruhr-University Bochum; Rainer Martin, Ruhr-University Bochum |
| Enhancement of Noisy Reverberant Speech by Linear Filtering Followed by Nonlinear Noise Suppression |
Takuya Yoshioka, NTT Communication Science Laboratories; Tomohiro Nakatani, NTT Communication Science Laboratories; Masato Miyoshi, NTT Communication Science Laboratories |
| Incremental Estimation of Reverberation with Uncertainty Using Prior Knowledge of Room Acoustics for Speech Dereverberation |
Tomohiro Nakatani, NTT Communication Science Laboratories; Takuya Yoshioka, NTT Communication Science Laboratories; Keisuke Kinoshita, NTT Communication Science Laboratories; Masato Miyoshi, NTT Communication Science Laboratories; Biing-Hwang Juang, Georgia Institute of Technology |
| Improving Quality Prediction Accuracy of P.563 for Noise Suppression |
Anders Ekman, Royal Institute of Technology; Bastiaan Kleijn, Royal Institute of Technology |
| Coherent Modulation Comb Filtering for Enhancing Speech in Wind Noise |
Brian King, University of Washington; Les Atlas, University of Washington |
| A Combined Approach for Estimating a Feature-Domain Reverberation Model in Non-Diffuse Environments |
Armin Sehr, University of Erlangen-Nuremberg; Jimi Y C Wen, Imperial College London; Walter Kellermann, University of Erlangen-Nuremberg; Patrick A Naylor, Imperial College London |
| Particle Filter Based Soft-Mask Estimation for Missing Feature Reconstruction |
Friedrich Faubel, Saarland University; Humza Raja, Saarland University; John McDonough, Saarland University; Dietrich Klakow, Saarland University |
| Automatic Optimization Scheme of Spectral Subtraction Based on Musical Noise Assessment via Higher-Order Statistics |
Yoshihisa Uemura, Nara Institute of Science and Technology; Yu Takahashi, Nara Institute of Science and Technology; Hiroshi Saruwatari, Nara Institute of Science and Technology; Kiyohiro Shikano, Nara Institute of Science and Technology; Kazunobu Kondo, YAMAHA Corp |
| Transducers, Acoustic Front-Ends, Hardware |
| Conventional and Distributed Mode Loudspeaker Arrays for the Application of Wave-Field Synthesis to Videoconference |
Jose Javier Lopez, Technical University of Valencia; Basilio Pueo, University of Alicante; Maximo Cobos, Technical University of Valencia |
| Development and Evaluation of Pocket-Size Blind Source Separation Microphone |
Takashi Hiekata, Kobe Steel, Ltd; Y Ikeda, Kobe Steel, Ltd; T Yamashita, Kobe Steel, Ltd; T Morita, Kobe Steel, Ltd; R Zhang, Feng Co, Ltd; Y Mori, Nara Institute of Science and Technology; Hiroshi Saruwatari, Nara Institute of Science and Technology; Kiyohiro Shikano, Nara Institute of Science and Technology |
| Voice Activity Detection, Double-Talk Detection & Signal Segmentation |
| Voice Activity Detection Based on the Adaptive Multi-Rate Speech Codec Parameters |
Daniele Giacobello, Aalborg University; Matteo Semmoloni, Nokia Siemens Networks; Danilo Neri, Nokia Siemens Networks; Luca Prati, Nokia Siemens Networks; Sergio Brofferio, Politecnico di Milano |
| Multi-Decision Sub-Band Voice Activity Detection for Speech Enhancement |
Alan Davis, Curtin University / Western Australian Telecommunications Research Institute; Siow Yong Low, Western Australian Telecommunications Research Institute; Sven Nordholm, Curtin University / Western Australian Telecommunications Research Institute |
| Handling Speaker Position Changes in a Meeting Diarization System by Combining DOA Clustering and Speaker Identification |
Tobias Hager, NTT Communication Science Laboratories; Shoko Araki, NTT Communication Science Laboratories; Kentaro Ishizuka, NTT Communication Science Laboratories; Masakiyo Fujimoto, NTT Communication Science Laboratories; Tomohiro Nakatani, NTT Communication Science Laboratories; Shoji Makino, NTT Communication Science Laboratories |
| New and Emerging Technologies |
| On the Reconstruction of Undersampled Wireless Acoustic Sensor Signals |
Piet Sommen, Eindhoven University of Technology; Kees Janse, Philips Research Laboratories |
