bluuebunny/crossref_metadata_2025 huggingface dataset:
Metrics for characterizing the sources of ocean anthropogenic noise.
Authors: Hildebrand John A.
Published: October 2013

Decibels are the standard shorthand for describing acoustic intensity and sound pressure level, but may lead to misunderstanding when applied as bioacoustic metrics. Acoustic power and source transmission energy are alternate metrics with intuitive appeal. Acoustic power, calculated from the acoustic intensity multiplied by the emitted solid angle, yields units of watts. Likewise, the energy per source transmission, given by multiplying acoustic power by the duration of the transmission, yields units of joules. For continuous (or quasicontinuous) signals, the standard procedure is to measure the root-mean-square (rms) of the signal. However, this presents problems for short duration signals where the duration of the signal being measured is an important parameter. In these cases it may be more appropriate to measure the peak-to-peak signal, rather than rms. Bandwidth is another important component of how the signal is described, typically in a narrow-band for ambient noise and broad-band for discrete sources. The characteristics of ocean anthropogenic noise sources in terms of these metrics will be discussed.
The importance of incorporating signal characteristics in the evaluation of noise exposure impacts on marine life.
Authors: Stocker Michael
Published: October 2013

Auditory thresholds are used to establish mitigation guidelines for anthropogenic noise exposure on marine animals. These thresholds are determined using either sinusoidal signals at specific frequencies or band limited, sinusoidal-derived Gaussian noise. Given that the preponderance of naturally occurring noise in the marine environment is sinusoidal, marine animals may have lower thresholds and, thus, lower tolerance to non-sinusoidal noise. Fast rise time impulse noise, continuous non-sinusoidal noise, or a combination of these characteristics may induce biological responses at lower levels than sinusoidal noise with an equivalent power density. The author proposes a metric to evaluate and express signal characteristics as a component of determining noise exposure impacts on marine animals.
Effects of soil water potential and moisture content on sound speed.
Authors: Lu Zhiqu , Sabatier James
Published: October 2013

In order to monitor and understand the temporal variations of sound speed due to the changes of soil physical properties under natural conditions, a long-term field soil survey has been conducted. Measurements of sound speed, soil temperature, soil moisture content, and water potential were performed continuously, along with the measurements of surface temperature and precipitation over a period of 2 years. The effects of water potential, moisture content, soil temperature, and depth on sound speed were assessed. Analysis of the data shows that there is a power law relationship between the sound speed and water potential. It is also found that the water potential is the dominant influence on the sound speed, whereas the moisture content and temperature have relatively minor impacts. Brutsaert and Luthin’s theory was employed to calculate the sound speed as a function of the effective stress. The theoretical predictions were compared with the experimental data and they are in good agreement. The study suggests that sound speed measurement could be used as a new and effective tool for water potential measurement.
Wideband multidomain ultrasonic transducer.
Authors: Nadtochiy Andriy , Ostrovskii Igor
Published: October 2013

Wideband multidomain ultrasonic transducer (WMUT) based on a ZX-cut aperiodically poled LiTaO3 is considered and demonstrated. For a wide band operation inside a frequency range of 40–80 MHz, the domain sizes within a multidomain array vary from 25–50 μ, and the total number of ferroelectric domains N varies from 25–266. The partial-waves method computation yields a strong effect of a narrow acoustic signal formation by a WMUT. For instance, an ultrasonic signal in water is about 10 times shorter than input rf chirp burst applied to the WMUT. As the number N increases, more frequencies in the acoustic signal are present. Experimentally it is demonstrated a wideband delay line for a chirp signal within the frequency range of 32–64 MHz. The delay line is fabricated in a 500-μ thick LiTaO3 wafer, and has two WMUTs with variable domain sizes from 25–50 μ. Ultrasound is generated by an input WMUT having N=14, and then detected with output WMUT having N=134. The experimental measurements are in agreement with theoretical computations. [Work supported by UM.]
Energy flux imaging method for time reversal in elastic media.
Authors: Anderson Brian E. , Guyer Robert A. , Ulrich Timothy J. , Le Bas Pierre-Yves , Larmat Carene , Griffa Michele , Johnson Paul A.
Published: October 2013

An energy flux imaging method for use in locating sources of wave energy during the back propagation stage of the time reversal process will be presented. During the back propagation phase of an ideal time reversal experiment, wave energy coalesces from all angles of incidence to recreate the source event; after the recreation, wave energy diverges in every direction. An energy flux imaging method based on this convergence/divergence behavior has been developed. The energy flux imaging method yields better spatial resolution in source reconstruction than is possible with traditional imaging methods. [This work is supported by Institutional Support (LDRD) at Los Alamos National Laboratory.]
Penetration depth for measurements with ultrasonic diffraction grating spectroscopy.
Authors: Greenwood Margaret S.
Published: October 2013

