Temporal coherence for pure tones in budgerigars (Melopsittacus undulatus) and humans (Homo sapiens).

PubMed

Neilans, Erikson G; Dent, Micheal L

2015-02-01

Auditory scene analysis has been suggested as a universal process that exists across all animals. Relative to humans, however, little work has been devoted to how animals perceptually isolate different sound sources. Frequency separation of sounds is arguably the most common parameter studied in auditory streaming, but it is not the only factor contributing to how the auditory scene is perceived. Researchers have found that in humans, even at large frequency separations, synchronous tones are heard as a single auditory stream, whereas asynchronous tones with the same frequency separations are perceived as 2 distinct sounds. These findings demonstrate how both the timing and frequency separation of sounds are important for auditory scene analysis. It is unclear how animals, such as budgerigars (Melopsittacus undulatus), perceive synchronous and asynchronous sounds. In this study, budgerigars and humans (Homo sapiens) were tested on their perception of synchronous, asynchronous, and partially overlapping pure tones using the same psychophysical procedures. Species differences were found between budgerigars and humans in how partially overlapping sounds were perceived, with budgerigars more likely to segregate overlapping sounds and humans more apt to fuse the 2 sounds together. The results also illustrated that temporal cues are particularly important for stream segregation of overlapping sounds. Lastly, budgerigars were found to segregate partially overlapping sounds in a manner predicted by computational models of streaming, whereas humans were not. PsycINFO Database Record (c) 2015 APA, all rights reserved.

Dynamic Spatial Hearing by Human and Robot Listeners

NASA Astrophysics Data System (ADS)

Zhong, Xuan

This study consisted of several related projects on dynamic spatial hearing by both human and robot listeners. The first experiment investigated the maximum number of sound sources that human listeners could localize at the same time. Speech stimuli were presented simultaneously from different loudspeakers at multiple time intervals. The maximum of perceived sound sources was close to four. The second experiment asked whether the amplitude modulation of multiple static sound sources could lead to the perception of auditory motion. On the horizontal and vertical planes, four independent noise sound sources with 60° spacing were amplitude modulated with consecutively larger phase delay. At lower modulation rates, motion could be perceived by human listeners in both cases. The third experiment asked whether several sources at static positions could serve as "acoustic landmarks" to improve the localization of other sources. Four continuous speech sound sources were placed on the horizontal plane with 90° spacing and served as the landmarks. The task was to localize a noise that was played for only three seconds when the listener was passively rotated in a chair in the middle of the loudspeaker array. The human listeners were better able to localize the sound sources with landmarks than without. The other experiments were with the aid of an acoustic manikin in an attempt to fuse binaural recording and motion data to localize sounds sources. A dummy head with recording devices was mounted on top of a rotating chair and motion data was collected. The fourth experiment showed that an Extended Kalman Filter could be used to localize sound sources in a recursive manner. The fifth experiment demonstrated the use of a fitting method for separating multiple sounds sources.

Underwater auditory localization by a swimming harbor seal (Phoca vitulina).

PubMed

Bodson, Anais; Miersch, Lars; Mauck, Bjoern; Dehnhardt, Guido

2006-09-01

The underwater sound localization acuity of a swimming harbor seal (Phoca vitulina) was measured in the horizontal plane at 13 different positions. The stimulus was either a double sound (two 6-kHz pure tones lasting 0.5 s separated by an interval of 0.2 s) or a single continuous sound of 1.2 s. Testing was conducted in a 10-m-diam underwater half circle arena with hidden loudspeakers installed at the exterior perimeter. The animal was trained to swim along the diameter of the half circle and to change its course towards the sound source as soon as the signal was given. The seal indicated the sound source by touching its assumed position at the board of the half circle. The deviation of the seals choice from the actual sound source was measured by means of video analysis. In trials with the double sound the seal localized the sound sources with a mean deviation of 2.8 degrees and in trials with the single sound with a mean deviation of 4.5 degrees. In a second experiment minimum audible angles of the stationary animal were found to be 9.8 degrees in front and 9.7 degrees in the back of the seal's head.

Toward a Neurophysiological Theory of Auditory Stream Segregation

ERIC Educational Resources Information Center

Snyder, Joel S.; Alain, Claude

2007-01-01

Auditory stream segregation (or streaming) is a phenomenon in which 2 or more repeating sounds differing in at least 1 acoustic attribute are perceived as 2 or more separate sound sources (i.e., streams). This article selectively reviews psychophysical and computational studies of streaming and comprehensively reviews more recent…

Separating pitch chroma and pitch height in the human brain

PubMed Central

Warren, J. D.; Uppenkamp, S.; Patterson, R. D.; Griffiths, T. D.

2003-01-01

Musicians recognize pitch as having two dimensions. On the keyboard, these are illustrated by the octave and the cycle of notes within the octave. In perception, these dimensions are referred to as pitch height and pitch chroma, respectively. Pitch chroma provides a basis for presenting acoustic patterns (melodies) that do not depend on the particular sound source. In contrast, pitch height provides a basis for segregation of notes into streams to separate sound sources. This paper reports a functional magnetic resonance experiment designed to search for distinct mappings of these two types of pitch change in the human brain. The results show that chroma change is specifically represented anterior to primary auditory cortex, whereas height change is specifically represented posterior to primary auditory cortex. We propose that tracking of acoustic information streams occurs in anterior auditory areas, whereas the segregation of sound objects (a crucial aspect of auditory scene analysis) depends on posterior areas. PMID:12909719

Separating pitch chroma and pitch height in the human brain.

PubMed

Warren, J D; Uppenkamp, S; Patterson, R D; Griffiths, T D

2003-08-19

Musicians recognize pitch as having two dimensions. On the keyboard, these are illustrated by the octave and the cycle of notes within the octave. In perception, these dimensions are referred to as pitch height and pitch chroma, respectively. Pitch chroma provides a basis for presenting acoustic patterns (melodies) that do not depend on the particular sound source. In contrast, pitch height provides a basis for segregation of notes into streams to separate sound sources. This paper reports a functional magnetic resonance experiment designed to search for distinct mappings of these two types of pitch change in the human brain. The results show that chroma change is specifically represented anterior to primary auditory cortex, whereas height change is specifically represented posterior to primary auditory cortex. We propose that tracking of acoustic information streams occurs in anterior auditory areas, whereas the segregation of sound objects (a crucial aspect of auditory scene analysis) depends on posterior areas.

The effects of spatially separated call components on phonotaxis in túngara frogs: evidence for auditory grouping.

PubMed

Farris, Hamilton E; Rand, A Stanley; Ryan, Michael J

2002-01-01

Numerous animals across disparate taxa must identify and locate complex acoustic signals imbedded in multiple overlapping signals and ambient noise. A requirement of this task is the ability to group sounds into auditory streams in which sounds are perceived as emanating from the same source. Although numerous studies over the past 50 years have examined aspects of auditory grouping in humans, surprisingly few assays have demonstrated auditory stream formation or the assignment of multicomponent signals to a single source in non-human animals. In our study, we present evidence for auditory grouping in female túngara frogs. In contrast to humans, in which auditory grouping may be facilitated by the cues produced when sounds arrive from the same location, we show that spatial cues play a limited role in grouping, as females group discrete components of the species' complex call over wide angular separations. Furthermore, we show that once grouped the separate call components are weighted differently in recognizing and locating the call, so called 'what' and 'where' decisions, respectively. Copyright 2002 S. Karger AG, Basel

Psychophysical investigation of an auditory spatial illusion in cats: the precedence effect.

PubMed

Tollin, Daniel J; Yin, Tom C T

2003-10-01

The precedence effect (PE) describes several spatial perceptual phenomena that occur when similar sounds are presented from two different locations and separated by a delay. The mechanisms that produce the effect are thought to be responsible for the ability to localize sounds in reverberant environments. Although the physiological bases for the PE have been studied, little is known about how these sounds are localized by species other than humans. Here we used the search coil technique to measure the eye positions of cats trained to saccade to the apparent locations of sounds. To study the PE, brief broadband stimuli were presented from two locations, with a delay between their onsets; the delayed sound meant to simulate a single reflection. Although the cats accurately localized single sources, the apparent locations of the paired sources depended on the delay. First, the cats exhibited summing localization, the perception of a "phantom" sound located between the sources, for delays < +/-400 micros for sources positioned in azimuth along the horizontal plane, but not for sources positioned in elevation along the sagittal plane. Second, consistent with localization dominance, for delays from 400 micros to about 10 ms, the cats oriented toward the leading source location only, with little influence of the lagging source, both for horizontally and vertically placed sources. Finally, the echo threshold was reached for delays >10 ms, where the cats first began to orient to the lagging source on some trials. These data reveal that cats experience the PE phenomena similarly to humans.

