Psychoacoustics publications

Purpose The two primary purposes of this report are (a) to compare the results of three brief cognitive screens in older adults and (b) to examine associations between performance on each of the screens and auditory function measured either concurrently or 9 years earlier. Method This was a prospective longitudinal study of 98 adults (66 women) with baseline ages ranging from 40 to 85 years. The mean interval between T1 baseline and T2 follow-up measurements was 8.8 years with a range from 7 to 11 years. Measures of hearing threshold, gap detection, and auditory temporal-order identification were completed at T1 and T2. The Mini-Mental State Examination was completed at T1 and T2, whereas the Montreal Cognitive Assessment (MoCA) and A Quick Test were completed at T2 only. Results Higher scores and pass rates were obtained for the Mini-Mental State Examination than for the MoCA or the A Quick Test. The measures were moderately correlated among themselves and with the Wechsler Adult Intelligence Scale–Third Edition. Significant associations emerged frequently between auditory and cognitive functions, most often for the auditory measure of temporal-order identification, including dichotic measures of this ability. Conclusions From this evaluation, the MoCA emerged as the preferred test for clinicians desiring a quick assessment of the cognitive function of their older patients. Auditory temporal-order identification is associated with cognitive function and explains about 10%–20% of the variation in cognitive function independent of age and hearing loss. Supplemental Material https://doi.org/10.23641/asha.12986021

The impact of an extraneous formant on intelligibility is affected by the extent (depth) of variation in its formant-frequency contour. Two experiments explored whether this impact also depends on masker spectro-temporal coherence, using a method ensuring that interference occurred only through informational masking. Targets were monaural three-formant analogues (F1+F2+F3) of natural sentences presented alone or accompanied by a contralateral competitor for F2 (F2C) that listeners must reject to optimize recognition. The standard F2C was created using the inverted F2 frequency contour and constant amplitude. Variants were derived by dividing F2C into abutting segments (100–200 ms, 10-ms rise/fall). Segments were presented either in the correct order (coherent) or in random order (incoherent), introducing abrupt discontinuities into the F2C frequency contour. F2C depth was also manipulated (0%, 50%, or 100%) prior to segmentation, and the frequency contour of each segment either remained time-varying or was set to constant at the geometric mean frequency of that segment. The extent to which F2C lowered keyword scores depended on segment type (frequency-varying vs constant) and depth, but not segment order. This outcome indicates that the impact on intelligibility depends critically on the overall amount of frequency variation in the competitor, but not its spectro-temporal coherence.

Speech audiometry in noise based on sentence tests is an important diagnostic tool to assess listeners’ speech recognition threshold (SRT), i.e., the signal-to-noise ratio corresponding to 50% intelligibility. The clinical standard measurement procedure requires a professional experimenter to record and evaluate the response (expert-conducted speech audiometry). The use of automatic speech recognition enables self-conducted measurements with an easy-to-use speech-based interface. This article compares self-conducted SRT measurements using smart speakers with expert-conducted laboratory measurements. With smart speakers, there is no control over the absolute presentation level, potential errors from the automated response logging, and room acoustics. We investigate the differences between highly controlled measurements in the laboratory and smart speaker-based tests for young normal-hearing (NH) listeners as well as for elderly NH, mildly and moderately hearing-impaired listeners in low, medium, and highly reverberant room acoustics. For the smart speaker setup, we observe an overall bias in the SRT result that depends on the hearing loss. The bias ranges from +0.7 dB for elderly moderately hearing-impaired listeners to +2.2 dB for young NH listeners. The intrasubject standard deviation is close to the clinical standard deviation (0.57/0.69 dB for the young/elderly NH compared with 0.5 dB observed for clinical tests and 0.93/1.09 dB for the mild/moderate hearing-impaired listeners compared with 0.9 dB). For detecting a clinically elevated SRT, the speech-based test achieves an area under the curve value of 0.95 and therefore seems promising for complementing clinical measurements.

