bluuebunny/crossref_metadata_2025 huggingface dataset:
Occlusion effects of hearing protection device on ear canal transfer function and cochlear injury from impulse noise exposure
Authors: Zagadou Brissi
Published: April 2018

The objective of this paper is to present a new finding on the effects of hearing protection devices (HPDs) on the ear canal (EC) transfer function (TFEC2ED). Specifically, we show that when an earmuff is worn, the TFEC2ED is not the same as that for the bare head, leading to a significant difference in cochlear injury prediction. We used shock tube blast with an acoustical test fixture (ATF) fitted with the recovered earmuffs from the historic blast overpressure project (BOP) to measure the TFEC2ED. A microphone was placed at the EC entrance under the earmuff (undermuff) to measure the pressure at the EC entrance, while the built-in ATF’s microphone was used to measure the eardrum (ED) pressure. The TFEC2ED is defined as the ratio of the pressure at the ED to the pressure at the EC entrance. We compared the cochlear integrated energy (ICE) values, when the TFEC2ED obtained from bare head (Case-1) and that measured using the ATF fitted with an earmuff (Case-2), respectively are used in the ICE-model, to estimate the effects of the difference between the TFEC2EDs on injury. The human BOP undermuff pressure data were used as model input. For Case-1, the input was at the EC. The ED pressure, reconstructed using the undermuff pressure and the TFEC2ED measured using the ATF, was used for Case-2. We found significant differences in the ICE-values as the result of the difference between the TFEC2EDs. We conclude that hearing protector models must correctly model the occlusion effects of HPDs by incorporating the change in the TFEC2ED.
Evaluation of the effect of aircraft size on indoor annoyance caused by sonic booms and rattle noise
Authors: Loubeau Alexandra
Published: April 2018

NASA plans to use a low-boom flight demonstration supersonic aircraft for community annoyance studies of sonic booms. As previously reported, laboratory studies were conducted in NASA's Interior Effects Room (IER) to help determine the relevance of using a sub-scale aircraft. Indoor annoyance was evaluated for a variety of sonic booms predicted for several different sizes of vehicles, whose levels were adjusted to the same range of loudness levels. Although no significant effect of aircraft size was found for equivalent loudness levels, a new laboratory study was conducted in the IER to extend this investigation to include the effect of secondary rattle noises. The rattle noise playback was determined from window rattle measurements performed in the IER that resulted in a database of rattle noises matched with sonic booms predicted for different aircraft, at different boom levels. The main objective was to test whether aircraft size is still not significant when realistic window rattles are included in the simulated indoor sound field. The data have also been used to evaluate a variety of noise metrics. Results using metrics computed on exterior boom levels show greater variation in subjective annoyance than in previous studies conducted without rattle.
Perceptual characterization of Mach cut-off sonic booms
Authors: Ortega Nicholas D. , Vigeant Michelle C. , Sparrow Victor
Published: April 2018

Mach cut-off refers to a set of flight conditions wherein the direct sound of a sonic boom does not reach the ground. This phenomenon occurs in the idealized lower atmosphere, which naturally refracts sound upwards. The current work, subdivided into two perceptual studies, investigates the perception of the evanescent sound field below the direct path of the boom. Both studies used Mach-cutoff ground signature recordings from NASA’s “Farfield Investigation of No-boom Thresholds” (FaINT). In the first study, subjects provided descriptors for Mach-cutoff ground signatures in order to develop a vocabulary useful for describing these sounds. In the second study, subjects rated the stimuli on multiple scales, comprised of descriptors from the first task and annoyance, to identify which perceptual attributes factor into annoyance. These findings were used to obtain an overall perceptual characterization of the Mach-cutoff ground signatures, and these studies may inform metrics for use in predicting reactions to Mach-cutoff flyovers. [Work supported by the FAA. The opinions, findings, conclusions and recommendations expressed in this material are those of the authors and do not necessarily reflect the views of ASCENT FAA Center of Excellence sponsor organizations.]
Determining a sufficient range of sound levels for single event analysis in quiet sonic boom community surveys
Authors: Lee Jasme , Rathsam Jonathan , Wilson Alyson G.
Published: April 2018

In order to enable quiet supersonic flight overland, NASA is establishing a quiet sonic boom database to present to national and international noise regulators. The database will contain dose-response models to provide evidence of a cause-and-effect relationship between boom sound levels and annoyance. In order to build the dose-response models, NASA is designing surveys for data collection. An important aspect in planning for future community surveys is the range of sound levels necessary for single event analysis. The dose-response models must be characterized at sound levels both above and below a notional threshold. The trade-offs for testing at high sound levels are the increased risk of negative community reaction and misconception of quiet sonic booms. On the other hand, testing only at low sound levels will result in unstable estimates in the dose-response models. To find a compromise between the two, analysis of a pilot community survey from 2011 explores how excluding high level booms will affect the dose-response analysis since both low and high level booms were present in the study. The data analysis will also provide insight into the appropriate range of sound levels for single event analysis in future community surveys.
Binaural perceptual weighting of reverberation level in normal hearing listeners
Authors: Ellis Gregory M. , Zahorik Pavel
Published: April 2018

