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This paper proposes a binaural-centered mode-matching (BCMM) method that performs sound field reproduction with two reproduction points, one at each ear. A sound field reproduced by higher-order Ambisonics converges to the target field around a sweet spot whose size is inversely proportional to the frequency when the truncated order of the spherical harmonic expansion is constant. By contrast, BCMM translates the spherical harmonic coefficients on the basis of the addition theorem to realize two reproduction points at both ear positions. The BCMM method thereby avoids degradation of reproduction due to a smaller sweet spot at higher frequencies, as occurs with the conventional method, and thereby, leads to an accurate reproduction at higher frequencies. A comparison of numerical simulations of the binaural signals obtained when rendering with BCMM and conventional methods shows that, compared with the conventional method, the BCMM method improves the reproduction accuracy.

The modified rhyme test [MRT; House, Williams, Hecker, and Kryter. (1965). J. Acoust. Soc. Am. 37, 158–166] is a widely used test for measuring the intelligibility of communication systems [ANSI (2009). S3.2 (American National Standards Institute, New York)] but has never gained widespread acceptance as a clinical test of speech intelligibility for listeners who are hearing impaired (HI). In this study, a clinical version of the MRT consisting of two 80-word lists was developed and tested on 2394 service members with varying levels of hearing loss. The test used a factorial design incorporating two speech levels [70 and 78 dB sound pressure level (SPL)], two signal-to-noise ratios (+4 and −4 dB), and two binaural conditions (diotic and binaural). High-frequency emphasis reduced the impact of audibility for HI listeners, focusing the test on the distortion component of hearing loss. The results show that listeners with normal hearing (NH) obtained an average score of 80% correct on the MRT80 test. Listeners with a moderate hearing impairment scored an average of 70% correct. The overall level had little impact on performance for either NH or HI listeners. The results demonstrate that the MRT80 test could be a useful test to assess the distortion effects of hearing loss on speech intelligibility, particularly in cases where it is desirable to use a closed-set test for automatic administration.

For abruptly gated sound, interaural time difference (ITD) cues at onset carry greater perceptual weight than those following. This research explored how envelope shape influences such carrier ITD weighting. Experiment 1 assessed the perceived lateralization of a tonal binaural beat that transitioned through ITD (diotic envelope, mean carrier frequency of 500 Hz). Listeners' left/right lateralization judgments were compared to those for static-ITD tones. For an 8 Hz sinusoidally amplitude-modulated envelope, ITD cues 24 ms after onset well-predicted reported sidedness. For an equivalent-duration "abrupt" envelope, which was unmodulated besides 20-ms onset/offset ramps, reported sidedness corresponded to ITDs near onset (e.g., 6 ms). However, unlike for sinusoidal amplitude modulation, ITDs toward offset seemingly also influenced perceived sidedness. Experiment 2 adjusted the duration of the offset ramp (25-75 ms) and found evidence for such offset weighting only for the most abrupt ramp tested. In experiment 3, an ITD was imposed on a brief segment of otherwise diotic filtered noise. Listeners discriminated right- from left-leading ITDs. In sinusoidal amplitude modulation, thresholds were lowest when the ITD segment occurred during rising amplitude. For the abrupt envelope, the lowest thresholds were observed when the segment occurred at either onset or offset. These experiments demonstrate the influence of envelope profile on carrier ITD sensitivity.

Bilateral cochlear-implant (CI) users struggle to understand speech in noisy environments despite receiving some spatial-hearing benefits. One potential solution is to provide acoustic beamforming. A headphone-based experiment was conducted to compare speech understanding under natural CI listening conditions and for two non-adaptive beamformers, one single beam and one binaural, called “triple beam,” which provides an improved signal-to-noise ratio (beamforming benefit) and usable spatial cues by reintroducing interaural level differences. Speech reception thresholds (SRTs) for speech-on-speech masking were measured with target speech presented in front and two maskers in co-located or narrow/wide separations. Numerosity judgments and sound-localization performance also were measured. Natural spatial cues, single-beam, and triple-beam conditions were compared. For CI listeners, there was a negligible change in SRTs when comparing co-located to separated maskers for natural listening conditions. In contrast, there were 4.9- and 16.9-dB improvements in SRTs for the beamformer and 3.5- and 12.3-dB improvements for triple beam (narrow and wide separations). Similar results were found for normal-hearing listeners presented with vocoded stimuli. Single beam improved speech-on-speech masking performance but yielded poor sound localization. Triple beam improved speech-on-speech masking performance, albeit less than the single beam, and sound localization. Thus, triple beam was the most versatile across multiple spatial-hearing domains.

Sound radiation of most natural sources, like human speakers or musical instruments, typically exhibits a spatial directivity pattern. This directivity contributes to the perception of sound sources in rooms, affecting the spatial energy distribution of early reflections and late diffuse reverberation. Thus, for convincing sound field reproduction and acoustics simulation, source directivity has to be considered. Whereas perceptual effects of directivity, such as source-orientation-dependent coloration, appear relevant for the direct sound and individual early reflections, it is unclear how spectral and spatial cues interact for later reflections. Better knowledge of the perceptual relevance of source orientation cues might help to simplify the acoustics simulation. Here, it is assessed as to what extent directivity of a human speaker should be simulated for early reflections and diffuse reverberation. The computationally efficient hybrid approach to simulate and auralize binaural room impulse responses [Wendt et al., J. Audio Eng. Soc. 62, 11 (2014)] was extended to simulate source directivity. Two psychoacoustic experiments assessed the listeners' ability to distinguish between different virtual source orientations when the frequency-dependent spatial directivity pattern of the source was approximated by a direction-independent average filter for different higher reflection orders. The results indicate that it is sufficient to simulate effects of source directivity in the first-order reflections.

