bluuebunny/crossref_metadata_2025 huggingface dataset:
Control modeling toward understanding articulatory disfluency in autism spectrum disorder
Authors: Talkar Tanya , Lammert Adam , Quatieri Thomas
Published: November 2019

Speech control models founded on principles of neuroscience have potential to create powerful clinical tools for personalized medicine and for understanding behavior. We explore two prominent speech control models: Directions into Velocities of Articulators (DIVA) (Guenther, 2006) and Task Dynamics (TD) (Saltzman, 1991) toward an understanding of speech disfluency of autism spectrum disorder (ASD). To introduce perturbations into both systems in the common framework of (Parrrell, 2018), we translate the TD model into MATLAB Simulink. For word stimuli, we compare both models’ acoustic outputs, under the hypothesis that speech impediments in ASD arise from overreliance on delayed sensory feedback (Lin, 2015). For DIVA, the output from a 50ms auditory feedback delay contain increased formant production errors and formant oscillations at about 10Hz, sometimes observed in disfluent ASD cases, but also seen in stuttering (Civier, 2010). The same delay introduced in the proprioceptive and tactile feedback in TD takes 5x the time to settle at the target, creating about 20Hz oscillations in TD articulators. Though neither fully capturing disfluency in ASD, together these models lay the groundwork for a more complete neurocomputational speech control model that can be used to longitudinally track and potentially guide ASD speech therapy.
Distribution of Tourette's verbal tics produced during active speech
Authors: Llorens Mairym
Published: November 2019

Tourette syndrome causes individuals to produce unwanted movements and vocalizations called tics. Verbal tics are those tics that closely mimic words/phrases—their execution requires the same set of articulators needed to produce speech. Tics respond to a preceding urge to tic and are akin to other urge-based actions like cough. Urges to tic can arise at any time but speech and tics must be deployed sequentially as they cannot be produced simultaneously. How are conflicts between the speech and tic motor systems resolved? This study tests the prediction that tics interfere with production of words and intonational phrases less frequently than expected by chance. As predicted, pilot data from one subject showed that the expected probability of words and tics co-occurring were these events independent was significantly higher than the observed likelihood of tics interrupting words. Correspondingly, the likelihood of intonational phrase-final boundaries co-occurring with tics if these events were independent was significantly lower than the observed probability of tics given intonational phrase-final boundaries. Data from multiple people with Tourette’s will be presented. Results suggest that utterances are “protected” from tic interruptions, providing evidence that the tic and speech motor systems are linked in production. [Work supported by NIH and NSF.]
Individual differences across caregivers in acoustic implementation of infant-directed and adult-directed speech: Modeling impacts on intelligibility in children with cochlear implants
Authors: Arjmandi Meisam K. , Houston Derek , Svirsky Mario , Wang Yuanyuan , Lehet Matt , Dilley Laura
Published: November 2019

Caregivers modify their speaking style from adult-directed speech (ADS) to infant-directed speech (IDS) when talking to infants. However, it is unclear how individual caregivers acoustically implement differences between ADS and IDS, and how these differences may affect experienced speech intelligibility by infants, particularly those with cochlear implants (CIs). Seven female talkers spoke fifteen utterances both in IDS and ADS. We analyzed these utterances and their cochlear implant-simulated versions (using 22-channel noise vocoders) to investigate how acoustic distances between ADS and IDS varied across talkers (based on Mel-frequency cepstral coefficients, MFCCs) and how the effect of these shifts in speaking style on utterance intelligibility were different between talkers (using the acoustic index of speech-to-reverberation modulation energy ratio tailored to CI devices, SRMR-CI). Results showed substantial variability across talkers comparing ADS and IDS for caregivers’ acoustic profiles from MFCCs and for speech intelligibility from SRMR-CI. These findings suggest that acoustic choices by individual mothers may differentially affect recoverability of intelligible words from speech signals by children with CIs, which may contribute to differences in these children’s language outcomes. [Work supported by NIH grant R01DC008581.]
Developmental changes in categorical perception of Mandarin Chinese tones
Authors: Qin Dandan , Cheng Bing , Zhang Wen , Zhang Yang
Published: November 2019

This study examined developmental changes in categorical perception of Mandarin Chinese lexical tones by 22 adults and 16 10-year-olds, who were all native Mandarin Chinese speakers. The speech stimuli included three types of continuum, tone 1 (T1)/tone 2 (T2), tone 1/tone 4 (T4) and tone 2/tone 4. The experimental protocol used conventional identification and discrimination tasks. The results confirmed categorical perception of the tonal continua by both adult and child groups. Refined tests further revealed the existence of T1 category with two phonetic boundaries in the T2/T4 continuum. More importantly, age-dependent differences were observed in discrimination accuracy and sensitivity to pitch height. Collectively, the data suggest that although adult-like categorical perception of lexical tones can be found by 10 years of age and earlier as the literature suggested, the dynamic developmental trajectory in fine-tuning the categorical perception of lexical tones extends beyond the age of 10.
Synthetic aperture towed-array processing: The Edmund J Sullivan Legacy
Authors: Candy James V.
Published: November 2019