The current status of ultrasonic diffraction grating spectroscopy was described at the Acoustics 08 Conference in Paris. An important question remains: What is the penetration depth of the ultrasound for such measurements? Although the spectral order m=1 at 90 deg is of primary interest, much insight can be gained by investigating a range of angles. The first step was to measure the amplitude of the m=1 transmitted beam in the liquid as a function of the transmitted angle. Data were obtained by sending a toneburst signal to the send transducer and measuring the response of a receive transducer immersed in the liquid. The angle of the receive transducer was varied in half-degree steps from about 50 to 90 deg. The data were compared with a formulation for the interference of N wavelets from N grooves and showed very good agreement. In addition, an experimental technique was developed to measure the penetration of ultrasound into the liquid. Experimental parameters that affect this penetration depth were also investigated. The results of these measurements will be discussed and compared with the theory.
The effectiveness of multiple harmonics in eliciting the precedence effect in normal-hearing listeners.
Authors: Ellison John , Braasch Jonas
Published: October 2013

Localization dominance due to the precedence effect refers to the ability of listeners to localize a primary auditory event in the presence of reflected sound. It has been shown that localization dominance becomes more stable as the bandwidth of a noise burst increases, which suggests the role of cross-frequency interactions on localization dominance. Currently, little is known about how stimuli containing multiple harmonics influence localization dominance. The present investigation uses a multiple harmonic complex to investigate localization dominance due to the precedence effect. Listeners locate the perceived lateral position of stimuli presented binaurally via headphones by adjusting the interaural-level difference (ILD) that approximates the location of the stimulus. Stimuli are presented as lead and lag signals that simulate direct and reflected sound, respectively. The interstimulus interval (ISI) is varied from 0 to 4 ms, and the phase of the harmonic complex is varied between ears. For both the test stimulus and broadband noise bursts (control condition) physical ILDs occur as a result of lead/lag interference, but in the case of the harmonic complexes, the ILD often does not average out over frequency, as is the case for broadband noise.
Are there specialized mechanisms for pitch interval perception?
Authors: McDermott Josh H. , Keebler Michael V. , Oxenham Andrew J.
Published: October 2013

Pitch intervals are a central feature of most musical systems. It has seemed likely that pitch uniquely permits precise perception of the interval by which two sounds differ, perhaps due to mechanisms specialized for music. We tested this notion by measuring interval discrimination thresholds for pitch, brightness, and loudness in the same subjects. We compared thresholds across dimensions by computing: (1) Weber fractions (the ratio of the interval threshold to the base interval size) and (2) the ratio of the interval threshold to the basic discrimination threshold for a dimension. We found that Weber fractions were comparable across dimensions (although Weber’s law did not hold, as thresholds were roughly constant across the range of interval sizes used), and that the threshold ratio was actually higher (worse) for pitch than for loudness or brightness. Interval thresholds thus provide no evidence that pitch interval perception has unusually high fidelity, as might be expected if it were the product of a specialized mechanism. The ubiquity of pitch in music may simply be due to basic discriminability, which, in contrast to interval discriminability, is far superior for pitch than for other dimensions. [Work supported by NIH grant R01DC05216.]
Relative contribution of frequency and duration cues to estimates of frequency change in tone sequences and glides.
Authors: Henry Molly J. , McAuley J. Devin
Published: October 2013

This research compared use of frequency and duration cues in magnitude estimates of frequency change (velocity). In two experiments, listeners rated how quickly tone sequences (Experiment 1) or glides (Experiment 2) changed in frequency on a scale ranging from 0 (not changing at all) to 100 (changing very quickly). In both experiments, velocity was specified in terms of semitones per second (ST/s), with ascending and descending stimuli presented at one of eight velocities on each trial (0, 2, 4, 6, 8, 10, 12, and 14 ST/s) with a variable starting frequency. Frequency and duration cues to velocity were varied orthogonally. The different stimulus velocities (Δf/Δt) were created by either holding Δt constant and varying Δf (frequency cue condition) or by holding Δf constant and varying Δt (duration cue condition). For tone sequences, magnitude estimates of velocity were based almost entirely on duration cues, with very little contribution of frequency cues. This finding contrasted with magnitude estimates of glides where both duration and frequency cues were equally effective at conveying velocity information.
The range of signal-to-noise ratios needed to equalize identification of different consonants.
Authors: Yund E. William , Divenyi Pierre L. , Ua Cruadhlaoich Matthew A. I. , Woods David L.
Published: October 2013

Individual consonants vary in intelligibility when speech is presented in constant-intensity speech-spectrum noise. This variation in consonant intelligibility is explained at various levels, ranging from bottom-up energetic masking of different-intensity auditory cues needed to identify different consonants, to the top-down use of context information. This study focuses on bottom-up auditory cues by measuring the differences in signal-to-noise ratio (SNR) needed to equate consonant identification in consonant-vowel-consonant (CVC) syllables. The goal is to develop a CVC test with high sensitivity to changes in consonant perception because each consonant is presented at SNRs above the perceptual floor and below the perceptual ceiling. In order to present the initial and final consonants of each CVC at independently controlled SNRs, the intensity of the noise was changed during 100 ms centered in the vowel. In 16 young normal-hearing listeners, the SNR range required to equate identification across the 20 leading and trailing consonants was greater than 40 dB. In spite of these large SNR variations occurring during the test (and sometimes even within a single CVC) the expected pattern of consonant confusions was obtained, suggesting that the mid-vowel adjustment of noise intensity did not interfere with normal consonant-perception processes.
Effects of on- and off-frequency maskers on speech recognition.
Authors: Apoux Frederic , Healy Eric W.
Published: October 2013