In general, perceived reverberation strength is related to physical reverberation level present at the two ears; however, this relation depends on listening condition. Pervious work using virtual auditory space techniques has demonstrated that when physical reverberation is reduced in one ear while leaving the other ear unchanged, listeners do not report a change in reverberation strength. Reducing physical reverberation equally in both ears under binaural listening or in the signal ear under monaural listening elicits a decrease in perceived reverberation strength. To better understand the relation between physical and perceived reverberation in different listening conditions, a perceptual weighting experiment was performed. Listeners reported perceived reverberation strength in a test stimulus using a magnitude estimate paradigm relative to a standard while listening over headphones. Test stimuli were generated by jittering reverberation level independently in the two channels of a binaural room impulse response (BRIR) before convolution with a speech token. BRIRs with reverberation levels of -12, -6, and 0 dB at each ear relative to the standard were tested. Results of a perceptual weighting analysis suggest that listeners use different weighting strategies in different listening conditions, and are generally consistent with a “better ear” hypothesis for reverberation listening.
Binaural speech unmasking and interference in adult bilateral cochlear-implant users
Authors: Goupell Matthew , Stakhovskaya Olga , Bernstein Joshua G.
Published: April 2018

Bilateral cochlear implants (BICIs) provide improved speech perception in noise, primarily derived from the head-shadow benefit. BICI listeners can also demonstrate binaural speech unmasking or squelch, particularly if the interferer is a single other speaker. However, under the same conditions, some BICI listeners can experience interference. This study tested 21 adult BICI listeners on speech understanding using a range of masker conditions (1, 2, 3, or 4 same-sex talkers, speech-modulated noise, or speech-shaped stationary noise). For listeners that showed binaural unmasking, the amount of unmasking was highest for the one-talker condition and least for the stationary noise, decreasing as the number of interfering talkers increased and the envelope modulation depth decreased. For listeners that showed interference, there were mainly two types of interference patterns: (1) where the amount of interference varied idiosyncratically across masker type, and (2) any stimulation in one ear essentially produced chance performance in the other ear for all masker conditions. The latter interference pattern occurred for the listeners with the largest asymmetries in speech understanding. This suggests, in these extreme cases, a possible central cortical processing bias for one ear and a more equal speech understanding between the ears may diminish interference.
The optimal noise-rejection threshold for normal and impaired hearing
Authors: Vasko Jordan , Healy Eric , Wang DeLiang
Published: April 2018

Binary masking represents a powerful tool for increasing speech intelligibility in noise. An essential aspect involves the local criterion (LC), which defines the signal-to-noise ratio below which time-frequency units are discarded. But binary masking is a victim of its own success in one regard—it produces ceiling sentence intelligibility across a broad range of LC values, making the exact optimal LC value difficult to determine. Further, the optimal value for hearing-impaired (HI) listeners is largely unknown. In the current study, the optimal LC was determined in normal-hearing (NH) and HI listeners using speech materials less likely to produce ceiling effects. The CID W22 words were mixed with noise consisting of recordings from a busy hospital cafeteria, then subjected to ideal binary masking. LC values ranged from -20 to + 5 dB relative to the overall SNR of -8 dB. NH subjects were tested at 65 dBA and HI subjects were tested at 65 dBA plus NAL-RP hearing-aid gains. Preliminary results suggest that the optimal LC is similar for NH and HI listeners. Additional conditions involving different speech materials and noise types suggest that the optimal LC can vary as a function of speech and/or noise type. [Work supported by NIH.]
Alternative methods for faster and more accurate estimation of hearing threshold
Authors: Biswas Amitava , Goshorn Jennifer
Published: April 2018

Determination of audiologic threshold is often necessary for variety of diagnoses and treatments. The usual method involves “10 dB down for response and 5 dB up for miss”. This common procedure lacks sufficient accuracy to identify smaller changes due to temporary threshold shift and for early diagnosis of degradation of hearing threshold due to ototoxicity or chemotherapy etc. This presentation will explore several alternative methods for faster and more accurate estimation of the threshold.
Exploring some questions on occlusion or the Bing effect during speech production with a microphone coupled to the speaker's external ear canal
Authors: Biswas Amitava
Published: April 2018