Cortical Transformation of Spatial Processing for Solving the Cocktail Party Problem: A Computational Model123

PubMed Central

Dong, Junzi; Colburn, H. Steven

2016-01-01

In multisource, “cocktail party” sound environments, human and animal auditory systems can use spatial cues to effectively separate and follow one source of sound over competing sources. While mechanisms to extract spatial cues such as interaural time differences (ITDs) are well understood in precortical areas, how such information is reused and transformed in higher cortical regions to represent segregated sound sources is not clear. We present a computational model describing a hypothesized neural network that spans spatial cue detection areas and the cortex. This network is based on recent physiological findings that cortical neurons selectively encode target stimuli in the presence of competing maskers based on source locations (Maddox et al., 2012). We demonstrate that key features of cortical responses can be generated by the model network, which exploits spatial interactions between inputs via lateral inhibition, enabling the spatial separation of target and interfering sources while allowing monitoring of a broader acoustic space when there is no competition. We present the model network along with testable experimental paradigms as a starting point for understanding the transformation and organization of spatial information from midbrain to cortex. This network is then extended to suggest engineering solutions that may be useful for hearing-assistive devices in solving the cocktail party problem. PMID:26866056

Cortical Transformation of Spatial Processing for Solving the Cocktail Party Problem: A Computational Model(1,2,3).

PubMed

Dong, Junzi; Colburn, H Steven; Sen, Kamal

2016-01-01

In multisource, "cocktail party" sound environments, human and animal auditory systems can use spatial cues to effectively separate and follow one source of sound over competing sources. While mechanisms to extract spatial cues such as interaural time differences (ITDs) are well understood in precortical areas, how such information is reused and transformed in higher cortical regions to represent segregated sound sources is not clear. We present a computational model describing a hypothesized neural network that spans spatial cue detection areas and the cortex. This network is based on recent physiological findings that cortical neurons selectively encode target stimuli in the presence of competing maskers based on source locations (Maddox et al., 2012). We demonstrate that key features of cortical responses can be generated by the model network, which exploits spatial interactions between inputs via lateral inhibition, enabling the spatial separation of target and interfering sources while allowing monitoring of a broader acoustic space when there is no competition. We present the model network along with testable experimental paradigms as a starting point for understanding the transformation and organization of spatial information from midbrain to cortex. This network is then extended to suggest engineering solutions that may be useful for hearing-assistive devices in solving the cocktail party problem.

Experimental assessment of theory for refraction of sound by a shear layer

NASA Technical Reports Server (NTRS)

Schlinker, R. H.; Amiet, R. K.

1978-01-01

The refraction angle and amplitude changes associated with sound transmission through a circular, open-jet shear layer were studied in a 0.91 m diameter open jet acoustic research tunnel. Free stream Mach number was varied from 0.1 to 0.4. Good agreement between refraction angle correction theory and experiment was obtained over the test Mach number, frequency and angle measurement range for all on-axis acoustic source locations. For off-axis source positions, good agreement was obtained at a source-to-shear layer separation distance greater than the jet radius. Measureable differences between theory and experiment occurred at a source-to-shear layer separation distance less than one jet radius. A shear layer turbulence scattering experiment was conducted at 90 deg to the open jet axis for the same free stream Mach numbers and axial source locations used in the refraction study. Significant discrete tone spectrum broadening and tone amplitude changes were observed at open jet Mach numbers above 0.2 and at acoustic source frequencies greater than 5 kHz. More severe turbulence scattering was observed for downstream source locations.

Minke whale song, spacing, and acoustic communication on the Great Barrier Reef, Australia

NASA Astrophysics Data System (ADS)

Gedamke, Jason

An inquisitive population of minke whale (Balaenoptera acutorostrata ) that concentrates on the Great Barrier Reef during its suspected breeding season offered a unique opportunity to conduct a multi-faceted study of a little-known Balaenopteran species' acoustic behavior. Chapter one investigates whether the minke whale is the source of an unusual, complex, and stereotyped sound recorded, the "star-wars" vocalization. A hydrophone array was towed from a vessel to record sounds from circling whales for subsequent localization of sound sources. These acoustic locations were matched with shipboard and in-water observations of the minke whale, demonstrating the minke whale was the source of this unusual sound. Spectral and temporal features of this sound and the source levels at which it is produced are described. The repetitive "star-wars" vocalization appears similar to the songs of other whale species and has characteristics consistent with reproductive advertisement displays. Chapter two investigates whether song (i.e. the "star-wars" vocalization) has a spacing function through passive monitoring of singer spatial patterns with a moored five-sonobuoy array. Active song playback experiments to singers were also conducted to further test song function. This study demonstrated that singers naturally maintain spatial separations between them through a nearest-neighbor analysis and animated tracks of singer movements. In response to active song playbacks, singers generally moved away and repeated song more quickly suggesting that song repetition interval may help regulate spatial interaction and singer separation. These results further indicate the Great Barrier Reef may be an important reproductive habitat for this species. Chapter three investigates whether song is part of a potentially graded repertoire of acoustic signals. Utilizing both vessel-based recordings and remote recordings from the sonobuoy array, temporal and spectral features, source levels, and associated contextual data of recorded sounds were analyzed. Two categories of sound are described here: (1) patterned song, which was regularly repeated in one of three patterns: slow, fast, and rapid-clustered repetition, and (2) non-patterned "social" sounds recorded from gregarious assemblages of whales. These discrete acoustic signals may comprise a graded system of communication (Slow/fast song → Rapid-clustered song → Social sounds) that is related to the spacing between whales.

The opponent channel population code of sound location is an efficient representation of natural binaural sounds.

PubMed

Młynarski, Wiktor

2015-05-01

In mammalian auditory cortex, sound source position is represented by a population of broadly tuned neurons whose firing is modulated by sounds located at all positions surrounding the animal. Peaks of their tuning curves are concentrated at lateral position, while their slopes are steepest at the interaural midline, allowing for the maximum localization accuracy in that area. These experimental observations contradict initial assumptions that the auditory space is represented as a topographic cortical map. It has been suggested that a "panoramic" code has evolved to match specific demands of the sound localization task. This work provides evidence suggesting that properties of spatial auditory neurons identified experimentally follow from a general design principle- learning a sparse, efficient representation of natural stimuli. Natural binaural sounds were recorded and served as input to a hierarchical sparse-coding model. In the first layer, left and right ear sounds were separately encoded by a population of complex-valued basis functions which separated phase and amplitude. Both parameters are known to carry information relevant for spatial hearing. Monaural input converged in the second layer, which learned a joint representation of amplitude and interaural phase difference. Spatial selectivity of each second-layer unit was measured by exposing the model to natural sound sources recorded at different positions. Obtained tuning curves match well tuning characteristics of neurons in the mammalian auditory cortex. This study connects neuronal coding of the auditory space with natural stimulus statistics and generates new experimental predictions. Moreover, results presented here suggest that cortical regions with seemingly different functions may implement the same computational strategy-efficient coding.

33 CFR 167.1702 - In Prince William Sound: Prince William Sound Traffic Separation Scheme.

Code of Federal Regulations, 2012 CFR

2012-07-01

... William Sound Traffic Separation Scheme. 167.1702 Section 167.1702 Navigation and Navigable Waters COAST... SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1702 In Prince William Sound: Prince William Sound Traffic Separation Scheme. The Prince William Sound...

33 CFR 167.1702 - In Prince William Sound: Prince William Sound Traffic Separation Scheme.

Code of Federal Regulations, 2014 CFR

2014-07-01

... William Sound Traffic Separation Scheme. 167.1702 Section 167.1702 Navigation and Navigable Waters COAST... SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1702 In Prince William Sound: Prince William Sound Traffic Separation Scheme. The Prince William Sound...

33 CFR 167.1702 - In Prince William Sound: Prince William Sound Traffic Separation Scheme.

Code of Federal Regulations, 2013 CFR

2013-07-01

... William Sound Traffic Separation Scheme. 167.1702 Section 167.1702 Navigation and Navigable Waters COAST... SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1702 In Prince William Sound: Prince William Sound Traffic Separation Scheme. The Prince William Sound...

33 CFR 167.1702 - In Prince William Sound: Prince William Sound Traffic Separation Scheme.

Code of Federal Regulations, 2010 CFR

2010-07-01

... William Sound Traffic Separation Scheme. 167.1702 Section 167.1702 Navigation and Navigable Waters COAST... SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1702 In Prince William Sound: Prince William Sound Traffic Separation Scheme. The Prince William Sound...

33 CFR 167.1702 - In Prince William Sound: Prince William Sound Traffic Separation Scheme.

Code of Federal Regulations, 2011 CFR

2011-07-01

... William Sound Traffic Separation Scheme. 167.1702 Section 167.1702 Navigation and Navigable Waters COAST... SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1702 In Prince William Sound: Prince William Sound Traffic Separation Scheme. The Prince William Sound...

Blind source separation by sparse decomposition

NASA Astrophysics Data System (ADS)

Zibulevsky, Michael; Pearlmutter, Barak A.

2000-04-01

The blind source separation problem is to extract the underlying source signals from a set of their linear mixtures, where the mixing matrix is unknown. This situation is common, eg in acoustics, radio, and medical signal processing. We exploit the property of the sources to have a sparse representation in a corresponding signal dictionary. Such a dictionary may consist of wavelets, wavelet packets, etc., or be obtained by learning from a given family of signals. Starting from the maximum a posteriori framework, which is applicable to the case of more sources than mixtures, we derive a few other categories of objective functions, which provide faster and more robust computations, when there are an equal number of sources and mixtures. Our experiments with artificial signals and with musical sounds demonstrate significantly better separation than other known techniques.