The present study investigated the relative influence of on- and off-frequency noise bands on speech recognition. The target speech stimuli consisted of 64 consonant-vowel consonants. The background noise was simplified speech spectrum-shaped noise or a sentence played backward. All stimuli were divided into 30 contiguous auditory filter width bands spanning 80 to 7563 Hz. Normal-hearing listeners were presented with 15 target speech bands having frequency locations determined randomly from trial to trial to simulate the quasi-random masking effect of noise on the distribution of speech cues across frequency. Results showed that high levels of noise had little impact on recognition when that noise resided only in the 15 off-target bands. Performance was similar when noise was present in all bands versus when present only in the 15 target bands, further indicating a modest effect of off-frequency noise on speech recognition. Performance with sine-wave vocoded speech bands was systematically lower than that obtained with the unprocessed speech bands, supporting the assumption that fine structure information plays a significant role in speech recognition in noise.
Effects of training format on perceptual learning of spectrally degraded voices.
Authors: Choi Sangsook , Kirk Karen , Talavage Tom , Krull Vidya , Smalt Chris , Baker Steve
Published: October 2013

There is evidence that familiarity with the talker facilitates speech recognition (see Nygaard, 2005 for reviews). Identifying talker voices from an auditory speech signal is difficult for cochlear implant users due to the lack of spectral detail conveyed by their processors. Recently, Sheffert and Olson (2004) demonstrated facilitation in voice learning when listeners with normal hearing were trained using audiovisual speech. In the present study, normal hearing listeners were presented with speech processed through a 12-channel noise band cochlear implant simulator and trained to recognize voices in an auditory-only or audiovisual format. Listeners in the audiovisual format required fewer sessions to reach criterion (80% accuracy) than those in the auditory-only format. Voice learning varied as a function of the talkers included in the training set. For the talkers that were more difficult to identify, audiovisual training yielded greater improvements in voice identification than did auditory-only training. After training, participants completed a word recognition task using processed sentences produced by training talkers and novel talkers. Talker familiarity did not influence word recognition performance for either training group. This may be due to ceiling effects with the 12-channel noise band processor. Additional testing in noise is underway.
Spectral modulation detection training: Different outcomes for different trained stimuli.
Authors: Sabin Andrew T. , Eddins David A. , Wright Beverly A.
Published: October 2013

The pattern of peaks and valleys of sound level spread across audio frequency (spectral modulation) is a key stimulus cue used for sound identification and localization. However, little is known about how practice can influence the ability to detect this modulation. Here we trained three separate groups of normal-hearing adults to detect one of three different spectral modulations. Listeners practiced distinguishing a noise with a flat spectrum from one with a sinusoidal spectral modulation 1 hr/day for 7 days. We varied the modulation depth adaptively to determine the detection threshold. On average, the groups trained to detect 0.5 cyc/oct or 1 cyc/oct spectral modulation across a 200–1600 Hz carrier improved significantly, while the group trained to detect 2 cyc/oct spectral modulation across a 400–3200 Hz carrier did not. Further, it took multiple sessions for the 0.5-cyc/oct group to reach optimal performance, but only one session for the 1-cyc/oct group. Thus it appears that the current training had a smaller and more rapid effect as spectral modulation frequency increased from 0.5 to 2 cyc/oct (if we assume no influence of carrier). Pre-training thresholds also decreased over this range, suggesting that there is less room to improve at higher spectral modulation frequencies.
Difference limen for dynamic center-of-gravity signals.
Authors: Hudson H. Dean , Hoglund Evelyn M. , Feth Lawrence L.
Published: October 2013

We hypothesize that central auditory processing of the dynamic center-of-gravity (COG) of a pair of tones is similar to that for a single tone that undergoes sinusoidal frequency modulation (FM). The signal COG can be moved dynamically by introducing amplitude modulation to two fixed-frequency anchor tones separated by one, or more, equivalent rectangular bandwidths (ERB). The phase relationship between the sinusoidal amplitude modulators of the anchor tones is used to control the magnitude of the COG deviation. In this study, the amplitude of the anchor tones and the depth of AM were identical between intervals. The phase relationship of the modulators was manipulated to control the deviation in the COG. To test our hypothesis, we compared FMDLs with COGDLs at three center frequencies and three modulation rates. First, difference limens were measured for the FM signals in partial replication of [Ozimek and Sek (1990)]. Next, COGDLs were measured. Initial attempts to match COG deviation to FM deviation were unsuccessful because the task was too difficult. Consequently, the modulation depth of the COGDL signals was fixed at 0.5, which resulted in replicable performance across subjects. Data for both the FMDL and COGDL task will be discussed. [Work supported by NIH and ASEE.]
Distortions in the perceived relative temporal positions of a signal and masker.
Authors: Wright Beverly A. , Fitzgerald Matthew B.
Published: October 2013

