bluuebunny/crossref_metadata_2025 huggingface dataset:
Using spectral constancy to encode temporal pitch and improve cochlear-implant melody perception.
Authors: Zhang Duo , Zeng Fan-Gang , Atal Bishnu S.
Published: July 2011

Despite good speech performance in quiet, melody perception remains a challenge to cochlear-implant (CI) users. Current CI pitch-encoding schemes significantly alter original signals, producing potentially detrimental effects on speech perception. The present study proposes to take advantage of spectral constancy to encode temporal pitch while minimizing signal distortion. Spectral constancy refers to unaltered timbre perception when the overall level of a signal is roved. For example, the vowel /a/ sounds the same independent of its overall level. The idea is to extract and encode pitch by increasing the gain of all channels proportionally and synchronously according to the pitch period. This scheme alters the temporal envelope of voiced sounds but preserves their spectral envelope in each time frame. It does not alter temporal envelope of unvoiced sounds, making it a good comprise between encoding pitch and preserving spectral and temporal envelopes. The new scheme has been implemented in a real-time speech processor and evaluated in nine nucleus CI users who identified ten melodies with minimal training and in a test protocol that randomized the stimulus presentation order. Compared with the standard continuous interleaved sampling strategy, the new scheme significantly improved CI melody perception by 10–20%.
Perceived imitation of regional dialects.
Authors: Phillips Sara C. , Clopper Cynthia G.
Published: July 2011

This study examined the effect of regional dialect on perceived spontaneous phonetic imitation. Participants shadowed CVC target words produced by talkers from two American English dialect regions, the North and the Midland. Imitation was measured acoustically in terms of vowel quality, vowel duration, midpoint f0, f0 trajectory, and onset and coda consonant duration. The productions and targets from the shadowing task were then used as stimulus materials in an AXB discrimination task to measure perceived imitation. Listeners selected whether the shadowed production of each word was more similar to the target (X) than a baseline recording of that same word. The AXB experiment included three conditions to distinguish between imitation of talker-specific and dialect-specific properties, in which X was the original target from the shadowing experiment, X was produced by another talker from the same region, or X was produced by a talker from the other region, respectively. Preliminary results suggest that listeners used talker-specific, rather than dialect-specific, properties when judging imitation. Among the acoustic measures, perceived imitation was correlated with vowel duration. These results suggest that talkers in a shadowing task imitate talker-specific variability more than general properties of regional dialect.
Phonetic convergence after perceptual exposure toward native and nonnative speakers.
Authors: Kim Midam
Published: July 2011

This study explores phonetic convergence by native English speakers after perceptual exposure to speech by either a native or a non-native English speaker. Three native and two Korean non-native English speakers read two sets of English words and sentences as the model speakers. A separate group of native English speakers read the two material sets, and was then exposed to one of the material sets either through auditory inputs read by a model speaker (experimental groups) or visual inputs (control groups), and read all materials again. Half of the participants were exposed to a native model speaker, and the other half to a non-native model speaker. Participants also conducted an implicit association test, which measured their attitudes toward native and foreign speakers of English. The first and second recordings of the test talkers were acoustically compared in terms of the Euclidian distances along acoustic-phonetic dimensions between participants’ postexposure and the model productions, and between their pre-exposure and the model productions. Preliminary results illustrate that, at least for some acoustic measures, (1) participants with relatively large distances between their pre-exposure recordings and the model’s recordings phonetically converged toward their models; and (2) participants generalized their convergence patterns to auditorily unexposed materials.
Final stop accommodation in married couples.
Authors: Miller Newman Sara E. , Heaton Hayley E.
Published: July 2011

This study investigates convergence between married couples to determine if there are patterns of convergence and if that convergence can be attributed to social indexing or speaker accommodation. Preliminary work done with sociolinguistic interviews of residents of Harkers Island, NC demonstrated that the release of the final stop consonant /t/ has social meaning: Those residents who most identified with island life deleted or failed to release their final /t/ in casual speech, and female speakers were much more likely to release final /t/ than male speakers in their cohort. The two married couples examined in the Harkers Island data revealed two separate patterns of accommodation. One couple exhibited convergence that mirrored the male, island identifying spectrum where his percentage of stop release was 5.4% and hers was 0%. The other couple exhibited convergence that mirrored the female /t/ release spectrum with her percentage of stop release of 29.6% and his of 30%. These preliminary results suggest that not only is there phonetic accommodation between spouses, but also that this accommodation is not mutual convergence. It is hypothesized that patterns of phonetic convergence will emerge, and that these patterns will fall along the continuum of social indexing in the larger community.
Variables influencing the size of carrier-phrase dependent effects on speech perception.
Authors: Vitela Antonia D. , Lotto Andrew J.
Published: July 2011