Sometimes some speakers prefer to use their palm or other objects to cover or occlude at least one ear during public address. This practice may be helpful to enhance their self monitoring of the sound production using the occlusion effect. The basic occlusion effect in the human auditory system has been explored and reported in the literature by several individuals. According to those reports, the speaker can hear his or her own voice significantly louder when the ear canal is blocked at the outer end. Many clinicians routinely utilize detectability of vibrations from a tuning fork when placed on the mastoid process and the ear canal is occluded. This is popularly known as the Bing effect. These empirical data suggest existence of an efficient acoustic connectivity from the vocal tract to the ear canal for healthy individuals. Therefore, this study will explore quantified effects of the classic Bing effect for several speakers across the audio spectrum with a microphone coupled to the speaker's external ear canal.
Online training for perceptual rehabilitation in cochlear implant users
Authors: Almaraz Damian , Een Elizabeth , Grafelman Elaine , Pacholec Justin , Quintanilla Yadi , Rodriguez Paula , Loebach Jeremy
Published: April 2018

Although cochlear implants have been demonstrated to be effective surgical treatments for deafness, new implant users must undergo an intense period of perceptual learning and adaptation to learn to hear with their prosthesis. Few adult cochlear implant users receive any formal training following implantation, and as a result, the perceptual skills that they develop are highly variable. This study assessed the efficacy of the high variability online training program for adult cochlear implant users dubbed the HiVOlT-CI in 31 normal hearing individuals listening to cochlear implant simulations and three adult cochlear implant users. The goal of our interactive, adaptive, high-variability training program is to provide empirically-based, adaptive and interactive perceptual training to new cochlear implant users to help them adjust to their prosthesis. We also aim to develop a common set of robust and adaptive cognitive auditory skills in cochlear implant users that will help them to hear in real world situations. We also hope to establish baseline levels of performance for experienced cochlear implant users to evaluate successful implant use by new users.
The development of sentence recognition in noise by cochlear implant users
Authors: James Chris J. , Karoui Chadlia , Marx Mathieu , Laborde Marie-Laurence , Tartayre Marjorie , Deguine Olivier , Fraysse Bernard
Published: April 2018

We previously established that under certain physiological conditions the sentence recognition scores in noise of cochlear implant (CI) users one month after activation are similar to those for normal listeners listening to 10–12 channel acoustic (vocoder) simulations (James et al., CIAP, Lake Tahoe, July 2017). We further analysed sentence recognition scores in noise over time in order to determine whether short term, 1-month scores predict long-term outcomes for sentence recognition in noise. Long-term sentence-in-noise data for fixed-SNR-level testing (10, 5, 2.5, 0 dB) were reduced to SNR50 values using a validated analytic sigmoid fit. Data were analysed for 106 adult CI users. 12-month SNR50s were highly correlated with 1-month sentence recognition scores in quiet and 10 dB SNR (r2 = 0.29 and r2 = 0.22, both p<0.001): CI users with initially poorer scores could catch up to some extent with early, better performers. However, early high performance (>80% correct score in quiet) was generally associated with 12-months SNR50s ≤2 dB. Early sentence recognition scores for CI users may predict their long-term recognition of speech in noise. The number of effective channels available to CI users may evolve to greater than 10-12 in the long term. Disclosure: C.J. is also an employee of Cochlear France. C.K. received partial doctoral funding from Cochlear France as part of the program “Conventions industrielles de formation par la recherche” (CIFRE).
The segregation of sequentially interleaved sentences by normal-hearing and cochlear-implant listeners using monaural level cues and the effect of dynamic-range compression
Authors: Archer-Boyd Alan , Carlyon Robert P.
Published: April 2018

Normal-hearing (NH) listeners can use level differences to segregate pairs of vocoded sentences that are presented to interleaved frequency channels; performance is better at both positive and negative target-to-masker ratios (TMRs) compared to a TMR of 0 dB (Ihlefeld and Shinn-Cunningham, 2008). The present study investigated whether unilateral cochlear-implant (CI) users could also use level differences alone to segregate two sentences. Due to the greatly reduced frequency resolution and increased TMR thresholds of CI users, the sentences were interleaved in time and presented sequentially. The Boston University (Kidd et al., 2008) corpus was used to construct interleaved, five-word sentences, consisting of name, verb, number, adjective, and object. The presentation order of the words alternated randomly for each word type. Each sentence began with the same two words and listeners were required to select the last three words in the target sentence. The level differences ranged from -12 to 12 dB. A monaural vocoder study showed that the task was possible for both with younger and older NH listeners, with performance increasing as level difference increased. The results of the CI user study will be presented, and the effects of dynamic-range compression on their performance discussed.
Effects of broad binaural fusion and hearing loss ondichotic concurrent vowel identification
Authors: Reiss Lina , Simmons Sara , Anderson Larissa , Molis Michelle R.
Published: April 2018

