bluuebunny/crossref_metadata_2025 huggingface dataset:
Intelligibility of sequentially presented speech signals.
Authors: Iyer Nandini , Brungart Douglas S. , Simpson Brian D.
Published: October 2013

In some operational environments, listeners are required to monitor several channels of simultaneous time-critical communication. There are two possible ways to reduce the transmission time for multiple speech messages: (1) present information from two or more speech channels simultaneously or (2) time compress the competing speech messages and present them sequentially. Previous studies have shown that the intelligibility of a single speech message is fairly impervious to high degrees of time compression. However, it is not clear if intelligibility is preserved when several such highly compressed messages are presented sequentially. The current experiment investigated speech intelligibility when two, three, or four time-compressed speech signals were presented sequentially. Three results were apparent: (1) speech intelligibility was near ceiling when time-compressed signals were presented sequentially and listeners were required to identify and respond to only one of the signals in the sequence; (2) speech intelligibility decreased dramatically when the listeners were required to respond to all the messages in the sequence, particularly at high compression rates; and (3) overall performance in the multiple-message identification task depended primarily on the total time interval between the onset of each new message, and not on the degree of time compression of the speech message contained in that interval.
Detection of warning signals while operating a motorcycle and wearing hearing protection.
Authors: DeSherlia Andrew D.
Published: October 2013

It has been shown that operators of motorcycles are exposed to dangerous levels of wind noise at high speeds [McCombe et al., J. Laryngol. Otol. 109, 599–604 (1995)]. Many governmental regulatory bodies have developed restrictions that ban the use of earplugs while operating a motorcycle, due to their obstruction of warning signals. In order to test this theory, 20 people with normal hearing were exposed to recordings of motorcycle wind noise accompanied by warning signals at different amplitudes. As the warning signal’s amplitude was adjusted, the listener was asked to indicate when it could be detected. This determined the threshold at which the warning signal could be detected for listeners with and without hearing protection. This study should broadly indicate whether hearing protection allows the listener to detect warning signals at a lower threshold than if no protection were used.
Extending the sawtooth wave form inspired pitch estimator with an auditory-based preprocessing stage.
Authors: Camacho Arturo
Published: October 2013

The sawtooth wave form inspired pitch estimator (SWIPE) [A. Camacho, J. Acoust. Soc. Am. 124, 1638–1652 (2008)] is extended by introducing an auditory-based preprocessing stage consisting of two main steps. First, a system consisting of a gammatone filterbank followed by half-wave rectifiers and low-pass filters is applied to the signal. This system tends to boost low-order harmonics of the signal, as shown in a work by Klapuri [A. Klapuri, IEEE Trans. Audio, Speech, Lang. Process. 16, 255–266 (2008)]. Second, the square root of the amplitude spectrum is computed on every channel, and they are summed to form a summary square-root spectrum (SSRS). SWIPE is then applied to this SSRS. Results show that the proposed method significantly outperforms the original SWIPE on music and telephonelike speech, albeit producing a small decrease in performance on high-fidelity speech.
An efficient code for environmental sound classification.
Authors: Lutfi Robert , Arora Raman , Liu Ching-Ju
Published: October 2013

Much work has been devoted in recent years to the goal of developing an automated sound recognition system that can accurately and efficiently classify a wide variety of common environmental sounds according to their generating source. The effort is driven, in part, by the desire to understand our own ability to reconstruct from sound an accurate perception of everyday objects and events present in our natural environment [cf. A. S. Bregman, Auditory Scene Analysis, MIT Press (1990)]. Here an efficient code is described for classifying environmental sounds that exploits a recent significant advance in sparse signal reconstruction known as compressed sensing (CS) [cf. D. L. Donoho, IEEE Trans. Inf. Theory, 52, 1289–1306 (2006)]. CS involves a novel approach to sampling in which the salient information in signals is recovered from the projection onto a small set of random basis functions. The advantage of CS over traditional Fourier or wavelet representations is its capacity to allow accurate classification at low signal-to-noise ratios based on few samples and little or no prior information about signals. To demonstrate we compare CS-based and Fourier-based classification for several representative environmental sound classification tasks. [Work supported by NIDCD].
A computational model for auditory saliency of environmental sound.
Authors: De Coensel Bert , Botteldooren Dick , Berglund Birgitta , Nilsson Mats E.
Published: October 2013