Dr. Edmund J. Sullivan developed synthetic aperture towed-array processing. Confronted by many nay-sayers Dr. Sullivan persisted with his work and developed the first approach as the signal processing group leader (SPGL) at the SACLANT ASW Center (now CMRE) in La Spezia, Italy. The first notional passive synthetic aperture processor using an overlapped correlation method was jointly developed [“Extended Towed Array Processing by Overlapped Correlator,” (JASA, 1989)]. This idea was to blossom even further in his collaborative works (Stergiopolous et al.) extending these ideas to a fully, recursive (in-time) model-based passive-synthetic aperture processor [“Space-time array processing: A model-based approach” (JASA, 1997)]. He also began collaborations with the Swedish Navy where he performed joint experiments in model-based passive synthetic aperture evaluating its performance in the ocean and demonstrating its effectiveness. Dr. Sullivan began mentoring more researchers and teaching at the University of Rhode Island where he advised students in underwater acoustics and processing leading researchers to the model-based approach (Cousins), synthetic aperture processing (Edelson) [“On the performance of the overlap-correlator synthetic aperture technique” (JASA, 1991)] and broadband processing (Holmes) [“Broadband passive synthetic aperture” (JASA, 2006) ]. His ideas were summarized in his recent text, [Model-Based Processing for Underwater Acoustic Arrays (Springer, 2015)].
Ed Sullivan and model-based ocean acoustic signal processing
Authors: Michalopoulou Zoi-Heleni
Published: November 2019

Ed Sullivan was a pioneer in model-based signal processing and his contributions have inspired numerous scientists in the fields of acoustic signal processing and ocean acoustics. Work by Ed and his colleagues is among the first efforts to formulate ocean acoustic inverse problems in a sequential filtering framework; methods were developed that overcome the destructive role of mismatch between assumed and true physics in inverse problems. In paper “Model-based ocean acoustic signal processing,” Acoustics Today, July 2011, Ed Sullivan identifies three main directions in signal processing, detection, classification, and estimation; he focuses on estimation and presents a study of model-based methods for inverse problems and how those methods can be successfully applied to the field of ocean acoustics. This presentation reviews these applications and discusses the impact of Ed Sullivan’s contributions on other advances in ocean acoustic signal processing.
Sullivan’s contributions to model-based acoustic array processing for underwater and beyond
Authors: Xiang Ning
Published: November 2019

Edmund Sullivan’s long and fruitful research efforts have spanned a number of fields, both underwater and ashore. Most notable to this author is his achievement on model-based processing for underwater acoustic arrays, documented in a 2015 Springer monograph. Through his unremitting dedication and long-time service to the Acoustical Society of America, and most prominently, the Technical Committee on Signal Processing in Acoustics, Ed’s achievements have long garnered utmost respect and reputability. This paper reviews his recent effort on model-based Bayesian signal processing with a rigorous handling of the Kalman filter, and acoustic array processing within the Bayesian framework. This paper also reflects personal remembrance of the mutually beneficial collaboration with Ed Sullivan in acoustic landmine detection and Bayesian analysis.
Trans-dimensional range-dependent geoacoustic inversion using modal dispersion data in the South China Sea
Authors: Weng Jin-Bao , Dosso Stan E. , Chapman N. Ross , Yang Yan-ming , Peng ZhaoHui , Wang GuangXu , Zhang Lingshan
Published: November 2019

This paper presents geoacoustic inversion of modal dispersion data in the South China Sea using a single hydrophone and multiple impulsive sources at ranges from 5–100 km along a shallow-water track with slowly varying bathymetry. As a first step, the source waveform and bubble pulse are deconvolved from the recorded time series using a short-range source recording, corrected for the surface reflection. A time-frequency warping analysis is used to filter individual modes and obtain dispersion (arrival time as a function of frequency) data for three modes. Trans-dimensional Bayesian inversion is applied to the modal dispersion data, based on probabilistic sampling over an unknown number of seabed layers. Range-dependent inversion is considered, based on separating the environment into a sequence of range-independent sections, with frequency-dependent modal propagation times summed over segments. The inversion results are compared to core samples collected at sites along the survey line and to an independent headwave arrival-time analysis of the impulsive-source data.
Characterization of acoustic attenuation in a coral reef environment
Authors: Thilges Kayla , Van Uffelen Lora J. , Freeman Simon , Freeman Lauren
Published: November 2019