Many hearing-impaired (HI) individuals have abnormally broad binaural pitch fusion, such that tones differing in pitch by up to 3-4 octaves are fused between ears (Reiss et al., 2017). Broad binaural fusion leads to averaging of the monaural pitches (Oh and Reiss, 2017), and may similarly lead to fusion and averaging of speech streams across ears. In this study, we examined the relationship between binaural fusion range and dichotic vowel identification in normal-hearing (NH) and HI listeners. Synthetic vowels /i/,/u/,/a/, and /ae/ were generated with three fundamental frequencies (F0) of 106.9, 151.2, and 201.8 Hz, and presented dichotically through headphones. For HI listeners, stimuli were shaped according to NAL-NL2 prescriptive targets. Listeners identified 1 vowel or 2 different vowels on each trial. Preliminary results indicate that NH listeners improved their percentage of both vowels identified as the ΔF0 increased, but HI listeners did not. HI listeners were more likely to fuse two vowels together even with large ΔF0. NH listeners with broad fusion also had poorer overall performance than those with sharp fusion. The findings suggest that broad binaural fusion can impede separation of voices and speech streams in both NH and HI listeners. [Work supported by NIH-NIDCD grant R01 DC013307.]
Modulation interference in cochlear implants
Authors: Chatterjee Monita , Kulkarni Aditya M.
Published: April 2018

Modulation interference, in which the temporal envelope of a signal is rendered less salient due to the presence of a competing temporal envelope of a masker, has been modeled as the result of modulation-tuned filters operating in the midbrain. In listeners with cochlear implants (CIs) who rely more heavily on the speech temporal envelope, such interference may be particularly problematic for speech perception in competing, fluctuating backgrounds. Here, we present evidence for modulation interference in CI users from our laboratory, using both non-speech and speech targets. Maskers were presented either in the same ear or in the contralateral ear relative to the target, and either concurrently with the target or preceding the target (forward masking). Tasks include the detection of the temporal envelope on the target as well as the discrimination of temporal envelopes between targets. Results suggest that modulation interference plays an important role in CI users, reducing the salience of the speech temporal envelope under a variety of masking conditions.
Improving temporal modulation sensitivity in cochlear implant users
Authors: Hossain Shaikat , Bissmeyer Susan R. , Goldsworthy Raymond L.
Published: April 2018

The present study investigated the limitations of temporal modulation sensitivity in cochlear implant (CI) users. Adult users of Cochlear Nucleus devices were tested by presenting sounds through their clinical processors using an auxiliary cable and using direct electrical stimulation provided through a custom research processor. Threshold and comfort current levels were determined by adjusting the phase duration of 10 Hz amplitude modulated pulse trains of 400 ms duration at a given stimulation rate. After mapping the participants, modulation detection thresholds were measured by means of a 3 alternative forced choice procedure which adaptively adjusted the modulation depth to determine the threshold in dB relative to full modulation. Participants completed the task using a graphical user interface to indicate which interval was different. Modulation frequencies from 10 to 800 Hz were tested at various stimulation rates on basal and apical electrodes in both psuedomonopolar and bipolar modes of stimulation. Preliminary findings indicate that modulation sensitivity drops off with increases in modulation frequency. Performance was significantly better with direct stimulation as compared to through the clinical processor and better at higher modulation frequencies than previous findings from CI users. Outcomes will have implications for the development of next generation CI devices.
Band importance functions of listeners with sensorineural hearing impairment
Authors: Yoho Sarah E. , Bosen Adam K.
Published: April 2018

The Speech Intelligibility Index (SII) includes a series of band importance functions for calculating the estimated intelligibility of speech under various conditions. Band importance functions describe the relative contribution of discrete spectral regions of speech to overall speech intelligibility. They are generally measured by filtering the spectrum into a series of frequency bands and determining intelligibility when each band is present, relative to intelligibility when other bands are present. The functions found in the SII are commonly used to provide estimations of intelligibility for listeners with hearing loss, even though they were developed with the use of normal-hearing listeners. In the current study, band importance functions were derived for individual listeners with sensorineural hearing impairment and a group of normal-hearing control participants. Functions were measured by filtering sentences to contain only random subsets of bands on each trial, and regressing speech recognition against the presence or absence of each band across trials. Preliminary results indicate that the shape of hearing loss plays an important role in the shape of band importance functions, with some listeners who have high-frequency sloping losses displaying decreased and even negative importance of high-frequency speech bands.
Natural signal statistics and the timecourse of spectral context effects in consonant categorization
Authors: Stilp Christian , Assgari Ashley
Published: April 2018

Speech perception is heavily influenced by surrounding sounds. When spectral properties differ between earlier (context) and later (target) sounds, this can produce spectral contrast effects (SCEs) that bias categorization of later sounds. For example, when context sounds have a low-F3 bias, listeners report more high-F3 responses to the target consonant (/d/); conversely, a high-F3 bias in context sounds produces more low-F3 responses (/g/). SCEs have been demonstrated using a variety of approaches, but most often, the context was a single sentence filtered two ways (e.g., low-F3 bias, high-F3 bias) to introduce long-term spectral properties that biased speech categorization. Here, consonant categorization (/d/-/g/) was examined following context sentences that naturally possessed desired long-term spectral properties without any filtering. Filtered sentences with equivalent spectral peaks were included as controls. For filtered context sentences, as average spectral peak magnitudes (i.e., filter gain) increased, SCE magnitudes (i.e., category boundary shifts) increased linearly. However, unfiltered context sentences showed no relationship between long-term average spectral energy and subsequent SCEs. Instead, spectral energy in the last 500 ms of unfiltered sentences predicted SCE magnitudes. Results highlight important considerations about how long-term versus short-term spectral energy in preceding sounds affects categorization of later sounds.
Non-reciprocal bilinear structures
Authors: Norris Andrew , Lu Zhaocheng , Wallen Samuel P. , Goldsberry Benjamin M. , Haberman Michael R. , Wiest Tyler J. , Seepersad Carolyn C.
Published: April 2018