Frequency selectivity in the amplitude modulation (AM) domain has been demonstrated using both simultaneous AM masking and forward AM masking. This has been explained using the concept of a modulation filter bank (MFB). Here, we assessed whether the MFB occurs before or after the point of binaural interaction in the auditory pathway by using forward masking in the AM domain in an ipsilateral condition (masker AM and signal AM applied to the left ear with an unmodulated carrier in the right ear) and a contralateral condition (masker AM applied to the right ear and signal AM applied to the left ear). The carrier frequency was 8 kHz, the signal AM frequency, fs, was 40 or 80 Hz, and the masker AM frequency ranged from 0.25 to 4 times fs. Contralateral forward AM masking did occur, but it was smaller than ipsilateral AM masking. Tuning in the AM domain was slightly sharper for ipsilateral than for contralateral masking, perhaps reflecting confusion of the signal and masker AM in the ipsilateral condition when their AM frequencies were the same. The results suggest that there might be an MFB both before and after the point in the auditory pathway where binaural interaction occurs.

This paper introduces and evaluates a speech signal manipulation scheme that generates transient speech induced binaural beats (SBBs). These SBBs can only be perceived when different signals are presented dichotically (to both ears). Event-related potentials were recorded in 22 normal-hearing subjects. Dichotic stimulus presentation reliably evoked auditory late responses (ALRs) in all subjects using such manipulated signals. As control measurements, diotic stimulation modalities were presented to confirm that the ALRs were not evoked by the speech signal itself or that the signal manipulation scheme created audible artifacts. Since diotic measurements evoked no ALRs, responses from dichotic stimulation are a pure correlate of binaural interaction. While there are several auditory stimuli (mostly modulated sinusoids or noise) that share this characteristic, none of them are based on running speech. Because SBBs can be added to any arbitrary speech signal, they could easily be combined with psychoacoustic tests, for example speech reception thresholds, adding an objective measure of binaural interaction.

Reverberation is essential for the realistic auralisation of enclosed spaces. However, it can be computationally expensive to render with high fidelity and, in practice, simplified models are typically used to lower costs while preserving perceived quality. Ambisonics-based methods may be employed to this purpose as they allow us to render a reverberant sound field more efficiently by limiting its spatial resolution. The present study explores the perceptual impact of two simplifications of Ambisonics-based binaural reverberation that aim to improve efficiency. First, a "hybrid Ambisonics" approach is proposed in which the direct sound path is generated by convolution with a spatially dense head related impulse response set, separately from reverberation. Second, the reverberant virtual loudspeaker method (RVL) is presented as a computationally efficient approach to dynamically render binaural reverberation for multiple sources with the potential limitation of inaccurately simulating listener's head rotations. Numerical and perceptual evaluations suggest that the perceived quality of hybrid Ambisonics auralisations of two measured rooms ceased to improve beyond the third order, which is a lower threshold than what was found by previous studies in which the direct sound path was not processed separately. Additionally, RVL is shown to produce auralisations with comparable perceived quality to Ambisonics renderings.

Empirical data are reported demonstrating how binaural detection is affected by joint variation of masker bandwidth, masker interaural correlation, and interaural time delay (ITD) of both masker and tonal signal. Most of the data were obtained with stimuli centered at 500 Hz; supplemental measures were obtained with stimuli centered at 4 kHz. The results indicate that as the interaural correlation of the masker (ρ) is decreased there is (1) an overall increase in threshold signal-to-noise ratio (S/N) and (2) a progressively smaller effect on threshold S/N as ITD is increased. All of the data were accounted for quite accurately using the same quantitative, interaural cross-correlation-based model that was recently shown to account for binaural detection and discrimination data obtained in previous experiments. Importantly, the new data were predicted and explained using values of model parameters that were identical or very close to those found to predict accurately the earlier data. The success of the enterprise attests to the robustness of the approach and the generality of the model's ability to make accurate predictions of binaural performance over a wide range of historically important stimulus conditions.

A binaural model predicting speech intelligibility in envelope-modulated noise for normal-hearing (NH) and hearing-impaired listeners is proposed. The study shows the importance of considering an internal noise with two components relying on the individual audiogram and the level of the external stimuli. The model was optimized and verified using speech reception thresholds previously measured in three experiments involving NH and hearing-impaired listeners and sharing common methods. The anechoic target, in front of the listener, was presented simultaneously through headphones with two anechoic noise-vocoded speech maskers (VSs) either co-located with the target or spatially separated using an infinite broadband interaural level difference without crosstalk between ears. In experiment 1, two stationary noise maskers were also tested. In experiment 2, the VSs were presented at different sensation levels to vary audibility. In experiment 3, the effects of realistic interaural time and level differences were also tested. The model was applied to two datasets involving NH listeners to verify its backward compatibility. It was optimized to predict the data, leading to a correlation and mean absolute error between data and predictions above 0.93 and below 1.1 dB, respectively. The different internal noise approaches proposed in the literature to describe hearing impairment are discussed.

This study examines the influence of an early lateral reflection on spatial perceptual attributes and speech reception. To this aim, a diffuse reflection is compared with a specular one. Although diffusive surfaces have widespread applications in room acoustics design, the knowledge of the perceptual and behavioral outcomes of these surfaces has yet to be fully developed. Two experiments were conducted to investigate how the reflection type, its temporal delay, and its azimuth affect spatial percepts (source distance, width, and focus) and speech intelligibility (SI) in diffuse stationary noise. The experimental setup included ecological elements: field measurements, a speaker-like source directivity, and real flat and diffusive surfaces. The results indicate that the presence of a single diffuse reflection reduces the perceived distance of a frontal speech source and makes it clearer. SI is higher with a diffuse reflection than with a specular one. Perceptual and behavioral outcomes both depend on the angle of reflection given the frequency- and angular-dependent properties of the diffusing surface and the directivity of the speech source. The results are interpreted with reference to loudness and binaural cues and to the precedence effect. Implications of the findings for acoustic design are also discussed.