A common method for measuring changes in temporal processing sensitivity in both humans and animals makes use of GaP-induced Inhibition of the Acoustic Startle (GPIAS). It is also the basis of a common method for detecting tinnitus in rodents. However, the link to tinnitus has not been properly established because GPIAS has not yet been used to objectively demonstrate tinnitus in humans. In guinea pigs, the Preyer (ear flick) myogenic reflex is an established method for measuring the acoustic startle for the GPIAS test, while in humans, it is the eye-blink reflex. Yet, humans have a vestigial remnant of the Preyer reflex, which can be detected by measuring skin surface potentials associated with the Post-Auricular Muscle Response (PAMR). A similar electrical potential can be measured in guinea pigs and we aimed to show that the PAMR could be used to demonstrate GPIAS in both species. In guinea pigs, we compare the GPIAS measured using the pinna movement of the Preyer reflex and the electrical potential of the PAMR to demonstrate that the two are at least equivalent. In humans, we establish for the first time that the PAMR provides a reliable way of measuring GPIAS that is a pure acoustic alternative to the multimodal eye-blink reflex. Further exploratory tests showed that while eye gaze position influenced the size of the PAMR response, it did not change the degree of GPIAS. Our findings confirm that the PAMR is a sensitive method for measuring GPIAS and suggest that it may allow direct comparison of temporal processing between humans and animals and may provide a basis for an objective test of tinnitus.

Differences in ear of presentation and level do not prevent effective integration of concurrent speech cues such as formant frequencies. For example, presenting the higher formants of a consonant-vowel syllable in the opposite ear to the first formant protects them from upward spread of masking, allowing them to remain effective speech cues even after substantial attenuation. This study used three-formant (F1+F2+F3) analogues of natural sentences and extended the approach to include competitive conditions. Target formants were presented dichotically (F1+F3; F2), either alone or accompanied by an extraneous competitor for F2 (i.e., F1±F2C+F3; F2) that listeners must reject to optimize recognition. F2C was created by inverting the F2 frequency contour and using the F2 amplitude contour without attenuation. In experiment 1, F2C was always absent and intelligibility was unaffected until F2 attenuation exceeded 30 dB; F2 still provided useful information at 48-dB attenuation. In experiment 2, attenuating F2 by 24 dB caused considerable loss of intelligibility when F2C was present, but had no effect in its absence. Factors likely to contribute to this interaction include informational masking from F2C acting to swamp the acoustic-phonetic information carried by F2, and interaural inhibition from F2C acting to reduce the effective level of F2.

Speech audiometry is an essential part of audiological diagnostics and clinical measurements. Development times of speech recognition tests are rather long, depending on the size of speech corpus and optimization necessity. The aim of this study was to examine whether this development effort could be reduced by using synthetic speech in speech audiometry, especially in a matrix test for speech recognition. For this purpose, the speech material of the German matrix test was replicated using a preselected commercial system to generate the synthetic speech files. In contrast to the conventional matrix test, no level adjustments or optimization tests were performed while producing the synthetic speech material. Evaluation measurements were conducted by presenting both versions of the German matrix test (with natural or synthetic speech), alternately and at three different signal-to-noise ratios, to 48 young, normal-hearing participants. Psychometric functions were fitted to the empirical data. Speech recognition thresholds were 0.5 dB signal-to-noise ratio higher (worse) for the synthetic speech, while slopes were equal for both speech types. Nevertheless, speech recognition scores were comparable with the literature and the threshold difference lay within the same range as recordings of two different natural speakers. Although no optimization was applied, the synthetic-speech signals led to equivalent recognition of the different test lists and word categories. The outcomes of this study indicate that the application of synthetic speech in speech recognition tests could considerably reduce the development costs and evaluation time. This offers the opportunity to increase the speech corpus for speech recognition tests with acceptable effort.

The frequency-specific tone-evoked auditory brainstem response (ABR) is an indispensable tool in both the audiology clinic and research laboratory. Most frequently, the toneburst ABR is used to estimate hearing thresholds in infants, toddlers, and other patients for whom behavioral testing is not feasible. Therefore, results of the ABR exam form the basis for decisions regarding interventions and hearing habilitation with implications extending far into the child’s future. Currently, responses are elicited by periodic sequences of toneburst stimuli presented serially to one ear at a time, which take a long time to measure multiple frequencies and intensities, and provide incomplete information if the infant wakes up early. Here, we describe a new method, the parallel ABR (pABR), which uses randomly timed toneburst stimuli to simultaneously acquire ABR waveforms to five frequencies in both ears. Here, we describe the pABR and quantify its effectiveness in addressing the greatest drawback of current methods: test duration. We show that in adults with normal hearing the pABR yields high-quality waveforms over a range of intensities, with similar morphology to the standard ABR in a fraction of the recording time. Furthermore, longer latencies and smaller amplitudes for low frequencies at a high intensity evoked by the pABR versus serial ABR suggest that responses may have better place specificity due to the masking provided by the other simultaneous toneburst sequences. Thus, the pABR has substantial potential for facilitating faster accumulation of more diagnostic information that is important for timely identification and treatment of hearing loss.