Structures that display non-reciprocal behavior resulting from nonlinear effects are presented. The nonlinear mechanism is bilinear stiffness, also known as bimodular elastic response. Specific realizations considered are in the form of bilinear springs in series making a finite degree of freedom uni-dimensional nonlinear structure. Recall that a system is reciprocal if FB uB,A= FA uA,B where is uB,A(respectively uA,B) is the displacement at point B (A) resulting from a force FA applied at A (FB applied at B). We say the system is fully non-reciprocal if the reciprocity relationship is violated for any sign of the applied forcing, positive or negative, compressive or tensile. Perhaps the simplest system displaying full non-reciprocity is a two degree-of-freedom spring-mass-spring-mass-spring structure, fixed at both ends. We first describe the static and low frequency behavior, illustrating full non-reciprocity. Similar systems with many bilinear springs and masses display nonlinear traveling wave effects, including pulse spreading and shock formation depending on whether the leading edge of the incident pulse is compressive or tensile, respectively. The talk will also discuss fabrication and optimization of the bilinear springs using additive manufacturing. Work supported by NSF-EFRI.
Interference of adaptation to foreign-accented speech from consecutive but not interleaved training of two accents
Authors: Woodruff Carr Kali , Wright Beverly
Published: April 2018

For perceptual learning on fine-grained discrimination tasks, improvement can be enhanced or disrupted when two tasks are trained, depending on how the training on those tasks is distributed. To investigate if this phenomenon extends to speech learning, we trained native-English speakers to transcribe both Mandarin- and Turkish-accented English sentences using one of three different configurations of the same training stimuli. After training, all trained groups performed better than untrained controls, and similarly to each other, on a novel talker of a trained accent (Mandarin). However, for a novel accent (Slovakian), performance was better than untrained controls when training alternated between the two accents, but not when the two accents were trained consecutively. Performance for the novel accent decreased as the number of contiguous sentences per accent during training increased. One interpretation of these results is that accent information is integrated during a restricted time window. If two accents are encountered within this window, information from both accents is integrated, yielding accent-general learning. If two accents are encountered in separate consecutive windows, accent-specific learning occurs for each accent and accent-general learning is prevented. These results mirror patterns for fine-grained discrimination learning, and illuminate the processes underlying the success of high-variability training.
Perception of non-native clear speech: The gap between speakers’ effort and actual increase in intelligibility
Authors: Kato Misaki , Baese-Berk Melissa M.
Published: April 2018

Non-native speakers are less effective at increasing intelligibility in clear speech than native speakers. However, examining only intelligibility may not fully address how non-native clear speech is perceived by listeners. We examined native English listeners’ perception of non-native English speakers’ clear speech in terms of intelligibility, comprehensibility, and speakers’ effort. Non-native English speakers produced English sentences in conversational and clear speaking styles. These productions were evaluated by 117 native English listeners. In an intelligibility task, listeners typed what they heard. In two rating tasks, listeners heard the same sentence produced by the same speaker in both a conversational and a clear speaking style. One group of listeners were asked to choose the sentence that was easier to understand (comprehensibility). Another group of listeners were asked to choose the sentence that the speaker was trying to say more clearly (speakers’ effort). The results showed that conversational-clear modifications did not result in increased intelligibility or comprehensibility. However, the rating task showed that listeners were sensitive to the speakers’ effort in the clear speaking style. These results suggest that non-native speakers’ effort to speak clearly may not necessarily result in intelligibility gains, but listeners may still be sensitive to this effort.
Speaker variability in spoken word recognition: Evidence from non-native listeners
Authors: Zhang Yu
Published: April 2018

Non-native listeners face many challenges during spoken word recognition. On the lexical level, the vocabulary size of non-native listeners is usually reduced relative to native listeners. Lexical competition can arise from the lexicon in both the native and the non-native language during speech perception. How do non-native listeners resolve speaker variability, a challenge that is presumably non-lexical in spoken word recognition? This study aims to explore non-native spoken word recognition in the face of speaker variability. Mandarin speakers using English as a second or foreign language are recruited to listen to pairs of English words (e.g., cat—dog). Lexical decision time and accuracy on the second item in a pair (e.g., dog) are measured. The pairs are either spoken by the same or different native speakers of American English to examine the effect of speaker change on reaction time and accuracy in a short-term priming paradigm. Non-native listeners’ pattern of response is explored as a function of the amount of priming received against speaker variability.
Unsupervised seabed characterization with Bayesian modeling of SAS imagery
Authors: Brandes T. S. , Ballard Brett
Published: April 2018