Previous work has provided evidence that the presence/absence of carrier phrase effects on target speech sound categorization can be well predicted by comparing the long-term average spectrum (LTAS) of the carrier phrase with the spectrum of the target. In this series of studies, we move beyond asking whether there is a significant effect or not to examining the relative size of these effects. Target series of minimal-pair words varying in vowel were preceded by carrier phrases synthesized to simulate a talker with a large or short vocal tract. Shifts in vowel categorization percentage as a function of change in talker served as the variable of interest. Results indicate that manipulations of the perception of the carrier as speech (by presenting forward or backwards), the LTAS of the carrier (by gating the duration of the carrier presented), and the local information in the carrier (by preserving or not the part of the carrier immediately preceding the target) affected the size of the resulting shifts. Of particular interest, the LTAS has an affect above and beyond the spectrum nearest to the target and the carriers perceived as speech resulted in larger shifts even when LTAS was held constant. [Work Supported by NIH-NIDCD].
Efference copy and context effects.
Authors: Scott Mark , Gick Bryan
Published: July 2011

One hypothesized component of speech production is “efference copy”; a signal carrying the predicted sensory-consequences of the motor-system’s actions [Wolpert and Ghahramani (2000)]. While brain-imaging studies [e.g., Aliu et al. (2009); Numminen et al. (1999)] have shown dampened auditory-cortex response to self-generated sounds (the predicted effect of efference copy), there are few behavioral demonstrations. This experiment will examine whether a context-effect (a common behavioral measure) is also dampened by efference copy. A context effect is a shift in categorization caused by surrounding sounds. For example, a syllable ambiguous between /da/ and /ga/ is perceived as more /da/-like when preceded by /ar/, but more /ga/-like when preceded by /al/ [Mann (1980)]. In this experiment, participants will silently mouth /ar/ or /al/ in time to a recording (it is assumed that mouthing engages efference copy). In one condition the recording will match what participants are mouthing; in another condition it will mismatch; in a third condition participants will hear the sounds without mouthing. After each mouthing, they will categorize a target syllable as /da/ or /ga/. The prediction is that the context effect will be dampened in the matching condition (due to efference copy), but not in the mismatching condition.
Effects of naturally-occurring segmental ambiguity on cross-modal priming of minimal pairs.
Authors: Clopper Cynthia G.
Published: July 2011

Facilitation is observed in cross-modal lexical decision tasks when the auditory prime and visual target are the same word relative to when the prime and target are unrelated words. In addition, segmentally ambiguous primes can facilitate lexical decision for both members of a minimal pair. The goal of the current study was to explore the effects of naturally-occurring ambiguity on cross-modal priming in a lexical decision task. The targets consisted of minimal pair words containing the vowels /h, \æ/. The auditory primes were either the target word, its minimal pair, or an unrelated word, produced by talkers from the Northern and Midland dialects of American English. The results confirmed facilitation when the prime and target matched relative to when they were unrelated. In addition, inhibition was observed when the prime was the minimal pair of the target relative to when they were unrelated, but only for the Midland dialect. In the Northern dialect, these vowels are undergoing a change that makes them more perceptually confusable. Thus, these findings suggest that the less ambiguous Midland primes led to competition between the minimal pairs, whereas the more ambiguous Northern primes neither facilitated nor inhibited lexical access of the target word.
Cue weighting for speech categorization changes based on regularities in short-term speech input.
Authors: Liu Ran , Soh Howard , Holt Lori L.
Published: July 2011

The ability to flexibly adapt long-term speech category representations to informative regularities in short-term input is critical for on-line speech perception. The present experiment investigates how short-term changes in the variability of two distinct acoustic cues affect the relative weighting of the cues for speech categorization. Native English adults distinguish the vowel categories /ae/ and /E/ using both spectral and duration cues. A spectral continuum from /ae/ to /E/ was crossed with a duration continuum between the same vowels to synthesize a 2-D grid of words ranging between “set” and “sat.” Baseline categorization data were collected from trials sampling the full grid of words; for each trial, listeners selected whether they heard set or sat. Then, they received short-term exposure to trials drawn only from subsets of stimuli for which one cue was held constant while the other cue exhibited the full range of variability. Results reveal that listeners shift their relative cue weights, compared to baseline, to rely more on the highly variable cue for categorization. This suggests that listeners track the variability present across multiple acoustic cues and dynamically adjust speech categorization to reflect this short-term statistical regularity. [Work supported by NIH and NSF.]
Perceptual and production training of Spanish intervocalic /r, d/ and tap.
Authors: Herd Wendy
Published: July 2011

