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Invited Talk |
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Modern Hearing Instrument Technology with a Focus on Signal Processing
Dr. Markus Hofbauer
Abstract Hearing aids have to cope with rigorous constraints: for the sake of cosmetics the devices have to be very small and the electronics has to be ultra power efficient to last several days on a coin cell battery. Furthermore the signal processing delay should not exceed 10ms for psychoacoustic reasons. Despite of these demanding conditions, modern hearing aids are small computers which provide substantial computational power that allows implementing several sophisticated signal processing algorithms. Hearing aids do not just amplify sound. Following the signal processing scheme of the auditory system, sound is processed on an auditory filterbank scale and is amplified with a non-linear compression scheme. In order to re-establish speech intelligibility, noise canceling algorithms attenuate disturbing noise sources and enhance the targeted speech source. Automatic sound scene classification allows to automatically adjust the hearing aid algorithm parameters accordingly to the current hearing situation. Wireless technology enables the streaming of audio and other data between and to the hearing aid devices and thus initiates prospects for new hearing aid features, applications and signal processing methods.
Biography Markus Hofbauer was born in Basel, Switzerland, on July 10, 1972. He attended the Gymnasium in Weil am Rhein, Germany, where he obtained the Abitur and spent an interim high-school year in Tacoma, WA, USA and received the high-school Diploma. In 1993 he joined the Swiss Federal Institute of Technology Zürich (ETH) to study electrical engineering and graduated with a Diploma degree in electrical engineering in 1998. Subsequently, he started as a research and teaching assistant at the Signal and Information Processing Laboratory (ISI) at ETH. In parallel with his work on his Ph.D., he was an appointed lecturer of the graduate course 'Adaptive Filters' at ETH from 1999-2004. He is co-author of the textbook 'Adaptive Filter'. In March 2005 he completed his Ph.D thesis entitled `Optimal Linear Separation and Deconvolution of Acoustical Convolutive Mixtures' and received the Ph.D. degree from ETH. From May 2005 till December 2007 he worked for SIEMENS Zürich as an R&D engineer in a technological consulting team. In January 2008 he joined the research and development department of Phonak AG where he develops hearing aid algorithms as a DSP senior engineer. |
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Volker Koch, 10/2009 |