Coral reefs are complicated and understudied acoustic propagation environments. In addition to geometric spreading, there are propagation losses due to bottom attenuation, volumetric scattering, and boundary scattering from bathymetric variability and sea surface roughness. On a Hawaiian coral reef, a field test was performed in shallow water during which low-level tones were projected from an underwater speaker and received by a single hydrophone at various ranges up to 500 m from the source. Acoustic data gathered during the field test were analyzed to characterize the sound propagation environment. Conventional geometric spreading assumptions were challenged for the sloping bathymetry characteristic of the Hawaiian coral reef environment. Geoacoustic parameters of the coral reef environment were extracted from transmission loss measurements using a nonlinear least-squares inversion. Losses were estimated for frequencies of 500 Hz, 2 kHz, 5 kHz, 10 kHz, and 15 kHz.
Creative reuse and renovation in educational performance environments
Authors: Hougland Dana
Published: November 2019

Educational institutions for the teaching and training students in the arts are often hampered by facitlity limitations. Shifting priotities for the space, outdated acoustical design direction, and budget restrictions place creative constraints on artists and educators. The talent of the performers and arts programs often far exceed the facilities. Funding for renovation is a slow process and may take decades before funds are available to modify and improve the acoustics of performance spaces particularly in rural areas. Volume, shaping, and poor reflected energy are the most common issues. Budget is a limiting factor at many institutions, however, creative collaborations with the design architects allow for significant acoustical improvement. Several case studies will be presented that examine improving the acoustical performance for both the performers and audience while working within the constraints of existing structures. Examples presented include conversion of a classic fan shaped theatre into a musical performance space, conversion of gymnasiums into rehearsal/performance spaces and acoustic enhancement of traditional auditoriums.
Establishment of ANSI/ASA S3/SC1.6—Standardization of evoked potential hearing test methods in toothed whales
Authors: Houser Dorian S.
Published: November 2019

The use of auditory evoked potentials (AEPs) to test the hearing of toothed whales has produced hundreds of toothed whale audiograms over the last two decades. Unfortunately, multiple approaches to obtaining AEP audiograms have been employed, including different methods for estimating the hearing threshold, the use of different acoustic test stimuli, different approaches to calibration, etc. Lack of methodological consistency has resulted in considerable variability in threshold estimates between researchers and laboratories and conclusions about differences in the hearing abilities of toothed whale populations that are likely erroneous. To address this issue, ANSI/ASA S3/SC1.6-2018 was developed. The document proposes standardized methods for performing AEP hearing tests in toothed whales. Implementation of the standard should reduce variability in threshold estimates obtained by different laboratories and researchers and make the resulting audiograms more comparable. Although AEP hearing tests are common for the clinical assessment of hearing in stranded whales and their use for this purpose is not substantially impacted by methodological variances, audiogram comparability is necessary for the broader implementation of AEP audiograms into environmental impact assessments and regulation. Thus, the standard should facilitate environmental stewardship efforts to manage the impact of anthropogenic sound on marine mammals.
Design of a miniature evoked potential testing system for animal assessment of hearing
Authors: Yonovitz Al , Smith Silas
Published: November 2019

A specially built apparatus has been designed to allow novel methodologies for simultaneous measurement of early, middle and late evoked potentials. Evoked potentials can be measured in active, unsedated, unrestrained rats. Typically, early evoked potentials are measured with animals that are anesthetized. Middle and late evoked potentials must be measured in awake animals. Chronically implanted electrodes provide the bioelectric potentials. To accomplish this active measurement, data are transferred wirelessly from a portable unit attached to the rat, to a computer via wireless Bluetooth. The system is controlled with a Raspberry Pi. Included are a differential input amplifier, A/D converter and low and high pass filters. The miniature system also includes a means to provide a wide range of auditory stimuli typically driving a small insert earphone. All programs and auditory stimuli are stored on micro SD cards. The pack stays in place and is not removable by the animal. This system is ideally suited for measurements of backward masking middle latency responses as a marker for Auditory Processing Disorders (APD). The pack is light and easily attachable.
Monitoring transitions between dispersed and school modes of sardines and anchovies with Bioacoustic Absorption Spectroscopy (BAS) measurements
Authors: Diachok Orest
Published: November 2019

BAS measurements at a biologically intense site in the Santa Barbara Channel permitted monitoring the temporal variation of absorption lines associated with fish in dispersed and school modes. Transmission Loss (TL) was measured at frequencies between 0.25 and 5 kHz between a fixed source and fixed receiver array separated by 3.7 km for 20 hours. Echo-sounder measurements: the average depth of fish at night was about 13 m. Trawls: 16 cm sardines and 10.5 cm anchovies dominated. The frequency dependence of bio-attenuation was derived through comparison of TL measurements and calculations. The latter were based on a normal mode code recently extended for broadband calculations (Porter, 2019). Bio-attenuation at 1.1 kHz, the resonance frequency of dispersed sardines, was 35 dB at 2200 l, 15 dB at 0100 l and 35 dB at 0400 l. When bio-attenuation at 1.1 kHz was minimum, bio-attenuation 0.4 kHz, attributed to schools of sardines, was maximum. The absorption line at 0.4 kHz is consistent with previously reported measurements and theoretical calculations of the resonance frequency of schools of 16 cm sardines (Diachok, 1999). Bio-attenuation at 1.8 and 0.6 kHz, attributed to 10.5 cm anchovies in dispersed and school modes respectively, were also anti-correlated. [Research supported by ONR.]
Effect of low intensity ultrasound on cancerous prostate cell viability compared to viability of healthy prostate cells
Authors: Herd Maria-Teresa , Marquez Barbara , Seay Hannah , Verducci Chloe
Published: November 2019