This study investigates the effectiveness of three different high variability training paradigms in training 42 speakers of American English to correctly perceive and produce Spanish intervocalic /r, d/ and tap. English speakers have difficulty producing /r/, because it is a novel phoneme. Although the tap exists as an allophone in American English, learners have difficulty recategorizing it as a phoneme. Similarly, learners experience difficulty acquiring the spirantization of /d/. Since Spanish spirantization and English flapping both affect /d/ intervocalically, the acquisition of the /d/-tap contrast proves difficult. Past research reported that high-variability perceptual training improves both perception and production and that production training improves both as well. However, trainees were able to listen to stimuli during production training, making it unclear whether production training transfers to perception. This study systematically controls both training modalities so they can be directly compared and introduces another training methodology that includes both perception and production. All three training paradigms improved English learners’ perception or production. While production trainees did not improve in their overall perception and declined in their perception of one contrast, perception trainees improved in their production and overall perception, indicating that perception transfers more effectively than production. [Research supported by the NSF.]
Auditory training: Stimulus exposure, task execution, and response feedback affect the neural detection of sound.
Authors: Tremblay Kelly , McClannahan Katrina
Published: July 2011

The P1-N1-P2 cortical auditory evoked potential (AEP) has been used to study the effects of training-related plasticity in the human central auditory system. One consistent finding is that enhanced P2 amplitudes coincide with improved perception. A typical interpretation of this result is that auditory training alters the physiological representation of the cue being trained. However, it is also possible that changes in P2 amplitude reflect other processes that are associated with, but not specific to, the improved representation of the trained cue itself. Here we isolated different components of auditory training by examining the effects of stimulus exposure, behavioral task execution, as well as task- plus-feedback on the AEPs of 30 normal-hearing young adults. For each group, AEP responses were passively obtained in response to two speech syllables differing in voice-onset-time, on three separate days. The first two sessions were on consecutive days and the third session occurred 1 week later. Results suggest that mere stimulus exposure alters the way sound is represented in the human auditory system, but the magnitude of P2 amplitude change is significantly greater when participants interact with the stimuli while executing a task. This point is especially true with the addition of feedback. [Work supported by NIH NIDCD Grant No. R01DC007705.]
Learning acoustically complex word-like units within a video-game training paradigm.
Authors: Lim Sung-joo , Holt Lori L. , Lacerda Francisco
Published: July 2011

Over the course of language development, infants learn native speech categories and word boundaries from speech input. Although speech category learning and word segmentation learning occur in parallel, most investigations have focused on one, assuming somewhat mature develop of the other. To investigate the extent to which listeners can simultaneously solve the categorization and segmentation learning challenges, we created an artificial, non-linguistic stimulus space that modeled the acoustic complexities of natural speech by recording a single talker’s multiple utterances of a set of sentences containing four keywords. There was acoustic variability across utterances, presenting a categorization challenge. The keywords were embedded in continuous speech, presenting a segmentation challenge. Sentences were spectrally rotated, rendering them wholly unintelligible, and presented within a video-game training paradigm that does not rely upon explicit feedback and yet is effective in training non-speech and non-native speech categorization [Wade & Holt (2005); Lim & Holt (submitted)]. With just 2 h of play, adult listeners could reliably extract word-length sound categories from continuous sound streams and generalized learning to novel tokens. The amount of “sentence” variability within training did not influence learning. [Research supported by NIH, NSF, and Riksbanken].
Effects of low-pass filtering and accent in a single-talker interference task.
Authors: Krause Miriam O. , Nelson Peggy B. , Kennedy Mary R. T.
Published: July 2011

Non-native-accented speech can be less intelligible than native-accented speech [Lang Learning 49, 285–310]. Furthermore, low-pass filtering can reproduce the reduced audibility of high-frequency hearing loss [JSLHR 21, 5–36]. The current study combines these two lines of inquiry using a sentence-repetition task in which monolingual listeners repeated sentences spoken by native- and non-native-accented speakers of English. The dependent measure was percent of keywords repeated accurately, and the study used a crossed design. One group repeated unfiltered speech and the other repeated speech low-pass filtered at 1400 Hz. For each group, listeners repeated sentences from each talker separately in the “alone” condition, while in the “attend” condition they heard both speakers simultaneously and repeated only one at a time. As expected, the filtered group performed significantly less accurate than the unfiltered group for both the alone and attending conditions. However, while the control group was able to maintain performance in the attending condition regardless of accent, listeners to filtered speech were roughly half as accurate for the non-native-accented speaker compared to the native-accented speaker in both conditions. Results suggest that peripheral hearing effects overwhelmed attentional effects in determining ability to repeat sentences accurately for these listeners. [Research supported by University of Minnesota Graduate School.]
Divergent patterns of voicing perception in various challenging listening conditions.
Authors: Winn Matthew B. , Chatterjee Monita
Published: July 2011