Simultaneous and non-simultaneous masking tasks are often introduced to listeners in terms of the veridical relative positions of the signal and masker. Here we report that the actual perceived temporal positions of the signal and masker do not always match those descriptions. We asked normal-hearing adults about the perceived presentation time of a brief “signal” that was presented either before, during, or after a longer “masker.” The task was either to mark the perceived presentation time of the signal on a schematic diagram of the masker, or simply to judge whether the signal occurred inside or outside of the masker. Listeners tended to perceive the signals as having occurred later than they actually had. This temporal distortion was most marked for signals that were presented near the end of the masker. Signals that were presented 25–100 ms before the end of the masker were perceived as occurring outside of the masker 50% of the time. Similar results were obtained with different masker durations (150, 300, and 450 ms), and various combinations of signal and masker spectra. This phenomenon may provide a measure of the time it takes to switch attention from a foreground to a background stimulus. [Work supported by NIH.]
Head shadow effect and pinna effect by sound intensity.
Authors: Chen Yen-Sheng , Yu Jen-Fang , Chen Chin-Kuo , Cheng Wei-De , Lin Yi-Hsin , Tsai Go-Long
Published: October 2013

This study is to develop a procedure of measuring the influences of the head shadow effect and pinna effect by stimulus intensities and stimulus frequencies (125 Hz to 8 kHz) based on the stimulus source at azimuth angles (0°–180°). In this study, the head shadow effect would be affected by the stimulus frequencies at 3–8 kHz, and the difference of sound pressure for bilateral ears was 7.46–41.40 dB SPL. The pinna effect would be affected by the stimulus frequencies at 3–8 kHz. The difference of sound pressure for all sound sources was 6.62–35.69 dB SPL. The pinna effect would occur most apparently at 135° of azimuth. Consequently, the sound intensity would not affect the head shadow effect and the pinna effect. On the contrary, the pinna effect, the head shadow effect, the frequency of stimuli and the sound location should be considered for hearing aids selection for hearing-impaired listeners.
Signal-processing strategies for electric-acoustic stimulation.
Authors: Brown Christopher A. , Bacon Sid P.
Published: October 2013

Electric-acoustic stimulation (EAS) occurs when a cochlear implant recipient receives some amount of acoustic stimulation, typically as a consequence of residual low-frequency hearing. This combined stimulation often leads to significant improvements in speech intelligibility, especially in the presence of a competing background. We have demonstrated EAS benefit when target speech in the low-frequency region is replaced with a tone modulated in frequency with the target talker’s F0, and in amplitude with the amplitude envelope of the target speech. We have hypothesized that in some instances, the modulated tone may be more beneficial than speech. For example, a patient possessing extremely limited residual hearing (e.g., only up to 100 Hz) may not show EAS benefit because the F0 of many talkers is too high in frequency to be audible. In support of this hypothesis, we demonstrated that the tone can be shifted down considerably in frequency without affecting its contribution to speech intelligibility. In the current experiment, while generating the tone, we manipulated various signal-processing parameters to examine their effects on the benefits of EAS. One goal of this work is to develop a real-time processor that provides EAS benefit to implant users with limited residual hearing. [Work supported by NIDCD.]
Parametric study on speaker identification biometric system using formant analysis.
Authors: Kovoor Binsu C. , Surpiya M. H. , Jacob K. Poulose
Published: October 2013

Speech is the most suitable bio-feature for environments where hands free is a requirement in the security system. Speaker identification is done successfully utilizing formant frequencies in the human speech. In the prototype system developed, which can be operated without the knowledge of the users, representative features are extracted and compared with speaker specific features stored in the knowledge base. The algorithm mainly consist of four modules, namely, recording of samples and building a knowledge base of the vowel sounds, building autoregressive models, recognizing vowel and thereby identifying the speaker. The important parameters to be fixed for obtaining optimum output for a given input data of the proposed model are envelope threshold and order of parametric AR model. The envelope threshold is the parameter that differentiates numerous peaks in the speech signal while an appropriate value of the order of AR model helps to compute the frequency response of the input signal accurately. The influences of various parameters have been studied and are discussed in this paper. It is expected that the results of the study will help a biometric system developer to choose appropriate value for the parameters in the algorithm for different applications.
Transition characteristics in speakers with dysarthria and in healthy controls: Part III: liquids and glides.
Authors: Weismer Gary , Kuo Christina , Kim Yunjung
Published: October 2013

