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Bioinspired methods for sound source localization offer opportunities for resource reduction as well as concurrent performance improvement in contrast to conventional techniques. Usually, sound source localization requires a large number of microphones arranged in irregular geometries, and thus has high resource requirements in terms of space and data processing. Motivated by biology and using digital signal processing methods, an approach that adapts the coupled hearing system of the fly Ormia ochracea with a minimally distant two-microphone array is presented. Despite its physiology, the fly is able to overcome physical limitations in localizing low-frequency sound sources. By exploiting the filtering effect of the coupling system, the direction-of-arrival of the sound is determined with two microphones at an intermediate distance of 0.06 m. For conventional beamforming algorithms, these physical limitations would result in degraded localization performance. In this work, the bioinspired coupling system is analyzed and subsequently parameterized direction-sensitive for different directions of incidence of the sound. For the parameterization, an optimization method is presented which can be adopted for excitation with plane as well as spherical sound wave propagation. Finally, the method was assessed using simulated and measured data. For 90% of the simulated scenarios, the correct direction of incidence could be determined with an accuracy of less than 1 ∘ despite the use of a minimal distant two-microphone array. The experiments with measured data also resulted in a correct determination of the direction of incidence, which qualifies the bioinspired method for practical use in digital hardware systems., (Creative Commons Attribution license.)

Acoustic dyadic sensors (ADSs) are a new type of acoustic sensor with higher directivity than microphones and acoustic vector sensors, which has great application potential in the fields of sound source localization and noise cancellation. However, the high directivity of an ADS is seriously affected by the mismatches between its sensitive units. In this article, (1) a theoretical model of mixed mismatches was established based on the finite-difference approximation model of uniaxial acoustic particle velocity gradient and its ability to reflect the actual mismatches was proven by the comparison of theoretical and experimental directivity beam patterns of an actual ADS based on MEMS thermal particle velocity sensors. (2) Additionally, a quantitative analysis method based on directivity beam pattern was proposed to easily estimate the specific magnitude of the mismatches, which was proven to be useful for the design of ADSs to estimate the magnitudes of different mismatches of an actual ADS. (3) Moreover, a correction algorithm based on the theoretical model of mixed mismatches and quantitative analysis method was successfully demonstrated to correct several groups of simulated and measured beam patterns with mixed mismatches.

In modern applications such as robotics, autonomous vehicles, and speaker localization, the computational power for sound source localization applications can be limited when other functionalities get more complex. In such application fields, there is a need to maintain high localization accuracy for several sound sources while reducing computational complexity. The array manifold interpolation (AMI) method applied with the Multiple Signal Classification (MUSIC) algorithm enables sound source localization of multiple sources with high accuracy. However, the computational complexity has so far been relatively high. This paper presents a modified AMI for uniform circular array (UCA) that offers reduced computational complexity compared to the original AMI. The complexity reduction is based on the proposed UCA-specific focusing matrix which eliminates the calculation of the Bessel function. The simulation comparison is done with the existing methods of iMUSIC, the Weighted Squared Test of Orthogonality of Projected Subspaces (WS-TOPS), and the original AMI. The experiment result under different scenarios shows that the proposed algorithm outperforms the original AMI method in terms of estimation accuracy and up to a 30% reduction in computation time. An advantage offered by this proposed method is the ability to implement wideband array processing on low-end microprocessors.

Sound source localization is crucial for communication and sound scene analysis. This study uses direction-of-arrival estimates of multiple ad hoc distributed microphone arrays to localize sound sources in a room. An affine mapping between the independent array estimates and the source coordinates is derived from a set of calibration points. Experiments show that the affine model is sufficient to locate a source and can be calibrated to physical dimensions. A projection of the local array estimates increases localization accuracy, particularly further away from the calibrated region. Localization tests in three dimensions compare the affine approach to a nonlinear neural network.

Acoustic images are an emergent data modality for multimodal scene understanding. Such images have the peculiarity of distinguishing the spectral signature of the sound coming from different directions in space, thus providing a richer information as compared to that derived from single or binaural microphones. However, acoustic images are typically generated by cumbersome and costly microphone arrays which are not as widespread as ordinary microphones. This paper shows that it is still possible to generate acoustic images from off-the-shelf cameras equipped with only a single microphone and how they can be exploited for audio-visual scene understanding. We propose three architectures inspired by Variational Autoencoder, U-Net and adversarial models, and we assess their advantages and drawbacks. Such models are trained to generate spatialized audio by conditioning them to the associated video sequence and its corresponding monaural audio track. Our models are trained using the data collected by a microphone array as ground truth. Thus they learn to mimic the output of an array of microphones in the very same conditions. We assess the quality of the generated acoustic images considering standard generation metrics and different downstream tasks (classification, cross-modal retrieval and sound localization). We also evaluate our proposed models by considering multimodal datasets containing acoustic images, as well as datasets containing just monaural audio signals and RGB video frames. In all of the addressed downstream tasks we obtain notable performances using the generated acoustic data, when compared to the state of the art and to the results obtained using real acoustic images as input.