Advances in audio source seperation and multisource audio content retrieval

NASA Astrophysics Data System (ADS)

Vincent, Emmanuel

2012-06-01

Audio source separation aims to extract the signals of individual sound sources from a given recording. In this paper, we review three recent advances which improve the robustness of source separation in real-world challenging scenarios and enable its use for multisource content retrieval tasks, such as automatic speech recognition (ASR) or acoustic event detection (AED) in noisy environments. We present a Flexible Audio Source Separation Toolkit (FASST) and discuss its advantages compared to earlier approaches such as independent component analysis (ICA) and sparse component analysis (SCA). We explain how cues as diverse as harmonicity, spectral envelope, temporal fine structure or spatial location can be jointly exploited by this toolkit. We subsequently present the uncertainty decoding (UD) framework for the integration of audio source separation and audio content retrieval. We show how the uncertainty about the separated source signals can be accurately estimated and propagated to the features. Finally, we explain how this uncertainty can be efficiently exploited by a classifier, both at the training and the decoding stage. We illustrate the resulting performance improvements in terms of speech separation quality and speaker recognition accuracy.

Sound Shell Model for Acoustic Gravitational Wave Production at a First-Order Phase Transition in the Early Universe.

PubMed

Hindmarsh, Mark

2018-02-16

A model for the acoustic production of gravitational waves at a first-order phase transition is presented. The source of gravitational radiation is the sound waves generated by the explosive growth of bubbles of the stable phase. The model assumes that the sound waves are linear and that their power spectrum is determined by the characteristic form of the sound shell around the expanding bubble. The predicted power spectrum has two length scales, the average bubble separation and the sound shell width when the bubbles collide. The peak of the power spectrum is at wave numbers set by the sound shell width. For a higher wave number k, the power spectrum decreases to k^{-3}. At wave numbers below the inverse bubble separation, the power spectrum goes to k^{5}. For bubble wall speeds near the speed of sound where these two length scales are distinguished, there is an intermediate k^{1} power law. The detailed dependence of the power spectrum on the wall speed and the other parameters of the phase transition raises the possibility of their constraint or measurement at a future space-based gravitational wave observatory such as LISA.

Sound Shell Model for Acoustic Gravitational Wave Production at a First-Order Phase Transition in the Early Universe

NASA Astrophysics Data System (ADS)

Hindmarsh, Mark

2018-02-01

A model for the acoustic production of gravitational waves at a first-order phase transition is presented. The source of gravitational radiation is the sound waves generated by the explosive growth of bubbles of the stable phase. The model assumes that the sound waves are linear and that their power spectrum is determined by the characteristic form of the sound shell around the expanding bubble. The predicted power spectrum has two length scales, the average bubble separation and the sound shell width when the bubbles collide. The peak of the power spectrum is at wave numbers set by the sound shell width. For a higher wave number k , the power spectrum decreases to k-3. At wave numbers below the inverse bubble separation, the power spectrum goes to k5. For bubble wall speeds near the speed of sound where these two length scales are distinguished, there is an intermediate k1 power law. The detailed dependence of the power spectrum on the wall speed and the other parameters of the phase transition raises the possibility of their constraint or measurement at a future space-based gravitational wave observatory such as LISA.

The Opponent Channel Population Code of Sound Location Is an Efficient Representation of Natural Binaural Sounds

PubMed Central

Młynarski, Wiktor

2015-01-01

In mammalian auditory cortex, sound source position is represented by a population of broadly tuned neurons whose firing is modulated by sounds located at all positions surrounding the animal. Peaks of their tuning curves are concentrated at lateral position, while their slopes are steepest at the interaural midline, allowing for the maximum localization accuracy in that area. These experimental observations contradict initial assumptions that the auditory space is represented as a topographic cortical map. It has been suggested that a “panoramic” code has evolved to match specific demands of the sound localization task. This work provides evidence suggesting that properties of spatial auditory neurons identified experimentally follow from a general design principle- learning a sparse, efficient representation of natural stimuli. Natural binaural sounds were recorded and served as input to a hierarchical sparse-coding model. In the first layer, left and right ear sounds were separately encoded by a population of complex-valued basis functions which separated phase and amplitude. Both parameters are known to carry information relevant for spatial hearing. Monaural input converged in the second layer, which learned a joint representation of amplitude and interaural phase difference. Spatial selectivity of each second-layer unit was measured by exposing the model to natural sound sources recorded at different positions. Obtained tuning curves match well tuning characteristics of neurons in the mammalian auditory cortex. This study connects neuronal coding of the auditory space with natural stimulus statistics and generates new experimental predictions. Moreover, results presented here suggest that cortical regions with seemingly different functions may implement the same computational strategy-efficient coding. PMID:25996373

The auditory P50 component to onset and offset of sound

PubMed Central

Pratt, Hillel; Starr, Arnold; Michalewski, Henry J.; Bleich, Naomi; Mittelman, Nomi

2008-01-01

Objective: The auditory Event-Related Potentials (ERP) component P50 to sound onset and offset have been reported to be similar, but their magnetic homologue has been reported absent to sound offset. We compared the spatio-temporal distribution of cortical activity during P50 to sound onset and offset, without confounds of spectral change. Methods: ERPs were recorded in response to onsets and offsets of silent intervals of 0.5 s (gaps) appearing randomly in otherwise continuous white noise and compared to ERPs to randomly distributed click pairs with half second separation presented in silence. Subjects were awake and distracted from the stimuli by reading a complicated text. Measures of P50 included peak latency and amplitude, as well as source current density estimates to the clicks and sound onsets and offsets. Results P50 occurred in response to noise onsets and to clicks, while to noise offset it was absent. Latency of P50 was similar to noise onset (56 msec) and to clicks (53 msec). Sources of P50 to noise onsets and clicks included bilateral superior parietal areas. In contrast, noise offsets activated left inferior temporal and occipital areas at the time of P50. Source current density was significantly higher to noise onset than offset in the vicinity of the temporo-parietal junction. Conclusions: P50 to sound offset is absent compared to the distinct P50 to sound onset and to clicks, at different intracranial sources. P50 to stimulus onset and to clicks appears to reflect preattentive arousal by a new sound in the scene. Sound offset does not involve a new sound and hence the absent P50. Significance: Stimulus onset activates distinct early cortical processes that are absent to offset. PMID:18055255

Development of an Acoustic Signal Analysis Tool “Auto-F” Based on the Temperament Scale

NASA Astrophysics Data System (ADS)

Modegi, Toshio

The MIDI interface is originally designed for electronic musical instruments but we consider this music-note based coding concept can be extended for general acoustic signal description. We proposed applying the MIDI technology to coding of bio-medical auscultation sound signals such as heart sounds for retrieving medical records and performing telemedicine. Then we have tried to extend our encoding targets including vocal sounds, natural sounds and electronic bio-signals such as ECG, using Generalized Harmonic Analysis method. Currently, we are trying to separate vocal sounds included in popular songs and encode both vocal sounds and background instrumental sounds into separate MIDI channels. And also, we are trying to extract articulation parameters such as MIDI pitch-bend parameters in order to reproduce natural acoustic sounds using a GM-standard MIDI tone generator. In this paper, we present an overall algorithm of our developed acoustic signal analysis tool, based on those research works, which can analyze given time-based signals on the musical temperament scale. The prominent feature of this tool is producing high-precision MIDI codes, which reproduce the similar signals as the given source signal using a GM-standard MIDI tone generator, and also providing analyzed texts in the XML format.

Single-channel mixed signal blind source separation algorithm based on multiple ICA processing

NASA Astrophysics Data System (ADS)

Cheng, Xiefeng; Li, Ji

2017-01-01

Take separating the fetal heart sound signal from the mixed signal that get from the electronic stethoscope as the research background, the paper puts forward a single-channel mixed signal blind source separation algorithm based on multiple ICA processing. Firstly, according to the empirical mode decomposition (EMD), the single-channel mixed signal get multiple orthogonal signal components which are processed by ICA. The multiple independent signal components are called independent sub component of the mixed signal. Then by combining with the multiple independent sub component into single-channel mixed signal, the single-channel signal is expanded to multipath signals, which turns the under-determined blind source separation problem into a well-posed blind source separation problem. Further, the estimate signal of source signal is get by doing the ICA processing. Finally, if the separation effect is not very ideal, combined with the last time's separation effect to the single-channel mixed signal, and keep doing the ICA processing for more times until the desired estimated signal of source signal is get. The simulation results show that the algorithm has good separation effect for the single-channel mixed physiological signals.

Combined visualization for noise mapping of industrial facilities based on ray-tracing and thin plate splines

NASA Astrophysics Data System (ADS)

Ovsiannikov, Mikhail; Ovsiannikov, Sergei

2017-01-01

The paper presents the combined approach to noise mapping and visualizing of industrial facilities sound pollution using forward ray tracing method and thin-plate spline interpolation. It is suggested to cauterize industrial area in separate zones with similar sound levels. Equivalent local source is defined for range computation of sanitary zones based on ray tracing algorithm. Computation of sound pressure levels within clustered zones are based on two-dimension spline interpolation of measured data on perimeter and inside the zone.