In two previous reports [Weismer, Kuo, & Chung, J. Acoust. Soc. Am. 121, 3135 (2007); Kuo and Weismer, J. Acoust. Soc. Am. 124, 2558 (2008)] the transition characteristics of speakers with dysarthria, secondary to Parkinson disease and amyotrophic lateral sclerosis (ALS), were compared to transition characteristics of healthy controls. The previous reports described the characteristics of obstruent-vowel [Weismer >etal/> (2007)] and diphthong [Kuo and Weismer (2008)] transitions extracted from an extended passage reading. Our hypothesis was that distributional analyses would show the CV, obstruent-vowel transitions to be less affected by the neurological disease as compared to diphthong transitions. The hypothesis was not supported by the data, which showed both types of transitions to be slower and less extensive than in healthy controls. The present report completes this analysis with documentation of transition characteristics of liquids and glides. Our expectation is that liquid and glide transition types will be slower and more extensive in talkers with dysarthria,as compared to healthy controls [Work supported by DC 003273.]
Comparison of voice intensity effects on electroglottographic versus acoustic jitter and shimmer.
Authors: Brockmann Meike , Drinnan Michael J. , Carding Paul N.
Published: October 2013

Electroglottographic (EGG) jitter and shimmer have been described as more sensitive than acoustic perturbation. Recent studies documented significant voice intensity effects on acoustic parameters even in “normal” voice intensity. The aim of this cross-sectional single cohort study was to compare voice intensity effects in acoustic and electroglottographic measurements in several voice tasks. Forty healthy adults (1:1 f:m) aged 20–40 years phonated /a/ at subjective “soft,” “medium,” and “loud” voice and at 65, 75, 85, and 95 dB. Voice intensity and perturbation measurement method effects were assessed with ANOVA. Electroglottographic and acoustic perturbation were significantly different (p<0.001) at all intensity levels and lowest at highest voice intensities. Biggest differences were found between “soft” and “loud” voice in EGG measurements. Electroglottographic results were significantly higher in all tasks, for example mean shimmer in “soft” voice: acoustic 4.4% (SEM: 0.58%) and electroglottographic 7.35% (SEM: 1.02%). The measurement difference was smaller in prescribed intensities (mean shimmer at 65dB: acoustic 2.5% (SEM: 0.23) and electroglottographic 3.0% (SEM: 0.28%). Voice intensity changes strongly influence acoustic and electroglottographic perturbation and are minimized when patients control for voice intensity. Electroglottographic measurements are generally higher and stronger influenced by voice intensity changes. Future work should cover if this reflects a better sensitivity or more measurement error.
Allophones, voice onset time, and aphasic speech errors.
Authors: Buchwald Adam , Rapp Brenda , Miozzo Michele
Published: October 2013

This paper examines the speech of DLE, an aphasic English speaker who regularly deletes /s/ from /s/-stop onset clusters in English words (e.g., /spik/→ [pik]). DLE’s production of stops in these words provides a window through which to examine the treatment of allophones by the speech production system. In English, the voiceless unaspirated stops that follow word-initial /s/ (e.g., the [p] in speak) are considered allophones of the voiceless stop (e.g., /p/), although the VOT more closely resembles that of phonologically voiced stops (e.g., /b/) which are often realized as voiceless unaspirated stops in word-initial position. Deletion errors occurring at a level before phonetic detail is generated should match the VOT of word-initial voiceless stop (e.g., peak), whereas those arising at a later phonetic level should match the VOT of the unaspirated stop (e.g., beak). In contrast to either of these possibilities, the partial activation of both aspirated and unaspirated allophones of /p/ should yield stops with a VOT that is intermediate between the voiced and voiceless phoneme productions. Thus, DLE’s errors can inform our understanding of the phonetic processing of allophones in speech production.
Perception of talker information by children with typical and impaired linguistic development.
Authors: Levi Susannah , Schwartz Richard
Published: October 2013

Despite the relatively large number of studies documenting the interaction of talker and linguistic information during speech perception in adults, such as talker-contingent phoneme processing or language-dependent processing of talker information, there is little work examining the development of talker processing in children. This study investigated the extent to which language familiarity affects the perception of the talker dimension of the speech signal by testing children’s discrimination of bilingual voices across two different languages. Children—aged 7 to 12—heard pairs of words either in English (familiar language) or German (unfamiliar language) and judged whether the two words were spoken by the same or different talkers. The results of this study indicate that children are able to perform this task at above-chance levels, revealing that the perception of indexical information appears early in development. Furthermore, children are able to process this information even when they are unfamiliar with the language and, therefore, lack access to linguistic information. A comparison between children with typical language development and those with specific language impairment suggests that the perception of indexical information can be retained even when linguistic perception is impaired.
Audiovisual speech perception in children with autism spectrum disorders and typical controls: Implications for development.
Authors: Irwin Julia
Published: October 2013

