The ability of cochlear implant users to use temporal envelope cues recovered from speech frequency modulation.
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ABSTRACT:
SUBMITTER: Won JH
PROVIDER: S-EPMC3427369 | biostudies-other | 2012 Aug
REPOSITORIES: biostudies-other
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Project description:RationalePrevious cochlear implant (CI) studies have shown that single-channel amplitude modulation frequency discrimination (AMFD) can be improved when coherent modulation is delivered to additional channels. It is unclear whether the multi-channel advantage is due to increased loudness, multiple envelope representations, or to component channels with better temporal processing. Measuring envelope interference may shed light on how modulated channels can be combined.MethodsIn this study, multi-channel AMFD was measured in CI subjects using a 3-alternative forced-choice, non-adaptive procedure ("which interval is different?"). For the reference stimulus, the reference AM (100 Hz) was delivered to all 3 channels. For the probe stimulus, the target AM (101, 102, 104, 108, 116, 132, 164, 228, or 256 Hz) was delivered to 1 of 3 channels, and the reference AM (100 Hz) delivered to the other 2 channels. The spacing between electrodes was varied to be wide or narrow to test different degrees of channel interaction.ResultsResults showed that CI subjects were highly sensitive to interactions between the reference and target envelopes. However, performance was non-monotonic as a function of target AM frequency. For the wide spacing, there was significantly less envelope interaction when the target AM was delivered to the basal channel. For the narrow spacing, there was no effect of target AM channel. The present data were also compared to a related previous study in which the target AM was delivered to a single channel or to all 3 channels. AMFD was much better with multiple than with single channels whether the target AM was delivered to 1 of 3 or to all 3 channels. For very small differences between the reference and target AM frequencies (2-4 Hz), there was often greater sensitivity when the target AM was delivered to 1 of 3 channels versus all 3 channels, especially for narrowly spaced electrodes.ConclusionsBesides the increased loudness, the present results also suggest that multiple envelope representations may contribute to the multi-channel advantage observed in previous AMFD studies. The different patterns of results for the wide and narrow spacing suggest a peripheral contribution to multi-channel temporal processing. Because the effect of target AM frequency was non-monotonic in this study, adaptive procedures may not be suitable to measure AMFD thresholds with interfering envelopes. Envelope interactions among multiple channels may be quite complex, depending on the envelope information presented to each channel and the relative independence of the stimulated channels.
| S-EPMC4592255 | biostudies-literature
Project description:For the perception of timbre of a musical instrument, the attack time is known to hold crucial information. The first 50 to 150 ms of sound onset reflect the excitation mechanism, which generates the sound. Since auditory processing and music perception in particular are known to be hampered in cochlear implant (CI) users, we conducted an electroencephalography (EEG) study with an oddball paradigm to evaluate the processing of small differences in musical sound onset. The first 60 ms of a cornet sound were manipulated in order to examine whether these differences are detected by CI users and normal-hearing controls (NH controls), as revealed by auditory evoked potentials (AEPs). Our analysis focused on the N1 as an exogenous component known to reflect physical stimuli properties as well as on the P2 and the Mismatch Negativity (MMN). Our results revealed different N1 latencies as well as P2 amplitudes and latencies for the onset manipulations in both groups. An MMN could be elicited only in the NH control group. Together with additional findings that suggest an impact of musical training on CI users' AEPs, our findings support the view that impaired timbre perception in CI users is at partly due to altered sound onset feature detection.
