Using long Med-El Combi40+ electrode arrays, it is now possible to cover the whole range of the cochlea, up to about two turns. Such insertion depths have received little attention. To evaluate the contribution of deeply inserted electrodes, five Med-El cochlear implant users were tested on vowel and consonant identification tests with fittings with first one, two, and up to five apical electrodes being deactivated. In addition, subjects performed pitch-ranking experiments, using loudness-balanced stimuli, to identify electrodes creating pitch confusions. Radiographs were taken to measure each electrode insertion depth. All subjects used each modified fitting for two periods of about 3 weeks. During the experiment, the same stimulation rate and frequency range were maintained across all the fittings used for each individual subject. After each trial period the subject had to perform three consonant and three vowel identification tests. All subjects showed deep electrode insertions ranging from 605-to 720-. The two subjects with the deepest electrode insertions showed significantly increased vowel-and consonant-identification performances with fittings with the two or three most apical electrodes deactivated compared to their standard fitting with all available electrodes activated. The other three subjects did not show significant improvements in performance when one or two of their most apical electrodes were deactivated. Four out of five subjects preferred to continue use of a fitting with one or more apical electrodes deactivated. The two subjects with the deepest insertions also showed pitch confusions between their most apical electrodes. Two possible reasons for these results are discussed. One is to reduce neural interactions related to electrodes producing pitch confusions. Another is to improve the alignment of the frequency components of sounds coded by the electrical signals delivered to each electrode to the overall pitch of the auditory perception produced by the electrical stimulation of auditory nerve fibers.
To investigate neural adaptive properties, near-field evoked potentials were recorded from a chronically implanted electrode in the ventral cochlear nucleus in awake Long-Evans rats exposed to acoustic stimuli or receiving intracochlear electric stimulation. Stimuli were 250-ms trains of repetitive acoustic clicks (10, 30 and 50 dB SPL) or biphasic electric pulses (30, 50 and 70 µA) with intratrain pulse rates ranging from 100 to 1000 pulses per second (pps). The amplitude of the first negative (N1) to positive (P1) component of the average evoked potentials was measured for each consecutive individual pulse in the train. While a progressive exponential decrease in N1–P1 amplitude was observed as a function of the position of the pulse within the train for both types of stimulation, the decrement of electric responses (adaptive pattern) was substantially less prominent than that observed for acoustic stimuli. Based on this difference, the present work was extended by modifying electric stimuli in order to try to restore normal adaptation phenomena. The results suggest the feasibility of mimicking acoustic adaptation by stimulation with exponentially decreasing electric pulse trains, which may be clinically applicable in the auditory implant field.
Abstrnct -Poor speech intelligibility in noise is a major source of dissatisfaction for users of both cochlear implants and conventional hearing aids. Many noise reduction schemes have been proposed so far. The most promising approaches assume that the target signal is emitted in front of the user, while signals from other directions are considered to be noise. These directional or beamforming systems can be realized either with directional microphones o r with microphone arrays and fixed or adaptive postprocessing. In this work, a novel combined fixedhdaptive beamforming noise reduction system with four head mounted microphones is proposed. Two microphones are mounted on either side of the head in a behind-the-ear hearing aid housing. Each of these pairs of microphones forms a fixed beamformer (Audio-Zoom) and the resulting outputs are then post-processed by an adaptive beamforming scheme. The system has been implemented in real time on a portable digital signal processor system. It was evaluated in a moderately reverberant room, and speech recognition tests with two normal hearing listeners were performed. Preliminary results demonstrate an improved directional pattern and significantly enhanced speech recognition in noise, corresponding to a signal-to-noise advantage of approximately 17 dB over a single omnidirectional microphone.
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