Further simulations of the effect of cochlear-implant pre-processing and head movement on interaural level differences

We simulated the effect of several automatic gain control (AGC) and AGC-like systems and head movement on the output levels, and resulting interaural level differences (ILDs) produced by bilateral cochlear-implant (CI) processors. The simulated AGC systems included unlinked AGCs with a range of parameter settings, linked AGCs, and two proprietary multi-channel systems used in contemporary CIs. The results show that over the range of values used clinically, the parameters that most strongly affect dynamic ILDs are the release time and compression ratio. Linking AGCs preserves ILDs at the expense of monaural level changes and, possibly, comfortable listening level. Multichannel AGCs can whiten output spectra, and/or distort the dynamic changes in ILD that occur during and after head movement. We propose that an unlinked compressor with a ratio of approximately 3:1 and a release time of 300-500 ms can preserve the shape of dynamic ILDs, without causing large spectral distortions or sacrificing listening comfort

Multifaceted evaluation of a binaural cochlear‐ implant sound‐processing strategy inspired by the medial olivocochlear reflex

[ES]El objetivo de esta tesis es evaluar experimentalmente la audición de los usuarios de implantes cocleares con una estrategia de procesamiento binaural de sonidos inspirada en el reflejo olivococlear medial, denominada "estrategia MOC". La tesis describe cuatro estudios dirigidos a comparar la inteligibilidad del habla en ruido, la localización de fuentes sonoras y el esfuerzo auditivo con procesadores de sonido estándar y con diversos procesadores MOC diseñados para reflejar de forma más o menos realista el tiempo de activación del reflejo olivococlear medial natural y sus efectos sobre la comprensión coclear humana

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

Gain optimization for cochlear implant systems

Cochlear implant systems need Automatic Gain Control (AGC) to compress the large dynamic range (~120 dB) of the acoustic environment into the small dynamic range (< 20 dB) of electrical stimulation. This thesis is concerned with the design, implementation and evaluation of AGC systems for cochlear implants. It investigated the effects of AGC on the speech intelligibility of cochlear implant recipients. Various AGC configurations were evaluated with sentences presented over a wide range of levels at different Signal-to-Noise Ratios (SNR) to identify important factors affecting the performance. Signal metrics were developed to quantify the effects of AGC on the channel envelopes. The goal was to improve speech intelligibility in adverse listening conditions. The performance-intensity functions of cochlear implant recipients with no AGC and with a front-end compression limiter were measured in noise. With no AGC, the proportion of envelope clipping grew monotonically with presentation level. The front-end limiter substantially reduced envelope clipping yet gave little improvement in speech intelligibility. The recipients were highly tolerant of envelope clipping when the background noise was low. SNR degradation was identified as the main factor reducing speech intelligibility. A front-end limiter cannot guarantee zero envelope clipping. In contrast, the proposed envelope profile limiter eliminated envelope clipping and hence preserved the spectral profile. The two AGCs were evaluated, with two release times (75 and 625 ms). The shorter release time gave worse speech intelligibility because it caused more waveform distortion and output SNR reduction. For a given release time, preserving spectral envelope profile gave additional benefits. In a take-home experiment, cochlear implant recipients rated a program with the envelope profile limiter equivalent to their everyday program. A conventional cochlear implant signal path uses a predetermined input dynamic range, which is shifted up or down by the AGC. In contrast, the proposed Adaptive Loudness Growth Function (ALGF) continually optimized the input dynamic range by estimating the noise floor and peak level in each channel. The ALGF gave better Speech Reception Threshold (SRT) than the existing state-of-the-art AGC system at the high presentation level when evaluated with a newly developed roving-level SRT test at three presentation levels

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

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

Listening to Music Through Hearing Aids: Potential Lessons for Cochlear Implants

Some of the problems experienced by users of hearing aids (HAs) when listening to music are relevant to cochlear implants (CIs). One problem is related to the high peak levels (up to 120 dB SPL) that occur in live music. Some HAs and CIs overload at such levels, because of the limited dynamic range of the microphones and analogue-to-digital converters (ADCs), leading to perceived distortion. Potential solutions are to use 24-bit ADCs or to include an adjustable gain between the microphones and the ADCs. A related problem is how to squeeze the wide dynamic range of music into the limited dynamic range of the user, which can be only 6–20 dB for CI users. In HAs, this is usually done via multi-channel amplitude compression (automatic gain control, AGC). In CIs, a single-channel front-end AGC is applied to the broadband input signal or a control signal derived from a running average of the broadband signal level is used to control the mapping of the channel envelope magnitude to an electrical signal. This introduces several problems: (1) an intense narrowband signal (e.g. a strong bass sound) reduces the level for all frequency components, making some parts of the music harder to hear; (2) the AGC introduces cross-modulation effects that can make a steady sound (e.g. sustained strings or a sung note) appear to fluctuate in level. Potential solutions are to use several frequency channels to create slowly varying gain-control signals and to use slow-acting (or dual time-constant) AGC rather than fast-acting AGC.</jats:p

