bluuebunny/crossref_metadata_2025 huggingface dataset:
Sounding out smart cities: Auralization and soundscape monitoring for environmental sound design
Authors: Southern Alex , Stevens Francis , Murphy Damian
Published: June 2017

Auralization is key in developing a better understanding of how significant changes or infrastructure planning in our urban environment can have an impact on our related environmental soundscape. It allows consultants, planners and other stakeholders to hear the potential acoustic changes that might result, so that designs might be better optimized; it is also a valuable dissemination tool for informing the public as to the nature of such changes. Auralization also facilitates subjective soundscape assessment of proposed developments at the design stage and once construction is complete, smart sensor networks enable soundscape monitoring and objective evaluated on an ongoing basis. Of particular interest is sound emitted from transportation, as it is generally considered as unwanted sound and hence defined as noise. Transportation noise and road traffic noise in particular, is considered a concern for public health by the WHO, and annoyance with some aspect of our daily soundscape is not uncommon. This work presents an overview of how auralization has been used in the context of some recent transportation noise related case studies. The complete auralization chain is presented including source measurement and soundscape monitoring, sound propagation modeling using numerical simulation, soundfield rendering, and the potential for immersive multimodal presentation.
Shape optimization of reactive mufflers using threshold Acceptance and FEM
Authors: Khamchane Abdelkader , Khelfaoui Youcef , Hamtache Brahim
Published: June 2017

In this paper, the acoustic performance of three different expansion-chamber mufflers with extended tube under space constraint is presented. A shape optimization analysis is performed using a novel scheme called Threshold Acceptance (TA), the best design obtained by the shape optimization method are analyzed by Finite Element Method (FEM). This numerical approach is based on the maximization of the sound transmission loss (STL) using the Transfer Matrix Method (TMM), a modeling method based on the plane wave propagation model. The FEM solution used to analyse the STL of the shape optimized mufflers is based on the Acoustic Power method, a standard computational code COMSOL Multiphysics is used to analyse in 3D the sound attenuation of the mufflers by the FE method. The acoustical ability of the mufflers obtained is than assessed by comparing the FEM solution with the analytical method. Results show that the maximal STL is precisely located at the desired targeted tone. In addition, the acoustical performance of muffler with outlet extended tube is found to be superiors to the other one. Consequently, this approach provides a quick and novel scheme for the shape optimization of reactive mufflers.
Binaural sound map of Malaga
Authors: Rosas Carmen , Luna Salvador
Published: June 2017

Streets, squares, parks, and even buildings and businesses, present a sonic footprint which characterises every city at a certain period. Despite this, in the field of acoustics these sounds are usually analysed as noises to be measured and reduced. This fact may have caused the study of city sounds to be deemed secondary and may also have led to prioritising the resolution of conflicts brought about by undesired noises. As it happens with the natural environment, however, taking proper care of an acoustic environment stems from its knowledge and appreciation. This sound map aims at creating a practical tool that collects all the most distinctive soundscapes of Malaga so that they can be listened to by people from all over the world, become part of the city's cultural heritage and be archived and catalogued for their conservation. Besides, its contrast with noise maps, a sound map allows for the characterisation of the territory from a different perspective, in which the identity of the depicted area is defined by all its sounds. All the recordings are being made with binaural microphones, so the sounds produce a more immersive experience when using headphones. These microphones were built for this project, and a series of HRTF measurements was obtained and applied to different audio signals for the realization of a psychoacoustic test, in order to assess the spatiality provided by the system. Another series of audio samples was generated from the MIT’s HRTFs, and both results have been compared.
Critical frequencies for the annoyance judgments of noises with tonal component, to develop an equal-annoyance contour
Authors: Pace Matias A. , Urquiza Nicolás , Masson Florent , Sato Shin-ichi
Published: June 2017

The influence of the tonal component in noise annoyance has been the focus of many previous researches. It has been found that the annoyance varies when the frequency of the tone changes. In this work a subjective tests on the noise annoyance were performed by using environmental, urban and industrial tonal noises. Stationary signals were used and modified by adding tones. The tonal noise was defined according to ISO 1996-2. An 11-point numerical scale test was conducted under the same sound pressure level (SPL) in dBA. The results of the tests showed that the perceived annoyance varied mainly due to the center frequency of the test stimuli rather than other psychoacoustic parameters of the signals. These results establish the critical frequencies of the spectrum at which the annoyance of the tonal noises varies considerably, toward the development of an equal-annoyance contour.
Performance analysis of the low-cost acoustic sensors developed for the DYNAMAP project: A case study in the Milan urban area
Authors: Alías Francesc , Alsina-Pagès Rosa M. , Socoró Joan Claudi , Orga Ferran , Nencini Luca
Published: June 2017