When listening to speech in the presence of concurrent talkers, listeners can benefit from glimpses that occur as a result of spectro-temporal modulations in the speech signals. These glimpses are characterized by a high local signal-to-noise ratio and allow listeners to collect relatively undistorted and reliable information on target speech features. A series of experiments was designed to measure the spatial advantage for binaurally presented speech when useful interaural time difference (ITD) information was provided only in glimpses of speech signals with otherwise ambiguous ITDs. For interaurally coherent signals, ITD information provided by target glimpses contributed substantially to the spatial advantage, but consistent target ITDs overall appeared to be of minor importance to speech intelligibility. For interaurally incoherent signals, a similarly large contribution of coherent ITD information in glimpses to the spatial advantage was not observed. Rather, target speech intelligibility depended on the interaural coherence of the interfering speech signals. While the previous observation conforms with models of auditory object formation, and the latter is consistent with equalization-cancellation theory modeling the spatial advantage, the two seem to be at odds for the presented set of experiments. A conceptual framework employing different strategies to process the perceptual foreground and background may solve this issue.

Listeners having, at most, "slight" hearing loss may exhibit deficits in binaural detection which appear to stem from increased levels of stimulus-dependent, additive internal noise [Bernstein and Trahiotis, J. Acoust. Soc. Am. 140, 3540-3548 (2016); J. Acoust. Soc. Am. 144, 292-307 (2018)]. This study reports that a small crew of such listeners also exhibits increased levels of low-level, stimulus-independent, additive internal noise. Detection thresholds were measured in: (1) the NoSπ configuration as a function of masker level; (2) the NρSπ configuration as a function of masker interaural correlation (ρ); (3) "the quiet" for So and Sπ tonal signals. Those measures were combined suitably to yield estimates of stimulus-independent, additive internal noise, separately, at center frequencies of 250, 500, and 4000 Hz. Derived levels of internal noise were found to be elevated, by about 5 dB at 250 and 500 Hz, and by about 9 dB at 4 kHz, for the group of listeners having no more than slight hearing loss and who exhibited deficits in binaural detection. The new findings, taken together with earlier investigations by the authors (which included data obtained from dozens of listeners), provide evidence that such listeners have greater levels of both stimulus-dependent and stimulus-independent, additive internal noise.

Previous studies have demonstrated that some species of odontocetes can be conditioned to reduce hearing sensitivity when warned of an impending intense sound; however, the underlying mechanisms remain poorly understood. In the present study, conditioned hearing attenuation was elicited in two bottlenose dolphins by pairing a 10-kHz tone (the conditioned stimulus) with a more intense tone (the unconditioned stimulus) at 28 kHz. Testing was performed in air, with sounds presented via contact transducers. Hearing was assessed via noninvasive measurement of monaural auditory nerve responses (ANR) and binaural auditory brainstem responses (ABR). ABRs/ANRs were measured in response to 40-kHz tone bursts, over 2 to 3-s time intervals before and after the conditioned and unconditioned stimuli. Results showed reductions in ABR/ANR amplitude and increases in latency after pairing the warning and more intense tones. Monaural ANRs from the left and right ears were attenuated by similar amounts when the warning and more intense sounds were preferentially applied to the right ear. The data support a neural mechanism operating at the level of the cochlea and/or auditory nerve and suggest the involvement of neural projections that can affect the contralateral ear.

This paper investigates classifying two target groups, surface reflectors (SR) and volume scatterers (VS), using echo envelope features. SR targets have convex surface patches that exhibit echo persistence over aspect angle, while VS targets are composed of random range-distributed and oriented reflectors producing echoes that become uncorrelated with small changes in aspect angle. The SR target group contains single-post (P1) and multiple-post (PM) types and the VS group contains Ficus benjamina (F) and Schefflera arboricola (S) foliage types with leaf areas that differ by a factor of 4. A biomimetic sonar emitting audible clicks acquired sequences of up to three binaural echoes from target views separated by 18°. Two artificial neural networks performing linear and nonlinear classification first differentiated SR/VS target groups and then P1/PM and F/S types. Classification performance improved with echo number, from a single monaural echo to three pairs of binaural echoes, demonstrating the benefit of sequential echoes. Linear and nonlinear classification of SR/VS targets achieved a minimum generalization error probability PEG = 0.003. Nonlinear P1/PM classification achieved PEG = 0.009 that was four times smaller than linear classification. Nonlinear F/S classification achieved PEG = 0.220, indicating that envelope features by themselves are inadequate to accurately differentiate foliage targets.

The impact of maskers on the receiving beam of a bottlenose dolphin, Tursiops truncatus, was investigated using the auditory evoked potential (AEP) method. The test signal was a train of tone pips with a 64 kHz carrier frequency. The stimulus produced AEPs as a sequence of waves replicating the pip rate: the rate following response (RFR). The masker was band-limited noise, with a passband of 45 to 90 kHz and a level of 105 dB re 1 μPa. Masker azimuths were at 0°, ±30°, ±60°, and ±90° relative to the head midline. The receiving beam was evaluated in terms of the RFR threshold dependence on the signal azimuth. The masked thresholds were higher than the baseline thresholds, which appeared mostly as a shift rather than a deformation in the receiving beam. The largest threshold shift appeared when the masker source was located in the most sensitive direction (zero azimuth); at lateral masker source positions, the threshold shift decreased. When the masker source was not at the head midline, the masked thresholds were higher at signal positions ipsilateral to the masker source than at positions contralateral to the masker source. The largest asymmetry was observed at the 30° masker azimuth in conjunction with the ±30° and ±120° signal azimuths; the asymmetries were 5.6 and 8.1 dB, respectively. This masking asymmetry was lower than expected from the previously found interaural intensity difference, which may be explained by the conflict between the test signal and the masker when it appeared at a binaural level of the auditory system.