Cancer continues to be a leading cause of death in the United States. Currently, the most commonly utilized treatment options in radiotherapy and chemotherapy raise serious biocompatibility and toxicity concerns. Given the adverse side effects of current treatment, it is important to explore modalities that can clearly discriminate between healthy and cancerous tissue. This investigation identifies low intensity ultrasound as a candidate for cancer therapy and tests whether it demonstrates a discriminatory response between cancerous prostate cells and healthy prostate cells. Cells were exposed to a 5 MHz, 0.126 W/cm2 wave for 60 minutes via a single element unfocused transducer in a water tank-controlled environment at 33 °C. Cell counts (alive and dead) before and after exposure to ultrasound were performed. On average cancerous prostate cells showed a viability of 53% compared to a viability of 99% for benign prostate cells, indicating a significant difference in observed death between healthy and cancerous prostate cells.
Remote hydrogelation by metal-ligand coordination using high intensity focused ultrasound
Authors: Jonnalagadda Umesh , Nguyen Minh , Li Feifei , Lee Jim , Liu Xu , Goto Atsushi , Kwan James J.
Published: November 2019

Smart materials have garnered interest due to their potential and versatility in a wide range of applications from tissue engineering, drug delivery, and cosmetics. It should be appreciated that while ultrasound has been utilized as a stimulus for smart materials, including organogels, its application has primarily been focussed below 100 kHz and for chemical scission. To investigate the potential of high intensity focussed ultrasound (HIFU) for gel formation, we synthesized a water-soluble comb-like polymethacrylate copolymer with randomly distributed tridentate ligands. We utilized reversible metal-ion non-covalent interactions in the design of our polymeric system to bind monomeric units together into temporary crosslinking networks or polymeric aggregates. To minimize the immediate formation of the metallogel, other functional groups within the polymer were involved in competitive reversible binding processes with the solvated metal ions. We show that HIFU triggers hydro-metallogelation by breaking the weaker labile interactions that in turn may allow for more favourable metal ion-ligand chelation, resulting in the formation of a gel network. Considering the established targeting potential of HIFU, our results here open a framework for targeted in situ triggered acoustic hydro-metallogelation for biomedical and industrial applications.
Determining structural acoustic properties from noise-based holographic measurements
Authors: Sternini Simone , Sarkar Jit , Rakotonarivo Sandrine , Bottero Alexis , Williams Earl G. , Tippmann Jeffery D. , Kuperman William A.
Published: November 2019

A facility has been constructed to determine the scattering properties of an arbitrary object in an external medium by measuring its response to a noise field in a laboratory. The measurements are made using a holographic array of a double layer of MEMS microphones placed in the near field of the object; the array is embedded in a 3-D printed scaffold structure which surrounds the object. Cross correlating the MEMS acoustic data together with holographic signal processing yields the structural impedance matrix which together with additional straightforward computation provides the scattering properties of the object when placed (loaded) in an external medium [J. Acoust. Soc. Am. 134, (6) (2013); J. Acoust. Soc. Am. 142(1) (2017)]. The theory underlying the experimental procedure is explained together with results from measurements of a capped cylindrical shell that are also compared to finite element calculations. [Work partially supported by the Office of Naval Research.]
Overview of efforts within ASA to promote effective communication of science to the public
Authors: Piacsek Andrew A. , Kloepper Laura
Published: November 2019

As the fruits of science and technology have permeated nearly every facet of modern life, and public policy decisions increasingly rely on knowledge created by the scientific process, there is growing concern that a deficiency of scientific literacy contributes to poor decisions in both private and public spheres. The ASA, consistent with its mission "to generate, disseminate, and promote the knowledge and practical applications of acoustics," is working to address this deficiency by facilitating communication between its members and journalists. The Public Relations Committee, together with Education in Acoustics and Student Council, have organized four special sessions since 2011 on the subject of effective communication of science to non-scientists. At the spring 2018 meeting in Minneapolis, Student Council organized a science communication workshop aimed at students and early career members. At the current meeting in San Diego, the Monday Tutorial Session is an active participation workshop, led by media professionals, designed to boost the skills and comfort level of ASA members in effectively communicating technical concepts, as well as the underlying scientific process, to members of the media. A summary of these efforts, and their significance for the Society, will be presented.
PhonoNet: Multi-stage deep learning for raga preservation in hindustani classical music
Authors: Chowdhuri Sauhaarda
Published: November 2019