Evaluation of patients who are unable to provide behavioral responses on standard clinical measures is challenging due to the lack of standard objective (non-behavioral) clinical audiological measures that assess the outcome of an intervention (e.g., hearing aids). Brainstem responses to short consonant-vowel stimuli (speech-auditory brainstem responses [speech-ABRs]) have been proposed as a measure of subcortical encoding of speech, speech detection, and speech-in-noise performance in individuals with normal hearing. Here, we investigated the potential application of speech-ABRs as an objective clinical outcome measure of speech detection, speech-in-noise detection and recognition, and self-reported speech understanding in 98 adults with sensorineural hearing loss. We compared aided and unaided speech-ABRs, and speech-ABRs in quiet and in noise. In addition, we evaluated whether speech-ABR F0 encoding (obtained from the complex cross-correlation with the 40 ms [da] fundamental waveform) predicted aided behavioral speech recognition in noise or aided self-reported speech understanding. Results showed that (a) aided speech-ABRs had earlier peak latencies, larger peak amplitudes, and larger F0 encoding amplitudes compared to unaided speech-ABRs; (b) the addition of background noise resulted in later F0 encoding latencies but did not have an effect on peak latencies and amplitudes or on F0 encoding amplitudes; and (c) speech-ABRs were not a significant predictor of any of the behavioral or self-report measures. These results show that speech-ABR F0 encoding is not a good predictor of speech-in-noise recognition or self-reported speech understanding with hearing aids. However, our results suggest that speech-ABRs may have potential for clinical application as an objective measure of speech detection with hearing aids.

Stream segregation for a test sequence comprising high-frequency (H) and low-frequency (L) pure tones, presented in a galloping rhythm, is much greater when preceded by a constant-frequency induction sequence matching one subset than by an inducer configured like the test sequence; this difference persists for several seconds. It has been proposed that constant-frequency inducers promote stream segregation by capturing the matching subset of test-sequence tones into an on-going, pre-established stream. This explanation was evaluated using 2-s induction sequences followed by longer test sequences (12–20 s). Listeners reported the number of streams heard throughout the test sequence. Experiment 1 used LHL– sequences and one or other subset of inducer tones was attenuated (0–24 dB in 6-dB steps, and ∞). Greater attenuation usually caused a progressive increase in segregation, towards that following the constant-frequency inducer. Experiment 2 used HLH– sequences and the L inducer tones were raised or lowered in frequency relative to their test-sequence counterparts (ΔfI = 0, 0.5, 1.0, or 1.5 × ΔfT). Either change greatly increased segregation. These results are concordant with the notion of attention switching to new sounds but contradict the stream-capture hypothesis, unless a “proto-object” corresponding to the continuing subset is assumed to form during the induction sequence.

This study explored the extent to which informational masking of speech depends on the frequency region and number of extraneous formants in an interferer. Target formants—monotonized three-formant (F1+F2+F3) analogues of natural sentences—were presented monaurally, with target ear assigned randomly on each trial. Interferers were presented contralaterally. In experiment 1, single-formant interferers were created using the time-reversed F2 frequency contour and constant amplitude, root-mean-square (RMS)-matched to F2. Interferer center frequency was matched to that of F1, F2, or F3, while maintaining the extent of formant-frequency variation (depth) on a log scale. Adding an interferer lowered intelligibility; the effect of frequency region was small and broadly tuned around F2. In experiment 2, interferers comprised either one formant (F1, the most intense) or all three, created using the time-reversed frequency contours of the corresponding targets and RMS-matched constant amplitudes. Interferer formant-frequency variation was scaled to 0%, 50%, or 100% of the original depth. Increasing the depth of formant-frequency variation and number of formants in the interferer had independent and additive effects. These findings suggest that the impact on intelligibility depends primarily on the overall extent of frequency variation in each interfering formant (up to ∼100% depth) and the number of extraneous formants.