Because the information flow received by the human auditory system exceeds the processing capacity of the brain, neural mechanisms engage and guide attention toward prominent parts of the auditory scene. Several computational models for auditory saliency have been proposed recently. Most of these are concerned with speech recognition, and therefore apply high temporal and spectral precision to relatively short sound fragments. Here, a simplified model is described that specifically targets the long exposure times usually considered in soundscape research. The model trades temporal and spectral accuracy for computational speed, but nevertheless implements the key elements that are present in the calculation of complex auditory saliency maps. A simplified “cochleagram” is calculated from the 1/3-octave band spectrogram using the Zwicker model for specific loudness. Saliency is determined based on spectro-temporal irregularities, extracted in parallel at different feature scales, using a center-surround mechanism. Finally, conspicuous peaks are selected using within-feature and between-feature competitions. The model is shown to behave as expected for a number of typical sounds. As an illustration, saliency calculation results for a set of recordings in urban parks are compared with other acoustical descriptors and with perceptual attribute scales from questionnaire studies.
Developing objective measures of listening effort.
Authors: Shrivastav Mini N. , Shrivastav Rahul , Harnsberger James D.
Published: October 2013

The goal of the present work is to develop valid measures of listening effort independent of task difficulty. Various markers of effort will be measured for listening conditions of varying difficulty but with equal performance. The experimental task will involve tracking a previously specified keyword(s) in a speech sample. The task difficulty will be manipulated by changing the number of keywords to be tracked and by changing the level of a noise masker so that the signal-to-noise-ratio changes from favorable to unfavorable. The participants will be a group of young normal-hearing individuals. Effort will be measured using several physiological markers such as galvanic skin response, heart rate, and blood oxygen level. Subjective ratings of listening effort will also be collected. For each level of difficulty, the percent correct detection performance will be calculated. Conditions that result in equal performance (for example, greater than 90% scores) will be selected for analyses. Any difference in physiological measures between these conditions will then likely reflect differences in listening effort. The study will provide preliminary candidates for developing an objective and real-time measure of listening effort.
Semiautomatic measurement of speech error data recorded with ultrasound.
Authors: Frisch Stefan
Published: October 2013

Ultrasound has been shown to be a useful articulatory research tool. However, ultrasound imaging produces video data that is time-consuming to analyze. In this study, previous work on imaging of articulatory errors between velar and alveolar stops is extended to a larger group of participants through the use of measures extracted from semiautomatically created tongue contours that were generated using EDGETRAK software. Speech errors were induced in word onset consonants using nonsense tongue twisters. Consistent with previous research, onset stop errors were quantified using raising of the tongue dorsum toward the palate and angle of elevation of the tongue blade toward the alveolar ridge. In the majority of cases, it was found that errors created productions that were statistically indistinguishable from normal productions of the wrong category. However, some intermediate or gradient errors between categories were also found. The semiautomated measurement technique appears to be an efficient alternative to manual measurement of the data, producing qualitatively similar results. [This work was supported by NIH R03 DC06164.]
An algorithm for token-by-token linear predictive coding analysis of oral monophthongs.
Authors: Oliveira Leonardo , Arantes Pablo
Published: October 2013

Linear Predictive Coding (LPC) is the most reliable vowel formant estimation method. LPC trajectories closely approximate formants and are preferable as a means of avoiding human bias. Nevertheless, formant analysis by LPC is not error free, as it can be affected by fundamental frequency, noise and vocal tract morphology. Error minimization is usually achieved through manual checking of LPC results with spectral contour. An algorithm to find optimal LPC parameters (Burg method) on a token by token basis was developed and implemented as a Praat script. The algorithm minimizes inter- and intra-vowel formant variability without knowledge of vowel identity. Algorithm results were compared with results of manual adjustment of LPC parameters on a trial and error basis on a corpus of 180 stressed and unstressed oral monophthongs of Brazilian Portuguese from a male speaker. First and second formants linear correlations are r2 = 0.98 and 0.99 respectively. The areas of /i, a, u/ vocalic triangles (mean values as vertices) of manual and automatic measurements showed only 4.5% difference. These results suggest the algorithm can be generalized to any speaker and speed up the measurement of vowel formants in large corpora. [Work supported by CNPq and Fapesp.]
Automatic acquisition device identification from speech recordings.
Authors: Garcia-Romero Daniel , Espy-Wilson Carol Y.
Published: October 2013

In the field of speaker recognition (SR), the artifacts left in the speech signal by the acquisition device are highly detrimental. When the speech signals used in the enrollment and testing phases come from different devices, the recognition accuracy decreases significantly. Hence, most SR systems try to remove these artifacts by using some general knowledge about the acquisition device. However, the degree of specificity used in the characterization of the devices is very broad (e.g., GSM handset as opposed to CDMA). This study focuses on the dual problem with a higher degree of specificity. That is, given a speech recording, how to automatically determine the exact make and model of the acquisition device regardless of the speaker and speech content. For this purpose, the HTIMIT and LLHDB databases, comprising short sentences from more than 300 speakers over 10 different acquisition devices, were used. Closed set identification experiments were conducted to assess the performance of different feature sets (i.e., MFCC and linear-FCC) along with a Gaussian mixture model classifier. The results show that statistical modeling of the device characteristics in terms of MFCC provides an excellent discriminative power among the 10 devices tested. Classification accuracies higher than 90% are reported.
Automatic analysis of vocal manifestations of psychological states.
Authors: Rosenfeld Elizabeth , Bernstein Jared
Published: October 2013

