Signal processing & audio processors

Abstract

Signal processing algorithms are the hidden components in the audio processor that converts the received acoustic signal into electrical impulses while maintaining as much relevant information as possible. Signal processing algorithms should be smart enough to mimic the functionality of external, middle and the inner-ear to provide the cochlear implant (CI) user with a hearing experience as natural as possible. Modern sound processing strategies are based on the continuous interleaved sampling (CIS) strategy proposed by B. Wilson in 1991, which provided envelope information over several intracochlear electrodes. The CIS strategy brought significant gains in speech perception. Translational research activities of MED-EL resulted in further improvements in speech understanding in noisy environments as well as enjoyment of music by not only coding CIS-based envelope information, but by also representing temporal fine structure information in the stimulation patterns of the apical channels. Further developments include "complete cochlear coverage" made possible by deep insertion of the intracochlear electrode, elaborate front end processing, anatomy based fitting (ABF), triphasic pulse stimulation instrumental in the suppression of facial nerve stimulation, and bimodal delay compensation allowing unilateral CI users to experience hearing with hearing aids on the contralateral ear. The large number of hardware developments might be exemplified by the RONDO, the world's first single unit audio processor in 2013. This article covers the milestones of translational research around the signal processing and audio processor topic that took place in association with MED-EL.

Keywords: Continuous interleaved sampling; anatomy-based fitting; bimodal delay compensation; fine structure processing; single unit processor; triphasic pulse stimulation.