Perception of consonant voicing is especially robust, even under conditions of harsh spectral degradation, hearing loss, and background noise. Considering the limitations and variability of the acoustic input, it is unlikely that the same perceptual strategy is used in all situations. A sequence of experiments was designed to assess the weighting of multiple acoustic cues to the voicing contrast word-initially and word-finally, and to reveal patterns unavailable in conventional confusion matrices and information transfer analysis. In preliminary results, the combination of background noise and low-pass filtering brought about a dramatic shift in weighting of the VOT and F0 cues for initial consonants; these degradations in isolation did not produce such a shift. Under conditions of spectral degradation (noise-band vocoding), listeners significantly increased their use of vowel duration, and significantly decreased their use of vowel formant transition, which is dominant in the optimal condition. These results suggest that in challenging conditions, listeners are apt to use secondary acoustic cues that are potentially less reliable or less perceptually salient than primary cues used in optimal conditions. It is concluded that comparison of performances by listeners in various conditions should be guarded, even when recognition scores appear to be equivalent on the surface.
Speech intelligibility in noise for adult and children users of cochlear implants.
Authors: Smiljanic Rajka , Sladen Douglas P.
Published: July 2011

Even though high levels of speech understanding in quiet can be developed postimplantation, speech recognition in noise is an especially challenging task for cochlear implant (CI) users. This study examined whether adult and children CI users benefit from signal clarity and contextual-semantic information to improve their speech recognition in noise. Ten adult and ten children CI users, and ten adults and ten children with normal hearing participated in sentence-in-noise listening tests. They heard sentences in which final word varied in predictability, i.e., high versus low semantic context, produced by one female and one male talker in conversational or clear speaking styles. The goal was to assess the effect of age and the length of experience with using cochlear implants on the ability to utilize the compensatory information at higher levels of speech processing independently and in combination with the acoustic-phonetic enhancements in speech perception. This research will allow us to explore the interaction between lower-level sensory and higher-level cognitive factors that affect speech processing in noise for adults and children with cochlear implants. The results of this study add to our current understanding of the development of speech recognition for individuals with cochlear implants and with normal hearing.
Talker variability in lexical access: Evidence from semantic priming.
Authors: Zhang Yu , Lee Chao-Yang
Published: July 2011

It is usually assumed that lexical representations are abstract and devoid of detailed information about various sources of acoustic variability. This assumption is evaluated in this study by examining the effect of talker variability on the access to word meaning in a short-term semantic priming experiment. Prime-target pairs that were semantically related (e.g., king-queen) or unrelated (e.g., bell-queen) were produced by the same talker or different talkers. Two interstimulus intervals (50 and 250 ms) were used between the prime and target to explore the time course of semantic priming. The auditory stimuli were presented to 60 listeners, whose task was to judge whether the target was a word or nonword item in English. It was hypothesized that a change in talker between the prime and target would influence the magnitude of semantic facilitation. Analysis of response accuracy and reaction time showed that the magnitude of semantic priming was attenuated in the different-talker condition, although the effect was obtained only for targets produced by the female speaker. There were no effects involving different interstimulus intervals. These results provided partial evidence that talker variability is encoded in lexical representations and affects lexical access.
Application of low-frequency methods for estimating object size.
Authors: McLaughlin Jack , Hamschin Brandon , Okopal Greg
Published: July 2011

Classification of submerged objects has traditionally been performed using high frequency sonars and imaging techniques. While this permits fine matching of target templates to images acquired in the field, HF methods are necessarily limited in range due to absorption of sound by the water. LF sonars, while offering increased detection range, come with some significant challenges related to the limited bandwidth available. Nonetheless, we show that it is feasible to estimate object size using nonimaging techniques. There are a number of low-frequency phenomena that can be exploited to this end. Among these are edge diffraction in which sharply angled facets of objects (“edges”) act like independent, radiating point sources, and helical waves, which can be set up in cylindrical objects. We show that with appropriate postprocessing of these returns, object edges can be localized thus allowing object extent to be assessed. In this paper, we describe our processing system, and then give results when this system is applied to over 40 sequences of returns from a rail system. In each sequence, a single solid, proud cylinder is insonified, and our system reports an estimate of cylinder length and radius. Histograms of these estimates cluster roughly around the true values.
Direction of arrival estimation of near-field sources using sparse representation framework.
Authors: Zhao Yinghui , Azimi-Sadjadi Mahmood R. , Dinstel Amanda
Published: July 2011