Previous research identified acoustic indicators of mood and emotion in speech [I. Murray and J. Arnott, J. Acoust. Soc. of Am. 93, 1097–1108 (1993)], suggesting that speech signals could be measured automatically to estimate the perceived mood or affect of the speaker. Forty experimental subjects produced 1584 spoken renditions of neutral sentences, either read-aloud or repeated verbatim from a spoken model. An automatic speech processing system extracted ten paralinguistic parameters from each of these spoken responses. Samples of spoken responses were then played to psychotherapists who assigned an apparent mood to each recording. Results indicate that very reliable (0.98) estimates of speaker mood can be derived from as few as 12 judgments made when listening to spoken renditions of fixed material that contains no indication of mood in the linguistic content. Furthermore, these reliable estimates can be predicted with a correlation coefficient of 0.79 from a linear combination of only three of the ten parameters: rate of speech, response latency, and maximum speech amplitude. This prediction accuracy was achieved without using any parameters based on fundamental frequency and without any sophisticated pattern matching techniques applied to the data.
Discriminating dysarthria type and predicting intelligibility from amplitude modulation spectra.
Authors: LeGendre Susan J. , Liss Julie M. , Lotto Andrew J.
Published: October 2013

Subtypes of dysarthria are often described in terms of particular disruptions in speech rhythms. In an attempt to quantify these differences, the envelope modulation spectrum (EMS) was computed on sentences read by speakers with hypokinetic, hyperkinetic, ataxic, and mixed flaccid-spastic dysarthria, and a group of normal speakers (43 speakers total). The EMS consists of the spectra of the slow-rate (up to 10 Hz) amplitude modulations of the full signal and within seven octave bands ranging in center frequency from 125–8000 Hz. From these spectra, six variables were calculated for each band relating to the peak frequency and amplitude and the relative energy above, below, and in the region of 4 Hz. According to stepwise discriminant analysis, six of these EMS variables could correctly discriminate speaker group membership with an accuracy of 84% (better performance was obtained for distinguishing individual groups). Further, 85% of the variance in the intelligibility of the dysarthric speakers (derived from a separate set of sentences) was predictable from a subset of six of these variables. These results suggest that there are reliable perturbations of speech temporal patterns associated with dysarthria subtypes and that these perturbations play an important role in intelligibility. [Work supported by NIH-NIDCD.]
Beyond acoustic data: Characterizing disordered speech using direct articulatory evidence from real time imaging.
Authors: Holsinger Edward , Ramanarayanan Vikram , Byrd Dani , Goldstein Louis , Gorno Tempini Maria Luisa , Narayanan Shrikanth
Published: October 2013

The traditional view of disordered speech characterizes speech errors as deletions, insertions, or transpositions of one or more phonetic symbols in sequence. Recent articulatory research, however, suggests that such a characterization of disordered speech is at best incomplete, and at worst incorrect. A consequence of abandoning the traditional view is that the value of phonetic transcription as a tool for analyzing speech errors is in question. We examine the discrepancy between the phonetic transcription and the actual articulatory events produced by an apraxic speaker whose articulation was imaged with real-time MRI with synchronous noise-canceled audio in a speak-after-me shadowing task. Results indicate that transcription was somewhat reliable between two trained transcribers, but several systematic discrepancies existed. In addition to the transcription being an incomplete record, we also found cases in which the transcription was misleading, unhelpful, or simply incorrect regarding the actual articulation produced by the speaker. Further examination of the acoustic signal revealed that the problems with the transcription cannot easily be overcome with detailed analysis of the acoustics. This study suggests that accurate characterization of disordered speech requires data of a type that cannot be provided solely by transcripts or acoustics. [Work supported by NIH]
Acoustical analysis of vowel production in people with Parkinson’s disease.
Authors: Richard Gabrielle , Menard Lucie , Cohen Henri , Pourcher Emmanuelle
Published: October 2013

Speech is the sound wave produced by the vibration of the vocal chords finely tuned by the supraglottal articulators: the tongue, the lips, and the jaw modulate the vocal tract shape to produce sounds that are distinctive enough to be distinguished. But how can people who suffer from a motor control disease such as Parkinson’s disease (PD) produce comprehensible speech? This work aims at studying the differences between speakers with PD and control speakers (C) regarding acoustical space, coarticulation, and contrastive emphasis. Eight speakers with PD and eight control speakers have been recorded in a speech production task. The Quebec French vowels /a/ and /i/ have been produced, in coarticulation with /b/ and /d/, in neutral and under emphasis contexts. The value of the first and second formants at the middle of the vowel, and the value of the second formant at the onset of the vowel have been extracted. Durations have also been extracted. The analysis showed that the coarticulation consonant has a greater impact on formantic values for the group with PD than the C group. Moreover, the formantic difference between the neutral and emphasis condition is greater for the group with PD than for the C group.
Articulatory modification induced by speaking rate change in Parkinson’s Disease.
Authors: Kim Yunjung , Weismer Gary
Published: October 2013