Dynamic noise maps are asked to accurately represent in real time the noise levels captured by acoustic sensors in urban environments. Thus, the measurement precision of the employed sensors becomes of paramount importance to obtain a reliable picture of the urban noise levels. In this work, a case study designed to validate the accuracy of the Leq values of the low-cost acoustic sensor developed for the DYNAMAP project in real-life conditions is presented. To that effect, simultaneous acoustic measurements have been conducted in the same locations in the Milan’s project pilot area using a Class I sound level meter and the low-cost sensor. The analysis of the parallel acoustic data collected during the recording campaign compares both measuring equipments accounting for traffic noise, background city noise, and non-traffic noise events.
Green’s function retrieval by crosscorrelation and multidimensional deconvolution: Application to atmospheric acoustics
Authors: Denis Max , Collier Sandra L. , Noble John , Alberts W. C. Kirkpatrick , Ligon David , Sim Leng , Reiff Christian G.
Published: June 2017

In this work, the Green’s function of an atmospheric acoustic propagation channel is extracted from experimental measurements. Of particular interest, is the accuracy of Green’s function retrieval methods in ideal and non-ideal situations. To this end, an emitting acoustic source (impulsive and audible) is placed at a distance of more than 90 meters from receiving tri-axis microphone arrays in open (ideal) and wooded areas (non-ideal) in southern Maryland. Green’s function retrieval methods are employed and investigated on the collected data. Green’s function retrieval by crosscorrelation has been successful in various applications despite the limitations of its lossless medium and equipartitioned wavefield assumptions. To overcome the violation of these assumptions, the multidimensional deconvolution has been proposed. Comparisons of the results between impulsive and audible sources using the two methods will be presented. The effects of multipath, source-receiver distance, temperature, and wind will be discussed.
Generation and measurement of shear waves in micellar fluids using ultrasound-based methods
Authors: Urban Matthew W. , Amador Carolina , Otilio Bruno , Kinnick Randall R.
Published: June 2017

Wormlike micellar fluids are viscoelastic and their mechanical properties have been evaluated with mechanically generated shear waves combined with optical detection as well as rheological testing. In this work, we describe the use of acoustic radiation force and ultrafast ultrasound imaging to generate shear waves and measure their propagation, respectively. This shear wave elastography (SWE) method has been used to measure the viscoelastic mechanical properties in tissue-mimicking phantoms and soft tissues. We tested micellar fluids made from cetrimonium bromide (CTAB) and sodium salicylate (NaSAL) with a 5:3 ratio with different concentrations (100, 200, 300, and 400 mM). We fit an extended Maxwell model to the rheological test data (0.001-15.91Hz) and used the model parameters to calculate the shear wave phase velocity in the range of the SWE data (100-500 Hz). The phase velocities calculated from the rheology testing compared well with the SWE results. The mean absolute error was less than 0.02 m/s for all the micellar fluids tested. The SWE method is nondestructive and can be used for characterization of the viscoelasticity of micellar fluids, which could be used as a model for biological tissues. [This work was supported in part by grant R01DK092255.]
Effect of hearing loss on spectral weighting of broadband signals
Authors: Joshi Suyash N. , Jesteadt Walt
Published: June 2017

Several studies have shown that normal hearing (NH) listeners place greater weight on the edge frequencies of broadband sounds when they are asked to pick the louder of two intervals in a sample discrimination task. However, little is known about the effects of hearing loss on spectral weighting for broadband sounds. In a classic study of sample discrimination, Doherty and Lutfi (1996) found that listeners with hearing loss (HL) placed greater weight on frequencies in the region of the HL and proposed that components that were less audible were weighted higher. The current study investigated the effect of audibility and hearing loss on spectral weighting for loudness judgments using two spectrally overlapping 18-tone complexes (from 208- to 8708-Hz). Four NH and nine HL listeners judged the loudness of tone complexes in two conditions where the mean level of each tone in the complex was set to either 10-dB SL or 75-dB SPL. The results show that listeners placed greater weight on the high-frequency edges of both tone complexes in 10-dB SL condition but not in 75-dB SPL condition, suggesting that the HL listeners place less weight on portions of the spectrum near the threshold of audibility. [Work supported by NIH.]
Consistent and inconsistent interaural cues don't differ for tone detection but do dffer for speech recognition
Authors: Gallun Frederick J. , Ellinger Rachel , Jakien Kasey
Published: June 2017

In 1965, Dr. Colburn and his advisor, Dr. Durlach, whose wide and deep influence on the field of acoustics we honor in another Special Session at this meeting, conducted a seminal experiment on binaural hearing. This experiment showed that detection of a 500-Hz tone in broadband noise is improved equally well by a “consistent” arrangement, where an interaural difference in time (ITD) favors the same ear as does a simultaneously presented interaural difference in intensity (IID) as it is by an “inconsistent” arrangement, in which ITD favors one ear and IID the other. In 2008, Gallun et al. extended these results by showing that consistent and inconsistent differences are equally effective for tone detection in multitone maskers, even though the source of masking is clearly different. Here we present data from a speech recognition task performed in the presence of speech maskers where inconsistent interaural differences produced less release from masking than did consistent cues. These results argue for the importance of a perceived location cue for speech recognition tasks (at least of this type) that is not needed for tone detection.
System identification of auditory processes using noise probes in isolation or embedded into soundscapes
Authors: Neri Peter
Published: June 2017