This paper uses the sparse representation framework to investigate localization of near-field sources (e.g., underwater bottom or buried targets) from the data captured using two uniform linear sensor (hydrophone) subarrays. The connection between the two array steering transformation matrices of the near-field sources corresponding to the two subarrays is first analyzed using the second Taylor expansion. This connection allows the construction of a new equivalent far-field steering matrix for each near-field source, hence converting the near-field source localization problem to a more convenient far-field one. The relationship between the signals observed by the two subarrays and the new constructed far-field directional matrix is investigated indicating that the DoA estimates of the sources can be cast into finding a solution to a sparse representation problem. The resolution of the sparse equation will also be discussed. Finally, simulation results are presented to demonstrate the effects of different noise levels on the accuracy of the DoA estimation for the cases of single and multiple sources.
Comparison of time-frequency distributions for target classification.
Authors: Arrieta Rodolfo , Holtzapple Richard , Lim Raymond
Published: July 2011

Time frequency representations characterize signals in the time-frequency plane and can aid in the physical interpretation of backscattered chirps. However, these 2-D representations force trade-offs in resolution, computational efficiency, noise reduction, and cross-term generation. We are mainly concerned with the ability of the transform to reject noise, but we are also concerned with the effect of the cross-term artifacts that do not represent real phenomena on the ability to classify targets using machine learning approaches. First, strong backscatter returns available for a particular target were identified. In the cluttered field under study, this was accomplished from a synthetic aperture image of the field. The processing reported herein used these returns from each studied target to produce a partition of the time frequency plane that contained disjoint sets of time-frequency regions. Each target was represented by a subset of these regions. To test the separability of the targets based on these subsets, random pings were selected from each target’s high backscatter region. These pings were then compared against the disjoint sets. This procedure was carried out with several different time-frequency methods in order to determine the importance of the method’s resolution, noise reduction quality, and cross-term artifacts, in producing robust time-frequency plane partitions that may be used to classify targets. [This work was sponsored by the SERDP.]
Bistatic detection and imaging of proud targets in shallow water using coherent space-time-frequency processing.
Authors: Anderson Shaun D. , Sabra Karim G.
Published: July 2011

An ongoing challenge for underwater sonar systems is to discriminate a man made target (typically having an elastic shell) from surrounding clutter returns especially in the presence of multipath. Network of autonomous systems of unmanned vehicles provides a practical means for bistatic measurements and thus potentially bistatic enhancement for target detection. One popular technique for automatic target detection is synthetic aperture imaging. An aspect that complicates this technique is the evolutional, time dependent aspect of the bistatic echo spectrum from elastic shells especially for the acoustic guided waves echo components at low to midfrequency. To this end, time-frequency analysis, and, in particular, Wigner-Ville analysis have been shown to provide a robust processing tool for interpreting the evolutional time dependent aspect of the scattered acoustic wave field especially around the midfrequency enhancement. Time-frequency methods will be used to enhance the sonar images created from monostatic or bistatic low-frequency measurements. The proposed approach will be tested using numerical simulations and experimental at-sea data. [Work supported by the ONR.]
Sound scattering by a small sphere.
Authors: Godin Oleg A.
Published: July 2011

Acoustic Green’s functions for a homogeneous medium with an embedded spherical obstacle arise in analyzes of scattering by objects on or near the seafloor, radiation by finite sources, sound attenuation in and scattering from clouds of suspended particles, etc. An exact solution of the problem of diffraction of a spherical sound wave on a sphere was obtained by Lord Rayleigh in 1872 and is given by an infinite series involving products of Bessel functions and Legendre polynomials. In this paper, a simple, closed-form solution is obtained for scattering by a sphere with a radius small compared to the wavelength. The solution is valid for arbitrary positions of the source and receiver relative to the scatterer. Low-frequency scattering is shown to be rather sensitive to boundary conditions on the surface of the obstacle. Relation is discussed of the asymptotic low-frequency solution to the exact solution due to Lord Rayleigh and to Lord Kelvin’s exact solution (1845) of a corresponding problem in electrostatics.
Investigation of the environmental factors affecting a horizontal underwater acoustic communication channel.
Authors: Pusey Grant M. , Duncan Alec J.
Published: July 2011

This study sought to characterize the long-term performance of a horizontal underwater acoustic communication system using experimentation and channel simulation. A long-term investigation was carried out off the coast of Western Australia in a depth of 100 m, simultaneously collecting environmental and modem communication performance data for over 16 days. Additionally, an underwater acoustic communication simulator based on the Bellhop propagation model was developed to provide performance predictions for the deployed modems. The signal-to-noise ratio was measured to be relatively high for the entire duration of the trial which helped classify environmental influences as being related to other parameters. Using multiple linear regression, it was determined that the sound speed profile, sea surface roughness, and signal-to-noise ratio contributed to the increased performance of underwater acoustic communication. This result was confirmed by implementing a simulation over the area of interest in a modeled environment and by updating the environmental parameters for each time step. By progressively adding more information to the simulator including ambient noise, wave height, and the sound speed profile, simulations provided effective predictions of the performance measured during the trials.
A fast algorithm for computing doppler introduced by sea surface gravity waves.
Authors: Peterson John C.
Published: July 2011