| S-EPMC3448664 | biostudies-literature
Project description:Spectrotemporal modulation (STM) detection performance was examined for cochlear implant (CI) users. The test involved discriminating between an unmodulated steady noise and a modulated stimulus. The modulated stimulus presents frequency modulation patterns that change in frequency over time. In order to examine STM detection performance for different modulation conditions, two different temporal modulation rates (5 and 10 Hz) and three different spectral modulation densities (0.5, 1.0, and 2.0 cycles/octave) were employed, producing a total 6 different STM stimulus conditions. In order to explore how electric hearing constrains STM sensitivity for CI users differently from acoustic hearing, normal-hearing (NH) and hearing-impaired (HI) listeners were also tested on the same tasks. STM detection performance was best in NH subjects, followed by HI subjects. On average, CI subjects showed poorest performance, but some CI subjects showed high levels of STM detection performance that was comparable to acoustic hearing. Significant correlations were found between STM detection performance and speech identification performance in quiet and in noise. In order to understand the relative contribution of spectral and temporal modulation cues to speech perception abilities for CI users, spectral and temporal modulation detection was performed separately and related to STM detection and speech perception performance. The results suggest that that slow spectral modulation rather than slow temporal modulation may be important for determining speech perception capabilities for CI users. Lastly, test-retest reliability for STM detection was good with no learning. The present study demonstrates that STM detection may be a useful tool to evaluate the ability of CI sound processing strategies to deliver clinically pertinent acoustic modulation information.
| S-EPMC4617902 | biostudies-literature
Project description:BackgroundOne factor which influences the speech intelligibility of cochlear implant (CI) users is the number and the extent of the functionality of spiral ganglion neurons (SGNs), referred to as "cochlear health." To explain the interindividual variability in speech perception of CI users, a clinically applicable estimate of cochlear health could be insightful. The change in the slope of the electrically evoked compound action potentials (eCAP), amplitude growth function (AGF) as a response to increased interphase gap (IPG) (IPGEslope) has been introduced as a potential measure of cochlear health. Although this measure has been widely used in research, its relationship to other parameters requires further investigation.MethodsThis study investigated the relationship between IPGEslope, demographics and speech intelligibility by (1) considering the relative importance of each frequency band to speech perception, and (2) investigating the effect of the stimulus polarity of the stimulating pulse. The eCAPs were measured in three different conditions: (1) Forward masking with anodic-leading (FMA) pulse, (2) Forward masking with cathodic-leading (FMC) pulse, and (3) with alternating polarity (AP). This allowed the investigation of the effect of polarity on the diagnosis of cochlear health. For an accurate investigation of the correlation between IPGEslope and speech intelligibility, a weighting function was applied to the measured IPGEslopes on each electrode in the array to consider the relative importance of each frequency band for speech perception. A weighted Pearson correlation analysis was also applied to compensate for the effect of missing data by giving higher weights to the ears with more successful IPGEslope measurements.ResultsA significant correlation was observed between IPGEslope and speech perception in both quiet and noise for between-subject data especially when the relative importance of frequency bands was considered. A strong and significant correlation was also observed between IPGEslope and age when stimulation was performed with cathodic-leading pulses but not for the anodic-leading pulse condition.ConclusionBased on the outcome of this study it can be concluded that IPGEslope has potential as a relevant clinical measure indicative of cochlear health and its relationship to speech intelligibility. The polarity of the stimulating pulse could influence the diagnostic potential of IPGEslope.
| S-EPMC10213548 | biostudies-literature
Project description:Purpose Although the speech intelligibility index (SII) has been widely applied in the field of audiology and other related areas, application of this metric to cochlear implants (CIs) has yet to be investigated. In this study, SIIs for CI users were calculated to investigate whether the SII could be an effective tool for predicting speech perception performance in a population with CI. Method Fifteen pre- and postlingually deafened adults with CI participated. Speech recognition scores were measured using the AzBio sentence lists. CI users also completed questionnaires and performed psychoacoustic (spectral and temporal resolution) and cognitive function (digit span) tests. Obtained SIIs were compared with predicted SIIs using a transfer function curve. Correlation and regression analyses were conducted on perceptual and demographic predictor variables to investigate the association between these factors and speech perception performance. Result Because of the considerably poor hearing and large individual variability in performance, the SII did not predict speech performance for this CI group using the traditional calculation. However, new SII models were developed incorporating predictive factors, which improved the accuracy of SII predictions in listeners with CI. Conclusion Conventional SII models are not appropriate for predicting speech perception scores for CI users. Demographic variables (aided audibility and duration of deafness) and perceptual-cognitive skills (gap detection and auditory digit span outcomes) are needed to improve the use of the SII for listeners with CI. Future studies are needed to improve our CI-corrected SII model by considering additional predictive factors. Supplemental Material https://doi.org/10.23641/asha.8057003.