Independence of Echo-Threshold and Echo-Delay in the Barn Owl

PubMed Central

Nelson, Brian S.; Takahashi, Terry T.

2008-01-01

Despite their prevalence in nature, echoes are not perceived as events separate from the sounds arriving directly from an active source, until the echo's delay is long. We measured the head-saccades of barn owls and the responses of neurons in their auditory space-maps while presenting a long duration noise-burst and a simulated echo. Under this paradigm, there were two possible stimulus segments that could potentially signal the location of the echo. One was at the onset of the echo; the other, after the offset of the direct (leading) sound, when only the echo was present. By lengthening the echo's duration, independently of its delay, spikes and saccades were evoked by the source of the echo even at delays that normally evoked saccades to only the direct source. An echo's location thus appears to be signaled by the neural response evoked after the offset of the direct sound. PMID:18974886

Speech segregation based-on binaural cue: interaural time difference (itd) and interaural level difference (ild)

NASA Astrophysics Data System (ADS)

Nur Farid, Mifta; Arifianto, Dhany

2016-11-01

A person who is suffering from hearing loss can be helped by using hearing aids and the most optimal performance of hearing aids are binaural hearing aids because it has similarities to human auditory system. In a conversation at a cocktail party, a person can focus on a single conversation even though the background sound and other people conversation is quite loud. This phenomenon is known as the cocktail party effect. In an early study, has been explained that binaural hearing have an important contribution to the cocktail party effect. So in this study, will be performed separation on the input binaural sound with 2 microphone sensors of two sound sources based on both the binaural cue, interaural time difference (ITD) and interaural level difference (ILD) using binary mask. To estimate value of ITD, is used cross-correlation method which the value of ITD represented as time delay of peak shifting at time-frequency unit. Binary mask is estimated based on pattern of ITD and ILD to relative strength of target that computed statistically using probability density estimation. Results of sound source separation performing well with the value of speech intelligibility using the percent correct word by 86% and 3 dB by SNR.

Model-free data analysis for source separation based on Non-Negative Matrix Factorization and k-means clustering (NMFk)

NASA Astrophysics Data System (ADS)

Vesselinov, V. V.; Alexandrov, B.

2014-12-01

The identification of the physical sources causing spatial and temporal fluctuations of state variables such as river stage levels and aquifer hydraulic heads is challenging. The fluctuations can be caused by variations in natural and anthropogenic sources such as precipitation events, infiltration, groundwater pumping, barometric pressures, etc. The source identification and separation can be crucial for conceptualization of the hydrological conditions and characterization of system properties. If the original signals that cause the observed state-variable transients can be successfully "unmixed", decoupled physics models may then be applied to analyze the propagation of each signal independently. We propose a new model-free inverse analysis of transient data based on Non-negative Matrix Factorization (NMF) method for Blind Source Separation (BSS) coupled with k-means clustering algorithm, which we call NMFk. NMFk is capable of identifying a set of unique sources from a set of experimentally measured mixed signals, without any information about the sources, their transients, and the physical mechanisms and properties controlling the signal propagation through the system. A classical BSS conundrum is the so-called "cocktail-party" problem where several microphones are recording the sounds in a ballroom (music, conversations, noise, etc.). Each of the microphones is recording a mixture of the sounds. The goal of BSS is to "unmix'" and reconstruct the original sounds from the microphone records. Similarly to the "cocktail-party" problem, our model-freee analysis only requires information about state-variable transients at a number of observation points, m, where m > r, and r is the number of unknown unique sources causing the observed fluctuations. We apply the analysis on a dataset from the Los Alamos National Laboratory (LANL) site. We identify and estimate the impact and sources are barometric pressure and water-supply pumping effects. We also estimate the location of the water-supply pumping wells based on the available data. The possible applications of the NMFk algorithm are not limited to hydrology problems; NMFk can be applied to any problem where temporal system behavior is observed at multiple locations and an unknown number of physical sources are causing these fluctuations.

Newborn infants detect cues of concurrent sound segregation.

PubMed

Bendixen, Alexandra; Háden, Gábor P; Németh, Renáta; Farkas, Dávid; Török, Miklós; Winkler, István

2015-01-01

Separating concurrent sounds is fundamental for a veridical perception of one's auditory surroundings. Sound components that are harmonically related and start at the same time are usually grouped into a common perceptual object, whereas components that are not in harmonic relation or have different onset times are more likely to be perceived in terms of separate objects. Here we tested whether neonates are able to pick up the cues supporting this sound organization principle. We presented newborn infants with a series of complex tones with their harmonics in tune (creating the percept of a unitary sound object) and with manipulated variants, which gave the impression of two concurrently active sound sources. The manipulated variant had either one mistuned partial (single-cue condition) or the onset of this mistuned partial was also delayed (double-cue condition). Tuned and manipulated sounds were presented in random order with equal probabilities. Recording the neonates' electroencephalographic responses allowed us to evaluate their processing of the sounds. Results show that, in both conditions, mistuned sounds elicited a negative displacement of the event-related potential (ERP) relative to tuned sounds from 360 to 400 ms after sound onset. The mistuning-related ERP component resembles the object-related negativity (ORN) component in adults, which is associated with concurrent sound segregation. Delayed onset additionally led to a negative displacement from 160 to 200 ms, which was probably more related to the physical parameters of the sounds than to their perceptual segregation. The elicitation of an ORN-like response in newborn infants suggests that neonates possess the basic capabilities of segregating concurrent sounds by detecting inharmonic relations between the co-occurring sounds. © 2015 S. Karger AG, Basel.

Audio Visual Instructional Materials for Distributive Education; a Classified Bibliography. Final Report.

ERIC Educational Resources Information Center

Levendowski, Jerry C.

The bibliography contains a list of 90 names and addresses of sources of audiovisual instructional materials. For each title a brief description of content, the source, purchase price, rental fee or free use for 16MM films, sound-slidefilms, tapes-records, and transparencies is given. Materials are listed separately by topics: (1) advertising and…

Effects of dynamic range compression on spatial selective auditory attention in normal-hearing listeners.

PubMed

Schwartz, Andrew H; Shinn-Cunningham, Barbara G

2013-04-01

Many hearing aids introduce compressive gain to accommodate the reduced dynamic range that often accompanies hearing loss. However, natural sounds produce complicated temporal dynamics in hearing aid compression, as gain is driven by whichever source dominates at a given moment. Moreover, independent compression at the two ears can introduce fluctuations in interaural level differences (ILDs) important for spatial perception. While independent compression can interfere with spatial perception of sound, it does not always interfere with localization accuracy or speech identification. Here, normal-hearing listeners reported a target message played simultaneously with two spatially separated masker messages. We measured the amount of spatial separation required between the target and maskers for subjects to perform at threshold in this task. Fast, syllabic compression that was independent at the two ears increased the required spatial separation, but linking the compressors to provide identical gain to both ears (preserving ILDs) restored much of the deficit caused by fast, independent compression. Effects were less clear for slower compression. Percent-correct performance was lower with independent compression, but only for small spatial separations. These results may help explain differences in previous reports of the effect of compression on spatial perception of sound.

33 CFR 167.1700 - In Prince William Sound: General.

Code of Federal Regulations, 2010 CFR

2010-07-01

... (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1700 In Prince William Sound: General. The Prince William Sound Traffic Separation Scheme consists of four parts: Prince William Sound Traffic Separation...

Single-sensor multispeaker listening with acoustic metamaterials

PubMed Central

Xie, Yangbo; Tsai, Tsung-Han; Konneker, Adam; Popa, Bogdan-Ioan; Brady, David J.; Cummer, Steven A.

2015-01-01

Designing a “cocktail party listener” that functionally mimics the selective perception of a human auditory system has been pursued over the past decades. By exploiting acoustic metamaterials and compressive sensing, we present here a single-sensor listening device that separates simultaneous overlapping sounds from different sources. The device with a compact array of resonant metamaterials is demonstrated to distinguish three overlapping and independent sources with 96.67% correct audio recognition. Segregation of the audio signals is achieved using physical layer encoding without relying on source characteristics. This hardware approach to multichannel source separation can be applied to robust speech recognition and hearing aids and may be extended to other acoustic imaging and sensing applications. PMID:26261314

Short-Latency, Goal-Directed Movements of the Pinnae to Sounds That Produce Auditory Spatial Illusions

PubMed Central

McClaine, Elizabeth M.; Yin, Tom C. T.