The technical advances in phase coherent underwater acoustic communications and the increasing availability of relatively inexpensive commercial modems have created a renewed interest in utilizing sound channel models for predicting modem performance. While accurate channel models already exist, their usefulness for modeling extensive “what-if” scenarios is limited by the computational resources required. Acoustic field models based on raytracing methods are a fast and efficient approach to sound channel modeling. They readily lend themselves to estimating the impulse response or time arrival structure of a hypothetical sound channel. The computed impulse response can then be convolved with the waveform transmitted by the modem to obtain an estimate of the timeseries observed at a hypothetical receiver. The existing models for addressing the Doppler effects introduced by sea surface gravity waves into the receiver timeseries are accurate, but relatively resource hungry. A fast algorithm for modeling Doppler based on simple postprocessing of the eigenrays computed by the raytracing based field model will be presented.
Simplified formulations for sea-surface scattering for use in modeling equalizer performance in underwater communications.
Authors: Rouseff Daniel , Jackson Darrell R.
Published: July 2011

Reflection off the rough sea surface typically introduces time spread in an acoustic signal. The details of channel response and hence the time spread will change as the sea surface evolves. Communications signals that are spread by the rough surface may still be recompressed and demodulated successfully by an equalizer providing that the channel response does not change too rapidly. To aid in designing an equalizer, it would be useful to know which acoustic paths should be treated as useful signal and which must be treated as noise because they change too rapidly. Viewed as a rough surface scattering problem, the classic Kirchhoff approximation should be appropriate for modeling reflected communications signals. Textbook descriptions of the Kirchhoff approximation are not promising, however, as they imply that the calculations depend on the details of the surface wave spectrum. In the present work, simplified expressions for the surface reflected communications signals are derived. The simplified results are tested in two ways: Predictions for the mutual coherence function are compared to numerically intense calculations, and predictions for communications performance are compared to experimental results. [Work supported by the ONR.]
Subspace dimension estimation for equalization of underwater acoustic channels.
Authors: Blair Ballard J. , Preisig James C.
Published: July 2011

The use of an array of receiving elements has several advantages such as increased signal to noise ratio due to array gain and the ability form beams, which reject interference and enhance a source. One common way to integrate an array receiver into a communication system is through the use of a multichannel decision feedback equalizer (DFE), which adaptively combines the signals from the receive elements and mitigates channel effects. However, the computational complexity of a multichannel DFE is too high for implementation into real-time systems. One solution is to use a beamformer to reduce the number of channels the DFE needs to combine and thus reducing the overall complexity. While there has been some research showing the optimal number of beams if the channel is known, it is still an open question how to determine this information from a received signal. This research examines approaches for determining the number of beams to use so that the performance of a reduced complexity beamformer-DFE system is at least as good as a system, which uses multichannel DFE with all of the sensors as inputs. Results will be presented from experimental data comparing the performance of the systems in different ocean environments.
Adaptive interference suppression and cancellation for underwater acoustic communications.
Authors: Cho Steve E. , Song Heechun , Hodgkiss William S.
Published: July 2011

An adaptive, time-reversal multichannel combiner is embedded within an iterative, successive interference cancellation framework to create a reduced-complexity, multiuser receiver with multiple-access interference suppression. The combined receiver achieves the goal of separating simultaneous, full-band transmissions from physically separated users transmitting to a base station in an underwater uplink setting. Although both receiver components could feasibly act alone, the combined receiver is shown to be more robust than its individuals, achieving both temporal and spatial interference mitigations in the presence of strong multiuser interference. Furthermore, with the addition of matching pursuit, a popular sparse channel estimation technique, and a block-by-block channel adaptation, the overall receiver is designed to be applicable to time-varying underwater acoustic environments. Analysis of experimental data collected during FAF-06 illustrates that for a two-user multiple-access system, multiuser separation can be achieved even in low signal-to-noise ratio (SNR) environments.
Sparse acoustic response function estimation via Gaussian-mixture model with application to M-ary spread spectrum communications.
Authors: Gendron Paul J.
Published: July 2011

