|Walter Kellermann is a Professor for communications at the Chair of Multimedia Communications and Signal Processing of the University of Erlangen-Nuremberg, Germany. He received the Dipl.-Ing. degree in electrical engineering from the University of Erlangen-Nuremberg in 1983, and the Dr.-Ing. degree from the Technical University Darmstadt, Germany, in 1988. From 1989 to 1990, he was a Postdoctoral Member of technical staff at AT&T Bell Laboratories, Murray Hill, NJ. In 1990, he joined Philips Kommunikations Industrie, Nuremberg, Germany. From 1993 to 1999, he was a Professor at the Fachhochschule Regensburg, before he had joined the University of Erlangen-Nuremberg as a Professor and Head of the Audio Research Laboratory in 1999. He authored or coauthored seven book chapters and more than 70 refereed papers in journals and conference proceedings. He served as a Guest Editor to various journals, as an Associate Editor and Guest Editor to IEEE Transactions on Speech and Audio Processing from 2000 to 2004, and presently serves as an Associate Editor to the EURASIP Journal on Signal Processing and EURASIP Journal on Advances in Signal Processing. He was the General Chair of the 5th International Workshop on Microphone Arrays in 2003 and the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics in 2005. His current research interests include speech signal processing, array signal processing, adaptive filtering, and its applications to acoustic human/machine interfaces.
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- 14:00, Monday, October 01, 2007
- Anfiteatro do Complexo Interdisciplinar (IST)
- Walter Kellermann, Erlangen-Nuremberg University, Distinguished Lecturer of the IEEE Signal Processing Society.
The acoustic interface for future multimedia and communication terminals should be hands-free and as natural as possible, which implies that the user should be free to move and and should not need to wear any devices. For digital signal processing this poses major challenges both for signal acquisition and reproduction, which reach far beyond the current state of the technology.
For ideal acquisition of an acoustic source signal in noisy and reverberant environments, we need to compensate acoustic echoes, suppress noise and interferences and we would like to dereverberate the desired source signal.
On the other hand, for a perfect reproduction of real or virtual acoustic scenes we need to create desired sound signals at the listeners ears, while at the same time we have to remove undesired reverberance and to suppress local noise.
In this talk we will briefly analyze the fundamental problems for signal processing in the framework of MIMO (multiple input - multiple output) systems and discuss current solutions.
In accordance with ongoing research we emphasize nonlinear and multichannel acoustic echo cancellation, as well as microphone array signal processing for beamforming, interference suppression, blind source separation, and source localization.