For patients with bilateral cochlear implants (BiCIs), understanding a target talker in a noisy situation can be difficult. Current efforts for improving speech-in-noise understanding have focused on improving signal-to-noise ratio by using multiple microphones or signal processing, with only moderate improvements in speech understanding performance. However, BiCI users typically report having a better ear for listening which can lead to an asymmetry in speech unmasking performance. This work proposes a novel listening strategy for improving speech-in-noise understanding by combining (a) a priori knowledge of a better ear and having a BiCI user selectively attend to a target talker in that ear with (b) signal processing that delivers the target talker to the better ear and the noisy background to the opposite ear. This strategy is different from traditional noise reduction strategies because it maintains situational awareness (background sounds are delivered to the ear contralateral to the better ear) while improving speech understanding. Speech recognition performance was evaluated with and without the better ear strategy in a speech-in-noise listening test using a virtual auditory space created from individualized head-related transfer functions. Listeners showed an average improvement of 4.4 dB signal-to-noise ratio in their speech reception threshold when using the better ear strategy with no listener showing a decrement in performance. This implies that the strategy has the potential to boost speech-in-noise recognition in BiCI users and may be useful in other hearing assistance devices such as hearing aids.

Hearing-impaired listeners are known to have difficulties not only with understanding speech in noise but also with judging source distance and movement, and these deficits are related to perceived handicap. It is possible that the perception of spatially dynamic sounds can be improved with hearing aids (HAs), but so far this has not been investigated. In a previous study, older hearing-impaired listeners showed poorer detectability for virtual left-right (angular) and near-far (radial) source movements due to lateral interfering sounds and reverberation, respectively. In the current study, potential ways of improving these deficits with HAs were explored. Using stimuli very similar to before, detailed acoustic analyses were carried out to examine the influence of different HA algorithms for suppressing noise and reverberation on the acoustic cues previously shown to be associated with source movement detectability. For an algorithm that combined unilateral directional microphones with binaural coherence-based noise reduction and for a bilateral beamformer with binaural cue preservation, movement-induced changes in spectral coloration, signal-to-noise ratio, and direct-to-reverberant energy ratio were greater compared with no HA processing. To evaluate these two algorithms perceptually, aided measurements of angular and radial source movement detectability were performed with 20 older hearing-impaired listeners. The analyses showed that, in the presence of concurrent interfering sounds and reverberation, the bilateral beamformer could restore source movement detectability in both spatial dimensions, whereas the other algorithm only improved detectability in the near-far dimension. Together, these results provide a basis for improving the detectability of spatially dynamic sounds with HAs.

Speech is an ecologically essential signal, whose processing crucially involves the subcortical nuclei of the auditory brainstem, but there are few experimental options for studying these early responses in human listeners under natural conditions. While encoding of continuous natural speech has been successfully probed in the cortex with neurophysiological tools such as electroencephalography (EEG) and magnetoencephalography, the rapidity of subcortical response components combined with unfavorable signal-to-noise ratios signal-to-noise ratio has prevented application of those methods to the brainstem. Instead, experiments have used thousands of repetitions of simple stimuli such as clicks, tone-bursts, or brief spoken syllables, with deviations from those paradigms leading to ambiguity in the neural origins of measured responses. In this study we developed and tested a new way to measure the auditory brainstem response (ABR) to ongoing, naturally uttered speech, using EEG to record from human listeners. We found a high degree of morphological similarity between the speech-derived ABRs and the standard click-evoked ABR, in particular, a preserved Wave V, the most prominent voltage peak in the standard click-evoked ABR. Because this method yields distinct peaks that recapitulate the canonical ABR, at latencies too short to originate from the cortex, the responses measured can be unambiguously determined to be subcortical in origin. The use of naturally uttered speech to measure the ABR allows the design of engaging behavioral tasks, facilitating new investigations of the potential effects of cognitive processes like language and attention on brainstem processing.