Normal-hearing listeners adapt to alterations in sound localization cues. This adaptation can result from the establishment of a new spatial map of the altered cues or from a stronger relative weighting of unaltered compared to altered cues. Such reweighting has been shown for monaural vs. binaural cues. However, studies attempting to reweight the two binaural cues, interaural differences in time (ITD) and level (ILD), yielded inconclusive results. This study investigated whether binaural-cue reweighting can be induced by lateralization training in a virtual audio-visual environment. Twenty normal-hearing participants, divided into two groups, completed the experiment consisting of 7 days of lateralization training, preceded and followed by a test measuring the binaural-cue weights. Participants' task was to lateralize 500-ms bandpass-filtered (2-4 kHz) noise bursts containing various combinations of spatially consistent and inconsistent binaural cues. During training, additional visual cues reinforced the azimuth corresponding to ITDs in one group and ILDs in the other group and the azimuthal ranges of the binaural cues were manipulated group-specifically. Both groups showed a significant increase of the reinforced-cue weight from pre- to posttest, suggesting that participants reweighted the binaural cues in the expected direction. This reweighting occurred within the first training session. The results are relevant as binaural-cue reweighting likely occurs when normal-hearing listeners adapt to new acoustic environments. Reweighting might also be a factor underlying the low contribution of ITDs to sound localization of cochlear-implant listeners as they typically do not experience reliable ITD cues with clinical devices., (© 2021. The Author(s).)

Beamforming is a type of audio array processing techniques used for interference reduction, sound source localization, and as pre-processing stage for audio event classification and speaker identification. The auditory scene analysis community can benefit from a systemic evaluation and comparison between different beamforming techniques. In this paper, five popular beamforming techniques are evaluated in two different acoustic environments, while varying the number of microphones, the number of interferences, and the direction-of-arrival error, by using the Acoustic Interactions for Robot Audition (AIRA) corpus and a common software framework. Additionally, a highly efficient phase-based frequency masking beamformer is also evaluated, which is shown to outperform all five techniques. Both the evaluation corpus and the beamforming implementations are freely available and provided for experiment repeatability and transparency. Raw results are also provided as a complement to this work to the reader, to facilitate an informed decision of which technique to use. Finally, the insights and tendencies observed from the evaluation results are presented.

To perform adaptive behaviours, animals have to establish a representation of the physical "outside" world. How these representations are created by sensory systems is a central issue in sensory physiology. This review addresses the history of experimental approaches toward ideas about sensory coding, using the relatively simple auditory system of acoustic insects. I will discuss the empirical evidence in support of Barlow's "efficient coding hypothesis", which argues that the coding properties of neurons undergo specific adaptations that allow insects to detect biologically important acoustic stimuli. This hypothesis opposes the view that the sensory systems of receivers are biased as a result of their phylogeny, which finally determine whether a sound stimulus elicits a behavioural response. Acoustic signals are often transmitted over considerable distances in complex physical environments with high noise levels, resulting in degradation of the temporal pattern of stimuli, unpredictable attenuation, reduced signal-to-noise levels, and degradation of cues used for sound localisation. Thus, a more naturalistic view of sensory coding must be taken, since the signals as broadcast by signallers are rarely equivalent to the effective stimuli encoded by the sensory system of receivers. The consequences of the environmental conditions for sensory coding are discussed.

Unlike other predators that use vision as their primary sensory system, bats compute the three-dimensional (3D) position of flying insects from discrete echo snapshots, which raises questions about the strategies they employ to track and intercept erratically moving prey from interrupted sensory information. Here, we devised an ethologically inspired behavioral paradigm to directly test the hypothesis that echolocating bats build internal prediction models from dynamic acoustic stimuli to anticipate the future location of moving auditory targets. We quantified the direction of the bat's head/sonar beam aim and echolocation call rate as it tracked a target that moved across its sonar field and applied mathematical models to differentiate between nonpredictive and predictive tracking behaviors. We discovered that big brown bats accumulate information across echo sequences to anticipate an auditory target's future position. Further, when a moving target is hidden from view by an occluder during a portion of its trajectory, the bat continues to track its position using an internal model of the target's motion path. Our findings also reveal that the bat increases sonar call rate when its prediction of target trajectory is violated by a sudden change in target velocity. This shows that the bat rapidly adapts its sonar behavior to update internal models of auditory target trajectories, which would enable tracking of evasive prey. Collectively, these results demonstrate that the echolocating big brown bat integrates acoustic snapshots over time to build prediction models of a moving auditory target's trajectory and enable prey capture under conditions of uncertainty., Competing Interests: The authors declare no competing interest.

Acoustic backscatter measurements were conducted on a stationary harbor porpoise (Phocoena phocoena) under controlled conditions. The measurements were made with the porpoise in the broadside aspect using three different types of signals: (1) a 475 μs linear frequency-modulated (FM) pulse with a frequency range from 23 to 160 kHz; (2) a simulated bottlenose dolphin (Tursiops "truncates") click with a peak frequency of 120 kHz; and (3) a simulated killer whale (Orcinus orca) click with a peak frequency of 60 kHz. The measurement with the FM pulse indicated that the mean target strength at the broadside aspect decreased from -26 to -50 dB as the frequency increased from 23 to 120 kHz in a nearly linear fashion (on a logarithm plot). Target strength variation with frequency was similar to a previous backscatter measurement on a bottlenose dolphin over a comparable frequency range (23-80 kHz). The porpoise seems to be a stealth body with low backscatter properties. The target strength of the porpoise was also about 15-16 dB lower than that of the bottlenose dolphin. The difference in lung volume of the two species when expressed in dB was also approximately 15 dB.

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%.