Hindustani classical music is an ancient improvisational form of music based on ragas, melodic frameworks which have no written representation and are passed down through a fading oral tradition. The proposed system, PhonoNet, aims to preserve Hindustani classical music by providing computational prediction of ragas, so singers can receive live feedback when learning. PhonoNet also creates a visual format for documenting and preserving raga information. First, the system computes the short-term Fourier Transform of the input audio data to form a chromagram representation of the notes being sung. These data are then augmented using a transpositional data augmentation algorithm and split into chunks for use as training inputs for a deep convolutional neural network. The convolutional network's filters are analyzed using a saliency visualization algorithm and modified with a recurrent layer to allow processing of full-length songs. The convolutional system achieves 78.9% validation accuracy for raga prediction on 150 second audio chunks. The joint raga prediction system achieves a new state-of-the-art 98.9% accuracy for raga prediction on full-length songs. Future work can extend the proposed hierarchical system to other tasks with long temporal sequences and extend the data augmentation and visualization algorithms to different applications of audio processing.
EMobility – The silent revolution!?
Authors: Genuit Klaus
Published: November 2019

Electric vehicle become more and more popular worldwide. A strong increase of new registered electric vehicles is expected. The interior and exterior sounds of electric vehicles will become an important aspect for sound engineering considering the psychoacoustics: what sounds do customers prefer, what kind of sounds are important for customers, how to consider the typical acoustical orientation known from combustion noises, should it be possible, that a customer can actively/individually influence the character of the vehicle interior noise, which new sounds arise and must be avoided or emphasized, how to design vehicle exterior noise (e.g., pedestrian protection)? Based on the ISO 12913 Soundscape standard the following topics will be discussed: concepts for the development of sounds, applying jury testing to predict human reactions towards the noise or sound? of new vehicle types, application of psychoacoustics analysis and simulation of warning signals in complex traffic scenarios to check their applicability and their annoyance.
Design and realisation of a directional electric vehicle warning sound system
Authors: Kournoutos Nikolaos , Cheer Jordan
Published: November 2019

The mandatory use of artificial warning sounds by electric vehicles at low speeds has generated concern over the resulting increase in urban noise levels. Systems have been proposed and tested that limit the vehicle’s environmental noise contributions by directing the radiated sound solely towards vulnerable road users. However, such solutions are often too costly to manufacture and maintain as they tend to feature loudspeaker arrays installed on the vehicle and exposed to the environment. This paper proposes a directional warning sound system for electric vehicles which uses an array of structural actuators attached to a panel of the vehicle’s body. The required warning sound is radiated by the vibrating panel and its directivity is controlled by adjusting the amplitude and phase of the signals driving the structural actuators. An analytical model is used to perform a simulation based parametric study of the proposed system, and this is then experimentally validated through measurements of a prototype using a flat metal panel. The effectiveness of the actuator array installed in an actual passenger car is then evaluated in a semi-anechoic environment. Directivity measurements are performed for different target zones using the array in a number of configurations on the vehicle’s body.
EV sound hacking Part 2: Creating an alternative soundtrack
Authors: Reuter Eric L.
Published: November 2019

The Hyundai VESS (detailed in Part 1) is controlled by one of the vehicle’s CAN (controller area network) buses. Messages containing speed, braking, power, and dozens of other parameters are continuously streamed to the bus and can be decoded by a simple microcontroller or single-board computer. The author, an avid railroad enthusiast, sought to replace the standard warning sounds in his Hyundai Kona EV with those of a locomotive. A microcontroller is used to translate CAN messages to DCC, a standard model railroad control protocol, to facilitate the use of an off-the-shelf model railroad sound module. This allows for speed-dependent locomotive sounds to be generated automatically by the electric vehicle.
Mimicking a perfectly matched layer with a porous medium
Authors: Wilson D. K. , Muhlestein Michael B.
Published: November 2019

The possibility of designing a porous medium that mimics a perfectly matched layer (PML) is discussed. In numerical methods, a PML provides essentially zero reflection and rapid attenuation in space. As a starting point to designing such a medium, the relaxation-model equations (10.54) and (10.55) from Ostashev and Wilson [Acoustics in Moving Inhomogeneous Media (CRC Press, Boca Raton, 2016)] are employed. In general, it does not appear to select the parameters to mimic a PML across a broad range of frequencies. For relatively high frequencies, however, it is shown that by setting the ratio of the vorticity and entropy shape factors to √Npr/(γ-1), where Npr is the Prandtl number and γ the ratio of specific heats, the impedance becomes real-valued while the medium remains attenuative. The reflected power is small provided that the porosity is close to one. It is unclear whether such media are physically realizable. Possible approaches to realization are discussed based on varying the pore shape, parallel configurations of pores with differing cross sections, and metamaterials.
Fundamental study on enlarge of a loop-tube-type thermoacoustic system—Measurement of onset temperature and sound field in cross-sectional area change
Authors: Sakamoto Shin-ichi , Inui Kenshiro , Katsuki Hidekazu
Published: November 2019

