bluuebunny/crossref_metadata_2025 huggingface dataset:
Guided wave approach for inline photovoltaic module component inspection
Authors: Meier Rico
Published: September 2012

Reliability of photovoltaic modules is one key factor for being financially attractive for customers all over the world. The further reduction in manufacturing costs lead to increased demands on module components and their materials to maintain acceptable mechanical yields and module reliability. Thus fast, economic and preferably non-destructive component characterization and manufacturing process control methods come more and more into focus. In the present work lamb wave based approaches for material characterization of plate-like photovoltaic module components were evaluated with respect to their precision and inline capabilities. The second-order elastic constants (Young’s modulus and Poisson’s ratio) were determined by automated numeric fitting of the Rayleigh-Lamb dispersion model on experimental data. Furthermore, the influence of mechanical stress on the ultrasound transport properties was investigated. Therefore the components were systematically loaded by mechanical and thermal stress. The resulting changes in ultrasound transport were correlated with results from mechanical testing. Lamb wave approaches turned out to be well suited methods for inline material characterization of plate-like module components. The elastic constants can be determined with high accuracy. The usage of plate-like structures in photovoltaic modules makes lamb waves an important tool for inline and non-destructive material characterization.
Context matters: Analyzing the necessity of context-dependent speech perception in complex models
Authors: Apfelbaum Keith S. , McMurray Bob
Published: September 2012

Speech perception faces variability in the realization of phonemes from vocal tract differences, coarticulation, and speaking rate. While the mechanisms accommodating this variability have long been debated, recent computational work suggests that listeners succeed by processing input relative to expectations about how context affects acoustics. McMurray and Jongman (2011) measured 24 cues for fricatives in 2880 tokens. When effects of talker voice and vowel context were removed with a relative encoding scheme, simple cue-integration/prototype models identified the fricatives at near-human levels of performance. These models were less successful with raw cue values, suggesting some form of compensation is required. However, contingent encoding may not be needed in all categorization architectures. Exemplar models account for context by holistically comparing incoming tokens to others in memory, and connectionist backpropagation models encode stimulus-specific information through their hidden units for better performance. We conducted several simulations using these classes of models. After training the models to identify fricatives from the McMurray and Jongman corpus, we assessed model performance on held-out tokens for which human categorization data was available. Both classes of models more closely matched human performance when trained on context-adjusted information than raw cues. This offers firmer support for context-dependent speech perception.
Linguistic and social effects on perceptions of voice onset time in Korean stops
Authors: Podesva Robert , D'Onofrio Annette , Acton Eric , Bowman Sam , Calder Jeremy , Chen Hsin-Chang , Lokshin Benjamin , Van Hofwegen Janneke
Published: September 2012

This paper investigates effects of linguistic and social factors on phoneme categorization of Seoul Korean stops. In an investigation of VOT in aspirated versus lenis stops of Korean, Oh (2011) reports VOT length in aspirated stops to be conditioned both linguistically and socially: bilabial stops exhibit shorter VOT than velars, following /a/ conditions shorter VOT than /i/, and female speakers exhibit shorter VOT than males. 10 native speakers of Seoul Korean (5 men, 5 women) were recorded producing bilabial and velar stops in the frame /CVn/. Recordings were manipulated to create a 10-step continuum of VOT length for each speaker, from 25ms to 115ms. 30 native speakers of Seoul Korean listened to each of these manipulated stimuli for every speaker and categorized them as containing either aspirated or lenis stops. Listeners were more likely to categorize a given VOT as aspirated when it occurred in a bilabial stop as opposed to a velar stop, when it preceded /a/ as opposed to /i/, and when it was produced by a female as opposed to a male. Results indicate that speakers exhibit knowledge of production patterns when categorizing phonemes, drawing upon both linguistic and social information.
Speakers of tonal and non-tonal Korean dialects use different cue weightings in the perception of the three-way laryngeal stop contrast
Authors: Lee Hyunjung , Jongman Allard , Politzer-Ahles Stephen
Published: September 2012

The current study investigates if and how listeners from tonal and non-tonal varieties of Korean (Kyungsang and Seoul) use different cue weighting in their perception of the three-way distinction among Korean voiceless stops (fortis, lenis, and aspirated). Forty-two Korean listeners (21 each from Seoul and Kyungsang) were tested with stimuli in which VOT and F0 were systematically manipulated. Analyses of the perceptual identification functions show that VOT and F0 cues trade off each other for the perception of the three stops. However, the trading relationship differs between the two dialects. Logistic regression analysis reported interactions among VOT, F0 and Dialect factors for perceiving one stop over the others, indicating the two dialects use the perceptual cues differently. Specifically, while Seoul listeners rely primarily on F0 for making lenis responses and on VOT and F0 for aspirated responses, F0 plays a less important role in modulating both lenis and aspirated responses for Kyungsang speakers. A similar pattern has been observed in production [Lee and Jongman 2012, JIPA 42 (2)]. The results will be discussed in terms of the close link between perception and production across the two different dialects.
The waiting is the hardest part: How asynchronous acoustic cues are integrated for fricative voicing and place of articulation
Authors: Galle Marcus E. , McMurray Bob
Published: September 2012