Seabed characterization has utility for numerous applications that seek to explore and interact with the seafloor, ranging from costal habitat monitoring and sub-bottom profiling to man-made object detection. In the work presented here, we characterize the seabed based on the texture patterns within SAS images constructed from high-frequency side-scan sonar. Features are measured from the SAS images (e.g., lacunarity, an established texture feature coding method, and a rotationally invariant histogram of oriented gradients). Based on these SAS image features, we perform unsupervised clustering with a hierarchical Bayesian model, which creates categories of seabed textures. Our clustering model is a new variant of the hierarchical Dirichlet process that is both adaptive to changes in the seabed and processes batches of SAS imagery in an online fashion. This allows the model to learn new seabed types as they are encountered and provides usable clustering results after each batch is processed, rather than having to wait for all the data to be collected before being processed. The model’s performance of seabed characterization by SAS image texture is demonstrated in the overall range and internal consistency of textures specific to each learned cluster. [Work supported by the Office of Naval Research.]
Use of adaptive thresholding to improve accuracy of small-aperture acoustic localization in shallow water
Authors: Murphy Paul , Polagye Brian
Published: April 2018

Passive acoustic localizing arrays are utilized in shallow water for applications ranging from environmental monitoring to marine security. Although large-aperture hydrophone arrays can provide accurate estimates of source position, they can be costly to deploy and maintain. Small-aperture arrays can be used to estimate the incident azimuth and elevation angles of acoustic sources, but their effectiveness may be hindered by multipath interference when deployed in reverberant environments. Here, a small-aperture hydrophone array is used to estimate the direction of acoustic sources with center frequencies of 69 and 120 kHz and durations less than 5 ms which were deployed at ranges of up to 75 m in a bay with a water depth on the order of 10 m. Time-differences of arrival are determined using an adaptive thresholding algorithm, which is shown to have superior performance to a cross-correlation method due to the prevalence of multipath interference in the signals. Errors in source bearing estimates and in the underlying time-differences of arrival calculations are evaluated by comparing to ground-truth values generated from the GPS position of the source.
Time reversal based frequency-domain equalization for underwater acoustic single-carrier transmissions
Authors: Tu Xingbin , Song Aijun , Xu Xiaomei
Published: April 2018

Time reversal processing generates a time-focused signal, after combining dispersive reception from a hydrophone array in the underwater acoustic channel. Because of this property, the time reversal method has been studied as a preprocessing scheme to support decision feedback equalization for underwater acoustic communications. Orthogonal frequency division multiplexing has also been combined with time reversal processing to improve spectral efficiency, as the equivalent compact channel impulse response allows short prefixes in the acoustic channel. Here, we show that time reversal processing can be used to facilitate prefix-free frequency domain equalization. Traditional frequency domain equalization schemes rely on the use of periodic prefixes, which bring a loss of spectral efficiency in the already-bandlimited underwater channel. The core idea is to use time reversal pre-processing to generate a delta-like equivalent impulse response to facilitate robust reconstruction of prefixes at the receiver for prefix-free single-carrier transmissions. The resultant receiver not only improves the spectral efficiency, but also supports communications over channels of variable fluctuation rates. The effectiveness of the receiver has been tested using the field measurements from the Gulf of Mexico in August 2016.
M-ary cyclic shift keying spread spectrum underwater acoustic communication using frequency domain energy detector
Authors: wei zhuoqun , Han Xiao , Yin Jingwei
Published: April 2018

M-ary cyclic shift keying (M-CSK) method can greatly improve the communication rate of conventional direct sequence spread spectrum, but the robustness is seriously influenced by the fluctuation of sea surface and Doppler effect. As to this problem, M-ary cyclic shift keying spread spectrum (M-CSKSS) underwater acoustic communication (UAC) based on frequency domain energy detector (FDED) algorithm is proposed in this paper. Time reversal (TR) is firstly adopted to suppress multipath spread using channel impulse response (CIR) estimated from the former symbol. And then the baseband signal after TR processing goes through a FDED, decisions are made according to the output energy later. Simulation results show the robustness of this algorithm under low signal-to-noise ration and multipath environment. In a field experiment, data transmission with very low bit error rate (BER) was achieved and the communication rate was 375bits/s.
Research on integrated positioning approach based on long/ultra-short baseline
Authors: Wang Jingfei , Zou Nan , Jin Fu
Published: April 2018