This talk will cover large experimental datasets spanning different levels of auditory analysis from low-level to higher-level, with specific focus on two inter-related problems: (1) how to exploit perceptual weights and their derivatives to infer the structure of the underlying auditory process; (2) how to transfer these tools from identification of isolated processes to situations where the process is embedded within complex sounds (e.g., speech). I will highlight connections with recent developments in both auditory and vision literature, for example (in relation to number 1 and 2 above, respectively) detailed understanding of how important nonlinearities are reflected in identifiable modulations of perceptual weights, and construction of noise probes for smooth embedding within structured contexts together with associated tasks specifically designed to meanigfully probe contextual phenomena. These developments demonstrate very substantial progress over the past two decades, and offer new exciting directions for further characterization of human sensory processes via tight integration of experimental/computational tools.
Spectral weighting of interaural time- and level differences for broadband signals
Authors: Epp Bastian , Ahrens Axel , Joshi Suyash Narendra
Published: June 2017

An important ability of the auditory system is to localize sound sources in complex acoustical environments. Two important cues for localization are interaural time- and level differences (ITD, ILD). The sensitivity to these cues differs across frequency and has previously been estimated through frequency-specific detection thresholds. Detection thresholds of ITDs/ILDs are, however, affected by stimulus energy in remote spectral regions, referred to as binaural interference. In this study, the spectral weights of ITD- and ILD cues in the lateralization of a broadband signal was investigated using regression analysis. The stimuli consisted of eleven 1-ERB-wide noise bands (442 Hz-5544 Hz) containing ITD or ILD cues. In experiment 1, ITDs or ILDs were applied to the noise bands and roved independently on every trial. In experiment 2, the noise bands centred at 442 Hz and 5544 Hz were removed to investigate the effect of stimulus bandwidth. In experiment 3, the same two noise bands were present, but contained uncorrelated noise, reducing the effective bandwidth of binaural information. The results show that the cue-bands with the lowest- and highest centre frequency received the highest weights, while the other bands are equally weighted. This indicates that these edge frequency bands play an important role in lateralizing sounds.
Neural encoding of vowel formant frequency in normal-hearing listeners
Authors: Svirsky Mario , Won Jong-Ho , Clinard Christopher G. , Wright Richard , Sagi Elad , Tremblay Kelly
Published: June 2017

Physiological correlates of speech acoustics are particularly important to study in humans because it is uncertain whether animals process speech the same way humans do. Studying the physiology of speech processing in humans, however, typically requires the use of noninvasive physiological measures. This is what we attempted in a recent study (Won, Tremblay, Clinard, Wright, Sagi, and Svirsky, JASA 2016) which examined the hypothesis that neural representations of formant frequencies may help predict vowel recognition. To test the hypothesis, the frequency-following response (FFR) and vowel recognition were obtained from 38 normal-hearing listeners using four different vowels. This allowed direct comparisons between behavioral and neural data in the same individuals. FFR was used because it reflects temporal encoding of formant frequencies below about 1500 Hz. Four synthetic vowels with formant frequencies below 1500 Hz were used. Duration was 70 ms for all vowels to eliminate temporal cues and to make identification more difficult. A mathematical model (Sagi et al., JASA 2010) was used to predict vowel confusion matrices based on the neural responses. The mathematical model was successful in predicting good vs poor vowel identification performers based exclusively on physiological data.
Envelope-following responses elicited by modified speech sounds for estimating temporal processing dysfunction
Authors: Aiken Steven J. , Purcell David , Wang Jian
Published: June 2017

Speech is an ideal stimulus for eliciting envelope-following responses (EFR) given its inherent periodicities and biological importance. However, the speech EFR may reflect multiple aspects of temporal coding in the auditory nerve and brainstem driven by phase-locking to temporal fine-structure (TFS; carried predominantly by low-frequency speech harmonics) and the periodicity envelope (carried predominantly by unresolved high-frequency harmonics). This limits its utility as a measure of specific encoding deficits as a function of frequency. Multiple-fundamental frequency (multi-f0) speech sounds give rise to EFR related to narrow ranges of speech harmonics and thus may allow for assessment more specific with respect to frequency and type of temporal coding. This study investigated the relationship between the EFR obtained from multi-f0 speech, and individual differences in release from masking thought to reflect poor TFS coding. Multi-f0 EFR was measured in adults across a wide age-range with normal and near-normal hearing. Speech-in-noise thresholds were measured adaptively while manipulating talker f0 and spatial location using a virtual sound field. Results indicate that release from masking based on talker f0 is associated with the EFR elicited from only low-frequency speech harmonics, suggesting that speech-in-noise difficulties reflect distinct deficits in the encoding of TFS in the auditory periphery.
EEG in response to running speech: Relation with intelligibility
Authors: Francart Tom , Vanthornhout Jonas , Decruy Lien , Wouters Jan
Published: June 2017