We have worked to improve the output of the thermoacoustic system to expand the range of system applications. The onset temperature of a loop-tube-type thermoacoustic system was experimentally examined under changes to cross-sectional area in a loop tube. The total length of the system was not varied. This cross-sectional area, i.e., inner diameter was set to values of 24.2, 42.6, and 95.6 mm; onset temperatures at these diameters were found to be 623, 329, and 121 °C, respectively. The sound field in the system was measured to investigate the reason why the onset temperature varies depending on the inner diameter. The sound pressure distribution, particle velocity distribution, sound intensity distribution, and phase difference distribution of sound pressure and particle velocity in the system required for evaluation of the thermoacoustic system were measured. It is confirmed that the sound field is different depending on the inner diameter. It is found that the oscillation frequency is different depending on the inner diameter. The oscillation frequency of system with an inner diameter of 24.2 mm shows two-wavelength resonance. The oscillation frequencies of the other system showed one-wavelength resonance. In addition, dissipation is confirmed to increase as the inner diameter decreased.
The perception of frequency modulated sounds in tone and non-tone language speakers: An electroencephalography study
Authors: Monahan Philip J. , Baskarasingham Mayoori , Pérez Alejandro
Published: November 2019

Speakers of tone languages acquire expertise in discriminating and identifying frequency modulated auditory signals [Chandrasekaran et al., Brain Res., 1128, 148–156 (2007); Kaan et al., Brain Res. 1148, 113–122 (2007)], as their native language utilizes such acoustic properties to cue lexical differences. How this expertise influences auditory neurophysiological responses to non-speech stimuli, however, remains poorly understood. We tested adult tone and non-tone language speakers in their automatic brain processing of non-speech frequency modulated (FM) tone chirps. Participants were presented with a series of FM tones in a many-to-one oddball mismatch negativity [MMN; Näätänen and Winkler, Psychol. Bull. 125, 826–856 (1999)] paradigm that varied in whether the modulation was concave or convex in nature and whether the difference between the tone chirp onset and offset frequencies was relatively large or small. The results revealed that the tone group produced a larger MMN than the non-tone group. Moreover, tone language participants produced significantly larger negative deflections in the event-related potential than the non-tone participants in response to both deviant types across a large post-stimulus time-window. Consequently, tone language speakers’ expertise in processing frequency cues impacts their neurophysiological responses to non-linguistic stimuli that vary along similar acoustic properties to linguistic stimuli.
Computational modelling of category stability in segment pairs participating in perceptual asymmetry
Authors: Calloway Ian
Published: November 2019

Certain unidirectional sound changes show a similarity to the laboratory phenomenon of asymmetrical misperception. The unidirectionality of these processes mirror the dissimilar confusion rates of the two segments. Despite the similarity of these processes, it is not clear what role perceptual asymmetry plays in conditioning these changes. This study employs modeling to simulate change in the characteristics of consonant pairs whose confusion rates pattern with a unidirectional sound change: /k/-to-/t/ (before /i/) and /k/-to-/p/ (before /u/). Ten native AmEng speakers were recorded producing CVC words, where the initial consonant was /p/, /t/, or /k/ and the vowel varied in height and backness. Reduced and unreduced variants were elicited. Acoustically relevant features distinguishing /k/ from /t/ and /k/ from /p/ were identified using random forests. Reduced productions of /k/ and /p/ and /k/ and /t/, respectively, show higher acoustic similarity in vocalic contexts favoring increased confusability. This tendency toward acoustic similarity is predicted to condition category convergence. However, a language learner’s category may be better informed by tokens that show less acoustic similarity to tokens of another category, predicted to condition divergence. Modelled results suggest that the stability of phonetic categories is sensitive to the relative weighting of these two forces.
Deep learning model for automated assessment of lexical stress of non-native english speakers
Authors: Korzekwa Daniel , Kostek Bozena
Published: November 2019

In this paper, we present a novel system to practice lexical stress in L2 English learning with Amazon Alexa home assistant. The language learning for non-native English speakers mostly focuses on practicing correct grammar, extending language vocabulary, and improving pronunciation. The system proposed enables a person to practice lexical stress skills at home by having conversations with Alexa assistant. The system assesses student's abilities to enunciate words with a correct lexical stress and automatically selects the next words to practice. After a series of exercises, the system informs the student on the improvement. The main scientific contribution of the work presented is a deep learning model for automated assessment of lexical stress of non-native English speakers. The model is based on a transfer learning technique. First, we train the model to predict the location of a lexical stress on a syllable level using a large corpus of native English speech. Then, we tune the model with a limited amount of a non-native speech. A corpus of non-native English speech obtained from Polish speakers is incorporated into the training and testing of the model. It is shown that the system enables to create a vocabulary for a particular speaker interactively.
Improved vowel labeling for prenasal merger using customized forced alignment
Authors: Shi Yuanming , Renwick Margaret E. , Maier Frederick
Published: November 2019

