Masking release for hearing-impaired listeners: The effect of increased audibility through reduction of amplitude variability.

The masking release (i.e., better speech recognition in fluctuating compared to continuous noise backgrounds) observed for normal-hearing (NH) listeners is generally reduced or absent in hearing-impaired (HI) listeners. One explanation for this lies in the effects of reduced audibility: elevated thresholds may prevent HI listeners from taking advantage of signals available to NH listeners during the dips of temporally fluctuating noise where the interference is relatively weak. This hypothesis was addressed through the development of a signal-processing technique designed to increase the audibility of speech during dips in interrupted noise. This technique acts to (i) compare short-term and long-term estimates of energy, (ii) increase the level of short-term segments whose energy is below the average energy, and (iii) normalize the overall energy of the processed signal to be equivalent to that of the original long-term estimate. Evaluations of this energy-equalizing (EEQ) technique included consonant identification and sentence reception in backgrounds of continuous and regularly interrupted noise. For HI listeners, performance was generally similar for processed and unprocessed signals in continuous noise; however, superior performance for EEQ processing was observed in certain regularly interrupted noise backgrounds.

Effect of Energy Equalization on the Intelligibility of Speech in Fluctuating Background Interference for Listeners With Hearing Impairment.

The masking release (MR; i.e., better speech recognition in fluctuating compared with continuous noise backgrounds) that is evident for listeners with normal hearing (NH) is generally reduced or absent for listeners with sensorineural hearing impairment (HI). In this study, a real-time signal-processing technique was developed to improve MR in listeners with HI and offer insight into the mechanisms influencing the size of MR. This technique compares short-term and long-term estimates of energy, increases the level of short-term segments whose energy is below the average energy, and normalizes the overall energy of the processed signal to be equivalent to that of the original long-term estimate. This signal-processing algorithm was used to create two types of energy-equalized (EEQ) signals: EEQ1, which operated on the wideband speech plus noise signal, and EEQ4, which operated independently on each of four bands with equal logarithmic width. Consonant identification was tested in backgrounds of continuous and various types of fluctuating speech-shaped Gaussian noise including those with both regularly and irregularly spaced temporal fluctuations. Listeners with HI achieved similar scores for EEQ and the original (unprocessed) stimuli in continuous-noise backgrounds, while superior performance was obtained for the EEQ signals in fluctuating background noises that had regular temporal gaps but not for those with irregularly spaced fluctuations. Thus, in noise backgrounds with regularly spaced temporal fluctuations, the energy-normalized signals led to larger values of MR and higher intelligibility than obtained with unprocessed signals.

Level variations in speech: Effect on masking release in hearing-impaired listeners.

Acoustic speech is marked by time-varying changes in the amplitude envelope that may pose difficulties for hearing-impaired listeners. Removal of these variations (e.g., by the Hilbert transform) could improve speech reception for such listeners, particularly in fluctuating interference. Léger, Reed, Desloge, Swaminathan, and Braida [(2015b). J. Acoust. Soc. Am. 138, 389-403] observed that a normalized measure of masking release obtained for hearing-impaired listeners using speech processed to preserve temporal fine-structure (TFS) cues was larger than that for unprocessed or envelope-based speech. This study measured masking release for two other speech signals in which level variations were minimal: peak clipping and TFS processing of an envelope signal. Consonant identification was measured for hearing-impaired listeners in backgrounds of continuous and fluctuating speech-shaped noise. The normalized masking release obtained using speech with normal variations in overall level was substantially less than that observed using speech processed to achieve highly restricted level variations. These results suggest that the performance of hearing-impaired listeners in fluctuating noise may be improved by signal processing that leads to a decrease in stimulus level variations.

Consonant identification in noise using Hilbert-transform temporal fine-structure speech and recovered-envelope speech for listeners with normal and impaired hearing.

Consonant-identification ability was examined in normal-hearing (NH) and hearing-impaired (HI) listeners in the presence of steady-state and 10-Hz square-wave interrupted speech-shaped noise. The Hilbert transform was used to process speech stimuli (16 consonants in a-C-a syllables) to present envelope cues, temporal fine-structure (TFS) cues, or envelope cues recovered from TFS speech. The performance of the HI listeners was inferior to that of the NH listeners both in terms of lower levels of performance in the baseline condition and in the need for higher signal-to-noise ratio to yield a given level of performance. For NH listeners, scores were higher in interrupted noise than in steady-state noise for all speech types (indicating substantial masking release). For HI listeners, masking release was typically observed for TFS and recovered-envelope speech but not for unprocessed and envelope speech. For both groups of listeners, TFS and recovered-envelope speech yielded similar levels of performance and consonant confusion patterns. The masking release observed for TFS and recovered-envelope speech may be related to level effects associated with the manner in which the TFS processing interacts with the interrupted noise signal, rather than to the contributions of TFS cues per se.

