Improvement of Speech Perception for Hearing-Impaired Listeners

Hearing impairment is becoming a prevalent health problem affecting 5% of world adult populations. Hearing aids and cochlear implant already play an essential role in helping patients over decades, but there are still several open problems that prevent them from providing the maximum benefits. Financial and discomfort reasons lead to only one of four patients choose to use hearing aids; Cochlear implant users always have trouble in understanding speech in a noisy environment. In this dissertation, we addressed the hearing aids limitations by proposing a new hearing aid signal processing system named Open-source Self-fitting Hearing Aids System (OS SF hearing aids). The proposed hearing aids system adopted the state-of-art digital signal processing technologies, combined with accurate hearing assessment and machine learning based self-fitting algorithm to further improve the speech perception and comfort for hearing aids users. Informal testing with hearing-impaired listeners showed that the testing results from the proposed system had less than 10 dB (by average) difference when compared with those results obtained from clinical audiometer. In addition, Sixteen-channel filter banks with adaptive differential microphone array provides up to six-dB SNR improvement in the noisy environment. Machine-learning based self-fitting algorithm provides more suitable hearing aids settings. To maximize cochlear implant users’ speech understanding in noise, the sequential (S) and parallel (P) coding strategies were proposed by integrating high-rate desynchronized pulse trains (DPT) in the continuous interleaved sampling (CIS) strategy. Ten participants with severe hearing loss participated in the two rounds cochlear implants testing. The testing results showed CIS-DPT-S strategy significantly improved (11%) the speech perception in background noise, while the CIS-DPT-P strategy had a significant improvement in both quiet (7%) and noisy (9%) environment

Coding Strategies for Cochlear Implants Under Adverse Environments

Cochlear implants are electronic prosthetic devices that restores partial hearing in patients with severe to profound hearing loss. Although most coding strategies have significantly improved the perception of speech in quite listening conditions, there remains limitations on speech perception under adverse environments such as in background noise, reverberation and band-limited channels, and we propose strategies that improve the intelligibility of speech transmitted over the telephone networks, reverberated speech and speech in the presence of background noise. For telephone processed speech, we propose to examine the effects of adding low-frequency and high- frequency information to the band-limited telephone speech. Four listening conditions were designed to simulate the receiving frequency characteristics of telephone handsets. Results indicated improvement in cochlear implant and bimodal listening when telephone speech was augmented with high frequency information and therefore this study provides support for design of algorithms to extend the bandwidth towards higher frequencies. The results also indicated added benefit from hearing aids for bimodal listeners in all four types of listening conditions. Speech understanding in acoustically reverberant environments is always a difficult task for hearing impaired listeners. Reverberated sounds consists of direct sound, early reflections and late reflections. Late reflections are known to be detrimental to speech intelligibility. In this study, we propose a reverberation suppression strategy based on spectral subtraction to suppress the reverberant energies from late reflections. Results from listening tests for two reverberant conditions (RT60 = 0.3s and 1.0s) indicated significant improvement when stimuli was processed with SS strategy. The proposed strategy operates with little to no prior information on the signal and the room characteristics and therefore, can potentially be implemented in real-time CI speech processors. For speech in background noise, we propose a mechanism underlying the contribution of harmonics to the benefit of electroacoustic stimulations in cochlear implants. The proposed strategy is based on harmonic modeling and uses synthesis driven approach to synthesize the harmonics in voiced segments of speech. Based on objective measures, results indicated improvement in speech quality. This study warrants further work into development of algorithms to regenerate harmonics of voiced segments in the presence of noise

Koklear İmplant Konuşma İşlemcileri için Optimum Parametrelerin Objektif Ölçütler Kullanılarak Belirlenmesi