Forced alignment is a popular technique for gaining phone-level audio transcriptions, but the pronunciation dictionaries used by it are typically based on standard varieties of US English, leading to errorful outputs for non-standard varieties. We employ a customized pronunciation dictionary with the Montreal Forced Aligner to increase labeling accuracy of the prenasal merger (a.k.a. pin-pen merger) in Southern US English. We allow the aligner to choose between IH (/ɪ/) and EH (/ɛ/) in words where the merger is expected, rather than enforcing a standard, unmerged pronunciation. We examine the tokens reclassified from EH to IH when using the new dictionary, and we use formant values to study the acoustic separation (measured by Pillai scores and Euclidean distances between centroids) between vowel formant clusters. When applied to the Digital Archive of Southern Speech (DASS), we find that the modification increases the separation between the prenasal allophones of IH and EH, and also that the proportion of prenasal EH tokens reclassified to IH is correlated with the original degree of separation between prenasal IH and EH for each DASS speaker. K-means clustering is also used to show the modification yields more accurate phonetic transcriptions, measured by increased precision and recall.
Robust neural network approach to source-range estimation in a simulated Arctic environment
Authors: Chen Rui , Schmidt Henrik
Published: November 2019

Underwater source-range estimation is conventionally performed via matched field processing (MFP), which relies on precise modelling of the propagation environment to produce accurate estimates. Here, we design a convolutional neural network (CNN) for long range (up to 50 km) source-range estimation in a complex Arctic propagation environment and compare its performance and robustness to MFP. The model parameters are optimized through cross-validation with regularization and other precautions implemented to deter overfitting. It is trained and tested with simulated data recorded on a vertical line array and its robustness to environmental mismatch is examined by introducing deviations to the original sound speed profile (SSP) when generating the test data. Results show that the CNN is more robust to environmental changes than MFP but at the expense of worse performance when the environmental parameters are accurately modelled. Insights into how the CNN performs range estimation is discussed as well. [Work supported by Office of Naval Research.]
Application of autoencoder to traffic noise analysis
Authors: Czyzewski Andrzej , Kurowski Adam , Zaporowski Szymon
Published: November 2019

The aim of an autoencoder neural network is to transform the input data into a lower-dimensional code and then to reconstruct the output from this code representation. Applications of autoencoders to classifying sound events in the road traffic have not been found in the literature. The presented research aims to determine whether such an unsupervised learning method may be used for deploying classification algorithms applied to the automatic annotation of road traffic-related events based on noise analysis. Two-dimensional representation of traffic sounds based on Mel Frequency Cepstral Coefficients (MFCC) was fed the autoencoder neural network, and after that classified with k-nearest neighbors algorithm, Support Vector Machines, and random forests. Obtained results show that sound recordings can help determine the number of vehicles passing on the road. However, instead of being treated as independent, this method output should be combined with another source of data, e.g., video processing results or microwave radar data readings. Comparative results of vehicle counting obtained with the use of autoencoder and different classifiers are shown in the paper. [The Polish National Centre finances the project for Research and Development (NCBR) from the European Regional Development Fund No. POIR.04.01.04-00-0089/16 entitled: "INZNAK: Intelligent Road Signs with V2X Interface for Adaptive Traffic Controlling."]
Alignment of canonical and realized acoustic cue labels for modification patterns in speech analysis
Authors: Soh Christine , Talkar Tanya , Choi Jeung-Yoon , Shattuck-Hufnagel Stefanie
Published: November 2019

Acoustic cues are properties of the speech signal that provide information about the distinctive features of the speaker’s intended words and phonemes. Analysis of acoustic cues can indicate reductions and modifications in speech, in which landmarks and other feature cues are deleted, inserted, or substituted for others, and can be informative in distinguishing underlying causes of speech impairments. To extract this information about modifications, we need to determine which predicted canonical labels the realized labels correspond to. We propose an algorithm that uses a time-based alignment method for the landmarks as well as a modified labeling scheme to more accurately find correspondences between realized landmarks and distinctive features to the canonical labels. The results show improved alignment not only for the realized landmark labels but also for the labels of other feature cues, enabling accurate and holistic analysis of modifications in speech, at the more detailed level of cues to distinctive features rather than the phoneme or phone level. Using this algorithm, we analyze a database of CN-REP (Children’s Nonword Repetition Task) recordings from children diagnosed with speech impairments and find several potential modification markers that may distinguish among different diagnoses.
A computationally efficient blind source separation for hearing aid applications and its real-time implementation on smartphone
Authors: Shreedhar Bhat Gautam , Shankar Nikhil , Panahi Issa
Published: November 2019

Conventional Blind Source Separation (BSS) techniques are computationally complex. This is due to the calculation of the demixing matrix for the entire signal or due to the frequent update of the demixing matrix at every time frame index, making them impractical to use in many real-time applications. In this paper, a robust, neural network based two-microphone sound source localization method is used as a criterion to enhance the efficiency of the Independent Vector Analysis (IVA), a BSS method. IVA is used to separate speech and noise sources which are convolutedly mixed. The practical usability of the proposed method is proved by implementing it on a smartphone to separate speech and noise in real-world scenarios for hearing-aid applications. The experimental results with objective and subjective tests reveal the usefulness of the developed method for real-world applications.
Fast parameter estimation of distributed sources in shallow water with a new UCA
Authors: Gu Qingyue , Wang Huigang , Sun Weitao , Xu Yifeng
Published: November 2019