It has recently been shown that the envelope of a running speech signal can be decoded from the EEG signal. The correlation between the decoded signal and the original envelope yields a measure of acuity of coding of the envelope (η). As the speech envelope is an important cue for speech intelligibility, we hypothesise that its neural representation is a prerequisite for speech intelligibility. So when factors related to higher order speech processing are ruled out, η should be highly correlated with speech intelligibility. We measured the EEG evoked by running speech masked by speech spectrum weighted noise at a number of signal-to-noise ratios (SNR), and (1) verified that η monotonically increased with SNR, and (2) compared a measure derived from η with behaviourally measured speech intelligibility. This was done for normal- and impaired-hearing listeners. To investigate the method's suitability for audiometry, we systematically varied the level of attention of the subject to the speech material, and found that the results were not strongly influenced by the level of attention when the right parameters are chosen for the decoder. We additionally measured listening effort using a new dual task paradigm and evaluated its effect on η.
Toward a cognitively controlled hearing aid
Authors: Dau Torsten , de Cheveigné Alain
Published: June 2017

The healthy auditory system can attend to weak sounds within complex acoustic scenes, a skill that degrades with aging and hearing loss. Recent technology such as microphone array processing should alleviate such impairment, but its uptake is limited by the lack of means to steer the processing towards one source among many. Within our auditory brain, efferent pathways put peripheral processing stages under the control of central stages, and ideally we would like such cognitive control to extend to the external device. Recent progress in the field of Brain Computer Interfaces (BCI) and some promising attempts at joint decoding of streams of audio and ECoG, EEG, or MEG suggests that such control might be possible. Is it? What scientific and technological hurdles need to be overcome to produce a “Cognitively Controlled Hearing Aid”? I will speak more specifically about our efforts to determine the reliability of EEG attention decoding in realistic acoustic scenes.
Robustness to real-world background noise increases between primary and non-primary human auditory cortex
Authors: Kell Alexander J. , McDermott Josh
Published: June 2017

In everyday listening, the sounds from sources we seek to understand are frequently embedded in background noise, which often profoundly alters auditory nerve spiking. To recognize sources of interest the brain must be somewhat robust to the effects of these background noises. To study the neural basis of listening in real-world background noise, we measured fMRI responses in human auditory cortex to a diverse set of thirty natural sounds, presented in quiet as well as embedded in thirty different everyday background noise textures (e.g., a bustling coffee shop, crickets chirping, etc.). We quantified the noise-robustness of neural responses by correlating each voxel’s response to the natural sounds in quiet with its response to those same sounds superimposed on background noise. Responses in core regions (commonly identified with primary auditory cortex) were substantially affected by background noise. However, noise-robustness increased with distance from primary auditory cortex: nearby non-primary areas were slightly more robust, while more distal areas were hardly affected by the background noises. Our results provide a neural correlate of the noise robustness of real-world listening, and offer evidence of a hierarchical organization in human auditory cortex.
Using visual cues to perceptually extract sonified data in collaborative, immersive big-data display systems
Authors: Lee Wendy , Chabot Samuel , Braasch Jonas
Published: June 2017

Recently, multi-modal presentation systems have gained much interest to study big data with interactive user groups. One of the problems of these systems is to provide a venue for both personalized and shared information. Especially, sound fields containing parallel audio streams can distract users from extracting necessary information. The way spatial information is processed in the brain allows humans to take complicated visuals and focus on details or the whole. However, temporal information, which can be better presented through audio, is processed differently, making dense sound environments difficult to segregate. In Rensselaer’s CRAIVE-Lab, sounds are presented spatially using an array of 134 loudspeakers to address individual participants who are working on analyzing data together. In this talk, we will present and discuss different methods to improve the ability of participants to focus on their designated audio streams using co-modulated visual cues. In this scheme, the virtual reality space is combined with see-through, augmented reality glasses to optimize the boundaries between personalized and global information. [Work supported by NSF #1229391 and the Cognitive and Immersive Systems Laboratory (CISL).]
Motion of intracochlear structures measured with a commercial optical coherence tomography (OCT) system
Authors: Ravicz Michael E. , Cho Nam H. , Maftoon Nima , Puria Sunil
Published: June 2017