2010-01-01

The precedence effect (PE) is an auditory spatial illusion whereby two identical sounds presented from two separate locations with a delay between them are perceived as a fused single sound source whose position depends on the value of the delay. By training cats using operant conditioning to look at sound sources, we have previously shown that cats experience the PE similarly to humans. For delays less than ±400 μs, cats exhibit summing localization, the perception of a “phantom” sound located between the sources. Consistent with localization dominance, for delays from 400 μs to ∼10 ms, cats orient toward the leading source location only, with little influence of the lagging source. Finally, echo threshold was reached for delays >10 ms, where cats first began to orient to the lagging source. It has been hypothesized by some that the neural mechanisms that produce facets of the PE, such as localization dominance and echo threshold, must likely occur at cortical levels. To test this hypothesis, we measured both pinnae position, which were not under any behavioral constraint, and eye position in cats and found that the pinnae orientations to stimuli that produce each of the three phases of the PE illusion was similar to the gaze responses. Although both eye and pinnae movements behaved in a manner that reflected the PE, because the pinnae moved with strikingly short latencies (∼30 ms), these data suggest a subcortical basis for the PE and that the cortex is not likely to be directly involved. PMID:19889848

Short-latency, goal-directed movements of the pinnae to sounds that produce auditory spatial illusions.

PubMed

Tollin, Daniel J; McClaine, Elizabeth M; Yin, Tom C T

2010-01-01

The precedence effect (PE) is an auditory spatial illusion whereby two identical sounds presented from two separate locations with a delay between them are perceived as a fused single sound source whose position depends on the value of the delay. By training cats using operant conditioning to look at sound sources, we have previously shown that cats experience the PE similarly to humans. For delays less than +/-400 mus, cats exhibit summing localization, the perception of a "phantom" sound located between the sources. Consistent with localization dominance, for delays from 400 mus to approximately 10 ms, cats orient toward the leading source location only, with little influence of the lagging source. Finally, echo threshold was reached for delays >10 ms, where cats first began to orient to the lagging source. It has been hypothesized by some that the neural mechanisms that produce facets of the PE, such as localization dominance and echo threshold, must likely occur at cortical levels. To test this hypothesis, we measured both pinnae position, which were not under any behavioral constraint, and eye position in cats and found that the pinnae orientations to stimuli that produce each of the three phases of the PE illusion was similar to the gaze responses. Although both eye and pinnae movements behaved in a manner that reflected the PE, because the pinnae moved with strikingly short latencies ( approximately 30 ms), these data suggest a subcortical basis for the PE and that the cortex is not likely to be directly involved.

An incentive-based source separation model for sustainable municipal solid waste management in China.

PubMed

Xu, Wanying; Zhou, Chuanbin; Lan, Yajun; Jin, Jiasheng; Cao, Aixin

2015-05-01

Municipal solid waste (MSW) management (MSWM) is most important and challenging in large urban communities. Sound community-based waste management systems normally include waste reduction and material recycling elements, often entailing the separation of recyclable materials by the residents. To increase the efficiency of source separation and recycling, an incentive-based source separation model was designed and this model was tested in 76 households in Guiyang, a city of almost three million people in southwest China. This model embraced the concepts of rewarding households for sorting organic waste, government funds for waste reduction, and introducing small recycling enterprises for promoting source separation. Results show that after one year of operation, the waste reduction rate was 87.3%, and the comprehensive net benefit under the incentive-based source separation model increased by 18.3 CNY tonne(-1) (2.4 Euros tonne(-1)), compared to that under the normal model. The stakeholder analysis (SA) shows that the centralized MSW disposal enterprises had minimum interest and may oppose the start-up of a new recycling system, while small recycling enterprises had a primary interest in promoting the incentive-based source separation model, but they had the least ability to make any change to the current recycling system. The strategies for promoting this incentive-based source separation model are also discussed in this study. © The Author(s) 2015.

Octopus Cells in the Posteroventral Cochlear Nucleus Provide the Main Excitatory Input to the Superior Paraolivary Nucleus

PubMed Central

Felix II, Richard A.; Gourévitch, Boris; Gómez-Álvarez, Marcelo; Leijon, Sara C. M.; Saldaña, Enrique; Magnusson, Anna K.

2017-01-01

Auditory streaming enables perception and interpretation of complex acoustic environments that contain competing sound sources. At early stages of central processing, sounds are segregated into separate streams representing attributes that later merge into acoustic objects. Streaming of temporal cues is critical for perceiving vocal communication, such as human speech, but our understanding of circuits that underlie this process is lacking, particularly at subcortical levels. The superior paraolivary nucleus (SPON), a prominent group of inhibitory neurons in the mammalian brainstem, has been implicated in processing temporal information needed for the segmentation of ongoing complex sounds into discrete events. The SPON requires temporally precise and robust excitatory input(s) to convey information about the steep rise in sound amplitude that marks the onset of voiced sound elements. Unfortunately, the sources of excitation to the SPON and the impact of these inputs on the behavior of SPON neurons have yet to be resolved. Using anatomical tract tracing and immunohistochemistry, we identified octopus cells in the contralateral cochlear nucleus (CN) as the primary source of excitatory input to the SPON. Cluster analysis of miniature excitatory events also indicated that the majority of SPON neurons receive one type of excitatory input. Precise octopus cell-driven onset spiking coupled with transient offset spiking make SPON responses well-suited to signal transitions in sound energy contained in vocalizations. Targets of octopus cell projections, including the SPON, are strongly implicated in the processing of temporal sound features, which suggests a common pathway that conveys information critical for perception of complex natural sounds. PMID:28620283

Speech intelligibility in complex acoustic environments in young children

NASA Astrophysics Data System (ADS)

Litovsky, Ruth

2003-04-01

While the auditory system undergoes tremendous maturation during the first few years of life, it has become clear that in complex scenarios when multiple sounds occur and when echoes are present, children's performance is significantly worse than their adult counterparts. The ability of children (3-7 years of age) to understand speech in a simulated multi-talker environment and to benefit from spatial separation of the target and competing sounds was investigated. In these studies, competing sources vary in number, location, and content (speech, modulated or unmodulated speech-shaped noise and time-reversed speech). The acoustic spaces were also varied in size and amount of reverberation. Finally, children with chronic otitis media who received binaural training were tested pre- and post-training on a subset of conditions. Results indicated the following. (1) Children experienced significantly more masking than adults, even in the simplest conditions tested. (2) When the target and competing sounds were spatially separated speech intelligibility improved, but the amount varied with age, type of competing sound, and number of competitors. (3) In a large reverberant classroom there was no benefit of spatial separation. (4) Binaural training improved speech intelligibility performance in children with otitis media. Future work includes similar studies in children with unilateral and bilateral cochlear implants. [Work supported by NIDCD, DRF, and NOHR.

Atmospheric sound propagation

NASA Technical Reports Server (NTRS)

Cook, R. K.

1969-01-01

The propagation of sound waves at infrasonic frequencies (oscillation periods 1.0 - 1000 seconds) in the atmosphere is being studied by a network of seven stations separated geographically by distances of the order of thousands of kilometers. The stations measure the following characteristics of infrasonic waves: (1) the amplitude and waveform of the incident sound pressure, (2) the direction of propagation of the wave, (3) the horizontal phase velocity, and (4) the distribution of sound wave energy at various frequencies of oscillation. Some infrasonic sources which were identified and studied include the aurora borealis, tornadoes, volcanos, gravity waves on the oceans, earthquakes, and atmospheric instability waves caused by winds at the tropopause. Waves of unknown origin seem to radiate from several geographical locations, including one in the Argentine.

Estimating the sound speed of a shallow-water marine sediment from the head wave excited by a low-flying helicopter.

PubMed

Bevans, Dieter A; Buckingham, Michael J

2017-10-01

The frequency bandwidth of the sound from a light helicopter, such as a Robinson R44, extends from about 13 Hz to 2.5 kHz. As such, the R44 has potential as a low-frequency sound source in underwater acoustics applications. To explore this idea, an experiment was conducted in shallow water off the coast of southern California in which a horizontal line of hydrophones detected the sound of an R44 hovering in an end-fire position relative to the array. Some of the helicopter sound interacted with seabed to excite the head wave in the water column. A theoretical analysis of the sound field in the water column generated by a stationary airborne source leads to an expression for the two-point horizontal coherence function of the head wave, which, apart from frequency, depends only on the sensor separation and the sediment sound speed. By matching the zero crossings of the measured and theoretical horizontal coherence functions, the sound speed in the sediment was recovered and found to take a value of 1682.42 ± 16.20 m/s. This is consistent with the sediment type at the experiment site, which is known from a previous survey to be a fine to very-fine sand.

A wavenumber approach to analysing the active control of plane waves with arrays of secondary sources

NASA Astrophysics Data System (ADS)

Elliott, Stephen J.; Cheer, Jordan; Bhan, Lam; Shi, Chuang; Gan, Woon-Seng

2018-04-01

The active control of an incident sound field with an array of secondary sources is a fundamental problem in active control. In this paper the optimal performance of an infinite array of secondary sources in controlling a plane incident sound wave is first considered in free space. An analytic solution for normal incidence plane waves is presented, indicating a clear cut-off frequency for good performance, when the separation distance between the uniformly-spaced sources is equal to a wavelength. The extent of the near field pressure close to the source array is also quantified, since this determines the positions of the error microphones in a practical arrangement. The theory is also extended to oblique incident waves. This result is then compared with numerical simulations of controlling the sound power radiated through an open aperture in a rigid wall, subject to an incident plane wave, using an array of secondary sources in the aperture. In this case the diffraction through the aperture becomes important when its size is compatible with the acoustic wavelength, in which case only a few sources are necessary for good control. When the size of the aperture is large compared to the wavelength, and diffraction is less important but more secondary sources need to be used for good control, the results then become similar to those for the free field problem with an infinite source array.