| S-EPMC6808321 | biostudies-literature
Project description:Speech understanding in noisy environments is still one of the major challenges for cochlear implant (CI) users in everyday life. We evaluated a speech enhancement algorithm based on neural networks (NNSE) for improving speech intelligibility in noise for CI users. The algorithm decomposes the noisy speech signal into time-frequency units, extracts a set of auditory-inspired features and feeds them to the neural network to produce an estimation of which frequency channels contain more perceptually important information (higher signal-to-noise ratio, SNR). This estimate is used to attenuate noise-dominated and retain speech-dominated CI channels for electrical stimulation, as in traditional n-of-m CI coding strategies. The proposed algorithm was evaluated by measuring the speech-in-noise performance of 14 CI users using three types of background noise. Two NNSE algorithms were compared: a speaker-dependent algorithm, that was trained on the target speaker used for testing, and a speaker-independent algorithm, that was trained on different speakers. Significant improvements in the intelligibility of speech in stationary and fluctuating noises were found relative to the unprocessed condition for the speaker-dependent algorithm in all noise types and for the speaker-independent algorithm in 2 out of 3 noise types. The NNSE algorithms used noise-specific neural networks that generalized to novel segments of the same noise type and worked over a range of SNRs. The proposed algorithm has the potential to improve the intelligibility of speech in noise for CI users while meeting the requirements of low computational complexity and processing delay for application in CI devices.
| S-EPMC5256482 | biostudies-literature
Project description:Speech-perception testing is essential for monitoring outcomes with a hearing aid or cochlear implant (CI). However, clinical care is time-consuming and often challenging with an increasing number of clients. A potential approach to alleviating some clinical care and possibly making room for other outcome measures is to employ technologies that assess performance in the home environment. In this study, we investigate 3 different speech perception indices in the same 40 CI users: phoneme identification (vowels and consonants), digits in noise (DiN) and sentence recognition in noise (SiN). The first two tasks were implemented on a tablet and performed multiple times by each client in their home environment, while the sentence task was administered at the clinic. Speech perception outcomes in the same forty CI users showed that DiN assessed at home can serve as an alternative to SiN assessed at the clinic. DiN scores are in line with the SiN ones by 3-4 dB improvement and are useful to monitor performance at regular intervals and to detect changes in auditory performance. Phoneme identification in quiet also explains a significant part of speech perception in noise, and provides additional information on the detectability and discriminability of speech cues. The added benefit of the phoneme identification task, which also proved to be easy to administer at home, is the information transmission analysis in addition to the summary score. Performance changes for the different indices can be interpreted by comparing against measurement error and help to target personalized rehabilitation. Altogether, home-based speech testing is reliable and proves powerful to complement care in the clinic for CI users.