In a cochlear implant (CI) speech processor, several parameters such as channel numbers, bandwidths, rectification type, and cutoff frequency play an important role in acquiring enhanced speech. The effective and general purpose CI approach has been a research topic for a long time. In this study, it is aimed to determine the optimum parameters for CI users by using different channel numbers (4, 8, 12, 16, and 22), rectification types (half and full) and cutoff frequencies (200, 250, 300, 350, and 400 Hz). The CI approaches have been tested on Turkish sentences which are taken from METU database. The optimum CI structure has been tested with objective quality that weighted spectral slope (WSS) and objective intelligibility measures such as short-term objective intelligibility (STOI) and perceptual evaluation of speech quality (PESQ). Experimental results show that 400 Hz cutoff frequency, full wave rectifier, and 16-channels CI approach give better quality and higher intelligibility scores than other CI approaches according to STOI, PESQ and WSS results. The proposed CI approach provides the ability to percept 91% of output vocoded Turkish speech for CI users. © 2022, TUBITAK. All rights reserved

A psychoacoustic "NofM"-type speech coding strategy for cochlear implants

We describe a new signal processing technique for cochlear implants using a psychoacoustic-masking model. The technique is based on the principle of a so-called "NofM" strategy. These strategies stimulate fewer channels (N) per cycle than active electrodes (NofM; N < M). In "NofM" strategies such as ACE or SPEAK, only the N channels with higher amplitudes are stimulated. The new strategy is based on the ACE strategy but uses a psychoacoustic-masking model in order to determine the essential components of any given audio signal. This new strategy was tested on device users in an acute Study, with either 4 or 8 channels stimulated per cycle. For the first condition (4 channels), the mean improvement over the ACE strategy was 17%. For the second condition (8 channels), no significant difference was found between the two strategies

Improving the Speech Intelligibility By Cochlear Implant Users

In this thesis, we focus on improving the intelligibility of speech for cochlear implants (CI) users. As an auditory prosthetic device, CI can restore hearing sensations for most patients with profound hearing loss in both ears in a quiet background. However, CI users still have serious problems in understanding speech in noisy and reverberant environments. Also, bandwidth limitation, missing temporal fine structures, and reduced spectral resolution due to a limited number of electrodes are other factors that raise the difficulty of hearing in noisy conditions for CI users, regardless of the type of noise. To mitigate these difficulties for CI listener, we investigate several contributing factors such as the effects of low harmonics on tone identification in natural and vocoded speech, the contribution of matched envelope dynamic range to the binaural benefits and contribution of low-frequency harmonics to tone identification in quiet and six-talker babble background. These results revealed several promising methods for improving speech intelligibility for CI patients. In addition, we investigate the benefits of voice conversion in improving speech intelligibility for CI users, which was motivated by an earlier study showing that familiarity with a talker’s voice can improve understanding of the conversation. Research has shown that when adults are familiar with someone’s voice, they can more accurately – and even more quickly – process and understand what the person is saying. This theory identified as the “familiar talker advantage” was our motivation to examine its effect on CI patients using voice conversion technique. In the present research, we propose a new method based on multi-channel voice conversion to improve the intelligibility of transformed speeches for CI patients

InterlACE Sound Coding for Unilateral and Bilateral Cochlear Implants

Objective: Cochlear implant signal processing strategies define the rules of how acoustic signals are converted into electrical stimulation patterns. Technological and anatomical limitations, however, impose constraints on the signal transmission and the accurate excitation of the auditory nerve. Acoustic signals are degraded throughout cochlear implant processing, and electrical signal interactions at the electrode-neuron interface constrain spectral and temporal precision. In this work, we propose a novel InterlACE signal processing strategy to counteract the occurring limitations. Methods: By replacing the maxima selection of the Advanced Combination Encoder strategy with a method that defines spatially and temporally alternating channels, InterlACE can compensate for discarded signal content of the conventional processing. The strategy can be extended bilaterally by introducing synchronized timing and channel selection. InterlACE was explored unilaterally and bilaterally by assessing speech intelligibility and spectral resolution. Five experienced bilaterally implanted cochlear implant recipients participated in the Oldenburg Sentence Recognition Test in background noise and the spectral ripple discrimination task. Results: The introduced alternating channel selection methodology shows promising outcomes for speech intelligibility but could not indicate better spectral ripple discrimination. Conclusion: InterlACE processing positively affects speech intelligibility, increases available unilateral and bilateral signal content, and may potentially counteract signal interactions at the electrode-neuron interface. Significance: This work shows how cochlear implant channel selection can be modified and extended bilaterally. The clinical impact of the modifications needs to be explored with a larger sample size