One of the most well-known treatment techniques to improve speech intelligibility of speakers with dysarthria involves the reduction of speaking rate (Duffy, 1995). The theory of this strategy is that a slower rate provides more time to attain articulatory targets. However, the underlying mechanism of enhanced speech intelligibility at slow speaking rate is not yet clear. In other words, it has not been directly examined whether this strategy causes measurable articulatory improvements or simply provides listeners with greater time to process impaired speech. This is an important question especially in clinical practice, related to establishing a treatment target, and furthermore, an overall treatment plan for the client. In this presentation, we will report some articulatory and acoustic indices of articulatory characteristics of speakers with Parkinson’s Disease and healthy controls, producing the same utterances at three different speaking rates. [Work supported by internal funds from LSU.]
Acoustic characteristics of vowels in dysarthria: A comparison of clear and conversational speech.
Authors: Tjaden Kris , Lam Jennifer , Sussman Joan , Blackburn Stacy , Fulcher Katrina
Published: October 2013

The impact of dysarthria on vowel production has been demonstrated in a variety of studies. Acoustically, these deficits often take the form of a reduced or collapsed vowel space area, reduced F2 interquartile range, and reduced vowel transition slopes and extents. Research further suggests that these segmental deficits may help to explain reduced intelligibility or overall speech severity, at least for some speakers with dysarthria. Clear speech shows some promise for improving vowel production in dysarthria as well as for maximizing intelligibility. Comparative group studies investigating clear speech in dysarthria are lacking, however. The current study will report acoustic measures of vowel production for individuals with dysarthria secondary to Multiple Sclerosis, Parkinson’s Disease, and healthy controls. As part of a larger study, speakers read 25 Harvard Sentences in a variety of speaking conditions, including Habitual or conversational and Clear. Static and dynamic segmental measures of vowel production will be reported. Group effects as well as individual speaker trends will be explored. [Work supported by NIH.]
The differential effects of dysarthria type on lexical segmentation.
Authors: Liss Julie M. , Spitzer Stephanie M. , Lansford Kaitlin , Caviness John N.
Published: October 2013

Although intelligibility deficits are common across various forms of dysarthria, it was hypothesized that the causes of intelligibility reductions depend on the ways the degraded signal challenges listener cognitive-perceptual strategies. This hypothesis was confirmed by analyzing the lexical boundary errors (LBE) elicited by 44 speakers with one of four forms of dysarthria (ataxic, hypokinetic, hyperkinetic, mixed spastic-flaccid). Error patterns from 60 listener transcripts revealed reliance on metrical stress to segment the 80 six-syllable phrases. However, reductions and disturbances in the acoustic cues to syllabic stress associated with each form of dysarthria resulted in different LBE patterns. Explanations for these unique patterns are found, in part, in the metrical structures of the dysarthric speech. For example, the temporal relationships among stressed and unstressed syllables varied widely in hyperkinetic speech, which ostensibly prevented listeners from optimally exploiting metrical stress as a cue for segmentation. The equal-even temporal structure of the spastic-flaccid speech resulted in the highest proportion of boundary insertions. The presence of interword pauses facilitated the assignment of word onset to strong syllables. Results will be presented relative to models of lexical segmentation and clinical implications for targets of remediation in the dysarthrias. [Work supported from NIH-NIDCD 5R01 DC6859.]
Acoustic effects of familiarity on speech.
Authors: MacAuslan Joel , Boyce Suzanne
Published: October 2013

In earlier work, we have reported subtle differences in acoustic measures of speech from 15 subjects who produced the Rainbow Passage at eight hour intervals over 36–42 hours of sleep deprivation. To determine whether results are due to sleep deprivation per se, or to the effect of increasing familiarity with the speech materials, we collected Rainbow Passage recordings from 15 subjects at eight hour intervals over three days while following their normal sleep/wake pattern. These data were subjected to the same measures (based on automatic detection of acoustic landmarks) as reported in our previous work. Results showed no consistent effect of familiarity with speech materials under these experimental conditions, supporting the contention that sleep deprivation affects speech articulation. [Work supported by NIH.]
Refinement of speech motor control in preschool-aged speakers.
Authors: Vick Jennell C. , Moore Christopher A. , Campbell Thomas F. , Shriberg Lawrence D. , Green Jordan R.
Published: October 2013