Personalizing transient noise reduction algorithm settings for cochlear implant users

Objectives: Speech understanding in noise is difficult for patients with a cochlear implant. One common and disruptive type of noise is transient noise. We have tested transient noise reduction (TNR) algorithms in cochlear implant users to investigate the merits of personalizing the noise reduction settings based on a subject's own preference. Design: The effect of personalizing two parameters of a broadband and a multiband TNR algorithm (TNRbb and TNRmb, respectively) on speech recognition was tested in a group of 15 unilaterally implanted subjects in cafeteria noise. The noise consisted of a combination of clattering dishes and babble noise. Each participant could individually vary two parameters, namely the scaling factor of the attenuation and the release time (tau). The parameter tau represents the duration of the attenuation applied after a transient is detected. As a reference, the current clinical standard TNR "SoundRelax" from Advanced Bionics was tested (TNRbb-std). Effectiveness of the algorithms on speech recognition was evaluated adaptively by determining the speech reception threshold (SRT). Possible subjective benefits of the algorithms were assessed using a rating task at a fixed signal-to-noise ratio (SNR) of SRT + 3 dB. Rating was performed on four items, namely speech intelligibility, speech naturalness, listening effort, and annoyance of the noise. Word correct scores were determined at these fixed speech levels as well. Results: The personalized TNRmb improved the SRT statistically significantly with 1.3 dB, while the personalized TNRbb degraded it significantly by 1.7 dB. For TNRmb, we attempted to further optimize its settings by determining a group-based setting, leaving out those subjects that did not experience a benefit from it. Using these group-based settings, however, TNRmb did not have a significant effect on the SRT any longer. TNRbb-std did not affect speech recognition significantly. No significant effects on subjective ratings were found for any of the items investigated. In addition, at a constant speech level of SRT + 3 dB, no effect of any of the algorithms was found on word correct scores, including TNRmb with personalized settings. Conclusions: Our study results indicate that personalizing noise reduction settings of a multiband TNR algorithm can significantly improve speech intelligibility in transient noise, but only under challenging listening conditions around the SRT. At more favorable SNRs (SRT + 3 dB), this benefit was lost. We hypothesize that TNRmb was beneficial at lower SNRs, because of more effective artifact detection under those conditions. Group-averaged settings of the multiband algorithm did not significantly affect speech recognition. TNRbb decreased speech recognition significantly using personalized parameter settings. Rating scores were not significantly affected by the algorithms under any condition tested. The currently available TNR algorithm for Advanced Bionics systems (SoundRelax) is a broadband filter that does not support personalization of its settings. Future iterations of this algorithm might benefit from upgrading it to a multiband variant with the option to personalize its parameter settings.Disorders of the head and nec

Effects of sound-induced hearing loss and hearing AIDS on the perception of music

This is the final version of the article. It first appeared from the Audio Engineering Society via https://doi.org/10.17743/jaes.2015.0081Exposure to high-level music produces several physiological changes in the auditory system that lead to a variety of perceptual effects. Damage to the outer hair cells within the cochlea leads to a loss of sensitivity to weak sounds, loudness recruitment (a more rapid than normal growth of loudness with increasing sound level) and reduced frequency selectivity. Damage to inner hair cells and/or synapses leads to degeneration of neurons in the auditory nerve and to a reduced flow of information to the brain. This leads to poorer auditory discrimination and may contribute to reduced sensitivity to the temporal fine structure of sounds and to poor pitch perception. Hearing aids compensate for the effects of threshold elevation and loudness recruitment via multi-channel amplitude compression, but they do not compensate for reduced frequency selectivity or loss of inner hair cells/synapses/neurons. Multi-channel compression can impair some aspects of the perception of music, such as the ability to hear out one instrument or voice from a mixture. The limited frequency range and irregular frequency response of most hearing aids is associated with poor sound quality for music. Finally, systems for reducing acoustic feedback can have undesirable side effects when listening to music.This work was supported by the Medical Research Council (UK, grant number G0701870), Action on Hearing Loss, Phonak, and Starkey

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