Cochlear Implant Research and Development in the Twenty-first Century: A Critical Update

Abstract: Cochlear implants (CIs) are the world’s most successful sensory prosthesis and have been the subject of intense research and development in recent decades. We critically review the progress in CI research, and its success in improving patient outcomes, from the turn of the century to the present day. The review focuses on the processing, stimulation, and audiological methods that have been used to try to improve speech perception by human CI listeners, and on fundamental new insights in the response of the auditory system to electrical stimulation. The introduction of directional microphones and of new noise reduction and pre-processing algorithms has produced robust and sometimes substantial improvements. Novel speech-processing algorithms, the use of current-focusing methods, and individualised (patient-by-patient) deactivation of subsets of electrodes have produced more modest improvements. We argue that incremental advances have and will continue to be made, that collectively these may substantially improve patient outcomes, but that the modest size of each individual advance will require greater attention to experimental design and power. We also briefly discuss the potential and limitations of promising technologies that are currently being developed in animal models, and suggest strategies for researchers to collectively maximise the potential of CIs to improve hearing in a wide range of listening situations

Electrical Stimulation of the Auditory System

In many healthcare systems electrical stimulation of the human auditory system, using cochlear implants, is a common treatment for severe to profound deafness. This chapter will describe how electrical stimulation manages to compensate for sensory-neural hearing loss by bypassing the damaged cochlea. The challenges involved in the design and application of cochlear implants will be outlined, including the programming of clinical systems to suit the needs of implanted patients. Today’s variety of patient will be reviewed: unilaterally and bilaterally implanted, bimodal users of a cochlear implant as well as a contralateral hearing aid, CROS device users having either asymmetrical hearing loss or single-sided deafness. Alternative devices such as auditory brainstem implants will be described, and additionally the more experimental auditory mid-brain implants and intraneural stimulation approaches. Research that is likely to bring medium term benefits to the clinical application of cochlear implants will also be described

The effect of a coding strategy that removes temporally masked pulses on speech perception by cochlear implant users.

Speech recognition in noisy environments remains a challenge for cochlear implant (CI) recipients. Unwanted charge interactions between current pulses, both within and between electrode channels, are likely to impair performance. Here we investigate the effect of reducing the number of current pulses on speech perception. This was achieved by implementing a psychoacoustic temporal-masking model where current pulses in each channel were passed through a temporal integrator to identify and remove pulses that were less likely to be perceived by the recipient. The decision criterion of the temporal integrator was varied to control the percentage of pulses removed in each condition. In experiment 1, speech in quiet was processed with a standard Continuous Interleaved Sampling (CIS) strategy and with 25, 50 and 75% of pulses removed. In experiment 2, performance was measured for speech in noise with the CIS reference and with 50 and 75% of pulses removed. Speech intelligibility in quiet revealed no significant difference between reference and test conditions. For speech in noise, results showed a significant improvement of 2.4 dB when removing 50% of pulses and performance was not significantly different between the reference and when 75% of pulses were removed. Further, by reducing the overall amount of current pulses by 25, 50, and 75% but accounting for the increase in charge necessary to compensate for the decrease in loudness, estimated average power savings of 21.15, 40.95, and 63.45%, respectively, could be possible for this set of listeners. In conclusion, removing temporally masked pulses may improve speech perception in noise and result in substantial power savings