This investigation explored whether meaningful subgroups of children would emerge from a heterogeneous population of typical preschool speakers as a result of convergences of measures from multiple speech production subsystems: behavioral, phonatory, articulatory, and respiratory. Because variability is a ubiquitous characteristic of preschool speech production, the goal of this classification was not only to minimize within-subgroup variability, but to identify the latent parameters across speech subsystems that characterized each resulting subgroup with the goal of better understanding the refinement of speech motor control that takes place at this stage of development. It is critical to understand the range of performance and interactions among speech production subsystems that exist in children with typical speech development so that we can better quantify the deviancies observed in children with speech disorders. Data were gathered from 60 speakers, aged 37–59 months, and analyzed using multivariate pattern-based classification analyses. The result was two distinct clusters of individuals whose performance across speech subsystems was differentiated by stability. One group was characterized by higher levels of variability across phonatory, articulatory, and respiratory subsystems compared to the second group. The groups did not differ significantly by composition in terms of age or sex.
Non-native speech processing in children with phonological disorders.
Authors: Yim Dongsun
Published: October 2013

This study examined how children with and without phonological disorders (PD) process accented speech spoken by a non-native speaker. Two research questions were asked in this project. First was to compare how children with and without PD perform on accented speech and second was to investigate whether the exposure to the accented speech allows children’s increase in overall performance. Children were aged from 3 to 8 years old who were native English speakers. The task was to identify the final words of low- and high-context sentences spoken by two non-native Chinese speakers. Overall, it is expected that older children perform better than younger children and children perform better on high- than low-context sentences which shows children’s ability to use contextual cues. For the first research question, it is hypothesized that typical children will outperform children with PD when processing accented speech which suggests unstable phonological representation in children with PD. For the second research question, it is hypothesized that children in both groups will benefit when exposed to the accented speech for the second time. However, whether children with PD will require more time to adapt to accented speech compared to typical children is of interest in the discussion.
Distinctive feature analysis of errant responses to Diagnostic Rhyme Test words.
Authors: Goshorn Edward L. , Kemker Brett E.
Published: October 2013

Ten subjects responded in a conventional multiple-choice format to two randomized lists of Diagnostic Rhyme Test (DRT) words in white noise at 3-dB signal-to-noise ratio. The DRT consists of 96 word-pairs whose initial phonemes differ by one feature: voice-unvoiced, nasal-oral, sustained-interrupted, sibilated-unsibilated, grave-acute, and compact-diffuse. List one consisted of DRT words counterbalanced for each feature and list two contained the contrasting feature. Six replicates were obtained for each subject. Errant responses were analyzed for DRT features as well as by a Chomsky-Halle feature system that permitted assigning numerical values that ranged from one to six to phonetic differences in DRT word-pairs. Results are consistent with previous findings for equality of percent correct across DRT word lists. However, DRT feature contrasts within word pairs were not equally difficult. For example, sibilated and unvoiced stimuli were significantly more difficult than their unsibilated or voiced counterpart, respectively. Also, there was no significant correlation between word error rate and the corresponding Chomsky-Halle feature value for differences in word-pairs. Thus, word difficulty appears to be unrelated to degree of feature difference in a word-pair, but rather is associated with relative phoneme audibility for the type of noise employed.
Context-related variability in adolescents’ misarticulated and typically produced /s/.
Authors: Koenig Laura L. , Preston Jonathan
Published: October 2013

Acquisition of the sound /s/ is widely variable in normal development, but misarticulations that persist longer than approximately age 10 are considered to represent residual speech sound errors (RSSEs). Recent studies suggest that RSSEs may reflect subtle underlying difficulties in developing phonological representations, which, in turn, correlate with reading and spelling skills. Thus, RSSEs, even if restricted to a few sounds, may have larger clinical relevance. The acoustic characteristics of /s/ have been investigated in normally developing adolescents, but less work exists on the acoustics of misarticulated /s/, particularly considering across phonetic contexts. The current work is an exploratory study of /s/ acoustics in adolescents who do and do not misarticulate /s/. Speech samples include word lists as well as connected speech. Syllable-initial /s/ is analyzed across words varying in the following context. Of particular interest is the extent of /s/ variability as a function of the upcoming vowels and consonants (i.e., the degree of anticipatory coarticulation), and how this changes over the duration of the fricative. Instrumental data of this sort may ultimately provide greater insight into the phonological systems of adolescents with RSSEs as compared to their normally speaking peers. [Work supported by NIH.]
Spectral differences between live and replayed speech measured in occluded ear.
Authors: Barac-Cikoja Dragana , Reyes Jose N.
Published: October 2013