With the development of underwater acoustic technology, the combination of multiple positioning means becomes the trend of underwater acoustic positioning technology. Based on an integrated long/ultra-short baseline location equipment, this paper discusses a multi-solution fusion method based on the minimum mean square error criterion. This method fuses long baseline and ultra-short baseline information on output, and it is relatively simple. The results of integrated long/ultra-short baseline location are compared with the results of long baseline location and ultra-short baseline location by simulation and experience. It shows that the integrated long/ultra-short baseline location system has the higher positioning accuracy than long baseline location and ultra-short baseline location. It realizes the high precision positioning of underwater target.
Study of long baseline optimization positioning algorithms Bbsed on redundancy measurement
Authors: Guo Ying , Wang Yan , Zou Nan
Published: April 2018

Generally, underwater vehicles real-time navigation system works through the principle of spherical intersection. The number of array elements required for this method is fixed, but in the case of redundant array elements, they can't be fully utilized. This paper presents two optimization methods based on redundancy measurement. One approach is the steepest descent method and the other is Gauss-Newton method. The former searches for the minimum value of the measurement’s sum of squared residuals along the negative gradient direction, and the later, a non-linear least-squares optimization method that constantly revise regression coefficients through multiple iterations to minimize the sum of squared residuals. Simulation show that both methods are greatly affected by the initial position, the closer the distance between initial position to the target is, the smaller positioning error is. On the same initial position, the Gauss-Newton method has less positioning error than the steepest descent method. The conclusions have been proved to be valid by sea trial.
A study on warning sound for drowsiness driving prevention system
Authors: Ann Ik-Soo
Published: April 2018

The Korea Expressway Corporation has studied various preventive measures to prevent drowsiness driving accidents, such as providing various drowsiness driven prevention measures, establishing drowsiness rest areas, and operating alarms on drowsiness driving in highway tunnel have been implemented. Recently, the Korea Transportation Safety has also used the monitoring of the driven state (DSM—Driven State Monitoring) and the driving information of the vehicle (VDI—Vehicle Driving Information) to confirm the state of the driver in the running car and perform a drowsiness driving and we are showing a good reaction by introducing a drowsiness driven prevention system which sounds vibration and warning sound. As described earlier, when a driver wakes up during a drowsiness operation, a warning sound plays an important role by installing it in a tunnel or in a car. In this paper, we further developed the warning sound for battle against drowsiness used in the tunnel of the existing highway and studied focusing on using inside the vehicle. This research aims to verify and further improve the warning sound part of the in-vehicle drowsiness driven prevention system which is just introduced and enforced, effectively and contribute effectively to prevent drowsiness driven accidents.
Human middle-ear muscles rarely contract in anticipation of acoustic impulses
Authors: Jones Heath , Greene Nate , Ahroon William A.
Published: April 2018

The Auditory Hazard Assessment Algorithm for Humans (AHAAH) is an electrical equivalence model of the human ear designed to predict the risk for auditory injury from a given impulse noise exposure. One concern with the model is that the middle-ear muscle contraction (MEMC) associated with the acoustic reflex is implemented as a protective mechanism for certain instances in which a person is “warned” prior to the impulse. The current study tested the assumption that the MEMC can be elicited by a conditioning stimulus prior to sound exposure (i.e., a “warned” response). In order to quantify the MEMC, we used laser-Doppler vibrometry to measure tympanic membrane motion in response to a reflex-eliciting acoustic impulse. After verifying the MEMC, we attempted to classically condition the response by pairing the reflex-eliciting acoustic impulse (unconditioned stimulus, UCS) with various preceding stimuli (conditioned stimulus, CS). Changes in the magnitude and/or time-course of the MEMC following repeated UCS-CS pairings would be evidence of MEMC conditioning. Out of the 55 subjects tested so far, both non-conditioned (n = 53) and conditioned (n = 2) responses have been observed. These findings suggest that a “warned” MEMC may not be present in enough people to justify inclusion for being considered protective.
Neurofeedback training of auditory selective attention enhances speech-in-noise understanding
Authors: Choi Inyong , Emory Caroline , Kim Subong , Schwalje Adam
Published: April 2018

Speech-in-noise (SiN) understanding involves multiple cortical processes including feature extraction, grouping, and selective attention. Among those processes, we aimed to investigate the causal relationship between auditory selective attention and SiN performance. Selective attention enhances the strength of cortical neural responses to attended sounds while suppresses the neural responses to ignored sounds, which forms an evidence of sensory gain control theory. We hypothesized that the cortical response-guided neurofeedback could strengthen the sensory gain control, which in turn will improve selective attention performance and may result in better SiN understanding. With a single-blinded, between-subjects design including a placebo group, subjects were asked to attend to one of two simultaneous but asynchronous streams. For the participants assigned to the experimental group, a visual feedback was provided after each trial to demonstrate whether their attention was correctly decoded using their single-trial EEG response. The experimental group participants with four weeks of this neurofeedback training exhibited amplified cortical evoked responses to target speech as well as improved SiN understanding, while the placebo group participants did not show consistent improvement. To our best knowledge, this is the first report of selective-attention training enhancing SiN performance.
Direct measurement of glottal flow waveform in an excised K9 larynx with a vocal tract
Authors: Maddox Alexandra , Oren Liran , Gutmark Ephraim , Farbos de Luzan Charles P. , Khosla Sid M.
Published: April 2018