Most knowledge of the motion of cochlear structures has been limited to measurements through the round window at the extreme base of the cochlea or through a hole made in the cochlear capsule, which can modify cochlear mechanics. Optical coherence tomography (OCT) provides the ability to measure shape or motion of structures through a thin layer of tissue or bone. The motion of cochlear structures has been measured in the mouse cochlear apex without making an opening into the cochlea, using a custom OCT system. Here we describe intracochlear vibrometry using a commercial OCT system. Specimens were prepared by opening the middle ear while maintaining part of the cartilaginous ear canal. A Thorlabs 905nm Ganymede III-HR OCT system with 100-kHz camera frame rate was used to measure cochlear anatomy in a 2-D radial slice (B-scan) and dynamic displacements along a line (A-line) that intersected several cochlear structures in response to tones presented to the ear canal. Differences in the magnitude and phase of the displacements along the A-line show differences in the motion of cochlear structures. These data will provide information for validating 3D finite element models of the cochlea. [Work supported by NIDCD R01 DC07910.]
Cochlear gain reduction in listeners with borderline normal quiet thresholds
Authors: Strickland Elizabeth A. , Holt Alexis , Morris Hayley
Published: June 2017

The medial olivocochlear reflex (MOCR) decreases the gain of the cochlear active process in response to sound. Evidence of the MOCR has been measured in humans using physiologic and psychoacoustic techniques. We have focused on psychoacoustic techniques, which convey the behavioral effects of gain reduction. We have modified forward masking paradigms understood to measure frequency selectivity and the input/output function at the level of the cochlea so that the stimuli (masker and signal) should be too short to evoke the MOCR. With this paradigm, a longer sound (precursor) is presented before these stimuli to evoke the MOCR. The present study examines the relationship between quiet threshold and the magnitude of the threshold shift (interpreted as gain reduction) produced by a pink noise precursor, as a function of precursor level. Gain reduction was measured at 2, 4, and 8 kHz in listeners with clinically normal quiet thresholds who had borderline normal quiet thresholds and low OAEs at one or more of these frequencies. Maximum gain was estimated by measuring the threshold masker level for a masker at the signal frequency and a masker well below the signal frequency. The relationship between quiet threshold, maximum gain, and gain reduction will be discussed.
Nonlinear response of human middle ear to high level sound
Authors: Cheng Jeffrey Tao , Remenschneider Aaron , Kozin Elliott , Furlong Cosme , Rosowski John J.
Published: June 2017

The human middle ear functions as an acoustic transformer that converts environmental sound energy into mechanical drive to stimulate the cochlea for hearing. This transformation has been demonstrated to be linear at sound levels at least as high as 125 dB. While this behavior can be expected to breakdown at some higher level, that level has not been demonstrated. In this study, we use cadaveric human ears to study middle ear nonlinear responses to high levels of sound. Tones between 200 and 10000 Hz with levels as high as 160 dB SPL are generated by a customized horn speaker and delivered to the ear canal opening. Sound pressure levels near the tympanic membrane surface are monitored by a calibrated pressure sensor. Simultaneously, Laser Doppler Vibrometry is used to record sound induced vibrations of the Umbo, in the center of the tympanic membrane, and the stapes, at the interface of the middle and inner ear. These data allow us to calculate the middle ear transfer function: the ratio of the stapes to umbo vibration. We use these measurements to investigate what is the threshold sound pressure above which the human middle ear response is no longer linear.
The decodability of sound sources and categories from their acoustic representations as they develop over time
Authors: Ogg Mattson , Slevc L. Robert
Published: June 2017

The temporal dynamics of sound impose constraints on how listeners can so rapidly and effectively identify objects in their environment. However, it is unclear what physical features distinguish acoustic sources, and how the roles of those features change as a sound unfolds. Potential differences between the optimal acoustic features for distinguishing sounds and those used by listeners could reveal new insights into how the auditory system functions and which aspects of sound it prioritizes. Thus, we investigated 216 high quality sound tokens decomposed into a set of acoustic features derived from an ERB filter bank in 5 millisecond increments. Support vector machine classifiers were iteratively trained and tested on sound categories (instrument, speech, environmental) and sound sources within those categories (e.g. instruments, speakers) at each time point. Decoding analyses were conducted on acoustic features (e.g. spectral centroid, aperiodicity) and the raw ERB filter output. Categories were decoded better using high level acoustic features, whereas individual exemplars were decoded better from the filter bank output, although these patterns varied over time. By examining how the classifiers weight individual features, these data reveal the relative contributions of specific acoustic features to sound discrimination as a function of time.
Investigating brain stimulation during amplitude modulation detection: A pilot study
Authors: Fogerty Daniel , Fridriksson Julius , Miller Rachel E. , Basilakos Alexandra , Rorden Chris , Bonilha Leonardo
Published: June 2017