To further our understanding of the transfer function between the mouth and the eardrum for occluded ears, we compared the recordings of live and played-back self-generated speech captured in the speaker’s ear canals. The airborne speech feedback (during the speech production) and its recording (during the playback phase) were delivered binaurally via earmolds or insert earphones (ER-2, Etymotic Research, Inc.). Three earmold styles were tested: Full shell with the canal portion extending either to the first or the second bend of the participant’s ear canal, and a custom open-fit (Microsonic 19A). In four separate conditions, in-the-canal speech recordings were made using a pair of probe tube microphones (ER-7C, Etymotic Research, Inc.) that were inserted at a fixed distance from the eardrum. Two female speakers provided samples of four word-imbedded vowels (heed, head, hawed, who’d). The participants spoke at a comfortable speaking level, and their speech was presented at a sound pressure level calibrated to match the level at the ears. The equivalent sound level and long-term average spectra of the corresponding pairs of speech samples were then analyzed as a function of the ear occlusion type. [Support provided by the Rehabilitation Engineering Research Center (RERC) on Hearing Enhancement.]
Acoustic-phonetic variation in word-final alveolar consonants in speech to infants with and without hearing loss.
Authors: Dilley Laura , DeMaison Laura
Published: October 2013

Relatively little work has examined acoustic-phonetic variability in consonants in infant-directed speech, particularly that directed to children with hearing loss. Assimilation is a form of variability which has been particularly well studied in adult-directed speech in which a word-final alveolar consonant takes the place of articulation of a following word-initial consonant (as when /n/ in cotton sounds like /m/ in the phrase cotton[m] balls). Two studies examined types of acoustic-phonetic variation in assimilable environments, defined as contexts in which a word-final alveolar stop (oral or nasal) is adjacent to a word-initial labial or velar consonant, in speech directed to adults and to infants with and without hearing loss. First, waveform, spectral, and perceptual information were used to classify tokens of word-final alveolar stops into four categories: assimilated, canonical, glottalized, or deleted. Next, the second formant, F2, was measured in a subset of these tokens, to gauge the extent of place assimilation in assimilable contexts. Preliminary results show substantial phonetic and acoustic variability in assimilable environments in infant-directed speech, but the extent of this variability appears similar to that in adult-directed speech. These findings have implications for understanding language development in normal-hearing children and children with hearing loss.
Effect of priming on age-related differences in release from informational masking.
Authors: Ezzatian Payam , Li Liang , Pichora-Fuller Kathy , Schneider Bruce
Published: October 2013

In a previous study, [Freyman et al. (2004)] showed that presenting listeners with all but the last word of a target nonsense sentence immediately prior to presenting the full sentence in a noisy background, produced a greater release from masking when the masker was two-talker nonsense speech than when it was speech-spectrum noise, thereby demonstrating that an auditory prime could produce a release from informational masking. In Experiment 1 of this study we showed that auditory priming produced an equivalent amount of release from informational masking in good-hearing younger and older adults. To investigate the extent to which this release from informational masking was due to the semantic content of the prime, in Experiment 2 we noise-vocoded the prime (using three bands) to remove semantic content, while retaining the prime’s amplitude envelope. This manipulation eliminated any release from informational masking. In Experiment 3, when the speech masker, but not the prime was vocoded, the performance of both age groups improved equivalently. These results indicate that younger and older adults benefit equally from semantic priming, and that both age groups make equivalent use of amplitude fluctuations in a masker in an informational masking paradigm.
Correlation of nonisotropically distributed ballistic scalar diffuse waves in two dimensions.
Authors: Weaver Richard , Froment Berenice , Campillo Michel
Published: October 2013

Theorems indicating that a fully equipartitioned random wave field will have correlations equivalent to the Greens function that would be obtained in an active measurement are now legion. Studies with seismic waves, ocean acoustics, and laboratory ultrasound have confirmed this. So motivated, seismologists have evaluated apparent seismic travel times in correlations of ambient seismic noise and constructed impressive tomographic maps of seismic wave velocity, and even detected fractional secular changes in seismic wave speed at a level of 0.0001. Inasmuch as the random seismic waves used in these evaluations are usually not fully equipartitioned, it seems right to ask why it works so well, or even if the results are trustworthy. The usual wave field used in long-period seismology is due to distant ocean storms and is, even in the presence of scattering, not isotropic. Here an asymptotic evaluation of the effect, on apparent travel time in a correlation waveform, is made of nonisotropic distributions of ballistic specific intensity. It is shown that for station pairs separated by one or more wavelengths, that the effect is small. A formula is derived permitting estimations of error, and corrections to travel times. It is successfully compared to errors seen in synthetic waveforms.
Geoacoustic inversion using particle swarm optimization.
Authors: Edelmann Geoffrey F. , Fialkowski Laurie T.
Published: October 2013

A novel optimization approach is applied to geoacoustic inversion with a towed horizontal array. Particle swarm optimization, a stochastic, population-based evolutionary algorithm, is applied to geoacoustic inversion problems in order to estimate unknown sediment and ocean parameters. Geoacoustic inversions with a towed array are applied to both synthetic data and experimental data from MAPEX 2000. Particle swarm optimization yields quick, consistent parameter estimates. These results will be compared with previously published results using an advanced rotated coordinate simulated annealing technique. [Work supported by ONR.]
Target detection under influence of sound wave scatterers using time reversal.
Authors: Tsurugaya Yoshiaki , Kikuchi Toshiaki , Mizutani Koichi
Published: October 2013