False vocal folds (FVF) or ventricular folds have been shown to impact the vibration of the true folds. Using computational and experimental models to examine the aerodynamic and acoustic effects the FVF, previous studies have shown that the presence of FVF lowers the phonation threshold pressure, the overall intraglottal pressure distribution, and enhances intraglottal vortical structures. However, the full effect of the FVF on the glottal flow waveform has never been measured in a tissue model of the larynx. Therefore, the objective of this study was to evaluate the impact of FVF on the glottal flow waveform in an excised canine model. A vocal tract model was placed over the larynx, and direct velocity measurements were taken at the glottal exit using tomographic particle image velocimetry. Measurements were taken with a systematic change of the FVF constriction and integrated to calculate the air flow waveform. The results show that a restriction above the folds increased the overall flow rate (dQ/dt) and change the maximum flow declination rate (MFDR). The 3-D measurements at the glottal exit can also be used to validate previous computational studies on how the vocal tract affects the phonation mechanism. The clinical significance of these findings will be further discussed.
Tongue muscle shortening patterns during speech
Authors: Stone Maureen , Leem Natalie , Garret Katie N. , Woo Jonghye , Prince Jerry L.
Published: April 2018

This study will used tagged-MRI to examine the 3D shortening patterns of the genioglossus, verticalis, and transversus muscles during speech tasks that move the tongue backwards, forwards, and in complex deformations. Tissue points at each muscle’s origin, insertion, and points in between are used to calculate length changes over time. To enhance the range of subject differences, 10 controls and 5 glossectomy subjects will be examined. The motivation is to understand how muscle behavior is linked to internal tongue motion patterns. Activated muscles typically shorten along their line of action. In the tongue, however, muscles interdigitate, so that when one muscle shortens, others are passively pulled out of linear alignment. Thus, muscle fibers may not form a straight line when activated or at rest. In addition, the mechanical consequences of activating a specific muscle can be complex since the tongue moves continuously during speech and different regions of the tongue experience different demands. Moreover, since the tongue stretches in three-dimensions, some muscles may activate to augment a primary muscle, that is, to prevent expansion in the wrong direction. Muscle shortening from tagged MRI does not capture actual muscle activity, but can provide insight into interactive muscle behaviors.
Automatic tongue contour extraction in ultrasound images with convolutional neural networks
Authors: Zhu Jian , Styler Will , Calloway Ian C.
Published: April 2018

Ultrasound imaging of the tongue can provide detailed articulatory information addressing a variety of phonetic questions. However, using this method often requires the time-consuming process of manually labeling tongue contours in noisy images. This study presents a method for the automatic identification and extraction of tongue contours using convolutional neural networks, a machine learning algorithm that has been shown to be highly successful in many biomedical image segmentation tasks. We have adopted the U-net architecture (Ronneberger, Fischer, & Brox 2015, U-Net: Convolutional Networks for Biomedical Image Segmentation, DOI:10.1007/978-3-319-24574-428), which learns from human-annotated splines using multiple, repeated convolution and max-pooling layers in the network for feature extraction, as well as deconvolution layers for generating spatially precise predictions of the tongue contours. Trained using a preliminary dataset of 8881 human-labeled tongue images from three speakers, our model generates discrete tongue splines comparable to those identified by human annotators (Dice Similarity Coefficient = 0.71). Although work is ongoing, this neural network based method shows considerable promise for the post-processing of ultrasound images in phonetic research. [Work supported by NSF grant BCS-1348150 to P.S. Beddor and A.W. Coetzee.]
Creaky phonation and Mandarin 214-Sandhi
Authors: Elkins Noah
Published: April 2018

Mandarin has four contrastive tones (55, 35, 214, and 51). The 214 tone is consistently produced at the lowest part of a speaker’s pitch range, and is therefore frequently accompanied by allophonic creaky phonation. Additionally, Mandarin has a complex system of tone sandhi; the type of sandhi being investigated in this paper is whereby, when two adjacent 214 tones are uttered, the first 214 changes to a 35, and the second 214 changes to a 21. Twenty speakers of Mandarin were recorded at the campus of Peking University using a sentence list constructed to elicit minimal pairs between sandhi-unaffected and sandhi-affected 214 tones. Age and gender were not considered a contributing factor to the presence of 214-linked creak, and so such factors were not controlled for. Preliminary results show that there is no significant difference between the amount of creak on affected and unaffected vowels. Therefore, the duration spent uttering the vowel at a lower pitch did not increase the intensity of creak. However, the continued presence of creak on a sandhi-affected 214 vowel may serve as a perceptual cue to that tone, whereby a Mandarin speaker can understand the meaning of the syllable even without a robust pitch contour.