Due to large number of reflections from surface and bottom in the shallow ocean environment, the radiated noise from the same source can be the distributed source model, instead of the traditional point model. The distributed source will appear two-dimension spatial distribution characteristic in azimuth and elevation. Four important parameters of two-dimensional spatial distribution source are the central angle and extended angle in both dimension, which is very important to identify the type of source. In this paper, a novel UCA array layout is proposed for an two-dimensional incoherent distributed(TDID) source, and a new ESPRIT algorithm for two-dimension distribution source is derived to estimate these parameters analytically with symmetrical structure of UCA. A large mount simulation of underwater environment shows this method can achieve high computational accuracy with low computational burden, and the results also verify the robustness to the different distribution forms and good performance with larger extended angle size.
Speaker identification using convolutional-long short-term memory neural networks
Authors: Tokgoz Serkan , Panahi Issa M.
Published: November 2019

Speaker identification (SI) techniques has been used in numerous commercial products over the last decades. In SI, the main purpose is to match a voice sample from an unknown speaker to one of the labeled speaker models. To be able accomplish this task, there are two operational phases, training (can be also termed as enrollment) and testing. In both phases, feature extraction and feature matching are the two key steps. In this work, we have extracted features with Mel-Frequency Cepstrum Coefficients (MFCC) because MFCC has accurate representation of the vocal tract, and a Spectral-Flux based Voice Activity Detector (VAD) is implemented to extract features from the speech segments. In feature matching task, we build a Convolutional Long short-term memory (LSTM) Neural Network for the speaker models. We examine main performance of the system in terms of identification rate and compare the proposed method with other SI methods under several noisy conditions at different signal to noise ratio (SNR) levels.
Ocean source localization with multi-frequency deep learning in uncertain environments
Authors: Niu Haiqiang , Gong ZaiXiao , Reeves Ozanich Emma , Gerstoft Peter , Wang Haibin , Li Zhenglin
Published: November 2019

A deep learning approach is proposed to locate broadband acoustic sources in ocean waveguides with uncertain bottom parameters. The residual neural networks, trained on a huge number of sound field replicas generated by an acoustic propagation model, are used to handle the environmental uncertainty in source localization. A two-step training strategy is presented to improve the training of the deep models. First, the range is discretized in a coarse (5 km) grid. Subsequently, the source range within the selected interval and source depth are discretized on a finer (0.1 km and 2 m) grid. The deep learning methods are demonstrated for simulated magnitude-only multi-frequency data in uncertain environments. Experimental results from the China Yellow Sea show that the approach is comparable with SAGA in performance while much faster in computation speed.
Study of transmission loss model using the phase of the source
Authors: Lee Dawoon , Chung Wookeen
Published: November 2019

MOE(Measure Of Effectiveness analysis) is needed to predict the sonar detection performance and maneuvering tactics in the ocean. To effect analysis, the detection probability is calculated by sonar characteristics and sound velocity distribution. The detection probability considering the sonar characteristics and the sound velocity distribution can be derived by using the FOM (figure of merit) and the transmission loss. Generally, the transmission loss used to calculate the detection probability is obtained by calculating the path of ray and the attention for the frequency of the source, sound velocity distribution. Although these models have advantages of being efficient and fast, it cannot consider the effect of constructive and destructive interference for the source. In this study, the propagation phenomena for source were calculated by solving the two-dimensional acoustic wave equation to obtain transmission loss. For the computational efficiency, the proposed method uses the geometric spreading correction for the amplitude correction of the three dimensions. The calculated transmission loss is compared with transmission loss using BELLHOP, which is a sound transmission model. As a result, it is confirmed that the proposed method can consider the phase of the source.
Using faster region-based convolutional neural network for automatic detection of baleen whale social calls
Authors: Rasmussen Jeppe H. , Širović Ana
Published: November 2019

Using blue and fin whale calls for estimating population density of the two species from passive acoustic data is an active research topic. However, manually analyzing long-term data is extremely time consuming yet a necessary first step for such work. Blue and fin whale social calls are highly variable and quite similarity, which has resulted in challenges in developing automatic detectors for these calls. The applicability of faster region-based convolutional neural network (Faster R-CNN) method for speeding up this detection and classification task was explored. A large dataset of blue whale D calls and fin whale 40 Hz calls from southern California was used for training the network: 1378 spectrograms were created from 10-s sound clips each containing a call. The resulting images were contrast-enhanced and manually labeled with region of interest (ROI) before being applied as training images. The probability score for each ROI was modified to favor detections within previously measured frequency range of the calls. Testing shows this Faster R-CNN to have a very low miss- and false positive rate for both call types and is thus a highly promising tool for detecting and classifying these baleen whale social calls.