In daily life, speech intelligibility is affected by masking caused by interferers and by reverberation. For a frontal target speaker and two interfering sources symmetrically placed to either side, spatial release from masking (SRM) is observed in comparison to frontal interferers. In this case, the auditory system can make use of temporally fluctuating interaural time/phase and level differences promoting binaural unmasking (BU) and better-ear glimpsing (BEG). Reverberation affects the waveforms of the target and maskers, and the interaural differences, depending on the spatial configuration and on the room acoustical properties. In this study, the effect of room acoustics, temporal structure of the interferers, and target-masker positions on speech reception thresholds and SRM was assessed. The results were compared to an optimal better-ear glimpsing strategy to help disentangle energetic masking including effects of BU and BEG as well as informational masking (IM). In anechoic and moderate reverberant conditions, BU and BEG contributed to SRM of fluctuating speech-like maskers, while BU did not contribute in highly reverberant conditions. In highly reverberant rooms a SRM of up to 3 dB was observed for speech maskers, including effects of release from IM based on binaural cues.

Limited abilities to localize sound sources and other reduced spatial hearing capabilities remain a largely unsolved issue in hearing devices like hearing aids or hear-through headphones. Hence, the impact of the microphone location, signal bandwidth, different equalization approaches, as well as processing delays in superposition with direct sound leaking through a vent was addressed in this study. A localization experiment was performed with normal-hearing subjects using individual binaural synthesis to separately assess the above-mentioned potential limiting issues for localization in the horizontal and vertical plane with linear hearing devices. To this end, listening through hearing devices was simulated utilizing transfer functions for six different microphone locations, measured both individually and on a dummy head. Results show that the microphone location is the governing factor for localization abilities with linear hearing devices, and non-optimal microphone locations have a disruptive influence on localization in the vertical domain, and an effect on lateral sound localization. Processing delays cause additional detrimental effects for lateral sound localization; and diffuse-field equalization to the open-ear response leads to better localization performance than free-field equalization. Stimuli derived from dummy head measurements are unsuited for evaluating individual localization abilities with a hearing device.

When two similar sounds are presented from different locations, with one (the lead) preceding the other (the lag) by a small delay, listeners typically report hearing one sound near the location of the lead sound source-this is called the precedence effect (PE). Several questions about the underlying mechanisms that produce the PE are asked. (1) How might listeners' relative weighting of cues at onset versus ongoing stimulus portions affect perceived lateral position of long-duration lead/lag noise stimuli? (2) What are the factors that influence this weighting? (3) Are the mechanisms invoked to explain the PE for transient stimuli applicable to long-duration stimuli? To answer these questions, lead/lag noise stimuli are presented with a range of durations, onset slopes, and lag-to-lead level ratios over headphones. Monaural, peripheral mechanisms, and binaural cue extraction are modeled to estimate the cues available for determination of perceived laterality. Results showed that all three stimulus manipulations affect the relative weighting of onset and ongoing cues and that mechanisms invoked to explain the PE for transient stimuli are also applicable to the PE, in terms of both onset and ongoing segments of long-duration, lead/lag stimuli.

A remote microphone (RM) links a talker's microphone to a listener's hearing aids (HAs). The RM improves intelligibility in noise and reverberation, but the binaural cues necessary for externalization are lost. Augmenting the RM signal with synthesized binaural cues and early reflections enhances externalization, but interactions of the RM signal with the HA processing could reduce its effectiveness. These potential interactions were evaluated using RM plus HA processing in a realistic listening simulation. The HA input was the RM alone, the augmented RM signal, the acoustic inputs at the HA microphones, including reverberation measured using a dummy head, or a mixture of the augmented RM and acoustic input signals. The HA simulation implemented linear amplification or independent dynamic-range compression at the two ears and incorporated the acoustic effects of vented earmolds. Hearing-impaired listeners scored sentence stimuli for intelligibility and rated clarity, overall quality, externalization, and apparent source width. Using the RM improved intelligibility but reduced the spatial impression. Increasing the vent diameter reduced clarity and increased the spatial impression. Listener ratings reflect a trade-off between the attributes of clarity and overall quality and the attributes of externalization and source width that can be explained using the interaural cross correlation.

For normal-hearing (NH) listeners, interaural information in both temporal envelope and temporal fine structure contribute to binaural unmasking of target signals in background noise; however, in many conditions low-frequency interaural information in temporal fine structure produces greater binaural unmasking. For bilateral cochlear-implant (CI) listeners, interaural information in temporal envelope contributes to binaural unmasking; however, the effect of encoding temporal fine structure information in electrical pulse timing (PT) is not fully understood. In this study, diotic and dichotic signal detection thresholds were measured in CI listeners using bilaterally synchronized single-electrode stimulation for conditions in which the temporal envelope was presented without temporal fine structure encoded (constant-rate pulses) or with temporal fine structure encoded (pulses timed to peaks of the temporal fine structure). CI listeners showed greater binaural unmasking at 125 pps with temporal fine structure encoded than without. There was no significant effect of encoding temporal fine structure at 250 pps. A similar pattern of performance was shown by NH listeners presented with acoustic pulse trains designed to simulate CI stimulation. The results suggest a trade-off across low rates between interaural information obtained from temporal envelope and that obtained from temporal fine structure encoded in PT.

Binaural auralization through proper room-acoustic simulation can produce a realistic listening experience as if the listener were sitting in a room with spatial perception, including enveloping reverberance. Based on analysis of experimentally measured binaural room-acoustic data, this paper discusses an approach to creating artificial but natural-sounding reverberation for binaural rendering that can be employed in simulating such an environment in an efficient way. Approaches to adjusting the spaciousness of enveloping reverberance within the context of artificially generated reverberation are investigated via hearing tests. This paper exploits the excellent pseudorandom properties of maximum-length sequences to generate deterministic and controllable decorrelations between binaural channels for artificial reverberation for room-acoustic simulations with high computational efficiency. To achieve natural-sounding enveloping reverberance in an enclosed space, and thereby an immersive environment, the shapes of both the reverberation energy decays and the spatial characteristics are found to be decisive. This paper discusses systematic hearing test results that support the mentioned finding.