The role of recovered envelope cues in the identification of temporal-fine-structure speech for hearing-impaired listeners.

Narrowband speech can be separated into fast temporal cues [temporal fine structure (TFS)], and slow amplitude modulations (envelope). Speech processed to contain only TFS leads to envelope recovery through cochlear filtering, which has been suggested to account for TFS-speech intelligibility for normal-hearing listeners. Hearing-impaired listeners have deficits with TFS-speech identification, but the contribution of recovered-envelope cues to these deficits is unknown. This was assessed for hearing-impaired listeners by measuring identification of disyllables processed to contain TFS or recovered-envelope cues. Hearing-impaired listeners performed worse than normal-hearing listeners, but TFS-speech intelligibility was accounted for by recovered-envelope cues for both groups.

Auditory and tactile gap discrimination by observers with normal and impaired hearing.

Temporal processing ability for the senses of hearing and touch was examined through the measurement of gap-duration discrimination thresholds (GDDTs) employing the same low-frequency sinusoidal stimuli in both modalities. GDDTs were measured in three groups of observers (normal-hearing, hearing-impaired, and normal-hearing with simulated hearing loss) covering an age range of 21-69 yr. GDDTs for a baseline gap of 6 ms were measured for four different combinations of 100-ms leading and trailing markers (250-250, 250-400, 400-250, and 400-400 Hz). Auditory measurements were obtained for monaural presentation over headphones and tactile measurements were obtained using sinusoidal vibrations presented to the left middle finger. The auditory GDDTs of the hearing-impaired listeners, which were larger than those of the normal-hearing observers, were well-reproduced in the listeners with simulated loss. The magnitude of the GDDT was generally independent of modality and showed effects of age in both modalities. The use of different-frequency compared to same-frequency markers led to a greater deterioration in auditory GDDTs compared to tactile GDDTs and may reflect differences in bandwidth properties between the two sensory systems.

Consonant identification using temporal fine structure and recovered envelope cues.

The contribution of recovered envelopes (RENVs) to the utilization of temporal-fine structure (TFS) speech cues was examined in normal-hearing listeners. Consonant identification experiments used speech stimuli processed to present TFS or RENV cues. Experiment 1 examined the effects of exposure and presentation order using 16-band TFS speech and 40-band RENV speech recovered from 16-band TFS speech. Prior exposure to TFS speech aided in the reception of RENV speech. Performance on the two conditions was similar (∼50%-correct) for experienced listeners as was the pattern of consonant confusions. Experiment 2 examined the effect of varying the number of RENV bands recovered from 16-band TFS speech. Mean identification scores decreased as the number of RENV bands decreased from 40 to 8 and were only slightly above chance levels for 16 and 8 bands. Experiment 3 examined the effect of varying the number of bands in the TFS speech from which 40-band RENV speech was constructed. Performance fell from 85%- to 31%-correct as the number of TFS bands increased from 1 to 32. Overall, these results suggest that the interpretation of previous studies that have used TFS speech may have been confounded with the presence of RENVs.

Two-microphone spatial filtering provides speech reception benefits for cochlear implant users in difficult acoustic environments.

This article introduces and provides an assessment of a spatial-filtering algorithm based on two closely-spaced (∼1 cm) microphones in a behind-the-ear shell. The evaluated spatial-filtering algorithm used fast (∼10 ms) temporal-spectral analysis to determine the location of incoming sounds and to enhance sounds arriving from straight ahead of the listener. Speech reception thresholds (SRTs) were measured for eight cochlear implant (CI) users using consonant and vowel materials under three processing conditions: An omni-directional response, a dipole-directional response, and the spatial-filtering algorithm. The background noise condition used three simultaneous time-reversed speech signals as interferers located at 90°, 180°, and 270°. Results indicated that the spatial-filtering algorithm can provide speech reception benefits of 5.8 to 10.7 dB SRT compared to an omni-directional response in a reverberant room with multiple noise sources. Given the observed SRT benefits, coupled with an efficient design, the proposed algorithm is promising as a CI noise-reduction solution.