Recent research provides evidence for a functional role of brain oscillations for perception. For example, auditory temporal resolution seems to be linked to individual gamma frequency of auditory cortex. Individual gamma frequency not only correlates with performance in between-channel gap detection tasks but can be modulated via auditory transcranial alternating current stimulation. Modulation of individual gamma frequency is accompanied by an improvement in gap detection performance. Aging changes electrophysiological frequency components and sensory processing mechanisms. Therefore, we conducted a study to investigate the link between individual gamma frequency and gap detection performance in elderly people using auditory transcranial alternating current stimulation. In a within-subject design, nine participants were electrically stimulated with two individualized transcranial alternating current stimulation frequencies: 3 Hz above their individual gamma frequency (experimental condition) and 4 Hz below their individual gamma frequency (control condition) while they were performing a between-channel gap detection task. As expected, individual gamma frequencies correlated significantly with gap detection performance at baseline and in the experimental condition, transcranial alternating current stimulation modulated gap detection performance. In the control condition, stimulation did not modulate gap detection performance. In addition, in elderly, the effect of transcranial alternating current stimulation on auditory temporal resolution seems to be dependent on endogenous frequencies in auditory cortex: elderlies with slower individual gamma frequencies and lower auditory temporal resolution profit from auditory transcranial alternating current stimulation and show increased gap detection performance during stimulation. Our results strongly suggest individualized transcranial alternating current stimulation protocols for successful modulation of performance.

There is evidence that the contribution of a given harmonic in a complex tone to residue pitch is influenced by the accuracy with which the frequency of that harmonic is encoded. The present study investigated whether listeners adjust the weights assigned to individual harmonics based on acquired knowledge of the reliability of the frequency estimates of those harmonics. In a two-interval forced-choice task, seven listeners indicated which of two 12-harmonic complex tones had the higher overall pitch. In context trials (60 % of all trials), the fundamental frequency (F0) was 200 Hz in one interval and 200 + ΔF0 Hz in the other. In different (blocked) conditions, either the 3rd or the 4th harmonic (plus the 7th, 9th, and 12th harmonics), were replaced by narrowband noises that were identical in the two intervals. Feedback was provided. In randomly interspersed test trials (40 % of all trials), the fundamental frequency was 200 + ΔF0/2 Hz in both intervals; in the second interval, either the third or the fourth harmonic was shifted slightly up or down in frequency with equal probability. There were no narrowband noises. Feedback was not provided. The results showed that substitution of a harmonic by noise in context trials reduced the contribution of that harmonic to pitch judgements in the test trials by a small but significant amount. This is consistent with the notion that listeners give smaller weight to a harmonic or frequency region when they have learned that this frequency region does not provide reliable information for a given task.

Development of electrophysiological means to assess the medial olivocochlear (MOC) system in humans is important to further our understanding of the function of that system and for the refinement and validation of psychoacoustical and otoacoustic emission methods which are thought to probe the MOC. Based on measurements in anesthetized animals it has been hypothesized that the MOC-reflex (MOCR) can enhance the response to signals in noise, and several lines of evidence support such a role in humans. A difficulty in these studies is the isolation of efferent effects. Efferent activation can be triggered by acoustic stimulation of the contralateral or ipsilateral ear, but ipsilateral stimulation is thought to be more effective. However, ipsilateral stimulation complicates interpretation of effects since these sounds can affect the perception of other ipsilateral sounds by mechanisms not involving olivocochlear efferents. We assessed the ipsilaterally evoked MOCR in human using a transtympanic procedure to record mass-potentials from the cochlear promontory or the niche of the round window. Averaged compound action potential (CAP) responses to masked probe tones of 4 kHz with and without a precursor (designed to activate the MOCR but not the stapedius reflex) were extracted with a polarity alternating paradigm. The masker was either a simultaneous narrow band noise masker or a short (20-ms) tonal ON- or OFF-frequency forward masker. The subjects were screened for normal hearing (audiogram, tympanogram, threshold stapedius reflex) and psychoacoustically tested for the presence of a precursor effect. We observed a clear reduction of CAP amplitude by the precursor, for different masking conditions. Even without an MOCR, this is expected because the precursor will affect the response to subsequent stimuli via neural adaptation. To determine whether the precursor also activated the efferent system, we measured the CAP over a range of masker levels, with or without precursor, and for different types of masker. The results show CAP reduction consistent with the type of gain reduction caused by the MOCR. These results generally support psychoacoustical paradigms designed to probe the efferent system as indeed activating the MOCR system, but not all observations are consistent with this mechanism.