A sonar cognitive map displays target components that are specified by signal features extracted from a single binaural echo pair. A biomimetic audible sonar probes targets configured using posts connected by tangential planes. Echo envelopes are processed to extract values of eight parameters that govern the mapping process. Being tuned to recognize posts and planes, a cognitive map is composed of these two components using the posts' centers and radii as landmarks. A platform with translational and rotational degrees of freedom implements a landmark-centric scanning trajectory whose step size adaptively changes with echo information. The sonar tracks the target surface by maintaining a constant first-echo arrival time and by equalizing binaural echo times to form singular echoes that identify landmarks. The mapping process employs five states from detection to termination that pass through the singular echo state. Separate states process echo interference caused by two posts and echoes from planar surfaces. Sonar scanning stops when the current landmark parameters match those of the first landmark. Two targets configured with three posts and an added plane illustrate the procedure. Cognitive maps exhibit landmark locations that are accurate to ±5% with post radius estimates accurate to ±20%.

Adults with bilateral cochlear implants (BiCIs) receive benefits in localizing stationary sounds when listening with two implants compared with one; however, sound localization ability is significantly poorer when compared to normal hearing (NH) listeners. Little is known about localizing sound sources in motion, which occurs in typical everyday listening situations. The authors considered the possibility that sound motion may improve sound localization in BiCI users by providing multiple places of information. Alternatively, the ability to compare multiple spatial locations may be compromised in BiCI users due to degradation of binaural cues, and thus result in poorer performance relative to NH adults. In this study, the authors assessed listeners' abilities to distinguish between sounds that appear to be moving vs stationary, and track the angular range and direction of moving sounds. Stimuli were bandpass-filtered (150-6000 Hz) noise bursts of different durations, panned over an array of loudspeakers. Overall, the results showed that BiCI users were poorer than NH adults in (i) distinguishing between a moving vs stationary sound, (ii) correctly identifying the direction of movement, and (iii) tracking the range of movement. These findings suggest that conventional cochlear implant processors are not able to fully provide the cues necessary for perceiving auditory motion correctly.

Preserved low-frequency acoustic hearing in cochlear implant (CI) recipients affords combined electric-acoustic stimulation (EAS) that could improve access to low-frequency acoustic binaural cues and enhance spatial hearing. Such benefits, however, could be undermined by interactions between electrical and acoustical inputs to adjacent (spectral overlap) or distant (binaural interference) cochlear places in EAS. This study simulated EAS in normal-hearing listeners, measuring interaural time difference (ITD) and interaural level difference (ILD) discrimination thresholds for a low-frequency noise (simulated acoustic target) in the presence or absence of a pulsatile high-frequency complex presented monotically or diotically (simulated unilateral or bilateral electric distractor). Unilateral distractors impaired thresholds for both cue types, suggesting influences of both binaural interference (which appeared more consistently for ITD than ILD) and physical spectral overlap (for both cue types). Reducing spectral overlap with an EAS gap between 1 and 3 kHz consistently improved binaural sensitivity. Finally, listeners displayed significantly lower thresholds with simulated bilateral versus unilateral electric stimulation. The combined effects revealed similar or better thresholds in bilateral full spectral overlap than in unilateral EAS gap conditions, suggesting that bilateral implantation with bilateral acoustic hearing preservation could allow for higher tolerance of spectral overlap in CI users and improved binaural sensitivity over unilateral EAS.

Accurate perception of binaural cues is essential for left-right sound localization. Much literature focuses on threshold measures of perceptual acuity and accuracy. This study focused on supra-threshold perception using an anticipatory eye movement (AEM) paradigm designed to capture subtle aspects of perception that might not emerge in behavioral-motor responses, such as the accumulation of certainty, and rapid revisions in decision-making. Participants heard interaural timing differences (ITDs) or interaural level differences in correlated or uncorrelated narrowband noises, respectively. A cartoon ball moved behind an occluder and then emerged from the left or right side, consistent with the binaural cue. Participants anticipated the correct answer (before it appeared) by looking where the ball would emerge. Results showed quicker and more steadfast gaze fixations for stimuli with larger cue magnitudes. More difficult stimuli elicited a wider distribution of saccade times and greater number of corrective saccades before final judgment, implying perceptual uncertainty or competition. Cue levels above threshold elicited some wrong-way saccades that were quickly corrected. Saccades to ITDs were earlier and more reliable for low-frequency noises. The AEM paradigm reveals the time course of uncertainty and changes in perceptual decision-making for supra-threshold binaural stimuli even when behavioral responses are consistently correct.

Pitch is a fundamental attribute in auditory perception involved in source identification and segregation, music, and speech understanding. Pitch percepts are intimately related to harmonic resolvability of sound. When harmonics are well-resolved, the induced pitch is usually salient and precise, and several models relying on autocorrelations or harmonic spectral templates can account for these percepts. However, when harmonics are not completely resolved, the pitch percept becomes less salient, poorly discriminated, with upper range limited to a few hundred hertz, and spectral templates fail to convey percept since only temporal cues are available. Here, a biologically-motivated model is presented that combines spectral and temporal cues to account for both percepts. The model explains how temporal analysis to estimate the pitch of the unresolved harmonics is performed by bandpass filters implemented by resonances in dendritic trees of neurons in the early auditory pathway. It is demonstrated that organizing and exploiting such dendritic tuning can occur spontaneously in response to white noise. This paper then shows how temporal cues of unresolved harmonics may be integrated with spectrally resolved harmonics, creating spectro-temporal harmonic templates for all pitch percepts. Finally, the model extends its account of monaural pitch percepts to pitches evoked by dichotic binaural stimuli.