Design and preliminary testing of a visually guided hearing aid.

An approach to hearing aid design is described, and preliminary acoustical and perceptual measurements are reported, in which an acoustic beam-forming microphone array is coupled to an eye-glasses-mounted eye-tracker. This visually guided hearing aid (VGHA)-currently a laboratory-based prototype-senses direction of gaze using the eye tracker and an interface converts those values into control signals that steer the acoustic beam accordingly. Preliminary speech intelligibility measurements with noise and speech maskers revealed near- or better-than normal spatial release from masking with the VGHA. Although not yet a wearable prosthesis, the principle underlying the device is supported by these findings.

Masking Release for Igbo and English.

In this research, we explored the effect of noise interruption rate on speech intelligibility. Specifically, we used the Hearing In Noise Test (HINT) procedure with the original HINT stimuli (English) and Igbo stimuli to assess speech reception ability in interrupted noise. For a given noise level, the HINT test provides an estimate of the signal-to-noise ratio (SNR) required for 50%-correct speech intelligibility. The SNR for 50%-correct intelligibility changes depending upon the interruption rate of the noise. This phenomenon (called Masking Release) has been studied extensively in English but not for Igbo - which is an African tonal language spoken predominantly in South Eastern Nigeria. This experiment explored and compared the phenomenon of Masking Release for (i) native English speakers listening to English, (ii) native Igbo speakers listening to English, and (iii) native Igbo speakers listening to Igbo. Since Igbo is a tonal language and English is a non-tonal language, this allowed us to compare Masking Release patterns on native speakers of tonal and non-tonal languages. Our results for native English speakers listening to English HINT show that the SNR and the masking release are orderly and consistent with other English HINT data for English speakers. Our result for Igbo speakers listening to English HINT sentences show that there is greater variability in results across the different Igbo listeners than across the English listeners. This result likely reflects different levels of ability in the English language across the Igbo listeners. The masking release values in dB are less than for English listeners. Our results for Igbo speakers listening to Igbo show that in general, the SNRs for Igbo sentences are lower than for English/English and Igbo/English. This means that the Igbo listeners could understand 50% of the Igbo sentences at SNRs less than those required for English sentences by either native or non-native listeners. This result can be explained by the fact that the perception of Igbo utterances by Igbo subjects may have been aided by the prediction of tonal and vowel harmony features existent in the Igbo language. In agreement with other studies, our results also show that in a noisy environment listeners are able to perceive their native language better than a second language. The ability of native language speakers to perceive their language better than a second language in a noisy environment may be attributed to the fact that: Native speakers are more familiar with the sounds of their language than second language speakers.One of the features of language is that it is predictable hence even in noise a native speaker may be able to predict a succeeding word that is scarcely audible. These contextual effects are facilitated by familiarity.

Auditory-filter characteristics for listeners with real and simulated hearing impairment.

Functional simulation of sensorineural hearing impairment is an important research tool that can elucidate the nature of hearing impairments and suggest or eliminate compensatory signal-processing schemes. The objective of the current study was to evaluate the capability of an audibility-based functional simulation of hearing loss to reproduce the auditory-filter characteristics of listeners with sensorineural hearing loss. The hearing-loss simulation used either threshold-elevating noise alone or a combination of threshold-elevating noise and multiband expansion to reproduce the audibility-based characteristics of the loss (including detection thresholds, dynamic range, and loudness recruitment). The hearing losses of 10 listeners with bilateral, mild-to-severe hearing loss were simulated in 10 corresponding groups of 3 age-matched normal-hearing listeners. Frequency selectivity was measured using a notched-noise masking paradigm at five probe frequencies in the range of 250 to 4000 Hz with a fixed probe level of either 70 dB SPL or 8 dB SL (whichever was greater) and probe duration of 200 ms. The hearing-loss simulation reproduced the absolute thresholds of individual hearing-impaired listeners with an average root-mean-squared (RMS) difference of 2.2 dB and the notched-noise masked thresholds with an RMS difference of 5.6 dB. A rounded-exponential model of the notched-noise data was used to estimate equivalent rectangular bandwidths and slopes of the auditory filters. For some subjects and probe frequencies, the simulations were accurate in reproducing the auditory-filter characteristics of the hearing-impaired listeners. In other cases, however, the simulations underestimated the magnitude of the auditory bandwidths for the hearing-impaired listeners, which suggests the possibility of suprathreshold deficits.