In contrast to static sounds, spatially dynamic sounds have received little attention in psychoacoustic research so far. This holds true especially for acoustically complex (reverberant, multisource) conditions and impaired hearing. The current study therefore investigated the influence of reverberation and the number of concurrent sound sources on source movement detection in young normal-hearing (YNH) and elderly hearing-impaired (EHI) listeners. A listening environment based on natural environmental sounds was simulated using virtual acoustics and rendered over headphones. Both near-far (‘radial’) and left-right (‘angular’) movements of a frontal target source were considered. The acoustic complexity was varied by adding static lateral distractor sound sources as well as reverberation. Acoustic analyses confirmed the expected changes in stimulus features that are thought to underlie radial and angular source movements under anechoic conditions and suggested a special role of monaural spectral changes under reverberant conditions. Analyses of the detection thresholds showed that, with the exception of the single-source scenarios, the EHI group was less sensitive to source movements than the YNH group, despite adequate stimulus audibility. Adding static sound sources clearly impaired the detectability of angular source movements for the EHI (but not the YNH) group. Reverberation, on the other hand, clearly impaired radial source movement detection for the EHI (but not the YNH) listeners. These results illustrate the feasibility of studying factors related to auditory movement perception with the help of the developed test setup.

Many audio applications perform perception-based time-frequency (TF) analysis by decomposing sounds into a set of functions with good TF localization (i.e. with a small essential support in the TF domain) using TF transforms and applying psychoacoustic models of auditory masking to the transform coefficients. To accurately predict masking interactions between coefficients, the TF properties of the model should match those of the transform. This involves having masking data for stimuli with good TF localization. However, little is known about TF masking for mathematically well-localized signals. Most existing masking studies used stimuli that are broad in time and/or frequency and few studies involved TF conditions. Consequently, the present study had two goals. The first was to collect TF masking data for well-localized stimuli in humans. Masker and target were 10-ms Gaussian-shaped sinusoids with a bandwidth of approximately one critical band. The overall pattern of results is qualitatively similar to existing data for long maskers. To facilitate implementation in audio processing algorithms, a dataset provides the measured TF masking function. The second goal was to assess the potential effect of auditory efferents on TF masking using a modeling approach. The temporal window model of masking was used to predict present and existing data in two configurations: (1) with standard model parameters (i.e. without efferents), (2) with cochlear gain reduction to simulate the activation of efferents. The ability of the model to predict the present data was quite good with the standard configuration but highly degraded with gain reduction. Conversely, the ability of the model to predict existing data for long maskers was better with than without gain reduction. Overall, the model predictions suggest that TF masking can be affected by efferent (or other) effects that reduce cochlear gain. Such effects were avoided in the experiment of this study by using maximally-compact stimuli.

The role of source properties in across-formant integration was explored using three-formant (F1+F2+F3) analogues of natural sentences (targets). In experiment 1, F1+F3 were harmonic analogues (H1+H3) generated using a monotonous buzz source and second-order resonators; in experiment 2, F1+F3 were tonal analogues (T1+T3). F2 could take either form (H2 or T2). Target formants were always presented monaurally; the receiving ear was assigned randomly on each trial. In some conditions, only the target was present; in others, a competitor for F2 (F2C) was presented contralaterally. Buzz-excited or tonal competitors were created using the time-reversed frequency and amplitude contours of F2. Listeners must reject F2C to optimize keyword recognition. Whether or not a competitor was present, there was no effect of source mismatch between F1+F3 and F2. The impact of adding F2C was modest when it was tonal but large when it was harmonic, irrespective of whether F2C matched F1+F3. This pattern was maintained when harmonic and tonal counterparts were loudness-matched (experiment 3). Source type and competition, rather than acoustic similarity, governed the phonetic contribution of a formant. Contrary to earlier research using dichotic targets, requiring across-ear integration to optimize intelligibility, H2C was an equally effective informational masker for H2 as for T2.