Positive effects of transcranial brain stimulation have been observed across a number of cognitive and sensory domains. This non-invasive external electric stimulation technique has been shown to induce changes in patterns of brain activity, and these changes have been suggested to be responsible for affecting sensory perception. Recent findings have suggested transcranial alternating current stimulation (tACS) can entrain neural oscillations, meaning the externally applied alternating current can affect firing rates by inducing oscillatory coherence. Thus, tACS introduces a methodology to causally investigate the effect of neural oscillations on auditory perception. The purpose of this study was to determine if tACS improves amplitude modulation detection for individuals with normal hearing. In a crossover design, young healthy adults with normal hearing were presented with sinusoidally amplitude-modulated (SAM) tones while undergoing tACS. SAM tones were presented with a modulation rate of 4 Hz or 8 Hz; the modulation depth was set to each participant’s modulation detection threshold. The frequency of tACS was varied across two sessions to match the modulation rate of one of the SAM tones. Preliminary results suggest variable responses to tACS across the participants tested. Considerations for auditory tACS research will be discussed.
Middle ear muscle reflex in children with auditory processing and/or attention deficit/hyperactivity disorders
Authors: Johnson Nicole E. , Morlet Thierry , Sklar Rachele , Grinstead Laura , Nemith Julianne , Nagao Kyoko
Published: June 2017

Previous studies suggest abnormal auditory efferent systems in children with Auditory Processing Disorder (APD) and children with Attention-Deficit/Hyperactivity Disorder (ADHD). The current study examined the middle ear muscle reflex (MEMR) of three groups of children. The APD group consisted of 32 children with auditory processing deficits and the ADHD group consisted of 50 children. The children with ADHD were either taking ADHD-related medication (n = 31) or not taking such medication (n = 19). All subjects were selected from an existing database. Ipsilateral MEMR responses at 0.5, 1, 2, and 4 kHz in each ear were categorized as normal or abnormal (defined as any response over 90dB HL, or no response). Contingency table analyses and corresponding chi-square tests were used to compare the proportion of abnormal MEMR responses between groups and ears. We found no significant difference between APD and ADHD groups. The children with APD and children with ADHD medication showed no significant ear difference (ps > 0.5), whereas children without ADHD medication exhibited a marginally significant difference between the left and right ear (p = 0.044). The results suggest abnormal asymmetry in auditory efferent pathways in ADHD. Further studies should include contralateral pathways to further evaluate asymmetrical patterns in these groups.
Power dissipation in the Organ of Corti
Authors: Prodanovic Srdjan , Gracewski Sheryl , Nam Jong-Hoon
Published: June 2017

In the cochlea, acoustic energy is transmitted toward the apex through the vibrations of a viscoelastic partition known as the organ of Corti complex. The dimensions of the vibrating structures range from a few hundred micrometers to a few micrometers. Vibrations of micro-structures in viscous fluid are subjected to energy dissipation. Because the viscous dissipation is considered to be detrimental to the function of hearing—sound amplification and frequency tuning, the cochlea is believed to use cellular actuators to overcome the dissipation. We have developed a computational model of the cochlea that incorporates viscous fluid dynamics, organ of Corti micro-structural mechanics, and electro-physiology of the outer hair cells. The model is validated by comparing with experimental results in the literature, such as the viscoelastic response of the tectorial membrane, and the cochlear input impedance. Using the model, we investigated how dissipation components in the cochlea affect its function. Our results suggest that most energy dissipation occurs within the organ of Corti complex, not in the scalar fluids. Our results suggest that appropriate dissipation enhances the tuning quality by confining the spread of energy from the amplification site.
A non-collinear mixing technique to measure the acoustic nonlinearity parameter of an adhesive bond from one side of the sample
Authors: Qu Jianmin , Ju Taeho , Achenbach Jan , Jacobs Laurence
Published: June 2017

he Acoustic Nonlinearity Parameter (ANLP) of adhesive bonds. One of the most significant features of the new method is that it requires only one-side access to the adhesive bond being measured, which significantly increases its utility in field measurements. To demonstrate the effectiveness of the newly developed technique, an adhesively jointed aluminum sample was measured with different heat loading time, using the non-collinear mixing technique with a longitudinal and a shear wave as incident waves to obtain the ANLP of the adhesive bond. The measured results clearly show that the ANLP varies with aging time. To verify that the signals received from the shear wave receiver are indeed the mixed wave, the finite element method was used to simulate the wave motion in the test sample. The simulation results clearly show that the signals recorded by the shear wave receiver are the desired mixed wave, whose amplitude is proportional to the ANLP of the adhesive bond.
Stochastic finite element on vibration analysis of composite plates having random damping parameters
Authors: Sepahvand Kheirollah , Marburg Steffen
Published: June 2017