In general, it is difficult to detect the forward scattering wave from the target located between the sound source and the array. In a word, because the sound wave from the sound source is strong, the forward scattering wave from the target cannot be isolated. Moreover, when the sound wave scatterers except the target exist, the scattered sound wave acts mutually, and the target cannot be isolated. Therefore, the propagation environment is divided into the reference sound field without the target and the target sound field including the target. Next, the forward scattering wave from the target is extracted by using the signal obtained in each sound field. The time-reversal processing was performed to the extracted scattered wave, and the target position was obtained. As a result, the forward scattering wave was extracted by selecting the referring sound field and the target sound field, and the position of the target could be obtained in the environment that scatterers except target influenced. To establish this method, it is necessary to confirm that the propagation environment retains the phase conjugate.
Instantaneous frequency analysis of ultrasound Doppler signal using Bayesian probability.
Authors: Mehmood Asif , Goggans Paul , Sabatier James
Published: October 2013

In ultrasonic Doppler vibrometry systems, acoustic returned from a moving target is frequency modulated by the components of the target’s velocity in the direction of the system’s co-located ultrasonic transducers. Because of this, instantaneous frequency (IF) analysis of the receiver transducer output can be used to reveal the velocities of moving components. In this paper, an IF estimation algorithm using Bayesian probability theory is presented. The acoustic returns are represented by suitable localized models that are constructed on the basis of a number of moving components of the target. Then the posterior probabilities of these localized models are computed and compared to determine which model best describes the data under observation. Once the preferred model is chosen, then its parameters are estimated. Our IF estimation method performs parameter estimation and model selection simultaneously, and extends to instantaneous estimation of these nonstationary signals. The calculations are implemented using Markov chain Monte Carlo in conjunction with thermodynamic integration with simulated annealing to draw samples from the joint posterior probability for the model and the parameters. Monte Carlo integration is then used to approximate the marginal posterior probabilities for all the parameters. The performance of this method is demonstrated by simulated and experimental results.
Signal processing agorithms for unmanned undersea vehicles navigation and collision avoidance.
Authors: Lee Hua
Published: October 2013

In this paper, signal processing algorithms for dynamic UUV homing and docking, navigation, and collision avoidance are presented. Many conventional techniques are image-based, which involved computation-intensive image reconstruction procedures, which is ineffective in dynamic real-time environment. Therefore, it is of great importance to develop computationally effective signal processing algorithms for high-precision UUV geolocation. To optimize the applicability, the algorithms presented in this paper cover all key modalities, including both active and passive, with temporal pulse and FMCW signal formats. In terms of operating configurations, the first case is classified as the passive-receiver systems. The transmitting signal format for this case is in the form of a twin-pulse set, with opposite polarity. The bearing angle is estimated from the temporal offset of the pulses. The dual version of this configuration is in the form of an active system and the double-integration algorithm can be effectively applied to both versions. The active configuration can then be extended to incorporate the FMCW signaling format. For the FMCW case, estimation of the bearing angle and range distance can be accomplished with the Fourier transform methods. In this paper, the presentation includes theoretical analysis, signal processing algorithms, and experiments.
Analysis of phased array transmitters and beamformers.
Authors: Coutu Gerard , Polonichko Vadim , Wanis Paul
Published: October 2013

As phased array ultrasonic transducers become more wide spread in underwater acoustic measurement applications, a better understanding of their basic performance characteristic is of benefit. The beam patterns of a phased array can be approximated by a steady state analysis. This approach indicates the directional response of an array in response to a sinusoidal excitation. However, a steady state examination does not properly describe transient behavior of the transmit waveform and also the received signal that is subsequently beamformed. We further investigate the transient performance of the beamformer using a closed-form solution to the acoustic wave equation for a multielement ultrasonic phased array transducer. The model produces transducer beam patterns and transient beamformer output for a series of reflective targets in the far-field of the transducer. The results show a close agreement with laboratory measurements and provide a tool for further investigation of phased array transducer design and performance.
Mathematical characterization of the acoustic brightness and contrast control problem in complex vector space.
Authors: Song Minho , Kim Yang-Hann
Published: October 2013