For typically developing perceivers, visual speech information influences what listeners hear. For example, when the place of articulation of visual and auditory speech tokens are incongruent, perceivers often report hearing a visually influenced response (the McGurk effect; McGurk and MacDonald, 1976). To better understand the role of visual speech information in development, children with autism spectrum disorders (ASD), children with typical development and adults with typical development were assessed on: (1) sensitivity to mismatched auditory and visual speech (McGurk); (2) visual gain in the presence of auditory noise; and (3) detection of auditory and visual asynchrony. Implications for the development of audiovisual speech processing in typically developing children and adults and children with ASD will be discussed. [Work supported by NIH.]
Predicting consonant recognition in quiet for listeners with normal hearing and hearing impairment using an auditory model.
Authors: Jürgens Tim , Brand Thomas , Kollmeier Birger
Published: October 2013

Contrary to listeners with normal hearing, listeners with high-frequency sensorineural hearing loss suffer especially from a poor consonant recognition. This is mainly attributed to the loss of audibility within high-frequency regions. In a former study [Jürgens and Brand, J. Acoust. Soc. Am. (submitted)] it was shown that consonant recognition rates in noise can be predicted well for normal-hearing listeners using an auditory model combined with a speech recognizer. The speech reception threshold (SRT) in noise could be predicted with an accuracy of 1–2 dB. In this study consonant recognition in quiet for normal-hearing and hearing-impaired listeners is investigated by regarding the individual absolute hearing threshold within the model. Observed and predicted psychometric functions of the consonant discrimination performance, consonant recognition rates of single consonants, and confusions are compared and discussed. The results show that the SRT in quiet can be predicted well for both normal-hearing and hearing-impaired listeners. Furthermore, the auditory model is modified to the signal processing of listeners with mild to moderate hearing impairment by, e.g., including a reduced dynamic compression, in order to investigate the influence on the predictions. [Work supported by DFG.]
Infant-directed speech to infants with a simulated hearing loss.
Authors: Lam Christa , Kitamura Christine
Published: October 2013

The acoustic/phonetic features of infant-directed speech (IDS) were measured during face-to-face interactions between mothers and their six-month-old normal hearing infants (N=32) using a setup in which the mother and infant are seated in separate rooms and interact over their respective video monitors. There were two sound conditions, (i) a hearing condition in which infants could hear their mothers at 55 dB SPL, and (ii) a hearing loss condition in which the sound was turned off, and the infants could not hear their mothers. All mothers were told your baby cannot hear you. Mothers and infants interacted for 10 min using three labeled toys (sheep, shoe, and shark) to facilitate the production of exemplars of the corner vowels /i/, /u/, and /a/. The results showed that mean fundamental frequency (F0), F0 range, and the duration of the IDS vowels were comparable across the two conditions. However, vowel hyperarticulation was only evident in the hearing condition, not in the hearing loss condition. These results suggest that mothers adapt their speech input according to signs of infant responsiveness. That is, when their infant cannot hear them, they maintain their modifications to IDS F0 and duration, but at the expense of vowel hyperarticulation.
The combined effects of distortions on vowel identification.
Authors: Molis Michelle R. , Teahen Melissa A. , Leek Marjorie R.
Published: October 2013

The effects of intrinsic and extrinsic distortions on vowel perception were evaluated. Potential sources of distortion were high presentation level, background noise, stimulus ambiguity, and hearing status. Vowel identifications were collected for three sets of synthesized vowels that varied in both F1 and F2 (front and back vowels), or F2 and F3 (central vowels). The systematic variation in formant frequencies produced stimulus sets that contained both clear, unambiguous category tokens as well as relatively ambiguous tokens that could be identified as more than one vowel category. As overall presentation level increased from 75–90 dB SPL, identification by normally hearing listeners became less consistent for ambiguous stimuli. Additional reductions in categorization precision occurred in the presence of background noise. The effects were highly dependent on stimulus set, with the largest effects observed for the central vowels. Data from hearing-impaired listeners at high presentation levels and in noise will also be reported. Evaluation of response patterns in the presence of combinations of these sources of distortion will help explore the difficulty people with hearing loss have processing speech in adverse listening conditions. [Work supported by OHSU Medical Research Foundation, Portland, OR.]
Acoustic correlates to perceived age in American English.
Authors: Harnsberger James D.
Published: October 2013

An acoustic analysis of voice, articulation, and prosodic cues to perceived age was completed for a speech database consisting of 150 speakers of American English. The speakers (75 male, 75 female) were recorded representing three age groups: young (18–30), middle-aged (40–55), and old (62 years and older). The database included several types of read speech: two passages (Rainbow, Grandfather), 16 sentences incorporating a wide phonemic repertoire, three sustained vowels ([a], [i], [u]), and multiple repetitions of nonsense words. Perceptual judgments of age were collected for the sentence materials from 160 listeners in total. These perceived ages were submitted to a multiple linear regression analysis with up to 34 acoustic measures representing: voice quality (e.g., jitter, shimmer, HNR, CPP), articulation (F1+F2 vowel space size, F1 and F2 of point vowels), fundamental frequency (mean, min, max, SD), intensity (SD), speaking rate, and others. An analysis of the sustained vowel data for the females showed that 14 variables were significantly correlated with perceived age (R-squared = 0.78, p < 0.01), while 13 significantly correlated for the male voices (R-squared = 0.57, p < 0.01). Results for the other classes of speech materials will also be reported.
Speech intelligence in a room via time reversal method.
Authors: Too Gee-Pinn , Hsieh Yu-Hao , Chou Chih-Hao , Wu Bo-Hsien , Lee Sony
Published: October 2013