Echolocating bats exhibit sophisticated sonar behaviors using ultrasounds with actively adjusted acoustic characteristics (e.g., frequency and time-frequency structure) depending on the situation. In this study, the utility of ultrasound in human echolocation was examined. By listening to ultrasonic echoes with a shifted pitch to be audible, the participants (i.e., sighted echolocation novices) could discriminate the three-dimensional (3D) roundness of edge contours. This finding suggests that sounds with suitable wavelengths (i.e., ultrasounds) can provide useful information about 3D shapes. In addition, the shape, texture, and material discrimination experiments were conducted using ultrasonic echoes binaurally measured with a 1/7 scaled miniature dummy head. The acoustic and statistical analyses showed that intensity and timbre cues were useful for shape and texture discriminations, respectively. Furthermore, in the discrimination of objects with various features (e.g., acrylic board and artificial grass), the perceptual distances between objects were more dispersed when frequency-modulated sweep signals were used than when a constant-frequency signal was used. These suggest that suitable signal design, i.e., echolocation sounds employed by bats, allowed echolocation novices to discriminate the 3D shape and texture. This top-down approach using human subjects may be able to efficiently help interpret the sensory perception, "seeing by sound," in bat biosonar.

Passive acoustic monitoring of wildlife requires sound recording systems. Several cheap, high-performance open-source solutions currently exist for recording soundscapes, but all of them are still reliant on commercial microphones. Commercial microphones are relatively expensive, specialized for particular taxa, and often have incomplete technical specifications. We designed Sonitor, an open-source microphone system to address all needs of ecologists that sample terrestrial wildlife acoustically. We evaluated the cost and durability of our system and measured trade-offs that are seldom acknowledged but which universally limit microphones' functions: weatherproofing versus sound attenuation, windproofing versus transmission loss after rain, signal loss in long cables, and analog sound amplification versus directivity with acoustic horns. We propose five microphone configurations suiting different budgets (from 8 to 33 EUR per unit), and fulfilling different sound quality and flexibility requirements. The Sonitor system consists of sturdy acoustic sensors that cover the entire sound frequency spectrum of sonant terrestrial wildlife at a fraction of the cost of commercial microphones., Competing Interests: No competing interests were disclosed., (Copyright: © 2019 Darras K et al.)

It is well known that reverberation plays an important role in perceived externalization of three-dimensional audio over headphones. In the case of the externalization of a frontal sound source, the reverberation heard by both ears is equally important. Relatively little is known about the relative influence of reverberation at the contralateral versus the ipsilateral ear on perceived externalization of a lateral sound source. For a lateral sound source, the direct sound energy is much higher at the ipsilateral ear than at the contralateral ear due to the shadowing effect, which dominates at mid to high frequencies. The reverberant energy does not have a major difference between two ears. Therefore, the direct-to-reverberant energy ratio is much lower for the contralateral ear than for the ipsilateral ear. In addition, the frequency-to-frequency variability, which describes the frequency variability in the magnitude spectrum, is more pronounced in the contralateral ear than in the ipsilateral ear. The results of two listening experiments and the analysis of short-term binaural cues suggested that the reverberation at the contralateral ear has more influence on perceived externalization of a lateral sound source than that at the ipsilateral ear.

Speech localization and enhancement involves sound source mapping and reconstruction from noisy recordings of speech mixtures with microphone arrays. Conventional beamforming methods suffer from low resolution, especially with a limited number of microphones. In practice, there are only a few sources compared to the possible directions-of-arrival (DOA). Hence, DOA estimation is formulated as a sparse signal reconstruction problem and solved with sparse Bayesian learning (SBL). SBL uses a hierarchical two-level Bayesian inference to reconstruct sparse estimates from a small set of observations. The first level derives the posterior probability of the complex source amplitudes from the data likelihood and the prior. The second level tunes the prior towards sparse solutions with hyperparameters which maximize the evidence, i.e., the data probability. The adaptive learning of the hyperparameters from the data auto-regularizes the inference problem towards sparse robust estimates. Simulations and experimental data demonstrate that SBL beamforming provides high-resolution DOA maps outperforming traditional methods especially for correlated or non-stationary signals. Specifically for speech signals, the high-resolution SBL reconstruction offers not only speech enhancement but effectively speech separation.

Introduction: This study compares a parabolic microphone to unaided hearing in detecting and comprehending hidden callers at ranges of 322 to 2510 m., Methods: Eight subjects were placed 322 to 2510 m away from a central listening point. The subjects were concealed, and their calling volume was calibrated. In random order, subjects were asked to call the name of a state for 5 minutes. Listeners with parabolic microphones and others with unaided hearing recorded the direction of the call (detection) and name of the state (comprehension)., Results: The parabolic microphone was superior to unaided hearing in both detecting subjects and comprehending their calls, with an effect size (Cohen's d) of 1.58 for detection and 1.55 for comprehension. For each of the 8 hidden subjects, there were 24 detection attempts with the parabolic microphone and 54 to 60 attempts by unaided listeners. At the longer distances (1529-2510 m), the parabolic microphone was better at detecting callers (83% vs 51%; P<0.00001 by χ 2 ) and comprehension (57% vs 12%; P<0.00001). At the shorter distances (322-1190 m), the parabolic microphone offered advantages in detection (100% vs 83%; P=0.000023) and comprehension (86% vs 51%; P<0.00001), although not as pronounced as at the longer distances., Conclusions: Use of a 66-cm (26-inch) parabolic microphone significantly improved detection and comprehension of hidden calling subjects at distances between 322 and 2510 m when compared with unaided hearing. This study supports the use of a parabolic microphone in search and rescue to locate responsive subjects in favorable weather and terrain., (Copyright © 2017 The Authors. Published by Elsevier Inc. All rights reserved.)

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.