A fundamental issue in speech perception is the fact that information is spread over time. This raises the question of how listeners integrate acoustic information in real-time. One possibility is that cues are utilized as soon as they arrive to partially activate lexical candidates. Alternatively, input could be a buffered, until sufficient information is available to make a decision. Between these extremes, listeners may vary depending on the usefulness of a given cue, and whether it directly cues a phonetic contrast, or serves as context (e.g., talker identity) for interpreting other cues. We examined this in word-final and initial fricative contrasts using the visual world paradigm. Participants selected a visual referent of an auditory word (ship), and we used the likelihood of fixating lexical competitors (sip/ship) at each point in time to determine when various factors affected higher level decision making. Several studies contrasted the order and utility of the cues (strong vs. weak in different positions), and direct cues vs. information like talker used for compensation. In general, listeners appear to use acoustic information immediately as it becomes available in real-time to lexical candidates.
The contribution of high-frequency information to fine-grained speech perception in cochlear-implant-simulated speech
Authors: Farris-Trimble Ashley , McMurray Bob
Published: September 2012

It has been observed that the contribution of information in the higher frequency ranges (>7000 Hz) to speech perception is negligible. This assumption has influenced cochlear implant processing strategies, which typically filter out high-frequency information. However, even if speech recognition is robust without high-frequency information, that information may nevertheless contribute to the actual processing of speech. We asked 15 normal-hearing participants to categorize words along 8-step b/p (e.g., beach to peach) and s/sh (e.g., self to shelf) continua in normal and CI-simulated speech. The CI simulations used a bandpass filter with 8 bands between 200 and 7000 Hz. The envelope for each band was extracted and used to amplify broadband noise within the band’s frequency range. Results showed that participants’ boundaries shifted significantly for both continua; the endpoints of each continuum were categorized consistently, but ambiguous tokens were more often categorized as sh and b when presented in CI-simulated speech. That is, eliminating high-frequency information during simulation also eliminated information necessary to the perception of sounds with high-frequency cues. These results suggest that while the loss of high-frequency information may not affect CI users’ ability to recognize speech, it may compromise their fine-grained discrimination.
Perceptual cues in Korean fricatives
Authors: Lee Goun , Jongman Allard
Published: September 2012

The current study explores the production and perception of two Korean fricatives - fortis [s’] and non-fortis (plain) [s]. Production data from 10 speakers were examined to investigate the acoustic cues that distinguish the two fricatives in different vowel contexts (high vowel /i/ vs. low vowel /a/). Measures included rise time, intensity, center of gravity (COG), F0, H1-H2, and CPP, as well as frication, aspiration, and subsequent vowel duration. COG and vowel duration consistently distinguished the two fricatives; additional cues varied across vowel contexts. For the /i/ context, intensity and F0 differed across the fricatives; for the /a/ context, rise time, H1-H2 and CPP differed across the fricatives. Four perceptual identification experiments were conducted with sixty native Korean listeners. Experiment 1 established that listeners can distinguish the two fricatives in intact natural syllables. In Experiments 2 and 3, listeners heard only the excised consonantal or vocalic segment. For the /a/ context, fricative identification was successful based on both consonantal and vocalic segments. In the /i/ context, fricative identification exceeded chance level only for the consonantal segment. In Experiment 4, cross-spliced stimuli revealed that speakers are more sensitive to vocalic cues than consonantal cues, but only in the /a/ context.
Timing of perception for all English diphones
Authors: Warner Natasha L. , McQueen James , Liu Priscilla Z. , Hoffmann Maureen , Cutler Anne
Published: September 2012

Information in speech does not unfold discretely over time; perceptual cues are gradient and overlapped. However, this varies greatly across segments and environments: listeners cannot identify the affricate in /pt∫/ until the frication, but information about the vowel in /li/ begins early. Unlike most prior studies, which have concentrated on subsets of language sounds, this study tests perception of every English segment in every phonetic environment, sampling perceptual identification at six points in time (13,470 stimuli/listener; 20 listeners). Results show that information about consonants after another segment is most localized for affricates (almost entirely in the release), and most gradual for voiced stops. In comparison to stressed vowels, unstressed vowels have less information spreading to neighboring segments and are less well identified. Indeed, many vowels, especially lax ones, are poorly identified even by the end of the following segment. This may partly reflect listeners' familiarity with English vowels' dialectal variability. Diphthongs and diphthongal tense vowels show the most sudden improvement in identification, similar to affricates among the consonants, suggesting that information about segments defined by acoustic change is highly localized. This large dataset provides insights into speech perception and data for probabilistic modeling of spoken word recognition.
Tone recognition in continuous Mandarin Chinese
Authors: Wu Jiang , Zahorian Stephen A. , Hu Hongbing
Published: September 2012