Damping parameters of composite structures possess significant uncertainty due to the structural complexity of such materials. Considering the parameters as random variables, this work uses the generalized polynomial chaos (gPC) expansion to capture the uncertainty in the damping and in the vibration responses of structures. A non-sampling based stochastic finite element formulation for damped vibration analysis is developed in which the gPC expansion is used to represent damping uncertainty and stochastic responses with unknown deterministic functions. The constructed gPC expansions for the parameters are used as random inputs to the FEM model to realize the responses on a few number of collocation points generated in random space. The realizations then are employed to estimate the unknown deterministic functions of the gPC expansion approximating the responses. The application of the proposed method is practiced on sample fiber-reinforced composite plates having random modal damping parameters. Utilizing a few random collocation points, the method indicates a very good agreement compared to the sampling-based Monte Carlo simulations with large number of realizations.
Not-quite-naïve listeners: Students as an audience for gamified crowdsourcing
Authors: McAllister Byun Tara , Harel Daphna , Hitchcock Elaine R. , Lopez Melissa
Published: June 2017

Collecting independent listeners’ judgments of speech accuracy/intelligibility is an essential component of research on speech disorders. Raters can be trained clinicians, students in speech-language pathology, or naïve listeners (now commonly recruited online via crowdsourcing platforms such as Amazon Mechanical Turk/AMT). However, limited comparison data exist to guide researchers in determining which rater population to use. We describe a study (Hitchcock et al., in prep) in which 2,256 tokens of English /r/ at the word level, produced by five children receiving intervention for /r/ misarticulation, were rated in a binary fashion using the online platform Experigen. Raters were certified clinicians (n = 3), students in speech-language pathology (n = 9 unique listeners per token), or naïve listeners recruited on AMT (n = 9 unique listeners per token). Interrater reliability was higher when comparing modal ratings between clinicians and students (Cohen’s kappa=.73, CI=.7-.77) than between clinicians and naïve listeners (Cohen’s kappa=.64, CI=.6-.68). However, students offered none of the speed/efficiency advantages available through AMT. We posit that students would be more motivated to complete ratings if they received feedback on accuracy. As a future direction, we propose a hybrid crowdsourcing platform/educational game, designed to sharpen important perceptual skills while also motivating students to contribute valid ratings for research data.
Use of crowdsourcing platforms to examine listener perception of disordered speech
Authors: Lansford Kaitlin L. , Borrie Stephanie A.
Published: June 2017

Crowdsourcing websites, such as Amazon’s Mechanical Turk (MTurk), offer a promising platform for researchers to examine the perceptual consequences of disordered speech production from a diverse sample of listeners. Although data collection via crowdsourcing mechanisms is largely unconstrained, and not without its limitations, it is cost-effective, time-efficient, and has been demonstrated to yield similar results to the same experiments conducted in the laboratory under the direct supervision of the researcher. For example, in our recently published perceptual learning experiment, MTurk and laboratory participants demonstrated equivalent intelligibility improvements following familiarization (perceptual training) with moderate hypokinetic dysarthric speech. Thus, in addition to supporting the ecological validity of perceptual training as a listener-focused means to reduce the intelligibility burden of dysarthria, these results support the continued use of crowdsourcing platforms to address empirical questions related to listener perception of disordered speech. The current presentation will review our previous findings of data equivalence between laboratory and MTurk perceptual experiments and will introduce the preliminary findings of an ongoing project in which perceptual data were collected exclusively from MTurk participants to examine generalization effects of perceptual training with different types and severities of dysarthric speech.
Extending the usable bandwidth of an acoustic beamforming array using phase unwrapping and array interpolation
Authors: Goates Caleb B. , Harker Blaine M. , Gee Kent L. , Neilsen Tracianne B.
Published: June 2017

The response of a frequency-domain beamformer, which is based on the beamforming array cross-spectral matrix, is usually limited by the spatial Nyquist frequency. This paper presents a method for overcoming grating lobes in beamforming on broadband signals using phase unwrapping and array interpolation. When the phase of each cross spectrum is successfully unwrapped across frequency, spatial array interpolation can be performed on both the magnitude and unwrapped phase of the cross spectral matrix at each frequency. This process can approximate the response of a dense array from one that is undersampled. For cases where the cross-spectral magnitude and phase vary smoothly, interpolation is straightforward, even above the spatial Nyquist frequency. Two example applications from anechoic measurements are presented: localization of a single broadband source, and characterization of a broadband source whose location changes with frequency. It is found that grating lobes are suppressed and the source is localized at frequencies up to at least eight times the spatial Nyquist frequency for these cases. [Work supported by the National Science Foundation and the Office of Naval Research.]
Toward the assessment of walls acoustic impedances from the analysis of first and second order reflections based on room impulse responses
Authors: Peic Tukuljac Helena , Pham Vu Thach , Lissek Herve , Vandergheynst Pierre
Published: June 2017

