The perception of the stereo effect in bilateral and bimodal cochlear implant users and its contribution to music enjoyment.

OBJECTIVES: In this clinical study, stereo perception of music samples and its contribution to music enjoyment in CI users is investigated. It is studied in free field as well as direct audio presentation. METHODS: 20 bilateral and 9 bimodal CI users performed stereo detection tests and music enjoyment ratings. Music was presented either in mono or in stereo in free field or with direct audio presentation. Stereo detection was assessed with a 3-AFC paradigm. Music enjoyment was studied with scale ratings. RESULTS: For bilateral CI users, stereo detection increased from 52% correct in free field to 86% with direct audio presentation. Increased music enjoyment with improved stereo detection was obtained. Bimodal CI users could not identify stereo sounds. Music enjoyment did not increase for stereo presentations in bimodal subjects. DISCUSSION: For bilateral CI users, improved stereo detection might increase music enjoyment with direct audio presentation, which is likely due to bypassing the room acoustics. In bimodal CI users, no clear improvement was found, which is likely attributed due to the different hearing losses and therefore individually different interaural frequency overlaps between the hearing aid and the cochlear implant. CONCLUSION: Direct audio presentation is an efficient method to improve music enjoyment in bilateral CI users.

Advanced beamformers for cochlear implant users: acute measurement of speech perception in challenging listening conditions.

OBJECTIVE: To investigate the performance of monaural and binaural beamforming technology with an additional noise reduction algorithm, in cochlear implant recipients. METHOD: This experimental study was conducted as a single subject repeated measures design within a large German cochlear implant centre. Twelve experienced users of an Advanced Bionics HiRes90K or CII implant with a Harmony speech processor were enrolled. The cochlear implant processor of each subject was connected to one of two bilaterally placed state-of-the-art hearing aids (Phonak Ambra) providing three alternative directional processing options: an omnidirectional setting, an adaptive monaural beamformer, and a binaural beamformer. A further noise reduction algorithm (ClearVoice) was applied to the signal on the cochlear implant processor itself. The speech signal was presented from 0° and speech shaped noise presented from loudspeakers placed at ±70°, ±135° and 180°. The Oldenburg sentence test was used to determine the signal-to-noise ratio at which subjects scored 50% correct. RESULTS: Both the adaptive and binaural beamformer were significantly better than the omnidirectional condition (5.3 dB±1.2 dB and 7.1 dB±1.6 dB (p<0.001) respectively). The best score was achieved with the binaural beamformer in combination with the ClearVoice noise reduction algorithm, with a significant improvement in SRT of 7.9 dB±2.4 dB (p<0.001) over the omnidirectional alone condition. CONCLUSIONS: The study showed that the binaural beamformer implemented in the Phonak Ambra hearing aid could be used in conjunction with a Harmony speech processor to produce substantial average improvements in SRT of 7.1 dB. The monaural, adaptive beamformer provided an averaged SRT improvement of 5.3 dB.

Factors affecting predicted speech intelligibility with cochlear implants in an auditory model for electrical stimulation.

A model of the auditory response to stimulation with cochlear implants (CIs) was used to predict speech intelligibility in electric hearing. The model consists of an auditory nerve cell population that generates delta pulses as action potentials in response to temporal and spatial excitation with a simulated CI signal processing strategy. The auditory nerve cells are modeled with a leaky integrate-and-fire model with membrane noise. Refractory behavior is introduced by raising the threshold potential with an exponentially decreasing function. Furthermore, the action potentials are delayed to account for latency and jitter. The action potentials are further processed by a central model stage, which includes spatial and temporal integration, resulting in an internal representation of the sound presented. Multiplicative noise is included in the internal representations to limit resolution. Internal representations of complete word sets for a sentence intelligibility test were computed and classified using a Dynamic-Time-Warping classifier to quantify information content and to estimate speech intelligibility. The number of auditory nerve cells, the spatial spread of the electrodes' electric field, and the internal noise intensity were found to have a major impact on the modeled speech intelligibility, whereas the influence of refractory behavior, membrane noise, and latency and jitter was minor.

Measurement and prediction of the acceptable noise level for single-microphone noise reduction algorithms.

OBJECTIVE: To measure the acceptable noise level (ANL) with and without noise reduction algorithms (NRAs), and to predict ΔANL, i.e. the difference in acceptable noise level with and without NRAs. DESIGN: The ANL test was applied to three NRAs. Furthermore, the measured ΔANL was predicted using several methods based on either the calculation of the signal-to-noise ratio or correlation methods of the processed signals with an unprocessed reference signal. STUDY SAMPLE: Ten normal-hearing and eleven hearing-impaired subjects accomplished the ANL test. RESULTS: In general, the ANL test could determine an increased acceptance of noise with some NRAs. However, great inter-individual differences also resulted that were attributed to audible distortions when an NRA was used. Prediction of the mean measured DANL was possible, but individual prediction of DANL failed due to inter-individual differences. Mean DANL was predicted more accurately for hearing-impaired subjects when individual hearing loss was taken into account. CONCLUSIONS: The ANL test is a suitable tool for measuring the advantage of one NRA. A prediction of the measured individual ΔANL failed. However, mean DANL could be predicted with some methods. Furthermore, the individual hearing loss should be taken into account for a more accurate prediction for hearing-impaired subjects.

Development and analysis of an International Speech Test Signal (ISTS).

For analysing the processing of speech by a hearing instrument, a standard test signal is necessary which allows for reproducible measurement conditions, and which features as many of the most relevant properties of natural speech as possible, e.g. the average speech spectrum, the modulation spectrum, the variation of the fundamental frequency together with its appropriate harmonics, and the comodulation in different frequency bands. Existing artificial signals do not adequately fulfill these requirements. Moreover, recordings from natural speakers represent only one language and are therefore not internationally acceptable. For this reason, an International Speech Test Signal (ISTS) was developed. It is based on natural recordings but is largely non-intelligible because of segmentation and remixing. When using the signal for hearing aid measurements, the gain of a device can be described at different percentiles of the speech level distribution. The primary intention is to include this test signal with a new measurement method for a new hearing aid standard (IEC 60118-15).