Tones are important characteristics of Mandarin Chinese for conveying lexical meaning. Thus tone recognition, either explicit or implicit, is required for automatic recognition of Mandarin. Most literature on machine recognition of tones is based on syllables spoken in isolation or even machine-synthesized voices. This is likely due to the difficulty of recognizing tones from syllables extracted from conversational speech, even for native speakers of Mandarin. In this study, human and machine recognition of tones from continuous speech is evaluated and compared for four conditions: 1, vowel portions of syllables; 2, complete syllables; 3, syllable pairs; 4, groupings of three syllables. The syllables are extracted from the RASC-863 continuous Mandarin Chinese database. The human listeners are all native speakers of Mandarin. The automatic recognition is based on either Hidden Markov Models, or neural networks, and a combination of spectral/temporal, energy, and pitch features. When very little context is used (i.e., vowel segments only) the human and machine performance is comparable. However, as the context interval is increased, the human performance is better than the best machine performance.
Lexical tone in Mandarin spoken word processing
Authors: Sereno Joan , Lee Hyunjung
Published: September 2012

Two priming experiments examined the separate contribution of lexical tone and segmental information in the processing of spoken words in Mandarin Chinese. Experiment 1 contrasted four types of prime-target pairs: tone-and-segment overlap (bo1-bo1), segment-only overlap (bo2-bo1), tone-only overlap (zhua1-bo1), and unrelated (han3-bo1) in an auditory lexical decision task with 48 native Mandarin listeners. Experiment 2 further investigated the minimal segmental overlap needed to trigger priming when tonal information is present. Four prime-target conditions were contrasted: tone-and-segment overlap (pa2-pa2), only onset segment overlap (ping2-pa2), only rime overlap (na2-pa2), and unrelated (kui4-pa2) in an auditory lexical decision task with 68 native Mandarin listeners. Results showed significant priming effects, as compared to the unrelated baseline, when both tonal and segmental information matched. Moreover, when only segmental information overlapped, priming was also observed while no priming for the tone-only overlap condition was found. Interestingly, partial segmental overlap, with matching tonal cues, did not produce priming. In fact, tonal overlap with matching onset segmental information resulted in significant inhibition. These data clearly indicate that tonal similarity only provides facilitation if there is complete segmental overlap. These findings will be discussed in terms of the segmental and suprasegmental information used in word recognition.
Cochlea-scaled entropy predicts intelligibility of Mandarin Chinese sentences
Authors: Jiang Yue , Stilp Christian E. , Kluender Keith R.
Published: September 2012

Cochlea-scaled spectral entropy (CSE; Stilp & Kluender, PNAS, 107(27):12387-12392 [2010]) is a measure of information-bearing change in complex acoustic signals such as speech. CSE robustly predicts English sentence intelligibility even amidst temporal distortion and widely different speaking rates (Stilp, Kiefte, Alexander, & Kluender, JASA, 128(4):2112-2126). However, CSE does not explicitly capture changes in fundamental frequency (f0) in any way distinct from that for other aspects of spectral shape (e.g., formant patterns, slope). This property of CSE could limit its ability to predict intelligibility of tone languages that use f0 for phonetic and morphological distinctions. The present study assesses the predictive power of CSE for Mandarin Chinese sentence intelligibility. Twenty-five native-Mandarin listeners transcribed Mandarin sentences in which consonant-length (80-ms) and vowel-length (112-ms) segments with either high or low CSE were replaced with speech-shaped noise. CSE reliably predicted listener performance; as greater amounts of CSE were replaced by noise, performance worsened. Results encourage information-theoretic approaches to speech perception, as change and not physical acoustic measures best predict sentence intelligibility across tonal and nontonal languages. [Work supported by NIDCD.]
The influence of a foreign accent on recall of spoken word lists
Authors: Ganeku Marissa , Bent Tessa
Published: September 2012

Talker and speaking rate variability influence spoken word recall (Nygaard et al., 1995; Goldinger et al., 1991; Martin et al., 1989) suggesting that indexical information affects memory processes. The current study investigated how another type of speech variability - the presence of a foreign accent - influenced spoken word recall. Recall of 10 spoken word lists with 10 words each was assessed at two presentation rates. Listeners received words from a native-accented talker, a non-native talker with a strong foreign accent, or a non-native talker with a mild foreign accent. Neighborhood density, neighborhood frequency, and word frequency were balanced across each list; all words were 100% intelligible (Bent, 2010). Recall was analyzed by: (1) accuracy in the correct serial position and (2) accuracy regardless of serial position. The first analysis demonstrated that listeners recalled more words with the slower presentation rate and in the early and late list positions, replicating earlier findings, but there was no talker effect. The second analysis demonstrated that listeners recalled fewer words from the strongly foreign-accented talker than the native talker or the mildly foreign-accented talker. Thus, foreign-accented speech is another source of variability that can influence processing and encoding in memory.
Adaptive modulation and power control for underwater acoustic communications
Authors: Radosevic Andreja , Rasheed Ahmed Rameez Ahmed , Duman Tolga M. , Proakis John G. , Stojanovic Milica
Published: September 2012