In order to address optimal acoustic control in rooms, an accurate characterization of the room shape and wall properties is required. There are only a few approaches that model the wall impedances. Most of them rely on finite difference time domain methods, which are limited to shoebox-shaped rooms and only valid at low frequencies (non-rectangular rooms and high frequencies lead to extremely high computational complexity). In order to overcome these limits, we propose the estimation of walls’ acoustic impedances based on the analysis of the room impulse responses. In room impulse responses, the early reflections’ amplitudes are proportional to the reflection coefficient of the corresponding wall element. The location of points on the walls (origins) for the first and second order echoes can be easily determined for a known room. The values of impedances at the origins of the first order echoes are determined directly from the room impulse response. The second order echoes' represent a product of the influence of wall points that belong to its path. Values of individual impedances are extracted from these echoes by overlapping origins of first order echoes of one receiver’s position with one origin of the second order echoes of another receiver’s position.
Experimental results on acoustic communication through drill strings using a strain sensor receiver
Authors: Alenezi Ali H. , Abdi Ali
Published: June 2017

Drilling oil wells is a costly, complex and high risk operation, especially for many wells whose depths can be at least several thousand feet. To minimize the cost and risk involved in drilling, typically there are several sensors mounted near the drill bit, to measure important parameters such as temperature and pressure around the drill bit. The collected information needs to be sent to the surface, to assist the driller with controlling and steering the drill bit. Compared to the use of highly expensive and vulnerable cables for wired communication in deep oil wells, or very low rate communication using pulses of mud in wells, transmission of information using acoustic signals thru the drill string is a feasible and promising method. In this paper, we use a drill string acoustic communication testbed, where the receiver is a strain sensor. This sensor measures local fractional displacements due to received vibrations. Using experimental data, we study key characteristics of a drill string channel impulse response, sensed by a strain sensor. We also demonstrate how the strain channel response structure can control the communication system performance. [This work was supported in part by the National Science Foundation (NSF), Grant IIP-1340415.]
Pitch pattern matching based speech enhancement
Authors: Wang Dongmei , Hansen John H. L.
Published: June 2017

The problem of speech enhancement in diverse noisy conditions has historically focused on the vocal tract spectral magnitude. However, studies have shown that improved quality, speaking style and speaker identity are all impacted by reliable prosody/F0 (pitch) information for human listening. In this study, we propose a speech enhancement algorithm based on pitch pattern matching. It can be considered as an example-based method since we attempt to replace the speech segment in the noisy speech with corresponding detected components from a dictionary which contains the clean speech signals. The average pitch value as well as overall pitch dynamic trends are used as features for pitch pattern matching. The speech segment in the dictionary with the best matched pitch pattern feature will be used to assist in the computation of an enhanced speech segment in the noisy speech sample. Here, Wiener filter is used for obtaining the target baseline speech from the noisy speech. The experimental results show that the pitch pattern feature is more computational efficient than a spectral based feature alone for speech enhancement, while obtaining similar speech enhancement performance in terms of both speech intelligibility and speech quality.
Impact of random failures and tolerances on ensemble average array beampattern and array gain
Authors: Berkman Evan F.
Published: June 2017

Analytic expressions for expected array beampattern with random element failures and random perturbations in element amplitude and phase response have been long available for arrays of omnidirectional elements. This work extends prior results to address random failures and tolerances of directive as well as omni-directional elements in curved as well as rectilinear array geometries and to blocks or groups of elements as well as single elements. The analytic result for both failures and multiple types of tolerances is concisely expressed in a single simple universal expression with explicit parametric dependencies. This expression is amenable to efficient numerical evaluation and quantitative results are trivially scaled to any combination of failure probabilities and standard deviation of various types of tolerances. Insightful interpretation of the results in terms of response of an ideal array plus a perturbation array is provided. The impact of failures and tolerances on array beampattern is translated to again simple expressions for array noise response and array gain for an incident diffuse acoustic noise field of arbitrary directionality. The emphasis on concise physically interpretable expressions is due to my training as a young engineer by Ira Dyer and the group of outstanding scientists and engineers he assembled at BBN.
Experimental results on synchronization with chirp signals using a vector sensor receiver
Authors: Zhang Erjian , Abdi Ali
Published: June 2017

Chirp signals, also known as linear frequency modulated signals, are widely used for synchronization, signal acquisition, and frame detection in underwater communication systems. This is due to the peak at the output of the chirp matched filter at the receive side. In low signal-to-noise ratio (SNR) scenarios, however, this peak can be buried in noise, which results in major synchronization errors and system performance loss. While a scalar array of spatially separated hydrophones can increase SNR to improve synchronization, the size of the array may not be suitable for small platforms. Acoustic vector sensors, on the other hand, are small-size devices that can serve as multichannel communication receivers. In this paper, performance of a vector sensor receiver for synchronization using a chirp signal is studied. Our experimental results indicate that a compact vector sensor receiver can significantly enhance the output of the filter matched to the chirp signal. This is because the proposed vector matched filter significantly suppresses the noise and provides a sharp peak at the output. This is particularly important for synchronization and signal acquisition in underwater communication systems operating in low SNR environments. [This work was supported in part by the National Science Foundation (NSF), Grant IIP-1500123.]