A Green Soundscape Index (GSI): The potential of assessing the perceived balance between natural sound and traffic noise.

PubMed

Kogan, Pablo; Arenas, Jorge P; Bermejo, Fernando; Hinalaf, María; Turra, Bruno

2018-06-13

Urban soundscapes are dynamic and complex multivariable environmental systems. Soundscapes can be organized into three main entities containing the multiple variables: Experienced Environment (EE), Acoustic Environment (AE), and Extra-Acoustic Environment (XE). This work applies a multidimensional and synchronic data-collecting methodology at eight urban environments in the city of Córdoba, Argentina. The EE was assessed by means of surveys, the AE by acoustic measurements and audio recordings, and the XE by photos, video, and complementary sources. In total, 39 measurement locations were considered, where data corresponding to 61 AE and 203 EE were collected. Multivariate analysis and GIS techniques were used for data processing. The types of sound sources perceived, and their extents make up part of the collected variables that belong to the EE, i.e. traffic, people, natural sounds, and others. Sources explaining most of the variance were traffic noise and natural sounds. Thus, a Green Soundscape Index (GSI) is defined here as the ratio of the perceived extents of natural sounds to traffic noise. Collected data were divided into three ranges according to GSI value: 1) perceptual predominance of traffic noise, 2) balanced perception, and 3) perceptual predominance of natural sounds. For each group, three additional variables from the EE and three from the AE were applied, which reported significant differences, especially between ranges 1 and 2 with 3. These results confirm the key role of perceiving natural sounds in a town environment and also support the proposal of a GSI as a valuable indicator to classify urban soundscapes. In addition, the collected GSI-related data significantly helps to assess the overall soundscape. It is noted that this proposed simple perceptual index not only allows one to assess and classify urban soundscapes but also contributes greatly toward a technique for separating environmental sound sources. Copyright © 2018 Elsevier B.V. All rights reserved.

Spatial selective attention in a complex auditory environment such as polyphonic music.

PubMed

Saupe, Katja; Koelsch, Stefan; Rübsamen, Rudolf

2010-01-01

To investigate the influence of spatial information in auditory scene analysis, polyphonic music (three parts in different timbres) was composed and presented in free field. Each part contained large falling interval jumps in the melody and the task of subjects was to detect these events in one part ("target part") while ignoring the other parts. All parts were either presented from the same location (0 degrees; overlap condition) or from different locations (-28 degrees, 0 degrees, and 28 degrees or -56 degrees, 0 degrees, and 56 degrees in the azimuthal plane), with the target part being presented either at 0 degrees or at one of the right-sided locations. Results showed that spatial separation of 28 degrees was sufficient for a significant improvement in target detection (i.e., in the detection of large interval jumps) compared to the overlap condition, irrespective of the position (frontal or right) of the target part. A larger spatial separation of the parts resulted in further improvements only if the target part was lateralized. These data support the notion of improvement in the suppression of interfering signals with spatial sound source separation. Additionally, the data show that the position of the relevant sound source influences auditory performance.

A Analysis of the Low Frequency Sound Field in Non-Rectangular Enclosures Using the Finite Element Method.

NASA Astrophysics Data System (ADS)

Geddes, Earl Russell

The details of the low frequency sound field for a rectangular room can be studied by the use of an established analytic technique--separation of variables. The solution is straightforward and the results are well-known. A non -rectangular room has boundary conditions which are not separable and therefore other solution techniques must be used. This study shows that the finite element method can be adapted for use in the study of sound fields in arbitrary shaped enclosures. The finite element acoustics problem is formulated and the modification of a standard program, which is necessary for solving acoustic field problems, is examined. The solution of the semi-non-rectangular room problem (one where the floor and ceiling remain parallel) is carried out by a combined finite element/separation of variables approach. The solution results are used to construct the Green's function for the low frequency sound field in five rooms (or data cases): (1) a rectangular (Louden) room; (2) The smallest wall of the Louden room canted 20 degrees from normal; (3) The largest wall of the Louden room canted 20 degrees from normal; (4) both the largest and the smallest walls are canted 20 degrees; and (5) a five-sided room variation of Case 4. Case 1, the rectangular room was calculated using both the finite element method and the separation of variables technique. The results for the two methods are compared in order to access the accuracy of the finite element method models. The modal damping coefficient are calculated and the results examined. The statistics of the source and receiver average normalized RMS P('2) responses in the 80 Hz, 100 Hz, and 125 Hz one-third octave bands are developed. The receiver averaged pressure response is developed to determine the effect of the source locations on the response. Twelve source locations are examined and the results tabulated for comparison. The effect of a finite sized source is looked at briefly. Finally, the standard deviation of the spatial pressure response is studied. The results for this characteristic show that it not significantly different in any of the rooms. The conclusions of the study are that only the frequency variations of the pressure response are affected by a room's shape. Further, in general, the simplest modification of a rectangular room (i.e., changing the angle of only one of the smallest walls), produces the most pronounced decrease of the pressure response variations in the low frequency region.

Development of a directivity controlled piezoelectric transducer for sound reproduction

NASA Astrophysics Data System (ADS)

Bédard, Magella; Berry, Alain

2005-04-01

One of the inherent limitations of loudspeaker systems in audio reproduction is their inability to reproduce the possibly complex acoustic directivity patterns of real sound sources. For music reproduction for example, it may be desirable to separate diffuse field and direct sound components and project them with different directivity patterns. Because of their properties, poly (vinylidene fluoride) (PVDF) films offer lot of advantages for the development of electroacoustic transducers. A system of piezoelectric transducers made with PVDF that show a controllable directivity was developed. A cylindrical omnidirectional piezoelectric transducer is used to produce an ambient field, and a piezoelectric transducers system, consisting of a series of curved sources placed around a cylinder frame, is used to produce a sound field with a given directivity. To develop the system, a numerical model was generated with ANSYS Multiphysics TM8.1 and used to calculate the mechanical response of the piezoelectric transducer. The acoustic radiation of the driver was then computed using the Kirchoff-Helmoltz theorem. Numerical and experimental results of the mechanical and acoustical response of the system will be shown.

Acoustic investigation of wall jet over a backward-facing step using a microphone phased array

NASA Astrophysics Data System (ADS)

Perschke, Raimund F.; Ramachandran, Rakesh C.; Raman, Ganesh

2015-02-01

The acoustic properties of a wall jet over a hard-walled backward-facing step of aspect ratios 6, 3, 2, and 1.5 are studied using a 24-channel microphone phased array at Mach numbers up to M=0.6. The Reynolds number based on inflow velocity and step height assumes values from Reh = 3.0 ×104 to 7.2 ×105. Flow without and with side walls is considered. The experimental setup is open in the wall-normal direction and the expansion ratio is effectively 1. In case of flow through a duct, symmetry of the flow in the spanwise direction is lost downstream of separation at all but the largest aspect ratio as revealed by oil paint flow visualization. Hydrodynamic scattering of turbulence from the trailing edge of the step contributes significantly to the radiated sound. Reflection of acoustic waves from the bottom plate results in a modulation of power spectral densities. Acoustic source localization has been conducted using a 24-channel microphone phased array. Convective mean-flow effects on the apparent source origin have been assessed by placing a loudspeaker underneath a perforated flat plate and evaluating the displacement of the beamforming peak with inflow Mach number. Two source mechanisms are found near the step. One is due to interaction of the turbulent wall jet with the convex edge of the step. Free-stream turbulence sound is found to be peaked downstream of the step. Presence of the side walls increases free-stream sound. Results of the flow visualization are correlated with acoustic source maps. Trailing-edge sound and free-stream turbulence sound can be discriminated using source localization.

33 CFR 167.1700 - In Prince William Sound: General.

Code of Federal Regulations, 2014 CFR

2014-07-01

... 33 Navigation and Navigable Waters 2 2014-07-01 2014-07-01 false In Prince William Sound: General... Schemes and Precautionary Areas Pacific West Coast § 167.1700 In Prince William Sound: General. The Prince William Sound Traffic Separation Scheme consists of four parts: Prince William Sound Traffic Separation...

33 CFR 167.1700 - In Prince William Sound: General.

Code of Federal Regulations, 2013 CFR

2013-07-01

... 33 Navigation and Navigable Waters 2 2013-07-01 2013-07-01 false In Prince William Sound: General... Schemes and Precautionary Areas Pacific West Coast § 167.1700 In Prince William Sound: General. The Prince William Sound Traffic Separation Scheme consists of four parts: Prince William Sound Traffic Separation...