Temporal masking functions for listeners with real and simulated hearing loss.

A functional simulation of hearing loss was evaluated in its ability to reproduce the temporal masking functions for eight listeners with mild to severe sensorineural hearing loss. Each audiometric loss was simulated in a group of age-matched normal-hearing listeners through a combination of spectrally-shaped masking noise and multi-band expansion. Temporal-masking functions were obtained in both groups of listeners using a forward-masking paradigm in which the level of a 110-ms masker required to just mask a 10-ms fixed-level probe (5-10 dB SL) was measured as a function of the time delay between the masker offset and probe onset. At each of four probe frequencies (500, 1000, 2000, and 4000 Hz), temporal-masking functions were obtained using maskers that were 0.55, 1.0, and 1.15 times the probe frequency. The slopes and y-intercepts of the masking functions were not significantly different for listeners with real and simulated hearing loss. The y-intercepts were positively correlated with level of hearing loss while the slopes were negatively correlated. The ratio of the slopes obtained with the low-frequency maskers relative to the on-frequency maskers was similar for both groups of listeners and indicated a smaller compressive effect than that observed in normal-hearing listeners.

Temporal modulation transfer functions for listeners with real and simulated hearing loss.

A functional simulation of hearing loss was evaluated in its ability to reproduce the temporal modulation transfer functions (TMTFs) for nine listeners with mild to profound sensorineural hearing loss. Each hearing loss was simulated in a group of three age-matched normal-hearing listeners through spectrally shaped masking noise or a combination of masking noise and multiband expansion. TMTFs were measured for both groups of listeners using a broadband noise carrier as a function of modulation rate in the range 2 to 1024 Hz. The TMTFs were fit with a lowpass filter function that provided estimates of overall modulation-depth sensitivity and modulation cutoff frequency. Although the simulations were capable of accurately reproducing the threshold elevations of the hearing-impaired listeners, they were not successful in reproducing the TMTFs. On average, the simulations resulted in lower sensitivity and higher cutoff frequency than were observed in the TMTFs of the hearing-impaired listeners. Discrepancies in performance between listeners with real and simulated hearing loss are possibly related to inaccuracies in the simulation of recruitment.

Speech reception by listeners with real and simulated hearing impairment: effects of continuous and interrupted noise.

The effects of audibility and age on masking for sentences in continuous and interrupted noise were examined in listeners with real and simulated hearing loss. The absolute thresholds of each of ten listeners with sensorineural hearing loss were simulated in normal-hearing listeners through a combination of spectrally-shaped threshold noise and multi-band expansion for octave bands with center frequencies from 0.25-8 kHz. Each individual hearing loss was simulated in two groups of three normal-hearing listeners (an age-matched and a non-age-matched group). The speech-to-noise ratio (S/N) for 50%-correct identification of hearing in noise test (HINT) sentences was measured in backgrounds of continuous and temporally-modulated (10 Hz square-wave) noise at two overall levels for unprocessed speech and for speech that was amplified with the NAL-RP prescription. The S/N in both continuous and interrupted noise of the hearing-impaired listeners was relatively well-simulated in both groups of normal-hearing listeners. Thus, release from masking (the difference in S/N obtained in continuous versus interrupted noise) appears to be determined primarily by audibility. Minimal age effects were observed in this small sample. Observed values of masking release were compared to predictions derived from intelligibility curves generated using the extended speech intelligibility index (ESII) [Rhebergen et al. (2006). J. Acoust. Soc. Am. 120, 3988-3997].

Evaluation of array-processing algorithms for a headband hearing aid.

Several array-processing algorithms were implemented and evaluated with experienced hearing-aid users. The array consisted of four directional microphones mounted broadside on a headband worn on the top of the listener's head. The algorithms included two adaptive array-processing algorithms, one fixed array-processing algorithm, and a reference condition consisting of binaural directional microphones. The algorithms were evaluated under conditions with both one and three independent noise sources. Performance metrics included quantitative speech reception thresholds and qualitative subject preference ratings for ease-of-listening measured using a paired-comparison procedure. On average, the fixed algorithm improved speech reception thresholds by 2 dB, while the adaptive algorithms provided 7-9-dB improvement over the reference condition. Subjects judging ease-of-listening generally preferred all array-processing algorithms over the reference condition. The results suggest that these adaptive algorithms should be evaluated further in more realistic acoustic environments.