In this work we explore the feasibility of a cognitive acoustic communication system that exploits a dynamic closed loop between the transmitter and receiver with the goal of maximizing the information throughput. Specifically, we design a power control mechanism and couple it with an adaptive modulation method based on orthogonal frequency division multiplexing (OFDM). We propose two methods: the first method is adaptive overall power adjustment in which the transmitter modifies the power gain to provide a target SNR at the receiver by exploiting a feedback link in a time-varying channel. The second method adaptively adjusts the modulation level on each individual sub-carrier to achieve a target bit error rate (BER) at the receiver. Crucial to both of these methods is the ability to predict the acoustic channel and the signal-to-noise ratio (SNR) one travel time ahead, which enables adaptive adjustment of the transmit power and modulation level. The performance of the proposed algorithms is demonstrated using an in-air test bed and further verified with real-time at-sea experiments conducted off the coast of Kauai, HI, in June 2011. Experimental results obtained using real-time at-sea experiments show significant savings in power, as well as improvement in the overall information throughput (bit rate) as compared to conventional, non-adaptive modulation with the same power consumption and target BER.
Speech enhancement via only mostly blind source separation
Authors: Goldhor Richard , Chenausky Karen , Boyce Suzanne , Gilbert Keith , Hamilton Sarah M. , Cin Joseph
Published: September 2012

In environments in which multiple simultaneously-active acoustic sources contribute to sensor responses, Blind Source Separation (BSS) signal processing techniques may be employed to separate (that is, estimate or reconstruct) the signal characteristics of “hidden” sources. Only Mostly Blind Source Separation (OMBSS) involves the estimation of similar sources in important contexts in which non-acoustic information is also available about one or more of the contributing sources. Recently-reported objective source separation performance measures confirm that non-acoustic information can be used effectively to support high-quality separation in situations in which traditional BSS methods perform poorly (e.g., when more sources are active than there are microphones available). Here we present the results of additional perceptual and objective tests showing that OMBSS processing enhances the intelligibility of speech recorded in the presence of multiple simultaneous speech-babble and non-speech maskers.
Use of pattern classification algorithms to interpret acoustic echolocation data from a walking-speed robotic sensor platform
Authors: Dieckman Eric A. , Hinders Mark
Published: September 2012

An unresolved issue for autonomous walking-speed robots in unstructured outdoor environments is maintaining situational awareness. One strategy is combining information from different sensors so the robot can function in a variety of conditions and environments. The very low-cost Microsoft Kinect accessory incorporates active infrared and RGB video sensors to provide real-time depth information, as well as a 4-channel microphone array. We validate the Kinect sensors and investigate the combination of infrared (passive and active) and non-linear acoustic echolocation sensors on our mobile robotic sensor platform. By using an acoustic parametric array to generate the audible echolocation signal, a tightly-controlled beam of low-frequency sound can interrogate targets at long distances, while infrared imaging works well in the nearfield and in difficult weather conditions. Sophisticated signal processing techniques are required to combine and interpret the collected data; we present an example using pattern classification on the acoustic echolocation data to differentiate between vehicle types.
Real-time active noise control of magnetic resonance imaging acoustic noise
Authors: Lin Chiao-Ying , Chen Jyh-Horng
Published: September 2012

Magnetic resonance imaging (MRI) is extensively used in clinical and medical researches because MRI is non-invasive and non-radiation. However, using the MRI cause the loud acoustic noise generated by gradient switching. The MRI noise is annoying patients. Therefore, active noise cancellation (ANC) is used to solve this problem in this article. In this article, the main work is to implement real-time system using DSP. The noise generated by each planar imaging (EPI), which is generally used in fMRI researches, is recorded from 3 Tesla MRI for testing ANC system. The EPI signal is regular and predictable, means the signal like a pattern, we using LMS algorithm to training the signal and learning predict the noise spectrum. After the calculated the noise, we output the reverse single to we home-built headset.
Digital signal processing in forensic acoustics cases
Authors: Yonovitz Al , Joe Herbert , Yonovitz Joshua
Published: September 2012

Forensic acoustics typically considers speaker identification, authenticity analysis, audio enhancement and transcript verification by analyzing and processing audio or speech signals. Other applications in forensic acoustics include different analyses of non-speech audio signals. This presentation will discuss three contentious litigation cases in which various digital processing techniques formed the foundation of scientifically-based conclusions. The first is a products liability case involving the death of a firefighter and his Personal Alert Safety System. When movement of a firefighter ceases, a high-pitched audible alert is emitted. When numbers of these devices were recorded, the task was to determine if the deceased firefighter’s device was included within the multiple devices examined. In the second case the order of the discharges of various caliber weapons based on a 9-1-1 call was determined via digital signal processing. The third study utilized processing techniques to differentiate and identify the type of sounds produced in a number of murder trials. These cases represent the varied challenges in forensic audio and acoustics investigations, and show the difficulty of applying broad theory to practice, the necessity for innovation in the field and the basis for establishing scientific reliability in order to secure admissibility into the courtroom.
Use of Galerkin's method using variable depth grids in the parabolic equation model
Authors: Sanders William
Published: September 2012