33 CFR 167.1700 - In Prince William Sound: General.

Code of Federal Regulations, 2012 CFR

2012-07-01

... 33 Navigation and Navigable Waters 2 2012-07-01 2012-07-01 false In Prince William Sound: General... Schemes and Precautionary Areas Pacific West Coast § 167.1700 In Prince William Sound: General. The Prince William Sound Traffic Separation Scheme consists of four parts: Prince William Sound Traffic Separation...

33 CFR 167.1700 - In Prince William Sound: General.

Code of Federal Regulations, 2011 CFR

2011-07-01

... 33 Navigation and Navigable Waters 2 2011-07-01 2011-07-01 false In Prince William Sound: General... Schemes and Precautionary Areas Pacific West Coast § 167.1700 In Prince William Sound: General. The Prince William Sound Traffic Separation Scheme consists of four parts: Prince William Sound Traffic Separation...

Separation and reconstruction of high pressure water-jet reflective sound signal based on ICA

NASA Astrophysics Data System (ADS)

Yang, Hongtao; Sun, Yuling; Li, Meng; Zhang, Dongsu; Wu, Tianfeng

2011-12-01

The impact of high pressure water-jet on the different materials target will produce different reflective mixed sound. In order to reconstruct the reflective sound signals distribution on the linear detecting line accurately and to separate the environment noise effectively, the mixed sound signals acquired by linear mike array were processed by ICA. The basic principle of ICA and algorithm of FASTICA were described in detail. The emulation experiment was designed. The environment noise signal was simulated by using band-limited white noise and the reflective sound signal was simulated by using pulse signal. The reflective sound signal attenuation produced by the different distance transmission was simulated by weighting the sound signal with different contingencies. The mixed sound signals acquired by linear mike array were synthesized by using the above simulated signals and were whitened and separated by ICA. The final results verified that the environment noise separation and the reconstruction of the detecting-line sound distribution can be realized effectively.

Auditory scene analysis in school-aged children with developmental language disorders

PubMed Central

Sussman, E.; Steinschneider, M.; Lee, W.; Lawson, K.

2014-01-01

Natural sound environments are dynamic, with overlapping acoustic input originating from simultaneously active sources. A key function of the auditory system is to integrate sensory inputs that belong together and segregate those that come from different sources. We hypothesized that this skill is impaired in individuals with phonological processing difficulties. There is considerable disagreement about whether phonological impairments observed in children with developmental language disorders can be attributed to specific linguistic deficits or to more general acoustic processing deficits. However, most tests of general auditory abilities have been conducted with a single set of sounds. We assessed the ability of school-aged children (7–15 years) to parse complex auditory non-speech input, and determined whether the presence of phonological processing impairments was associated with stream perception performance. A key finding was that children with language impairments did not show the same developmental trajectory for stream perception as typically developing children. In addition, children with language impairments required larger frequency separations between sounds to hear distinct streams compared to age-matched peers. Furthermore, phonological processing ability was a significant predictor of stream perception measures, but only in the older age groups. No such association was found in the youngest children. These results indicate that children with language impairments have difficulty parsing speech streams, or identifying individual sound events when there are competing sound sources. We conclude that language group differences may in part reflect fundamental maturational disparities in the analysis of complex auditory scenes. PMID:24548430

Assessment and control design for steam vent noise in an oil refinery.

PubMed

Monazzam, Mohammad Reza; Golmohammadi, Rostam; Nourollahi, Maryam; Momen Bellah Fard, Samaneh

2011-06-13

Noise is one of the most important harmful agents in work environment. Noise pollution in oil refinery industries is related to workers' health. This study aimed to determine the overall noise pollution of an oil refinery operation and its frequency analysis to determine the control plan for a vent noise in these industries. This experimental study performed in control unit of Tehran Oil Refinery in 2008. To determine the noise distributions, environmental noise measurements were carried out by lattice method according to basic information and technical process. The sound pressure level and frequency distribution was measured for each study sources subject separately was performed individually. According to the vent's specification, the measured steam noise characteristics reviewed and compared to the theoretical results of steam noise estimation. Eventually, a double expansion muffler was designed. Data analysis and graphical design were carried out using Excel software. The results of environmental noise measurements indicated that the level of sound pressure was above the national permitted level (85 dB (A)). The Mean level of sound pressure of the studied steam jet was 90.3 dB (L). The results of noise frequency analysis for the steam vents showed that the dominant frequency was 4000 Hz. To obtain 17 dB noise reductions, a double chamber aluminum muffler with 500 mm length and 200 mm diameter consisting pipe drilled was designed. The characteristics of steam vent noise were separated from other sources, a double expansion muffler was designed using a new method based on the level of steam noise, and principle sound frequency, a double expansion muffler was designed.

Sound stream segregation: a neuromorphic approach to solve the “cocktail party problem” in real-time

PubMed Central

Thakur, Chetan Singh; Wang, Runchun M.; Afshar, Saeed; Hamilton, Tara J.; Tapson, Jonathan C.; Shamma, Shihab A.; van Schaik, André

2015-01-01

The human auditory system has the ability to segregate complex auditory scenes into a foreground component and a background, allowing us to listen to specific speech sounds from a mixture of sounds. Selective attention plays a crucial role in this process, colloquially known as the “cocktail party effect.” It has not been possible to build a machine that can emulate this human ability in real-time. Here, we have developed a framework for the implementation of a neuromorphic sound segregation algorithm in a Field Programmable Gate Array (FPGA). This algorithm is based on the principles of temporal coherence and uses an attention signal to separate a target sound stream from background noise. Temporal coherence implies that auditory features belonging to the same sound source are coherently modulated and evoke highly correlated neural response patterns. The basis for this form of sound segregation is that responses from pairs of channels that are strongly positively correlated belong to the same stream, while channels that are uncorrelated or anti-correlated belong to different streams. In our framework, we have used a neuromorphic cochlea as a frontend sound analyser to extract spatial information of the sound input, which then passes through band pass filters that extract the sound envelope at various modulation rates. Further stages include feature extraction and mask generation, which is finally used to reconstruct the targeted sound. Using sample tonal and speech mixtures, we show that our FPGA architecture is able to segregate sound sources in real-time. The accuracy of segregation is indicated by the high signal-to-noise ratio (SNR) of the segregated stream (90, 77, and 55 dB for simple tone, complex tone, and speech, respectively) as compared to the SNR of the mixture waveform (0 dB). This system may be easily extended for the segregation of complex speech signals, and may thus find various applications in electronic devices such as for sound segregation and speech recognition. PMID:26388721

Sound stream segregation: a neuromorphic approach to solve the "cocktail party problem" in real-time.

PubMed

Thakur, Chetan Singh; Wang, Runchun M; Afshar, Saeed; Hamilton, Tara J; Tapson, Jonathan C; Shamma, Shihab A; van Schaik, André

2015-01-01

The human auditory system has the ability to segregate complex auditory scenes into a foreground component and a background, allowing us to listen to specific speech sounds from a mixture of sounds. Selective attention plays a crucial role in this process, colloquially known as the "cocktail party effect." It has not been possible to build a machine that can emulate this human ability in real-time. Here, we have developed a framework for the implementation of a neuromorphic sound segregation algorithm in a Field Programmable Gate Array (FPGA). This algorithm is based on the principles of temporal coherence and uses an attention signal to separate a target sound stream from background noise. Temporal coherence implies that auditory features belonging to the same sound source are coherently modulated and evoke highly correlated neural response patterns. The basis for this form of sound segregation is that responses from pairs of channels that are strongly positively correlated belong to the same stream, while channels that are uncorrelated or anti-correlated belong to different streams. In our framework, we have used a neuromorphic cochlea as a frontend sound analyser to extract spatial information of the sound input, which then passes through band pass filters that extract the sound envelope at various modulation rates. Further stages include feature extraction and mask generation, which is finally used to reconstruct the targeted sound. Using sample tonal and speech mixtures, we show that our FPGA architecture is able to segregate sound sources in real-time. The accuracy of segregation is indicated by the high signal-to-noise ratio (SNR) of the segregated stream (90, 77, and 55 dB for simple tone, complex tone, and speech, respectively) as compared to the SNR of the mixture waveform (0 dB). This system may be easily extended for the segregation of complex speech signals, and may thus find various applications in electronic devices such as for sound segregation and speech recognition.

33 CFR 167.1320 - In Puget Sound and its approaches: General.

Code of Federal Regulations, 2011 CFR

2011-07-01

... SECURITY (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1320 In Puget Sound and its approaches: General. The traffic separation scheme in Puget Sound and its approaches consists of three parts: Rosario...

Thermal management of thermoacoustic sound projectors using a free-standing carbon nanotube aerogel sheet as a heat source.

PubMed

Aliev, Ali E; Mayo, Nathanael K; Baughman, Ray H; Avirovik, Dragan; Priya, Shashank; Zarnetske, Michael R; Blottman, John B

2014-10-10

Carbon nanotube (CNT) aerogel sheets produce smooth-spectra sound over a wide frequency range (1-10(5) Hz) by means of thermoacoustic (TA) sound generation. Protective encapsulation of CNT sheets in inert gases between rigid vibrating plates provides resonant features for the TA sound projector and attractive performance at needed low frequencies. Energy conversion efficiencies in air of 2% and 10% underwater, which can be enhanced by further increasing the modulation temperature. Using a developed method for accurate temperature measurements for the thin aerogel CNT sheets, heat dissipation processes, failure mechanisms, and associated power densities are investigated for encapsulated multilayered CNT TA heaters and related to the thermal diffusivity distance when sheet layers are separated. Resulting thermal management methods for high applied power are discussed and deployed to construct efficient and tunable underwater sound projector for operation at relatively low frequencies, 10 Hz-10 kHz. The optimal design of these TA projectors for high-power SONAR arrays is discussed.