Time reversal method has been widely used in ultrasound and underwater communication. In the present study, this method is applied in a room to enhance the received signal by increasing the signal-to-noise ratio (SNR). A passive time reversal process allows the estimation of source signal at the original source location. Both theoretical simulations and experiments are performed in the present study. The experiments are performed in an anechoic room as well as in reverberation room. The results indicate that SNR has been greatly enhanced by this approach. Also, array length, cross correlation coefficents and other parameters are also considered and discussed in the present study.
Environmental variability and acoustic uncertainty in the New England Shelfbreak Test Experiments.
Authors: Hornick Heather R. , Lynch James F.
Published: October 2013

Two New England Shelfbreak Test experiments were conducted off the coast of New Jersey in the Middle Atlantic Bight, a complex and highly variable environment. Water column observations were collected from moorings, CTDs, and a Scanfish. Such extensive observations are critical to determining an accurate depiction of the sound speed field variability with which to determine the uncertainty of acoustic transmission loss models. Specific environmental parameters are identified as critical keys affecting acoustic uncertainty. By determining which keys have the most significant effect on changes to sound propagation and measuring transmission loss variability based on these parameters, the uncertainty within the region is estimated comparing several previously developed methods. Given the limitations of collecting data of adequate spatial and temporal coverage, it is necessary to determine how increased numbers of observations can reduce the acoustic uncertainty and quantify the improvement in order to balance the cost of observations with an acceptable level of uncertainty.
Observations of micromultipath formation at very long ranges in shallow water.
Authors: Wylie Jennifer , DeFerrari Harry A. , Saintval Wendy
Published: October 2013

Long- and very-long-range shallow-water transmission is observed for low-frequency m-sequences transmissions with center frequencies of 50, 100, and 250 Hz. The source is suspended from a ship that is holding position at each of three ranges: 10, 20, and 80 km. The receiver is a bottomed horizontal array. The array is used to resolve out-of-plane arrival angles to a small fraction of a degree by observing phase rolls of the spatial coherence function of the various arriving wavefronts. At the 10 and 20 km ranges, distinct pulse arrivals are observed in one-to-one correspondence to normal mode arrivals predicted with PE propagation models. But at the 80 km range, a large number of distinct arrivals appear with intensities or arrival times inconsistent with deterministic normal mode or ray arrivals. One possibility is that these are micromultipaths from out-of-plane paths after reflections from facets or sloping bottom contours. In fact, each of the arrivals has a unique horizontal arrival angle, and the steeper the out-of-plane angle, the later the arrival—an observation that is verified with broadband PE propagation model computations for both random faceted and gently sloping bottoms.
Calibration of a passive acoustic target scattered data from the Gulf of Maine.
Authors: Küsel Elizabeth T. , Ratilal Purnima
Published: October 2013

The scattered field from long vertical cylindrical targets, such as BBN targets, can be calculated by direct application of Green’s theorem and making use of appropriate boundary conditions. In a recent paper submitted to J. Acoust. Soc. Am., the authors show that the scattered levels from BBN-type targets in ocean waveguides with depth-dependent sound speed profile are influenced by the refractive nature of the incident field over the target depth extent. In the current work, the scattering model for vertically extended cylindrical (VEC) targets in range-dependent ocean waveguides is validated with experimental data from the 2006 ONR-sponsored ocean acoustic waveguide remote sensing (OAWRS) experiment in the Gulf of Maine. In order to take into account the scintillation in the measured scattered intensity caused by spatial and temporal fluctuations of the ocean waveguide, a stochastic approach is adopted. Monte Carlo simulations of the scattered field statistics are calculated by implementing the VEC model in a range-dependent environment randomized by internal waves. The simulated scattered field statistics for different source-receiver-target configurations is compared to the scattered returns from BBN targets measured during the 2006 Gulf of Maine experiment. Results show good agreement between the simulated and experimental scattered field statistics.
Considering modal aliasing in the implementation of an acoustic echo canceller in the wave domain.
Authors: Schneider Martin , Kellermann Walter
Published: October 2013

Traditional multichannel acoustic echo cancellation applied to wave field synthesis systems is computationally extremely expensive due to the large number of channels to be identified. By using wave domain adaptive filtering (WDAF) one is able to choose an approximation which neglects the interaction (exchange of energy) between different modes. This reduces the number of necessary adaptive filter coefficients drastically. Furthermore, it allows to use computationally relatively inexpensive single-channel filter adaptation algorithms. Then, however, modal aliasing due to the limited resolution of the used microphone array becomes a significant problem since it constitutes an interaction between different modes with respect to the microphone signals. The effects of modal aliasing on the adaptation behavior of a WDAF-based echo cancellation system are analyzed and a modified wave domain representation of the synthesized wave field is proposed which reflects the limited resolution of the microphone array.
Imaging room acoustics with the audio camera.
Authors: O’Donovan Adam , Duraiswami Ramani , Gumerov Nail A. , Zotkin Dmitry N.
Published: October 2013