Choice of a depth grid affects the accuracy of a parabolic equation (PE) propagation model, as well as the speed of execution. Choice of a fine grid step may result in lower discretization errors, but will lengthen computation times. Moreover, application of a sampling requirement (e.g. N samples per wavelength) to the water column results in oversampling in the bottom. Since an artificial absorbing later in the bottom is often employed, a large part of the computational domain is oversampled. Fine depth sampling is only needed in the region about the water sediment interface. Hence, Galerkin’s method using a variable depth grid is implemented. This is demonstrated to achieve the same error as a uniform grid with fine spacing over the entire depth domain, while taking a fraction of the run time. This is particularly important in models where nested PE models are required, as with noise models (so called N by 2D runs), full 3D, or broadband models. The variable grid Galerkin’s method is also used in an elastic PE model, in which sampling requirements for low-shear speed sediments require much finer sampling than that in the water column.
Modeling low-frequency seismo-acoustic propagation in the Arctic using a parabolic equation solution
Authors: Metzler Adam M. , Collis Jon M. , Siegmann William L.
Published: September 2012

Propagation in Arctic environments is complicated by three-dimensional variations in the waveguide. The sound speed minimum occurs at or near the ice-covered surface, and the upward refracting profile causes long-range propagation to interact repeatedly with the ice cover. Propagation in the Arctic waveguide needs to include an elastic ice cover of variable thickness which may terminate, overlying an ocean layer and an elastic sediment bottom. Parabolic equation solutions are accurate and efficient for elastic layers, although currently, solutions do not exist for Arctic environments. In this paper, elastic parabolic equation solutions will be obtained for range-dependent Arctic waveguides with elastic ice cover and an elastic bottom. Particular interest will be directed to environments where the ice cover terminates. Results will be benchmarked against normal mode and wave number integration solutions.
Special tools and procedures for measuring ground vibration
Authors: Phillips James E.
Published: September 2012

This talk will present some of the tools and procedures used to analyze ground vibration and the propagation of vibration through soils. These tools and procedures are typically used to evaluate and control groundborne noise and vibration associated ground based transportation such as trains.
Advancements in instrumentation for source identification
Authors: Golden Matthew V.
Published: September 2012

In the beginning, acousticians only had their ears to identify sources. Today we have many more advanced instrumentation, even more than the simple intensity probe that we had just a few decades ago. This paper will review several of these new devices. The first will be an Intensity Tracking System that uses machine vision to produce sound intensity maps. The second is an Acoustic Camera that uses 256 microphones in a two dimension array to create real time videos of the sound field overlaid with standard video. The final instrument will be an environmental noise monitor that uses the arrival times at multiple microphones to detect the location of sources in three dimensional space. Real life measurement results will be shown from each instrument. These new tools give acousticians far more tools than they had just a few years ago.
Spatial release from electronic clutter masking in FM bat echolocation
Authors: Warnecke Michaela , Bates Mary E. , Simmons James A.
Published: September 2012

For big brown bats, angular separation of target and clutter echoes causes spatial release from clutter masking. Experiments using echoes that are electronically-generated by loudspeakers show that lowpass-filtering of normally masking echoes also causes clutter masking to disappear. Such lowpass-filtering induces amplitude-latency trading, which retards neural response times from clutter echoes at higher frequencies relative to lower frequencies. Countervailing changes in presentation-times of higher frequencies in electronically-generated clutter echoes restores masking. We present new results showing that moving the clutter-delivering loudspeakers to a different azimuth and elevation causes clutter masking to disappear. But, similar to the earlier experiments, the countervailing changes in presentation of higher frequencies reintroduce masking. In the bat’s inferior colliculus, FM sounds that mimic broadcasts and echoes evoke ~1 spike per sound at each neuron’s best frequency. However, amplitude-tuning is very broad, so bats work in the latency domain instead, to exploit their high acuity for detecting coherence or non-coherence of echo responses. Overall, the results indicate that big brown bats use neuronal response timing for virtually all auditory computations of echo delay, including those involved in clutter rejection derived from echo spectra. [Work supported by ONR and NSF.]
Inventory size and complexity in the song of the American Robin
Authors: Reyes Kayleigh E.
Published: September 2012

This paper discusses American Robin morning song and offers a component system to explain its complex structure and discuss the relationship between inventory size and syllable complexity. Previous research has noted that Robin song is combinatorial, meaning Robins are capable of creating different songs by concatenating elements called syllables. For each robin, all unique syllables produced during a 40 minute recording were grouped into a syllable inventory. Inspired by observations linguists have made about human phoneme inventories, a component system was devised that broke each syllable into its smallest parts. These parts were found by using minima in intensity contours to correspond to natural breaks in the robin's utterances. This component system included all unique syllable parts, 7 total, from 11 syllable inventories with between 8 and 24 different syllables. This system showed certain components occurred more often than others and robins with larger syllable inventories had more rare components on average. It also showed each token of the same syllable type had the same component structure. This component system will also be used to test for patterns in sequencing of components in syllable structure to determine where components can occur and constraints on syllable sequencing in overall song structure.
Use of the radon transform for estimation of shear wave speed
Authors: Urban Matthew W. , Greenleaf James F.
Published: September 2012