Body posture has an effect on sleeping quality and breathing disorders and therefore it is important to be recognized for the completion of the sleep evaluation process. Since humans have a directional acoustic radiation pattern, it is hypothesized that microphone arrays can be used to recognize different body postures, which is highly practical for sleep evaluation applications that already measure respiratory sounds using distant microphones. Furthermore, body posture may have an effect on distant microphone measurement; hence, the measurement can be compensated if the body posture is correctly recognized. A spherical harmonics decomposition approach to the spatial acoustic radiation is presented, assuming an array of eight microphones in a medium-sized audiology booth. The spatial sampling and reconstruction of the radiation pattern is discussed, and a final setup for the microphone array is recommended. A case study is shown using recorded segments of snoring and breathing sounds of three human subjects in three body postures in a silent but not anechoic audiology booth.

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.

Sound propagation encompasses various acoustic phenomena including reverberation. Current virtual acoustic methods ranging from parametric filters to physically accurate solvers can simulate reverberation with varying degrees of fidelity. The effects of reverberant sounds generated using different propagation algorithms on acoustic distance perception are investigated. In particular, two classes of methods for real time sound propagation in dynamic scenes based on parametric filters and ray tracing are evaluated. The study shows that ray tracing enables more distance accuracy as compared to the approximate, filter-based method. This suggests that accurate reverberation in virtual reality results in better reproduction of acoustic distances.

Previous headphone experiments have shown that listeners can lateralize high-frequency sine-wave amplitude-modulated (SAM) tones based on interaural time differences in the envelope. However, when SAM tones are presented to listeners in free field or in a room, diffraction by the head or reflections from room surfaces alter the modulation percentages and change the shapes of the envelopes, potentially degrading the envelope cue. Amplitude modulation is transformed into mixed modulation. This article presents a mathematical transformation between the six spectral parameters for a modulated tone and six mixed-modulation parameters for each ear. The transformation was used to characterize the stimuli in the ear canals of listeners in free-field localization experiments. The mixed modulation parameters were compared with the perceived changes in localization attributable to the modulation for five different listeners, who benefited from the modulation to different extents. It is concluded that individual differences in the response to added modulation were not systematically related to the physical modulation parameters themselves. Instead, they were likely caused by individual differences in processing of envelope interaural time differences.

Objective: To design a test battery and conduct a proof-of-concept experiment of a test method that can be used to measure the detection performance afforded by military advanced hearing protection devices (HPDs) and tactical communication and protective systems (TCAPS)., Design: The detection test was conducted with each of the four loudspeakers located at front, right, rear and left of the participant. Participants wore 2 in-ear-type TCAPS, 1 earmuff-type TCAPS, a passive Combat Arms Earplug in its "open" or pass-through setting and an EB-15LE™ electronic earplug. Devices with electronic gain systems were tested under two gain settings: "unity" and "max". Testing without any device (open ear) was conducted as a control., Study Sample: Ten participants with audiometric requirements of 25 dBHL or better at 500, 1000, 2000, 4000, 8000 Hz in both ears., Results: Detection task performance varied with different signals and speaker locations. The test identified performance differences among certain TCAPS and protectors, and the open ear., Conclusions: A computer-controlled detection subtest of the Detection-Recognition/Identification-Localisation-Communication (DRILCOM) test battery was designed and implemented. Tested in a proof-of-concept experiment, it showed statistically-significant sensitivity to device differences in detection effects with the small sample of participants (10). This result has important implications for selection and deployment of TCAPS and HPDs on soldiers and workers in dynamic situations.

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.

Internally coupled ears (ICEs) allow small animals to reliably determine the direction of a sound source. ICEs are found in a variety of taxa, but crickets have evolved the most complex arrangement of coupled ears: an acoustic tracheal system composed of a large cross-body trachea that connects two entry points for sound in the thorax with the leg trachea of both ears. The key structure that allows for the tuned directionality of the ear is a tracheal inflation (acoustic vesicle) in the midline of the cross-body trachea holding a thin membrane (septum). Crickets are known to display a wide variety of acoustic tracheal morphologies, most importantly with respect to the presence of a single or double acoustic vesicle. However, the functional relevance of this variation is still not known. In this study, we investigated the peripheral directionality of three co-occurring, closely related cricket species of the subfamily Gryllinae. No support could be found for the hypothesis that a double vesicle should be regarded as an evolutionary innovation to (1) increase interaural directional cues, (2) increase the selectivity of the directional filter or (3) provide a better match between directional and sensitivity tuning. Nonetheless, by manipulating the double acoustic vesicle in the rainforest cricket Paroecanthus podagrosus, selectively eliminating the sound-transmitting pathways, we revealed that these pathways contribute almost equally to the total amount of interaural intensity differences, emphasizing their functional relevance in the system., (© 2016. Published by The Company of Biologists Ltd.)

Artificially designed acoustic meta-surfaces have the ability to manipulate sound energy to an extraordinary extent. Here, we report on a new type of directional reflective surface consisting of an array of sub-wavelength Helmholtz resonators with varying internal coiled path lengths, which induce a reflection phase gradient along a planar acoustic meta-surface. The acoustically reshaped reflective surface created by the gradient-impeding meta-surface yields a distinct focal line similar to a parabolic cylinder antenna, and is used for directive sound beamforming. Focused beam steering can be also obtained by repositioning the source (or receiver) off axis, i.e., displaced from the focal line. Besides flat reflective surfaces, complex surfaces such as convex or conformal shapes may be used for sound beamforming, thus facilitating easy application in sound reinforcement systems. Therefore, directional reflective surfaces have promising applications in fields such as acoustic imaging, sonic weaponry, and underwater communication.