The acoustic brightness and contrast control methods [J.-W. Choi and Y.-H. Kim, J. Acoust. Soc. Am. 111, 1695 (2002)] is one of the methods which can generate higher acoustic energy in desired space by controlling multiple control sources. Since the problem is represented with complex valued multidimensional vector, the relation between input control values (magnitude and phase information of control sources) and output value (acoustic energy level in desired zone) is somewhat hard to predict intuitively. In this paper, a visualized analysis method for the acoustic brightness and contrast control problems is proposed. A geometric structure is given to the problem and acoustic variables are corresponded with geometric quantities. To give a geometric structure to acoustic brightness and contrast problem, the problem is reformulated with inner product. From the geometric structure, the acoustic brightness and contrast control problems are interpreted as a simple geometry problem. From the proposed methods, various 2D simulation results with different conditions can be interpreted with simple geometry characteristic.
Analysis of source denoising techniques.
Authors: Bahr Chris , Yardibi Tarik , Liu Fei , Cattafesta Lou
Published: October 2013

A denoising technique for acoustic signals is analyzed. Previous work derived a methodology where three microphones are assumed to view a single signal field, each under the influence of independent channel noise. Statistical relations are used to determine the coherent acoustic field at each microphone [J. Y. Chung, J. Acoust. Soc. Am. 62, 388–395 (1977)]. The technique is validated using a simulation of a directive acoustic source, specifically a baffled piston, with multiple microphones placed in an arc array in the far field. While the solution is analytically exact for truly incoherent noise, weakly coherent noise is shown to have a significant impact on the technique output. The sensitivity of the denoised output to signal-to-noise ratio and cross-channel noise coherence is studied. In a high-channel-count array, different microphone combinations may be used for the technique. Different microphone selection algorithms are studied to find a rule set most likely to determine the true acoustic signal strength. Outlier rejection is applied to the data and shown to improve simulation results. Additional optimization schemes are applied to the data in an attempt to find alternative methods to three-microphone denoising. [Work partially supported by GE Global Research.]
Source characterization study 2007: The three-dimensional primary field of a seismic airgun array.
Authors: Tashmukhambetov Arslan M. , Ioup George E. , Ioup Juliette W. , Sidorovskaia Natalia A. , Newcomb Joal J. , Stephens James M. , Rayborn Grayson H.
Published: October 2013

During September 2007 the Littoral Acoustic Demonstration Center collected acoustic and related data from three moored arrays and ship-deployed hydrophones spanning the full water column to measure the 3-D acoustic field of a seismic airgun array. A seismic source vessel shot a series of lines to give detailed angle and range information concerning the field of the primary arrival. The data were collected in the western Gulf of Mexico between the East Break and Alamos Canyon regions. Peak pressures, sound exposure levels, total energy spectra, one-third octave band, and source directivity analyses are measures used to characterize the field. Three-dimensional maps of these quantities are generated to show dependence on emission and azimuthal angles and range. Three-dimensional visualizations are being developed using a visualization cave and software for 2-D cave emulation. [Research supported by the Joint Industry Programme through the International Association of Oil and Gas Producers.]
Aspect determination using a transducer with a spiral wavefront: Prototype and experimental results.
Authors: Dzikowicz Benjamin R. , Hefner Brian T.
Published: October 2013

An underwater acoustical navigational beacon is built whose operation is similar to the VHF omnidirectional ranging systems used in aircraft navigation. The beacon consists of two disk transducers whose direction of motion is radial: a circular transducer with constant radius and a spiral transducer whose radius varies linearly by one wavelength over 360 deg. To determine aspect, short tone bursts are then sent from each of the transducers. A remote hydrophone can then determine its aspect to the beacon by comparing the phase between the two transducers, with the circular transducer acting as reference. Results from tank experiments demonstrate the viability of this technique. The vertical beam patterns are also determined and compared to numerical results. In addition, other techniques, such as producing a spiral wavefront using a phased array, are also discussed. [Work sponsored by the Office of Naval Research.]
High frequency scattering from inclusions in marine sediments: Data/model comparisons.
Authors: Ivakin Anatoliy
Published: October 2013

Recent theoretical modeling has shown that the size-distribution of inclusions in marine sediments, such as gravel-sized grains and shells embedded in sandy bottoms, is critical for high-frequency scattering. Analysis of existing data on the bottom scattering strength and sediment ground truth also shows that the backscatter data are primarily responsive to the coarse contents of the sediment. Previous ground truth data analysis, usually based on the sediment sieving, assumed that particles are nearly spherical, so that the sieve size is approximately equal to the equivalent diameter (diameter of sphere with the same volume). Coarse particles, such as shell fragments, however, appear to be very nonspherical. Therefore, a relationship between the particle sieve size and equivalent diameter, defined by the particle shape, is needed to provide required size-distributions for acoustic scattering models. Recent statistical analysis of sediment samples collected during the SAX04 experiment resulted in a set of such empiric relationships, and also provided estimating the particle shape factor, known to be an important parameter in the sediment transport processes. Obtained size distributions were used for calculations of frequency-angular dependencies of the bottom backscattering strength and provided a good agreement in the SAX04 scattering data/model comparisons. [Work supported by ONR.]