Many methods in the field of elasticity imaging use shear waves to investigate the material properties of various soft tissues. The accurate and robust measurement of the shear wave speed is necessary for reliable clinical measurements. We propose using the Radon transformation on the spatiotemporal shear wave data to estimate the shear wave speed. A similar algorithm called the Radon sum transformation was proposed by Rouze, et al (IEEE Trans. Ultrasonics Ferroelectr. Freq. Control. 2010. pp. 2662-70), but this algorithm requires different input parameters that can affect the results. The use of the full Radon transform allows for the differentiation of waves that are traveling in different directions, thus this method can be used as a directional filter. We will also demonstrate the connection between estimating shear wave speeds in the Radon transform domain and estimating shear wave speeds from Fourier transform k-space. Results from shear wave measurements using shear waves induced by ultrasound radiation force will be shown. We will examine the accuracy of measurements made in calibrated elasticity phantoms, and show examples of shear wave speed estimation in arteries and kidneys. [This research was supported in part by NIH grants EB002640 and DK082408.]
Radiation-force-based estimation of acoustic attenuation using harmonic motion imaging
Authors: Chen Jiangang , Hou Gary Y. , Marquet Fabrice , Konofagou Elisa
Published: September 2012

Tissue characterization such as attenuation estimation remains challenging but important. Attenuation represents the energy loss during wave propagation through biological tissues, thus affects both therapeutic and diagnostic ultrasound applications. In this study, a novel attenuation estimation approach was developed using radiation-force-based method of Harmonic Motion Imaging (HMI). The HMI set-up comprised of a forcing transducer (fcenter = 4.7 MHz, AM frequency = 25 Hz) in conjunction with a confocal pulse-echo transducer (fcenter = 7.5 MHz), with the former inducing tissue displacement and the latter simultaneously acquiring RF signals. Tissue displacements were estimated from the RF signals using a 1-D cross-correlation method (window size: 1 mm; overlap: 90%). 2D displacement images were obtained through raster-scan (10 × 10 mm2). A linear regression model was applied to the displacements at different depths for calculating attenuation. Gel phantoms with known attenuation (n = 5) (CIRS Inc.) and in vitro canine livers (n = 3) were tested. Results demonstrated that attenuations obtained from the phantoms showed good correlation (R2 ≈ 99%) with the independently obtained values (0.28–1.45 dB/cm/MHz) (13 ± 8% underestimated), while those from the canine liver were 0.32 ± 0.03 dB/cm/MHz, within the normal range reported in the literature (0.28–1.01 dB/cm/MHz) (Duck, Academic Press 1990). Future studies will entail attenuation measurements in pathological tissues and HIFU monitoring.
Rayleigh wave propagation method for the characterization of viscoelastic properties of biomaterials
Authors: Kazemirad Siavash , Mongeau Luc
Published: September 2012

The frequency-dependent viscoelastic properties of injectable biomaterials used for vocal fold augmentation and repair must be characterized to ensure the integrity with the vibrating tissue throughout the frequency range of vocalization. Experimental methods for quantifying the frequency-dependent viscoelastic properties of biomaterials over a broad frequency range (i.e., up to 4 kHz) using Rayleigh wave propagations were investigated. Appropriate models for Rayleigh wave propagations in single and layered media were developed. Different silicone rubber samples were made and tested to evaluate the proposed methods. Rayleigh waves at different frequencies were launched on the surface of different samples; i.e., single layer samples and samples composed of a substrate with known material properties coated with a thin layer of the soft material that is to be characterized. The input vibrations of the actuator and the motion of the sample surface were measured using an accelerometer and a laser Doppler vibrometer, respectively. A transfer function method was used to obtain the complex Rayleigh wavenumbers. Finally, the complex shear and elastic moduli and the loss factor of samples were calculated through the proper modelling using the measured wavenumbers. The results were compared and shown to be in good agreement with those obtained from other measurement methods.
Sound intensity measurements in vehicle interiors
Authors: Gade Svend , Hald Jo/rgen , Mo/rkholt Jakob
Published: September 2012

In some cases it is important to be able to measure not only the total sound intensity on a panel surface in a vehicle cabin, but also the components of that intensity due to sound radiation and due to absorption from the incident field. For example, these intensity components may be needed for calibration of energy flow models of the cabin noise. A robust method based on surface absorption coefficient measurement is presented in this paper.
Ultrasound characterization of Chinese hamster ovary cells
Authors: Mahajan Aditya D. , Daniels Aislinn R. , Rodriguez Yim J. , Herd Maria-Teresa
Published: September 2012