A narrowband MEMS direction finding sensor has been developed based on the mechanically coupled ears of the Ormia Ochracea fly. The sensor consists of two wings coupled at the middle and attached to a substrate using two legs. The sensor operates at its bending resonance frequency and has cosine directional characteristics similar to that of a pressure gradient microphone. Thus, the directional response of the sensor is symmetric about the normal axis making the determination of the direction ambiguous. To overcome this shortcoming two sensors were assembled with a canted angle similar to that employed in radar bearing locators. The outputs of two sensors were processed together allowing direction finding with no requirement of knowing the incident sound pressure level. At the bending resonant frequency of the sensors (1.69 kHz) an output voltage of about 25 V/Pa was measured. The angle uncertainty of the bearing of sound ranged from less than 0.3° close to the normal axis (0°) to 3.4° at the limits of coverage (±60°) based on the 30° canted angle used. These findings indicate the great potential to use dual MEMS direction finding sensor assemblies to locate sound sources with high accuracy.

The growing availability of efficient and relatively inexpensive virtual auditory display technology has provided new research platforms to explore the perception of auditory motion. At the same time, deployment of these technologies in command and control as well as in entertainment roles is generating an increasing need to better understand the complex processes underlying auditory motion perception. This is a particularly challenging processing feat because it involves the rapid deconvolution of the relative change in the locations of sound sources produced by rotational and translations of the head in space (self-motion) to enable the perception of actual source motion. The fact that we perceive our auditory world to be stable despite almost continual movement of the head demonstrates the efficiency and effectiveness of this process. This review examines the acoustical basis of auditory motion perception and a wide range of psychophysical, electrophysiological, and cortical imaging studies that have probed the limits and possible mechanisms underlying this perception., (© The Author(s) 2016.)

Both reward based operant conditioning (OC) and reflex-based prepulse inhibition (PPI) procedures are used in sound localisation studies in mice. Since the results of both procedures are compared in the literature, it is important to assess whether they provide similar results if the same stimulus paradigm is applied. Here, we compare the sensitivity of C57BL/6 mice in OC and PPI procedures for detecting a switch in speaker location using broadband and narrowband noise stimuli and determined their minimum audible angle (MAA). In the OC procedure, we calculated d' values from the hit and false alarm rates. In the PPI procedure, we calculated the area under ROC curves from the startle response amplitudes and derived da values to obtain a sensitivity measure that corresponds to d'. For both procedures, the mean sensitivity to the speaker switch increased with an increase in angular separation. For broadband noise stimuli, a d' of up to 3.3 (OC) and a da of up to 1.1 (PPI) were observed at large speaker separations. Narrowband noise stimuli resulted in lower sensitivities in both procedures, resulting in a maximum d' of 2.0 (OC) and a maximum da of 0.3 (PPI). Using a sensitivity of 1.0 as the threshold criterion, broadband noise MAAs in the range from 32° to 46° were observed in the OC procedure whereas a broadband noise MAAs of 108° or higher were observed in the PPI procedure. In the OC procedure, narrowband noise MAAs in the range from 37° to 62° were observed. In the PPI procedure, no narrowband noise MAA could be determined since none of the subjects reached the threshold. Thus, OC procedures result in a better performance of the subjects in the sound localization task than PPI procedures, challenging the view that both procedures can be used interchangeably., (Copyright © 2016 Elsevier B.V. All rights reserved.)

Blind humans echolocate nearby targets by emitting palatal clicks and perceiving echoes that the auditory system is not able to resolve temporally. The mechanism for perceiving near-range echoes is not known. This paper models the direct mouth-to-ear signal (MES) and the echo to show that the echo enhances the high-frequency components in the composite MES/echo signal with features that allow echolocation. The mouth emission beam narrows with increasing frequency and exhibits frequency-dependent transmission notches in the backward direction toward the ears as predicted by the piston-in-sphere model. The ears positioned behind the mouth detect a MES that contains predominantly the low frequencies contained in the emission. Hence the high-frequency components in the emission that are perceived by the ears are enhanced by the echoes. A pulse/echo audible sonar verifies this model by echolocating targets from 5 cm range, where the MES and echo overlap significantly, to 55 cm. The model predicts that unambiguous ranging occurs over a limited range and that there is an optimal range that produces the highest range resolution.

An equivalent analog circuit is designed to mimic the coupled ears of the fly Ormia ochracea for sound source localization. This coupled circuit receives two signals with tiny phase difference from a space closed two-microphone array, and produces two signals with obvious intensity difference. The response sensitivity can be adjusted through the coupled circuit parameters. The directional characteristics of the coupled circuit have been demonstrated in the experiment. The miniature microphone array can localize the sound source with low computational burden by using the intensity difference. This system has significant advantages in various applications where the array size is limited.

Designing a "cocktail party listener" that functionally mimics the selective perception of a human auditory system has been pursued over the past decades. By exploiting acoustic metamaterials and compressive sensing, we present here a single-sensor listening device that separates simultaneous overlapping sounds from different sources. The device with a compact array of resonant metamaterials is demonstrated to distinguish three overlapping and independent sources with 96.67% correct audio recognition. Segregation of the audio signals is achieved using physical layer encoding without relying on source characteristics. This hardware approach to multichannel source separation can be applied to robust speech recognition and hearing aids and may be extended to other acoustic imaging and sensing applications.