Auditory spatial perception relies on more than one spatial cue. This study investigated the effects of cue congruence on auditory localization and the extent of visual bias between two binaural cues-interaural time differences (ITDs) and interaural level differences (ILDs). Interactions between these binaural cues were manipulated by stereophonic techniques. The results show that incoherent binaural information increased auditory response noise and amplified visual bias. The analysis further suggests that although ILD is not the dominant cue for low-frequency localization, it may strengthen the position estimate by combining with the dominant ITD information to minimize estimation noise.

In contrast to the well-known benefits for speech intelligibility, the advantage afforded by binaural stimulus presentation for reducing listening effort has not been thoroughly examined. This study investigated spatial release of listening effort and its relation to binaural speech intelligibility in listeners with normal hearing. Psychometric functions for speech intelligibility of a frontal target talker masked by a stationary speech-shaped noise were estimated for several different noise azimuths, different degrees of reverberation, and by maintaining only interaural level or time differences. For each of these conditions, listening effort was measured using a categorical scaling procedure. The results revealed that listening effort was significantly reduced when target and masker were spatially separated in anechoic conditions. This effect extended well into the range of signal-to-noise ratios (SNRs) in which speech intelligibility was at ceiling, and disappeared only at the highest SNRs. In reverberant conditions, spatial release from listening effort was observed for high, but not low, direct-to-reverberant ratios. The findings suggest that listening effort assessment can be a useful method for revealing the benefits of spatial separation of sources under realistic listening conditions comprising favorable SNRs and low reverberation, which typically are not apparent by other means.

International audience; Mammals use binaural or monaural (spectral) cues to localize acoustic sources. While the sensitivity of terrestrial mammals to changes in source elevation is relatively poor, the accuracy achieved by the odontocete cetaceans' biosonar is high, independently of where the source is. Binaural/spectral cues are unlikely to account for this remarkable skill. In this paper, bone-conducted sound in a dolphin's mandible is studied, investigating its possible contribution to sound localization. Experiments are conducted in a water tank by deploying, on the horizontal and median planes of the skull, ultrasound sources that emit synthetic clicks between 45 and 55 kHz. Elastic waves propagating through the mandible are measured at the pan bones and used to localize source positions via either binaural cues or a correlation-based full-waveform algorithm. Exploiting the full waveforms and, most importantly, reverberated coda, it is possible to enhance the accuracy of source localization in the vertical plane and achieve similar resolution of horizontal- vs vertical-plane sources. The results noted in this paper need to be substantiated by further experimental work, accounting for soft tissues and making sure that the data are correctly mediated to the internal ear. If confirmed, the results would favor the idea that dolphin's echolocation skills rely on the capability to analyze the coda of biosonar echoes.

Binaural loudness balancing is performed in research and clinical practice when fitting bilateral hearing devices, and is particularly important for bimodal listeners, who have a bilateral combination of a hearing aid and a cochlear implant. In this study, two psychophysical binaural loudness balancing procedures were compared. Two experiments were carried out. In the first experiment, the effect of procedure (adaptive or adjustment) on the balanced loudness levels was investigated using noise band stimuli, of which some had a frequency shift to simulate bimodal hearing. In the second experiment, the adjustment procedure was extended. The effect of the starting level of the adjustment procedure was investigated and the two procedures were again compared for different reference levels and carrier frequencies. Fourteen normal hearing volunteers participated in the first experiment, and 38 in the second experiment. Although the final averaged loudness balanced levels of both procedures were similar, the adjustment procedure yielded smaller standard deviations across four test sessions. The results of experiment 2 demonstrated that in order to avoid bias, the adjustment procedure should be conducted twice, once starting from below and once from above the expected balanced loudness level. ispartof: Journal Of The Acoustical Society Of America vol:143 issue:6 pages:3720-3729 ispartof: location:United States status: published

In a remote microphone (RM) system, a talker speaks into a microphone and the signal is transmitted to the hearing aids worn by the hearing-impaired listener. A difficulty with remote microphones, however, is that the signal received at the hearing aid bypasses the head and pinna, so the acoustic cues needed to externalize the sound source are missing. The objective of this paper is to process the RM signal to improve externalization when listening through earphones. The processing is based on a structural binaural model, which uses a cascade of processing modules to simulate the interaural level difference, interaural time difference, pinna reflections, ear-canal resonance, and early room reflections. The externalization results for the structural binaural model are compared to a left-right signal blend, the listener's own anechoic head-related impulse response (HRIR), and the listener's own HRIR with room reverberation. The azimuth is varied from straight ahead to 90° to one side. The results show that the structural binaural model is as effective as the listener's own HRIR plus reverberation in producing an externalized acoustic image, and that there is no significant difference in externalization between hearing-impaired and normal-hearing listeners.

Masked detection thresholds were measured for a noise target in the presence of a masker composed of (1) a lead/lag noise pair with the lead interaural time difference (ITD) set the same or opposite to the target, (2) a diotic masker, and (3) a dichotic pair of decorrelated noises. If the precedence effect actually eliminates a second, later arriving stimulus, a spatial release from masking would be expected when the lead ITD is opposite that of the target. Results for a range of lead/lag delays suggest that the precedence effect is not the result of a perceptual removal of the lag.