Ultrasonic characteristics of various tissues are currently not known in enough detail to be used reliably for tissue identification or diagnosis. Analysis at a cellular level as opposed to a tissue level can examine these qualities specific to a cell type. The purpose of this research is to find the ultrasonic tissue characterization of Chinese hamster ovary (CHO) cells to develop a general test for modeling cells. To analyze the characteristics, CHO cells are cultured and prepared into a pellet-sized sample, which are then scanned with single element transducers at high frequencies (5–50 MHz). The speed of sound and attenuation of the pellets are measured using through transmissions with unfocused transducers, and backscatter coefficients are measured using pulse/echo transmissions with focused transducers. This study may establish a possible model and experimental method, in addition to providing a control for the characteristics of other cell types, specifically comparing normal and cancerous cells.
Quantitative ultrasound characterization and comparison of prostate cancer cells and normal prostate cells
Authors: Rodriguez Yim J. , Daniels Aislinn R. , Mahajan Aditya D. , Herd Maria-Teresa
Published: September 2012

Ultrasound plays an important role in helping diagnose prostate cancer as part of Ultrasound Guided Biopsies; however by better characterizing normal and cancerous prostate cells - and not the actual tumor- this study enhances ultrasound as a first-hand diagnostic tool. Using quantitative ultrasound, normal and cancerous prostate cells were analyzed and compared. Experiments to determine tissue characteristics were performed using single element transducers ranging from 5-50 MHz. Measurements of speed of sound, attenuation, and backscatter coefficients were made. The current results present a valuable insight on the differences between benign and malignant formations by analyzing them at the cellular level. Analysis of cellular behavior at smaller scales provides significant information for better understanding the properties of tumors at a larger scale. These findings contribute to enhance tissue characterization. Moreover, the results obtained present relevant data regarding speed of sound, attenuation, and backscatter coefficients useful for comparative studies and further analysis.
Nonlinear scattering of crossed ultrasonic beams in a constricted flow for the detection of a simulated deep vein thrombosis: Part II
Authors: Rebersak Markus S. , Korman Murray S.
Published: September 2012

Experiments show that the turbulence generated by a thrombosis (ping pong ball) lodged into a polyethylene cylindrical tube (leg vein) can be detected by the nonlinear scattering at the combination (3.8 MHz sum frequency) from two mutually perpendicular primary wave components (f1=1.8 MHz and f2 = 2.0 MHz). In study (1) the nonlinear scattering at the sum frequency is measured vs. angle from turbulence generated by a submerged water jet. In (2) the thrombosis model vein is submerged in the turbulent flow field, while in (3) the vein remains in place but the thrombosis is removed. Nonlinear scattering at the combination frequency in (1) shows significant Doppler shift and frequency broadening vs. angle. In (2) nonlinear scattering exists but is diminished in amplitude, Doppler shift and spectral broadening, as was expected. In case (3), there is virtually no scattering at the sum frequency, since the vein mitigates the turbulent flow. Results are presented at fine angular increments, and improved alignments to measure mean Doppler shift, standard deviation, skewness and kurtosis vs. scattering angle and thus characterize certain aspects of the turbulence behind the clot. Results extend the original work of Sean M. Mock [J. Acoust. Soc. Am., 129, 2410 (2011)].
Electro-dynamic soil-plate-oscillator transducer for monitoring the buried vibration response to airborne sound excitation
Authors: Nardini Amie E. , Korman Murray S.
Published: September 2012

A plastic drum-like anti-personal mine simulant (2 inch diam, by 1 inch tall, by ¼ inch thick aluminum tube, capped by a ¼ Al bottom circular plate and an elastic acrylic 1/16 inch thick circular top plate) was constructed. It was then modified to generate an electrical response to top-plate vibration. The mine simulant was converted to a transducer by fastened a sewing machine size bobbin wound with fine enamel copper wire onto the inside face on the bottom plate of the simulant. Next, a rare earth magnet was fastened to the inside surface of the elastic top plate (using a tacky wax). Leads were connected using insulated feed through connectors to complete the transducer design. In testing out the electro-dynamic transducer landmine simulant, results showed that the design was adequate to detect the vibration resonant response using a 2 inch burial depth in a concrete soil box filled with dry masonry sand. Two 12 inch diameter subwoofers were located a meter above the soil box and radiated sound levels on the order of 80-90 dB re 20 micro Pa, were measured near the soil surface. The swept sinusoidal transducer/microphone response exhibited a fundamental resonance near 190 Hz.
Physics of the blues—Scales, harmony, and the origin of blues piano styles
Authors: Gibson J. Murray
Published: September 2012

The development of the equal-temperament scale was driven, not by compromise, but by the need to extend the composer’s “palette” and increase the harmonic sophistication of western music. Many interesting musical idioms emerged from harmonic artifacts associated with equal temperament. In particular the blues “crushed-note” piano style resulted during the birth to the blues from the melding of this scale with the pure melodies and harmonies. In the presentation I will relate the history from a scientific perspective and illustrate with short keyboard performances. In homage to Kansas - the home of Scott Joplin -related ragtime piano idioms will also be covered. I’ll conclude by emphasizing how music is an excellent medium that displays the intimate relationship between science and art, and is a great vehicle to teach science.
Pitch bending in the diatonic harmonica
Authors: Cottingham James P.
Published: September 2012