Passive seismic monitoring in the oceans uses long-term deployments of Ocean Bottom Seismometers (OBSs). An OBS usually records the three components of ground motion and pressure, typically at 100 Hz. This makes the OBS an ideal tool to investigate fin and blue whales that vocalize at frequencies below 45 Hz. Previous applications of OBS data to locate whale calls have relied on single channel analyses that disregard the information that is conveyed by the horizontal seismic channels. Recently, Harris, Matias, Thomas, Harwood, and Geissler [J. Acoust. Soc. Am. 134, 3522-3535 (2013)] presented a method that used all four channels recorded by one OBS to derive the range and azimuth of fin whale calls. In this work, the detection, classification, and ranging of calls using this four-channel method were further investigated, focusing on methods to increase the accuracy of range estimates to direct path arrivals. Corrections to account for the influences of the sound speed in the water layer and the velocity structure in the top strata of the seabed were considered. The single station method discussed here is best implemented when OBSs have been deployed in deep water on top of seabed strata with low P-wave velocity. These conditions maximize the ability to detect and estimate ranges to fin whale calls.

Sound focusing is to create a concentrated acoustic field in the region surrounded by a loudspeaker array. This problem was tackled in the previous research via the Helmholtz integral approach, brightness control, acoustic contrast control, etc. In this paper, the same problem was revisited from the perspective of beamforming. A source array model is reformulated in terms of the steering matrix between the source and the field points, which lends itself to the use of beamforming algorithms such as minimum variance distortionless response (MVDR) and linearly constrained minimum variance (LCMV) originally intended for sensor arrays. The beamforming methods are compared with the conventional methods in terms of beam pattern, directional index, and control effort. Objective tests are conducted to assess the audio quality by using perceptual evaluation of audio quality (PEAQ). Experiments of produced sound field and listening tests are conducted in a listening room, with results processed using analysis of variance and regression analysis. In contrast to the conventional energy-based methods, the results have shown that the proposed methods are phase-sensitive in light of the distortionless constraint in formulating the array filters, which helps enhance audio quality and focusing performance.

This work presents a method to control the perceived distance of an auditory object by changing the directivity pattern of a loudspeaker and consequently the direct-to-reverberant ratio at the listening spot. Control of the directivity pattern is achieved by beamforming using a compact multi-driver loudspeaker unit. A small-sized cubic array consisting of six drivers is assembled, and per driver beamforming filters are derived from directional measurements of the array. The proposed method is evaluated using formal listening tests. The results show that the perceived distance can be controlled effectively by directivity pattern modification.

A physical structure such as a cylindrical pipe controls the propagated sound spectrum in a predictable way that can be used to localize the sound source. This paper designs a monaural sound localization system based on multiple pyramidal horns around a single microphone. The acoustic resonance within the horn provides a periodicity in the spectral domain known as the fundamental frequency which is inversely proportional to the radial horn length. Once the system accurately estimates the fundamental frequency, the horn length and corresponding angle can be derived by the relationship. The modified Cepstrum algorithm is employed to evaluate the fundamental frequency. In an anechoic chamber, localization experiments over azimuthal configuration show that up to 61% of the proper signal is recognized correctly with 30% misfire. With a speculated detection threshold, the system estimates direction 52% in positive-to-positive and 34% in negative-to-positive decision rate, on average.

In this paper, we address the problems of modeling the acoustic space generated by a full-spectrum sound source and using the learned model for the localization and separation of multiple sources that simultaneously emit sparse-spectrum sounds. We lay theoretical and methodological grounds in order to introduce the binaural manifold paradigm. We perform an in-depth study of the latent low-dimensional structure of the high-dimensional interaural spectral data, based on a corpus recorded with a human-like audiomotor robot head. A nonlinear dimensionality reduction technique is used to show that these data lie on a two-dimensional (2D) smooth manifold parameterized by the motor states of the listener, or equivalently, the sound-source directions. We propose a probabilistic piecewise affine mapping model (PPAM) specifically designed to deal with high-dimensional data exhibiting an intrinsic piecewise linear structure. We derive a closed-form expectation-maximization (EM) procedure for estimating the model parameters, followed by Bayes inversion for obtaining the full posterior density function of a sound-source direction. We extend this solution to deal with missing data and redundancy in real-world spectrograms, and hence for 2D localization of natural sound sources such as speech. We further generalize the model to the challenging case of multiple sound sources and we propose a variational EM framework. The associated algorithm, referred to as variational EM for source separation and localization (VESSL) yields a Bayesian estimation of the 2D locations and time-frequency masks of all the sources. Comparisons of the proposed approach with several existing methods reveal that the combination of acoustic-space learning with Bayesian inference enables our method to outperform state-of-the-art methods.

Pressure-differential microphones inspired by the hearing mechanism of a special parasitoid fly have been described previously. The designs employ a beam structure that rotates about two pivots over an enclosed back volume. The back volume is only partially enclosed due to open slits around the perimeter of the beam. The open slits enable incoming sound waves to affect the pressure profile in the microphone's back volume. The goal of this work is to study the net moment applied to pressure-differential microphones by an incoming sound wave, which in-turn requires modeling the acoustic pressure distribution within the back volume. A lumped-element distributed transmission-line model of the back volume is introduced for this purpose. It is discovered that the net applied moment follows a low-pass filter behavior such that, at frequencies below a corner frequency depending on geometrical parameters of the design, the applied moment is unaffected by the open slits. This is in contrast to the high-pass filter behavior introduced by barometric pressure vents in conventional omnidirectional microphones. The model accurately predicts observed curvature in the frequency response of a prototype pressure-differential microphone 2 mm × 1 mm × 0.5 mm in size and employing piezoelectric readout.