In a complex auditory scene, signals of interest can be distinguished from masking sounds by differences in source location [spatial release from masking (SRM)] and by differences between masker-alone and masker-plus-signal envelopes. This study investigated interactions between those factors in release of masking of 700-Hz tones in an open sound field. Signal and masker sources were colocated in front of the listener, or the signal source was shifted 90° to the side. In Experiment 1, the masker contained a 25-Hz-wide on-signal band plus flanking bands having envelopes that were either mutually uncorrelated or were comodulated. Comodulation masking release (CMR) was largely independent of signal location at a higher masker sound level, but at a lower level CMR was reduced for the lateral signal location. In Experiment 2, a brief signal was positioned at the envelope maximum (peak) or minimum (dip) of a 50-Hz-wide on-signal masker. Masking was released in dip more than in peak conditions only for the 90° signal. Overall, open-field SRM was greater in magnitude than binaural masking release reported in comparable closed-field studies, and envelope-related release was somewhat weaker. Mutual enhancement of masking release by spatial and envelope-related effects tended to increase with increasing masker level.

Interaural differences are important for listeners to be able to maintain focus on a sound source of interest in the presence of multiple sources. Because interaural differences are sound localization cues, most binaural-cue-based source separation algorithms attempt separation by localizing each time-frequency (T-F) unit to one of the possible source directions using interaural differences. By assembling T-F units that are assigned to one direction, the sound stream from that direction is enhanced. In this paper, a different type of binaural cue for source-separation purposes is proposed. For each T-F unit, the target-direction signal is cancelled by applying the equalization-cancellation (EC) operation to cancel the signal from the target direction; then, the dominance of the target in each T-F unit is determined by the effectiveness of the cancellation. Specifically, the energy change from cancellation is used as the criterion for target dominance for each T-F unit. Source-separation performance using the target-EC cue is compared with performance using localization cues. With simulated multi-talker and diffuse-babble interferers, the algorithm based on target-EC cues yields better source-separation performance than the algorithm based on localization cues, both in direct comparison with the ideal binary mask and in measured speech intelligibility for the separated target streams.

The COVID-19 pandemic has resulted in widespread use of facial masks that cover the nose and mouth areas to prevent viral transmission. While masks have helped provide protection from spreading illness, they have also had an impact on how people communicate, with many finding it more difficult to understand speech from those who are wearing masks. Facial masks affect the directivity and sound levels of the wearer’s verbal communication, and if opaque, they also cover visual cues that aid understanding. Studies have documented the effect of different mask types on speech intelligibility, speech transmission index, talker directivity, and resulting speech features. This talk presents the first of measurements towards composing a corpus of impulse responses gathered between two head and torso simulators (HATS) in an anechoic chamber at varying directional positions and distances. One HATS with a mouth simulator serves as the talker and is outfitted with different mask types, including cloth, N95, and surgical paper masks, in combination with face shields. The resulting impulse response data base may be used by researchers to simulate masked verbal communication scenarios such as in virtual soundfield environments, allowing systematic testing of how masks impact speech perception in numerous situations.

Military operations often require Service Members (SMs) to make life-and-death decisions based on information they can only obtain auditorily. However, the wide variety of critical sounds, essential spoken messages, and masking environments encountered while performing military missions makes it difficult to use clinical tests to predict the operational performance of hearing-impaired SMs. In this study, ≈2400 SMs performed listening tasks derived from binaural recordings made during realistic battlefield exercises at the US Army Joint Readiness Training Center. The listening tasks included standardized speech recognition tasks based on the Oldenburg Matrix Test and Modified Rhyme Test (MRT), as well as scenario-based multiple-choice tasks where questions were asked about unmodified verbal exchanges that occurred between SMs engaged in the battlefield simulation. The scores SMs achieved on these operational tasks were then compared to their hearing thresholds and to their scores on the MRT80. The results show that an auditory fitness-for-duty standard that requires individuals with thresholds that exceed the values defined in the new US Army H3 Profile to obtain a minimum score on the MRT80 is generally able to identify individuals who are most at risk of performing abnormally poorly on militarily relevant listening tasks. [The views expressed in this abstract are those of the authors and do not reflect the official policy of the Department of the Army, Department of Defense, or U.S. Government.]

In this presentation, we describe the Binaural Filterbank Hearing Aid Simulation (BFHAS) which is a MATLAB-based package that simulates commonly available hearing aid signal processing. This package allows the user to process custom audio signals (e.g., noisy speech; music) with a suite of hearing aid algorithms. Linear or non-linear fitting prescription fitting formulae can be used for gain compensation. The hearing-aid algorithms include spectral-subtraction noise suppression, wide dynamic-range compression (WDRC), and frequency compression based on sinusoidal modeling of the high frequencies of the signal. Users can vary signal processing settings like number of bands; add/modify spectral subtraction; gains, knee points, attack and release times for WDRC; compression ratios and cutoffs for frequency compression etc. The simulation provides the processed hearing aid output that includes the compensation and hearing aid quality and intelligibility metrics for the input stimuli. Future updates will include the room-simulations and built-in noise generators to simulate different levels of signal degradation. We also provide a user guide and examples illustrating the use of different package modules. This package will be made available for download. [Work supported by NIH R01 DC012289 and by a grant from GN ReSound to University of Colorado.]

This study used prepulse inhibition (PPI) of the acoustic startle reflex as a physiological measure to determine the sensitivity to a break in interaural correlation (BIC) in binaural gap detection and compared this physiological measure with the traditional behavioral measure. Results show that BIC could be used as a prepulse to induce PPI at different interaural delays. Furthermore, the PPIs induced by BIC at an interaural delay of 4 ms, but not 0, 2, or 8 ms, were correlated with the BIC delay threshold. These findings suggest that the BIC-induced PPI paradigm may serve as an objective measure of binaural gap detection.

This article deals with the assessment and prediction of the reproduction quality when binaurally auralizing spherical microphone array data for room simulation applications. The auralization is perceptually assessed in a listening experiment using the two attributes, apparent source width (ASW) and listener envelopment (LEV), for spatial quality description, whereas the technical analysis employs a psychoacoustically motivated model for room acoustical perception (RAP) which is specifically designed to estimate ASW and LEV. Both analyses focus on the array configuration in terms of varying modal resolutions and its influence on the spatial reproduction quality. The auralizations comprise three simulated environments, i.e., free-field sound fields as well as a dry and a reverberant room. Ten different audio signals are used as test material. Perceptual results show that the array configuration strongly influences the perception of ASW and LEV which also depends on the reflection properties of the simulated room. The ASW and LEV predictions by the RAP model correlate well with the results from the listening experiment.