Pitch bending by wind instrument players is standard practice in many genres of music, but it is considered essential in blues harmonica playing. Some simple pitch bends involve the coupling of a single reed to a resonator such as the vocal tract of the player, but a full description of pitch bending in the harmonica involves consideration of the coupling between the two reeds, one for each direction of airflow, that share a single reed chamber. The most common pitch bends are those for which the primary reed in the reed chamber sounds a note higher in frequency than that sounded by the secondary reed. In these cases notes can be bent downward to pitches between those sounded by the two reeds. In addition, some players use more advanced techniques to bend notes beyond the frequencies of the chamber reeds. This paper reviews experimental work and theoretical modeling done on pitch bending in the harmonica during the last thirty years. This includes measurements made while pitch bends are produced by players as well as experiments on harmonicas in more conventional laboratory settings.
Extension of a launch pad noise prediction model to multiple engines and directional receivers
Authors: Plotkin Kenneth J. , Vu Bruce T.
Published: September 2012

A model, PAD, has been developed for prediction of noise in the vicinity of launch vehicles, with original application to the mobile launcher and tower for the Ares I launch vehicle. It follows the basic principles of a traditional NASA model (NASA SP-8072, 1971), with updated source components, including impingement, water suppression and acoustic shielding by three dimensional launcher configurations. For application to Space Launch System, the model has been extended to multi-engine vehicles, using the plume merging model developed by Kandula, Vu and Lindsay (AIAA Paper 2005-3091) and accommodating multiple flame holes in the deck. The capability has also been added to account for receiver directivity. This can be significant when predicting the load on the surfaces of enclosures on the launch tower. It is also an issue for model scale tests (such as ASMAT) where microphones and their mounts are not small enough to be omnidirectional, and thus do not measure free field levels. (Work supported by the National Aeronautics and Space Administration.)
Scale model acoustic test overview
Authors: Counter Douglas , Houston Janice
Published: September 2012

Launch environments, such as lift-off acoustic (LOA) and ignition overpressure (IOP), are important design factors for any vehicle and are dependent upon the design of both the vehicle and the ground systems. LOA environments are used directly in the development of vehicle vibro-acoustic environments and IOP is used in the loads assessment. The Scale Model Acoustic Test (SMAT) program was implemented to verify the Space Launch Systems LOA and IOP environments for the vehicle and ground systems including the Mobile Launcher (ML) and tower. The SMAT is currently in the design and fabrication phase. The SMAT program is described in this presentation.
Acoustic measurement of 1:42 scale booster and launch pad
Authors: Ishii Tatsuya , Tsutsumi Seiji , Ui Kyoichi , Tokudome Shinichiro , Ishii Yutaka , Wada Kei , Nakamura Satoru
Published: September 2012

This paper describes the acoustic measurement of the subscale booster and launch pad. The 1:42 scale solid propellant booster was settled over the launch pad model. The launch pad model was designed to deflect the hot and high speed plume, aimed at mitigating the feedback Mach waves toward the vehicle. The launch pad plays a role in attenuating the sound due to the impingement of the plume and the deflector. To investigate the acoustic field with a different booster height, acoustic measurement was carried out. The measurement involved the conventional acoustic measurement and the sound source localization. The conventional measurement employed the near-field microphones around the booster model and the far-field microphones on an arc centered on the booster nozzle or the impingement point of the plume and the launch pad. In the sound source localization, a phased array microphone system was settled to focus the deflector exit. The obtained acoustic data helped revise the design of the launch pad model.
Decomposition of military jet aircraft mixing noise into fine and large-scale turbulent components
Authors: Neilsen Tracianne B. , Gee Kent L. , Wall Alan T. , James Michael M.
Published: September 2012

Many far-field measurements of laboratory-scale jet noise have shown good agreement with the two similarity spectra developed to represent the contributions of fine-scale and large-scale turbulent structures [Tam et al., AIAA paper 96-1716, 1996]. Measurements near an F-22A Raptor provide a means to study how accurately the similarity spectra describe the noise from a full-scale, high-performance, jet engine. Comparisons have been made using ground-based microphones at 60° to 150° for three engine conditions: intermediate, military and afterburner, with more detailed analyses than described previously [Neilsen et al., J. Acoust. Soc. Am. 129, 2242 (2011) ]. The good agreement with Tam’s predictions - the fine-scale spectrum at upstream and sideline angles and the large-scale spectrum in the maximum radiation direction - permits a quantitative analysis of the contributions of the two spectra at other angles. The radiated spectra and overall levels for all three engine conditions have been decomposed into contributions from the two spectra as functions of angle. Of particular interest is the appreciable contribution of fine-scale turbulence aft of the maximum radiation directions at military and afterburner conditions. [Work sponsored by the Office of Naval Research.]