The spectral localization cues contained in the head-related transfer function are known to play a contributory role in the sound source localization abilities of humans. However, existing localization techniques are unable to fully exploit this diversity to accurately localize a sound source. The availability of just two measured signals complicates matters further, and results in front to back confusions and poor performance distinguishing between the source locations in a vertical plane. This study evaluates the performance of a source location estimator that retains the frequency domain diversity of the head-related transfer function. First, a method for extracting the directional information in the subbands of a broadband signal is described, and a composite estimator based on signal subspace decomposition is introduced. The localization performance is experimentally evaluated for single and multiple source scenarios in the horizontal and vertical planes. The proposed estimator's ability to successfully localize a sound source and resolve the ambiguities in the vertical plane is demonstrated, and the impact of the source location, knowledge of the source and the effect of reverberation is discussed.

Given scarce number of recorded signals, spatial sound field synthesis with an extended sweet spot is a challenging problem in acoustic array signal processing. To address the problem, a synthesis and upmixing approach inspired by the equivalent source method (ESM) is proposed. The synthesis procedure is based on the pressure signals recorded by a microphone array and requires no source model. The array geometry can also be arbitrary. Four upmixing strategies are adopted to enhance the resolution of the reproduced sound field when there are more channels of loudspeakers than the microphones. Multi-channel inverse filtering with regularization is exploited to deal with the ill-posedness in the reconstruction process. The distance between the microphone and loudspeaker arrays is optimized to achieve the best synthesis quality. To validate the proposed system, numerical simulations and subjective listening experiments are performed. The results demonstrated that all upmixing methods improved the quality of reproduced target sound field over the original reproduction. In particular, the underdetermined ESM interpolation method yielded the best spatial sound field synthesis in terms of the reproduction error, timbral quality, and spatial quality.

The activity of sensory neural populations carries information about the environment. This may be extracted from neural activity using different strategies. In the auditory brainstem, a recent theory proposes that sound location in the horizontal plane is decoded from the relative summed activity of two populations in each hemisphere, whereas earlier theories hypothesized that the location was decoded from the identity of the most active cells. We tested the performance of various decoders of neural responses in increasingly complex acoustical situations, including spectrum variations, noise, and sound diffraction. We demonstrate that there is insufficient information in the pooled activity of each hemisphere to estimate sound direction in a reliable way consistent with behavior, whereas robust estimates can be obtained from neural activity by taking into account the heterogeneous tuning of cells. These estimates can still be obtained when only contralateral neural responses are used, consistently with unilateral lesion studies. DOI: http://dx.doi.org/10.7554/eLife.01312.001.

Head-related transfer functions (HRTFs) are generally large datasets, which can be an important constraint for embedded real-time applications. A method is proposed here to reduce redundancy and compress the datasets. In this method, HRTFs are first compressed by conversion into autoregressive-moving-average (ARMA) filters whose coefficients are calculated using Prony's method. Such filters are specified by a few coefficients which can generate the full head-related impulse responses (HRIRs). Next, Legendre polynomials (LPs) are used to compress the ARMA filter coefficients. LPs are derived on the sphere and form an orthonormal basis set for spherical functions. Higher-order LPs capture increasingly fine spatial details. The number of LPs needed to represent an HRTF, therefore, is indicative of its spatial complexity. The results indicate that compression ratios can exceed 98% while maintaining a spectral error of less than 4 dB in the recovered HRTFs.

Parametric methods for modeling the perceptually relevant features of head-related transfer functions (HRTFs) are very important for the development of low-cost immersive sound applications. This letter describes an efficient method based on a low-order infinite impulse response filter implemented by a chain of second order sections of conventional shelving and peak audio filters. The parameters (central frequency, gain, and quality factor) are numerically adjusted by iteratively fitting the frequency response of the filter to the desired HRTF. Besides allowing for low-order binaural models, the proposed approach provides an efficient way to synthesize HRTFs for non-measured angles by applying a simple interpolation between the parameters from neighboring responses. Additionally, the HRTF database size is significantly reduced.

Sound localization in the sagittal planes, including the ability to distinguish front from back, relies on spectral features caused by the filtering effects of the head, pinna, and torso. It is assumed that important spatial cues are encoded in the frequency range between 4 and 16 kHz. In this study, in a double-blind design and using audio-visual training covering the full 3-D space, normal-hearing listeners were trained 2 h per day over three weeks to localize sounds which were either band limited up to 8.5 kHz or spectrally warped from the range between 2.8 and 16 kHz to the range between 2.8 and 8.5 kHz. The training effect for the warped condition exceeded that for procedural task learning, suggesting a stable auditory recalibration due to the training. After the training, performance with band-limited sounds was better than that with warped ones. The results show that training can improve sound localization in cases where spectral cues have been reduced by band-limiting or remapped by warping. This suggests that hearing-impaired listeners, who have limited access to high frequencies, might also improve their localization ability when provided with spectrally warped or band-limited sounds and adequately trained on sound localization.

Four experiments investigated the effects on speech intelligibility of reverberation, sound source locations, and amplitude modulation of the interferers. Speech reception thresholds (SRTs) were measured using headphones and stimuli that simulated real-room listening, considering one or two interferers which were stationary or speech-modulated noises. In experiment 1, SRTs for modulated noises showed little variation with increasing interferer reverberation. Reverberation might have increased masking by filling in the modulated noise gaps, but simultaneously changed the noise spectra making them less effective maskers. In experiment 2, SRTs were lower when measured using a unique one-voice modulated interferer rather than a different interferer for each target sentence, suggesting that listeners could take advantage of the predictability of the interferer gaps. In experiment 3, increasing speech reverberation did not significantly affect the difference of SRTs measured with stationary and modulated noises, indicating that the ability to exploit noise modulations was still useful for temporally smeared speech. In experiment 4, spatial unmasking remained constant when applying modulations to the interferers, suggesting an independence of the abilities to exploit these modulations and the spatial separation of sources. Finally, a model predicting binaural intelligibility for modulated noises was developed and provided a good fit to the experimental data.