Experimental binaural masking-pattern data are presented together with model simulations for 12- and 600-ms signals. The masker was a diotic 11-Hz wide noise centered on 500 Hz. The tonal signal was presented either diotically or dichotically (180° interaural phase difference) with frequencies ranging from 400 to 600 Hz. The results and the modeling agree with previous data and hypotheses; simulations with a binaural model sensitive to monaural modulation cues show that the effect of duration on off-frequency binaural masking-level differences is mainly a result of modulation cues which are only available in the monaural detection of long signals.

Both bimodal cochlear implant and bilateral hearing aid users can exhibit broad binaural pitch fusion, the fusion of dichotically presented tones over a broad range of pitch differences between ears [Reiss, Ito, Eggleston, and Wozny. (2014). J. Assoc. Res. Otolaryngol. 15(2), 235–248; Reiss, Eggleston, Walker, and Oh. (2016). J. Assoc. Res. Otolaryngol. 17(4), 341–356; Reiss, Shayman, Walker, Bennett, Fowler, Hartling, Glickman, Lasarev, and Oh. (2017). J. Acoust. Soc. Am. 143(3), 1909–1920]. Further, the fused binaural pitch is often a weighted average of the different pitches perceived in the two ears. The current study was designed to systematically measure these pitch averaging phenomena in bilateral hearing aid users with broad fusion. The fused binaural pitch of the reference-pair tone combination was initially measured by pitch-matching to monaural comparison tones presented to the pair tone ear. The averaged results for all subjects showed two distinct trends: (1) The fused binaural pitch was dominated by the lower-pitch component when the pair tone was either 0.14 octaves below or 0.78 octaves above the reference tone; (2) pitch averaging occurred when the pair tone was between the two boundaries above, with the most equal weighting at 0.38 octaves above the reference tone. Findings from two subjects suggest that randomization or alternation of the comparison ear can eliminate this asymmetry in the pitch averaging range. Overall, these pitch averaging phenomena suggest that spectral distortions and thus binaural interference may arise during binaural stimulation in hearing-impaired listeners with broad fusion.

Speech reception thresholds (SRTs) decrease as target and maskers are spatially separated (spatial release from masking, SRM). The current study systematically assessed how SRTs and SRM for a frontal target in a spatially symmetric masker configuration depend on spectro-temporal masker properties, the availability of short-time interaural level difference (ILD) and interaural time difference (ITD), and informational masking. Maskers ranged from stationary noise to single, interfering talkers and were modified by head-related transfer functions to provide: (i) different binaural cues (ILD, ITD, or both) and (ii) independent maskers in each ear ("infinite ILD"). Additionally, a condition was tested in which only information from short-time spectro-temporal segments of the ear with a favorable signal-to-noise ratio (better-ear glimpses) was presented. For noise-based maskers, ILD, ITD, and spectral changes related to masker location contributed similarly to SRM, while ILD cues played a larger role if temporal modulation was introduced. For speech maskers, glimpsing and perceived location contributed roughly equally and ITD contributed less. The "infinite ILD" condition might suggest better-ear glimpsing limitations resulting in a maximal SRM of 12 dB for maskers with low or absent informational masking. Comparison to binaural model predictions highlighted the importance of short-time processing and helped to clarify the contribution of the different binaural cues and mechanisms.

This study introduces a model for solving three different auditory tasks in a multi-talker setting: target localization, target identification, and word recognition. The model was used to simulate psychoacoustic data from a call-sign-based listening test involving multiple spatially separated talkers [Brungart and Simpson (2007). Percept. Psychophys. 69(1), 79-91]. The main characteristics of the model are (i) the extraction of salient auditory features ("glimpses") from the multi-talker signal and (ii) the use of a classification method that finds the best target hypothesis by comparing feature templates from clean target signals to the glimpses derived from the multi-talker mixture. The four features used were periodicity, periodic energy, and periodicity-based interaural time and level differences. The model results widely exceeded probability of chance for all subtasks and conditions, and generally coincided strongly with the subject data. This indicates that, despite their sparsity, glimpses provide sufficient information about a complex auditory scene. This also suggests that complex source superposition models may not be needed for auditory scene analysis. Instead, simple models of clean speech may be sufficient to decode even complex multi-talker scenes.

Children who are deaf and receive bilateral cochlear implants (BiCIs) perform better on spatial hearing tasks using bilateral rather than unilateral inputs; however, they underperform relative to normal-hearing (NH) peers. This gap in performance is multi-factorial, including the inability of speech processors to reliably deliver binaural cues. Although much is known regarding binaural sensitivity of adults with BiCIs, less is known about how the development of binaural sensitivity in children with BiCIs compared to NH children. Sixteen children (ages 9–17 years) were tested using synchronized research processors. Interaural time differences and interaural level differences (ITDs and ILDs, respectively) were presented to pairs of pitch-matched electrodes. Stimuli were 300-ms, 100-pulses-per-second, constant-amplitude pulse trains. In the first and second experiments, discrimination of interaural cues (either ITDs or ILDs) was measured using a two-interval left/right task. In the third experiment, subjects reported the perceived intracranial position of ITDs and ILDs in a lateralization task. All children demonstrated sensitivity to ILDs, possibly due to monaural level cues. Children who were born deaf had weak or absent sensitivity to ITDs; in contrast, ITD sensitivity was noted in children with previous exposure to acoustic hearing. Therefore, factors such as auditory deprivation, in particular, lack of early exposure to consistent timing differences between the ears, may delay the maturation of binaural circuits and cause insensitivity to binaural differences.