Using a spherical microphone array and real time signal processing using a graphical processing unit (GPU), an audio camera has been developed. This device provides images of the intensity of the sound field arriving at a point from a specified direction to the spherical array. Real-time performance is achieved via use of GPUs. The intensity can be displayed integrated over the whole frequency band of the array, or in false color, with different frequency bands mapped to different color bands. The resulting audio camera may be combined with video cameras to achieve multimodal scene capture and analysis. A theory of registration of audio camera images with video camera images is developed, and joint analysis of audio and video images performed. An interesting application of the audio camera is the imaging of concert hall acoustics. The individual reflections that constitute the impulse response measured at a particular seat may be imaged, and their spatial origin determined. Other applications of the audio camera to people tracking, noise suppression, and camera pointing are also presented. [Work partially supported by NVIDIA and the VA.]
Software for bioacoustic analysis of passive acoustic data.
Authors: Mellinger David K. , Gillespie Douglas , Figueroa Harold , Stafford Kate , Yack Tina
Published: October 2013

Software analysis systems comprise an important stage in passive acoustic research. Here we compare and evaluate three animal acoustic analysis software systems: Ishmael, PAMGUARD, and XBAT. These packages are compared and evaluated for their capabilities at some of the common tasks in animal sound analysis: recording, display, detection, classification, measurement, localization, and tracking of animal sounds. They are also compared for their extensibility (how easy is it, say, to add a new detection algorithm), their ease of use (how hard is the software to learn to use, how quick to use it once one knows how), their hardware interfacing (what types of sound acquisition hardware can they receive sound from), their sound file interfaces (can they read everyone’s sound files), their documentation, and their software interfaces (how well do they share data with other programs, other methodologies, like visual surveys). Special features of each system will also be discussed.
Passive acoustic monitoring of marine ecosystems in the Pacific Islands Region.
Authors: Lammers Marc O. , Wong Kevin , Brainard Russell E. , Au Whitlow W. L. , Fisher-Pool Pollyanna
Published: October 2013

Monitoring the changing state of marine habitats is a challenging task that is exacerbated when the habitats in question are in remote locations. Passive acoustic monitoring is sometimes the best, if not the only, means of gauging levels of biological and anthropogenic activities in such areas. Since 2006, the Pacific Islands Fisheries Science Center, in partnership with the University of Hawaii, has been building a network of long-term acoustic monitoring stations across the Pacific Islands Region using Ecological Acoustic Recorders (EARs). The network is currently composed of 29 long-term monitoring stations and spans the Hawaiian Archipelago, American Samoa, the Line Islands, Johnston Atoll, Wake Atoll, Guam, and the Commonwealth of the Northern Mariana Islands. EARs are diver-deployed and refurbished by local partners and/or during annual or biennial research cruises to deployment locations. A wide range of natural and anthropogenic acoustic signals are monitored, including sounds produced by invertebrates, fish, cetaceans, vessels, and surface weather events. The long-term trends in biological acoustic activity obtained through this network will be used to gauge the relative stability of the ecosystems associated with each location. Detections of vessels are common at many locations and provide a quantitative means of establishing levels of anthropogenic activity.
Autonomous seafloor recorders and autodetection buoys to monitor whale activity for long-term and near-real-time applications.
Authors: Tremblay Christopher , Calupca Tom , Clark Christopher W. , Robbins Matt , Spaulding Eric , Warde Ann , Kemp John , Newhall Kristopher
Published: October 2013

Biologists and engineers from Cornell have installed arrays of autonomous seafloor recorders in multiple ocean habitats. These systems are designed to continuously monitor for the occurrence of endangered whales, particularly right whales; provide critical data on whale seasonal occurrence, distribution and relative numbers; and evaluate potential noise impacts from manmade activities and commercial shipping. Recent studies have been conducted in Massachusetts Bay around commercial LNG ports, the Arctic Ocean at seismic study locations, and in New York waters extending to the shelf edge. In combination with Cornell-developed MATLAB- and JAVA-based software applications, these seafloor recorders have been utilized to evaluate seasonal whale presence, calculate levels of commercial shipping and construction noise, and locate and track the movements and behaviors of individual whales under different acoustic conditions. Cornell and ocean engineers from the Woods Hole Oceanographic Institution have installed an operational network of autodetection buoys off New England to provide near-real-time data on right whale presence to transiting LNG vessels, government agencies, and public entities. When combined, the seafloor and autodetection buoy systems provide a powerful mechanism for understanding whale behavioral ecology and describing their acoustic habitats, while mitigating the risks from ship strike and noise exposure.