Enhancement of interaural level differences for bilateral cochlear implant users

A Fused Deep Denoising Sound Coding Strategy for Bilateral Cochlear Implants

Cochlear implants (CIs) provide a solution for individuals with severe sensorineural hearing loss to regain their hearing abilities. When someone experiences this form of hearing impairment in both ears, they may be equipped with two separate CI devices, which will typically further improve the CI benefits. This spatial hearing is particularly crucial when tackling the challenge of understanding speech in noisy environments, a common issue CI users face. Currently, extensive research is dedicated to developing algorithms that can autonomously filter out undesired background noises from desired speech signals. At present, some research focuses on achieving end-to-end denoising, either as an integral component of the initial CI signal processing or by fully integrating the denoising process into the CI sound coding strategy. This work is presented in the context of bilateral CI (BiCI) systems, where we propose a deep-learning-based bilateral speech enhancement model that shares information between both hearing sides. Specifically, we connect two monaural end-to-end deep denoising sound coding techniques through intermediary latent fusion layers. These layers amalgamate the latent representations generated by these techniques by multiplying them together, resulting in an enhanced ability to reduce noise and improve learning generalization. The objective instrumental results demonstrate that the proposed fused BiCI sound coding strategy achieves higher interaural coherence, superior noise reduction, and enhanced predicted speech intelligibility scores compared to the baseline methods. Furthermore, our speech-in-noise intelligibility results in BiCI users reveal that the deep denoising sound coding strategy can attain scores similar to those achieved in quiet conditions.

A Mixed-Rate Strategy on a Bilaterally-Synchronized Cochlear Implant Processor Offering the Opportunity to Provide Both Speech Understanding and Interaural Time Difference Cues

Background/Objective: Bilaterally implanted cochlear implant (CI) users do not consistently have access to interaural time differences (ITDs). ITDs are crucial for restoring the ability to localize sounds and understand speech in noisy environments. Lack of access to ITDs is partly due to lack of communication between clinical processors across the ears and partly because processors must use relatively high rates of stimulation to encode envelope information. Speech understanding is best at higher stimulation rates, but sensitivity to ITDs in the timing of pulses is best at low stimulation rates. Methods: We implemented a practical "mixed rate" strategy that encodes ITD information using a low stimulation rate on some channels and speech information using high rates on the remaining channels. The strategy was tested using a bilaterally synchronized research processor, the CCi-MOBILE. Nine bilaterally implanted CI users were tested on speech understanding and were asked to judge the location of a sound based on ITDs encoded using this strategy. Results: Performance was similar in both tasks between the control strategy and the new strategy. Conclusions: We discuss the benefits and drawbacks of the sound coding strategy and provide guidelines for utilizing synchronized processors for developing strategies.

A Compact Two-Loudspeaker Virtual Sound Reproduction System for Clinical Testing of Spatial Hearing With Hearing-Assistive Devices

Exciting developments in hearing aid and cochlear implant technology for linking signal processing across the ears have improved spatial hearing outcomes. This has resulted in an increased emphasis on clinical assessment of the spatial hearing abilities of hearing-assistive device users. Effective assessment of spatial hearing currently requires a large and costly loudspeaker array system, housed in a heavily acoustically treated testing room. This imposes economic and logistical constraints that limit proliferation of array systems, particularly in developing nations. Despite their size and cost, the ability of current clinical array systems to reproduce realistic spatial sound fields is limited, which substantially reduces the range of realistic acoustic scenes that can be used for diagnostic testing. We propose an alternative low-cost, compact virtual acoustics system with just two loudspeakers. This system uses crosstalk cancelation to reproduce pressure signals at the device microphones that match those for real-world sound sources. Furthermore, in contrast to clinical array systems, the system can adapt to different room acoustics, removing the requirement for a heavily acoustically treated testing environment. We conducted a proof-of-concept study in two stages: in the first, we evaluated the physical performance of the system for a stationary listener in anechoic conditions and in a small audiological testing booth with moderate acoustic treatment. To do this, a head and torso simulator was fitted with specially adapted hearing-assistive devices that allowed direct access to the microphone signals. These microphone signals were compared for real and virtual sound sources at numerous source locations. In the second stage, we quantified the system’s robustness to head rotations with and without the system adapting for head position. In the stationary case, the system was found to be highly effective at reproducing signals, such as speech, at all tested source locations. When head rotation was added, it performed well for rotations of up to 2°, even without adapting. However, performance improved markedly for larger rotations when the system adapted. These findings suggest that a compact, low-cost virtual acoustics system can give wider access to advanced and ecologically valid audiological testing, which could substantially improve clinical assessment of hearing-assistive device users.