An important feature of underwater acoustic response functions is their sparsity in delay-Doppler. Models of the response must have the complexity to exploit this sparsity and the flexibility to adapt to the diverse environmental conditions that impart it. The Gaussian-mixture assignment over delay-Doppler slots can provide such a framework and leads to adaptive shrinkage operators that effectively attenuate small noise-like delay-Doppler paths and leaves larger occupied paths unaffected [Gendron, IEEE Trans. SP 53 2005]. Accurate treatment of the latent parameters of the model aids performance across environments and extends its usefulness in underwater acoustic applications. This model is applied to M-ary orthogonal spread spectrum signaling where necessary SNR/bit at low received SNR is obtained with array, processing, and coding gain. The approach allows for bulk path dilation estimation and compensation that extends the effective channel coherence duration to that associated with the natural Doppler variations between the various acoustic paths. Symbol decisions are iterative with channel estimates and bulk time varying dilation compensation. These receiver structures are tested on very shallow water acoustic transmissions in Buzzard’s Bay MA. Probability of bit error is reported under 10−5 with two elements combining at −16 dB received SNR.
Underwater image transmission performance in multi-path flat fading channel.
Authors: Yoon Jong R. , Kim Jongwook
Published: July 2011

Time-variant sea surface scattering, Doppler spreading due to source/receiver motion, and grazing angle dependent bottom reflection in the multi-path underwater acoustic channel cause the signal fading in amplitude and phase fluctuation. Consequently, the performance of underwater acoustic communication systems is degraded, and high-speed digital communication is disrupted. In this study, two different multi-path channels such as a water tank and a river are considered and the experiments are conducted under flat fading condition. Rayleigh and Rice fading channel models are adopted and their distributions are compared with the measured in experiments. Finally, the standard Lenna image using binary frequency shift keying modulation is transmitted and the quality of the received image is analyzed based on the channel models. It is shown that the quality of the received image and the channel model is consistent with each other.
Communication performance analysis according to seasons in Yellow Sea.
Authors: Kim Juho , Bok Tae-Hoon , Lee Chong-Hyuen , Paeng Dong-Guk , Bae Jin-Ho , Kim Seongil
Published: July 2011

Communication environments vary with time in the context of underwater channel because multipaths are affected by the sound speed profile (SSP) in the ocean. In this paper, the average SSPs for 10 years (2000–2009) were calculated in the Yellow Sea so that the eigen-ray paths with the channel impulse responses were determined. The computer simulation was conducted to compare the performance of underwater communication near the surface according to the variation of season. The performance of underwater communication was analyzed using the BPSK modulation and time reversal method. The significant differences of bit error rate (BER) were shown according to the change of season. In many cases, the received signals were distorted by the multi-path channel whose features are frequency selected fading. The time reversal method was applied to compare the BER without frequency selected effect. It turns out that the good performance was shown in the autumn because of the direct path through the mixed layer channel which made the multi-path effects relatively weaker. [Work supported by UD100014DD.]
A new signal processing method for multipath compensation in shallow underwater acoustic communication.
Authors: Gao Da Zhi , Wang Ning , Wang Hao Zhong
Published: July 2011

The most difficult problem in shallow underwater acoustic communications is considered to be the the time spread inherent to acoustic propagation in guide. The standard approach is to design an equalizer that attempts to correct for these propagation effects. In the presentation, a novel approach to multipath compensation in shallow water acoustic communication is proposed. The method is based on a dedispersion transformation technique, which was proposed in our previous paper [N. Wang, 9th Western Pacific Acoustic Conference, Beijing (2009)]. After dedispersion transform to a long range acoustic propagation signal, the field sampling data of a single hydrophone are transformed to several normal modes with the same waveforms. The dispersion of each modes, especially higher modes, can be removed after the transform. Then the computation cost of equalization can be decreased dramatically. The proposed approach requires less prior environmental information except an approximate beta-value of waveguide invariant. The validity of the present approach is verified in simulation.
An analysis on fractional Fourier transform in Doppler and multipath mobile underwater acoustic channel.
Authors: Yin Jing Wei , Yang Sen , Zhang Xiao
Published: July 2011

Fractional Fourier transform is a kind of time-frequency analysis method, which combines the information of signals in time domain and frequency domain. FRFT has chirp-based decomposition characteristics, for a given chirp signal, there is a fraction which gathers the energy of linear frequency-modulated signals to a maximum, which provides a basis for the detection of chirp signal with Doppler frequency offset and can also be used in the synchronization detection of mobile communication in underwater acoustic and the channel parameter estimation. This paper describes the theoretical basis of FRFT and its characteristics analysis, then by the computer simulation, performance analysis of FRFT and correlation detection in the conditions of Doppler and multipath are worked out, and the facts are verified that fractional Fourier transform is more suitable than copy correlator for Doppler frequency offset and multipath underwater acoustic channel.
Room acoustics simulation of hospital patient care-unit.
Authors: Utami Sentagi S. , Navvab Mojtaba
Published: July 2011