| S-EPMC8669965 | biostudies-literature
Project description:The upper limit of rate-based pitch perception and rate discrimination can differ substantially across cochlear implant (CI) users. One potential reason for this difference is the presence of a biological limitation on temporal encoding in the electrically-stimulated auditory pathway, which can be inherent to the electrical stimulation itself and/or to the degenerative processes associated with hearing loss. Electrophysiological measures, like the electrically-evoked frequency following response (eFFR) and auditory change complex (eACC), could potentially provide valuable insights in the temporal processing limitations at the level of the brainstem and cortex in the electrically-stimulated auditory pathway. Obtaining these neural responses, free from stimulation artifacts, is challenging, especially when the neural response is phase-locked to the stimulation rate, as is the case for the eFFR. In this study we investigated the feasibility of measuring eFFRs, free from stimulation artifacts, to stimulation rates ranging from 94 to 196 pulses per second (pps) and eACCs to pulse rate changes ranging from 36 to 108%, when stimulating in a monopolar configuration. A high-sampling rate EEG system was used to measure the electrophysiological responses in five CI users, and linear interpolation was applied to remove the stimulation artifacts from the EEG. With this approach, we were able to measure eFFRs for pulse rates up to 162 pps and eACCs to the different rate changes. Our results show that it is feasible to measure electrophysiological responses, free from stimulation artifacts, that could potentially be used as neural correlates for rate and pitch processing in CI users.
| S-EPMC8052190 | biostudies-literature
Project description:ObjectivesPrevious studies investigating speech perception in noise have typically been conducted with static masker positions. The aim of this study was to investigate the effect of spatial separation of source and masker (spatial release from masking, SRM) in a moving masker setup and to evaluate the impact of adaptive beamforming in comparison with fixed directional microphones in cochlear implant (CI) users.DesignSpeech reception thresholds (SRT) were measured in S0N0 and in a moving masker setup (S0Nmove) in 12 normal hearing participants and 14 CI users (7 subjects bilateral, 7 bimodal with a hearing aid in the contralateral ear). Speech processor settings were a moderately directional microphone, a fixed beamformer, or an adaptive beamformer. The moving noise source was generated by means of wave field synthesis and was smoothly moved in a shape of a half-circle from one ear to the contralateral ear. Noise was presented in either of two conditions: continuous or modulated.ResultsSRTs in the S0Nmove setup were significantly improved compared to the S0N0 setup for both the normal hearing control group and the bilateral group in continuous noise, and for the control group in modulated noise. There was no effect of subject group. A significant effect of directional sensitivity was found in the S0Nmove setup. In the bilateral group, the adaptive beamformer achieved lower SRTs than the fixed beamformer setting. Adaptive beamforming improved SRT in both CI user groups substantially by about 3 dB (bimodal group) and 8 dB (bilateral group) depending on masker type.ConclusionsCI users showed SRM that was comparable to normal hearing subjects. In listening situations of everyday life with spatial separation of source and masker, directional microphones significantly improved speech perception with individual improvements of up to 15 dB SNR. Users of bilateral speech processors with both directional microphones obtained the highest benefit.
| S-EPMC4430230 | biostudies-literature
Project description:ObjectiveTo analyze speech reading through Internet video calls by profoundly hearing-impaired individuals and cochlear implant (CI) users.MethodsSpeech reading skills of 14 deaf adults and 21 CI users were assessed using the Hochmair Schulz Moser (HSM) sentence test. We presented video simulations using different video resolutions (1280 × 720, 640 × 480, 320 × 240, 160 × 120 px), frame rates (30, 20, 10, 7, 5 frames per second (fps)), speech velocities (three different speakers), webcameras (Logitech Pro9000, C600 and C500) and image/sound delays (0-500 ms). All video simulations were presented with and without sound and in two screen sizes. Additionally, scores for live Skype™ video connection and live face-to-face communication were assessed.ResultsHigher frame rate (>7 fps), higher camera resolution (>640 × 480 px) and shorter picture/sound delay (<100 ms) were associated with increased speech perception scores. Scores were strongly dependent on the speaker but were not influenced by physical properties of the camera optics or the full screen mode. There is a significant median gain of +8.5%pts (p = 0.009) in speech perception for all 21 CI-users if visual cues are additionally shown. CI users with poor open set speech perception scores (n = 11) showed the greatest benefit under combined audio-visual presentation (median speech perception +11.8%pts, p = 0.032).ConclusionWebcameras have the potential to improve telecommunication of hearing-impaired individuals.
| S-EPMC3554620 | biostudies-literature