In performing arts venues, the spectra of direct and reflected sound at a receiving location differ, due to seat dip effect, diffusive and absorptive surfaces, and source directivity. This paper examines the influence of differing lead and lag spectral contents on echo suppression threshold. The results indicate, that for a highpass filtered direct sound and a broadband reflection, attenuation of low frequencies initially results in an increase in echo suppression threshold, while for higher cutoff frequencies echo suppression threshold drastically decreases. For broadband direct sound and filtered reflections, the echo suppression threshold is inversely related to high frequency content.

In musical or theatrical performance, some venues allow listeners to individually localize and segregate individual performers, while others produce a well blended ensemble sound. The room acoustic conditions that make this possible, and the psycho-acoustic effects at work are not fully understood. This research utilizes auralizations from measured and simulated performance venues to investigate spatial discrimination of multiple acoustic sources in rooms. Signals were generated from measurements taken in a small theater, and listeners in the audience area were asked to distinguish pairs of speech sources on stage with various spatial separations. This experiment was repeated with the proscenium splay walls treated to be flat, diffusive, or absorptive. Similar experiments were conducted in a simulated hall, utilizing 11 early reflections with various characteristics, and measured late reverberation. The experiments reveal that discriminating the lateral arrangement of two sources is possible at narrower separation angles when reflections come from flat or absorptive rather than diffusive surfaces.

The process of echolocation is accomplished by bats partly using the beam profiles associated with their ear shapes that allow for discrimination between different echo directions. Indeed, knowledge of the emitted signal characteristic and measurement of the echo travel time from a target make it possible to compensate for attenuation due to distance, and to focus on filtering through the receivers' beam profiles by comparing received echoes to the original signal at all frequencies in the spectrum of interest. From this basis, a beam profile method to localize a target in three-dimensional space for an ultrasonic sensor system equipped with an emitter and two receivers is presented. Simulations were conducted with different noise levels, and only the contribution of the receivers' beam profiles was considered to estimate the orientation of the target with respect to the receivers. The beam pattern of the Phyllostomus discolor's ear was adopted as that of a receiver. Analyses of beam resolution and frequency ranges were conducted to enhance the accuracy of orientation estimates. The choice of appropriate resolution and frequency ranges guarantee that error mean values for most of the orientations are within [0.5°, 1.5°], even in noisy situations: Signal-to-noise ratio values considered in this work are 35 and 50 dB.

The present investigation assessed fusion and localization dominance aspects of the precedence effect under headphones across a variety of stimulus conditions in 10 normal-hearing listeners. Listeners were presented with "lead-lag" pairs of brief (123 μs) impulses or trains of such pairs lateralized by interaural time or level differences (ITD or ILD). Listeners used a touch-sensitive display to indicate for the final lead-lag pair presented on each trial (1) whether one or two locations were perceived and (2) the location perceived. In the event two locations were perceived, subjects were further instructed to indicate the left-most location perceived. Results demonstrated that lead-lag fusion was more robust for stimuli lateralized by ITD than ILD, particularly when cues of the test stimulus differed from cues of the preceding "buildup" stimulus, consistent with Krumbholz and Nobbe [(2002). J. Acoust. Soc. Am. 112, 654-663]. Unexpectedly, results also demonstrated reduced localization dominance with increasing lead-lag delay, suggesting that the fusion aspect of the precedence effect may be dissociated from the localization dominance aspect under buildup. It is thus argued that buildup of fusion might be understood more generally as an example of auditory object formation rather than a special facility for enhanced sound localization.

The area of sound field synthesis has significantly advanced in the past decade, facilitated by the development of high-quality sound-field capturing and re-synthesis systems. Spherical microphone arrays are among the most recently developed systems for sound field capturing, enabling processing and analysis of three-dimensional sound fields in the spherical harmonics domain. In spite of these developments, a clear relation between sound fields recorded by spherical microphone arrays and their perception with a re-synthesis system has not yet been established, although some relation to scalar measures of spatial perception was recently presented. This paper presents an experimental study of spatial sound perception with the use of a spherical microphone array for sound recording and headphone-based binaural sound synthesis. Sound field analysis and processing is performed in the spherical harmonics domain with the use of head-related transfer functions and simulated enclosed sound fields. The effect of several factors, such as spherical harmonics order, frequency bandwidth, and spatial sampling, are investigated by applying the repertory grid technique to the results of the experiment, forming a clearer relation between sound-field capture with a spherical microphone array and its perception using binaural synthesis regarding space, frequency, and additional artifacts. The experimental study clearly shows that a source will be perceived more spatially sharp and more externalized when represented by a binaural stimuli reconstructed with a higher spherical harmonics order. This effect is apparent from low spherical harmonics orders. Spatial aliasing, as a result of sound field capturing with a finite number of microphones, introduces unpleasant artifacts which increased with the degree of aliasing error.

This letter presents results from a study on diffusive architectural surfaces and auditory perception. Spatial discrimination of multiple sources is investigated in a simulated performance venue with various diffusive surface treatments. Simulations were generated with closely spaced sound sources on the stage of a concert hall and a listener in the audience area. Subjects were asked to distinguish signals in which pairs of simultaneous talkers were presented at various lateral separations, in halls with flat or diffusive surfaces. The experiments reveal that discriminating differences in the lateral arrangement of sources is possible at narrower separation angles when reflections come from flat rather than diffusive surfaces.