Acoustical design challenges in hospitals are providing better communication to reduce medical errors and speech privacy that has also become a legal issue according to the Health Information Portability and Accountability Act (HIPAA). Some industrial products performance criteria are being exaggerated in the claims as solutions for better room acoustics of healthcare spaces. This study demonstrates an acoustical quality improvement of a patient care-unit environment. Building regulation codes, health service activities, space openness, and confined space layout are used as parametric studies for effective room acoustics design solutions. Current techniques in computer simulation are utilized along with field measurements to localize sources and identify critical elements that create the most impact. Acoustical treatments with characteristics adopting the industrial products are applied within the architectural elements including adjustable hung curtains, furniture, and art displays on wall surfaces. The improvement results are subjectively evaluated within VR environment of selected scenes to obtain real time spatial experience given the auralization capabilities. These results show the detailed comparison of each design variable with respect to their ability to fulfill the speech communication requirements. The study shows the limits and possibilities for acoustic design improvement of patient care-units at various stages of design.
Application of constrained-layer damping to improve speech privacy in healthcare facilities.
Authors: Shafer Benjamin M. , Tinianov Brandon
Published: July 2011

Contemporary wall assemblies for demising partitions in commercial buildings, such as hospitals, are commonly designed using heavy-gauge (versus light-gauge) steel framing components and gypsum drywall. Recent studies indicate that these types of partitions do not provide adequate speech privacy between rooms [A. Bétit, Sound & Vibration, 14–16 (2010)]. Former testing has proven that constrained-layer damping technology may be used to achieve much higher transmission loss values in the speech range than gypsum drywall. Measurements were made to explore the use of constrained-layer damping technology as a single-layer solution on heavy-gauge steel framing versus multiple layers of gypsum drywall, specifically for speech privacy. Theoretical results are also presented and discussed.
Tools to predict binaural speech intelligibility in complex listening environments for normal and hearing-impaired listeners.
Authors: Kollmeier Birger , Rennies Jan , Brand Thomas
Published: July 2011

Appropriate models for speech intelligibility in acoustically “difficult” situations (i.e., background noise from several interfering sound sources and reverberation) are an important tool not only for research, but also to characterize the intelligibility achieved in special-purpose rooms such as healthcare spaces. This contribution presents an overview on existing and new models that have mostly been implemented in the “speech intelligibility prediction toolbox” in order to allow for a comparison of model performance and to assess the expected speech intelligibility for a predefined situation as a function of individual hearing loss. The extention of a binaural speech intelligibility model with room acoustical parameters is introduced, which quantitatively predicts the combined influence of reverberation and different positions of target speaker and interferers both in quiet and in noise. The accuracy in prediction for individuals can be increased by including the respective individual audiogram. Taken together, a comparatively exact prediction of speech intelligibility can be done in comparatively “easy” acoustical situations for listeners with a mild-to-moderate hearing loss. The current models still show room for improvement as soon as more complex listening situations, hearing loss types, and more detailed properties of the speech reception process are considered.
On the acoustics of operating rooms.
Authors: Harrison Seth M. , Sacco Nicholas R.
Published: July 2011

Designing modern operating rooms to meet al.l design criteria can be a challenge. Creating a sterile, well-lit, temperature and humidity appropriate environment for medical operations typically takes precedence over the acoustics of the operating room. The sterile, washable operating room surfaces are commonly hard and reflective, offering little sound absorption. These surface finishes result in strong discrete reflections, increased background noise, and reduced speech intelligibility. Code mandated energy efficiency and ventilation requirements coupled with the owner-driven desire for economical mechanical systems can result in excessive HVAC noise, adding to background noise levels. Yet, speech intelligibility and speech privacy levels are expected to be superb to minimize errors due to miscommunication and to prevent the unintentional spread of confidential patient information. The acoustical challenge is to design operating rooms to control noise and to enhance room acoustics without affecting the performance of other systems. Speech intelligibility and impulse response measurements from modern hospital operating rooms and design strategies will be discussed.
Methods of flanking path control in party walls.
Authors: LoVerde John , Dong Wayland
Published: July 2011

It is generally appreciated in the acoustical community that flanking paths can control the overall noise isolation, especially for high-STC wall designs [Nightingale, et al., NRC-IRC, Flanking Transmission in Framed Buildings, Phases I, II, and III (1997–2010)]. A recent condominium retrofit into an existing structure produced complaints of very low privacy between some units, in spite of a double-stud wall design that would typically provide acceptable isolation. Airborne noise reduction testing revealed a variety of unique flanking paths due to the constraints of the existing structure and the intersection of disparate materials (the construction includes wood, concrete, steel, and masonry components). A number of methods were implemented, and additional testing performed to quantify the flanking paths and the effectiveness of the mitigation. The methods and results are presented.