Source splitting via the point source method

NASA Astrophysics Data System (ADS)

Potthast, Roland; Fazi, Filippo M.; Nelson, Philip A.

2010-04-01

We introduce a new algorithm for source identification and field splitting based on the point source method (Potthast 1998 A point-source method for inverse acoustic and electromagnetic obstacle scattering problems IMA J. Appl. Math. 61 119-40, Potthast R 1996 A fast new method to solve inverse scattering problems Inverse Problems 12 731-42). The task is to separate the sound fields uj, j = 1, ..., n of n \\in \\mathbb {N} sound sources supported in different bounded domains G1, ..., Gn in \\mathbb {R}^3 from measurements of the field on some microphone array—mathematically speaking from the knowledge of the sum of the fields u = u1 + sdotsdotsdot + un on some open subset Λ of a plane. The main idea of the scheme is to calculate filter functions g_1, \\ldots, g_n, n\\in \\mathbb {N} , to construct uell for ell = 1, ..., n from u|Λ in the form u_{\\ell }(x) = \\int _{\\Lambda } g_{\\ell,x}(y) u(y) {\\,\\rm d}s(y), \\qquad \\ell =1,\\ldots, n. We will provide the complete mathematical theory for the field splitting via the point source method. In particular, we describe uniqueness, solvability of the problem and convergence and stability of the algorithm. In the second part we describe the practical realization of the splitting for real data measurements carried out at the Institute for Sound and Vibration Research at Southampton, UK. A practical demonstration of the original recording and the splitting results for real data is available online.

Study of Noise-Certification Standards for Aircraft Engines. Volume 2. Procedures for Measuring Far Field Sound Pressure Levels around an Outdoor Jet-Engine Test Stand.

DTIC Science & Technology

1983-06-01

60 References ........................................................... 79 AccesSqlon For NTIS rFA&I r"!’ TAU U: .,P Dist r A. -. S iv...separate exhaust nozzles for discharge of fan and turbine exhaust flows (e.g., JT15D, TFE731 , ALF-502, CF34, JT3D, CFM56, RB.211, CF6, JT9D, and PW2037...minimum radial distance from the effective source of sound at 40 Hz should then be approxi- mately 69 m. At 60 Hz, the minimum radial distance should be

Estimated communication range of social sounds used by bottlenose dolphins (Tursiops truncatus).

PubMed

Quintana-Rizzo, Ester; Mann, David A; Wells, Randall S

2006-09-01

Bottlenose dolphins, Tursiops truncatus, exhibit flexible associations in which the compositions of groups change frequently. We investigated the potential distances over which female dolphins and their dependent calves could remain in acoustic contact. We quantified the propagation of sounds in the frequency range of typical dolphin whistles in shallow water areas and channels of Sarasota Bay, Florida. Our results indicated that detection range was noise limited as opposed to being limited by hearing sensitivity. Sounds were attenuated to a greater extent in areas with seagrass than any other habitat. Estimates of active space of whistles showed that in seagrass shallow water areas, low-frequency whistles (7-13 kHz) with a 165 dB source level could be heard by dolphins at 487 m. In shallow areas with a mud bottom, all whistle frequency components of the same whistle could be heard by dolphins travel up to 2 km. In channels, high-frequency whistles (13-19 kHz) could be detectable potentially over a much longer distance (> 20 km). Our findings indicate that the communication range of social sounds likely exceeds the mean separation distances between females and their calves. Ecological pressures might play an important role in determining the separation distances within communication range.

Similar Sound Separation and Cumulative Introduction in Learning Letter-Sound Correspondences

ERIC Educational Resources Information Center

Carnine, Douglas W.

1976-01-01

Since similarity inhibits learning, children will have greater difficulty learning responses that are more similar to each other; the instructional procedure of separating similar sounds was evaluated in two experiments. (DMT)

Turbofan noise generation. Volume 2: Computer programs

NASA Technical Reports Server (NTRS)

Ventres, C. S.; Theobald, M. A.; Mark, W. D.

1982-01-01

The use of a package of computer programs developed to calculate the in duct acoustic mods excited by a fan/stator stage operating at subsonic tip speed is described. The following three noise source mechanisms are included: (1) sound generated by the rotor blades interacting with turbulence ingested into, or generated within, the inlet duct; (2) sound generated by the stator vanes interacting with the turbulent wakes of the rotor blades; and (3) sound generated by the stator vanes interacting with the velocity deficits in the mean wakes of the rotor blades. The computations for three different noise mechanisms are coded as three separate computer program packages. The computer codes are described by means of block diagrams, tables of data and variables, and example program executions; FORTRAN listings are included.

Turbofan noise generation. Volume 2: Computer programs

NASA Astrophysics Data System (ADS)

Ventres, C. S.; Theobald, M. A.; Mark, W. D.

1982-07-01

The use of a package of computer programs developed to calculate the in duct acoustic mods excited by a fan/stator stage operating at subsonic tip speed is described. The following three noise source mechanisms are included: (1) sound generated by the rotor blades interacting with turbulence ingested into, or generated within, the inlet duct; (2) sound generated by the stator vanes interacting with the turbulent wakes of the rotor blades; and (3) sound generated by the stator vanes interacting with the velocity deficits in the mean wakes of the rotor blades. The computations for three different noise mechanisms are coded as three separate computer program packages. The computer codes are described by means of block diagrams, tables of data and variables, and example program executions; FORTRAN listings are included.

Judging sound rotation when listeners and sounds rotate: Sound source localization is a multisystem process.

PubMed

Yost, William A; Zhong, Xuan; Najam, Anbar

2015-11-01

In four experiments listeners were rotated or were stationary. Sounds came from a stationary loudspeaker or rotated from loudspeaker to loudspeaker around an azimuth array. When either sounds or listeners rotate the auditory cues used for sound source localization change, but in the everyday world listeners perceive sound rotation only when sounds rotate not when listeners rotate. In the everyday world sound source locations are referenced to positions in the environment (a world-centric reference system). The auditory cues for sound source location indicate locations relative to the head (a head-centric reference system), not locations relative to the world. This paper deals with a general hypothesis that the world-centric location of sound sources requires the auditory system to have information about auditory cues used for sound source location and cues about head position. The use of visual and vestibular information in determining rotating head position in sound rotation perception was investigated. The experiments show that sound rotation perception when sources and listeners rotate was based on acoustic, visual, and, perhaps, vestibular information. The findings are consistent with the general hypotheses and suggest that sound source localization is not based just on acoustics. It is a multisystem process.

33 CFR 167.1320 - In Puget Sound and its approaches: General.

Code of Federal Regulations, 2014 CFR

2014-07-01

... 33 Navigation and Navigable Waters 2 2014-07-01 2014-07-01 false In Puget Sound and its approaches... Separation Schemes and Precautionary Areas Pacific West Coast § 167.1320 In Puget Sound and its approaches: General. The traffic separation scheme in Puget Sound and its approaches consists of three parts: Rosario...

33 CFR 167.1320 - In Puget Sound and its approaches: General.

Code of Federal Regulations, 2013 CFR

2013-07-01

... 33 Navigation and Navigable Waters 2 2013-07-01 2013-07-01 false In Puget Sound and its approaches... Separation Schemes and Precautionary Areas Pacific West Coast § 167.1320 In Puget Sound and its approaches: General. The traffic separation scheme in Puget Sound and its approaches consists of three parts: Rosario...

33 CFR 167.1320 - In Puget Sound and its approaches: General.

Code of Federal Regulations, 2012 CFR

2012-07-01

... 33 Navigation and Navigable Waters 2 2012-07-01 2012-07-01 false In Puget Sound and its approaches... Separation Schemes and Precautionary Areas Pacific West Coast § 167.1320 In Puget Sound and its approaches: General. The traffic separation scheme in Puget Sound and its approaches consists of three parts: Rosario...

Examination of propeller sound production using large eddy simulation

NASA Astrophysics Data System (ADS)

Keller, Jacob; Kumar, Praveen; Mahesh, Krishnan

2018-06-01

The flow field of a five-bladed marine propeller operating at design condition, obtained using large eddy simulation, is used to calculate the resulting far-field sound. The results of three acoustic formulations are compared, and the effects of the underlying assumptions are quantified. The integral form of the Ffowcs-Williams and Hawkings (FW-H) equation is solved on the propeller surface, which is discretized into a collection of N radial strips. Further assumptions are made to reduce FW-H to a Curle acoustic analogy and a point-force dipole model. Results show that although the individual blades are strongly tonal in the rotor plane, the propeller is acoustically compact at low frequency and the tonal sound interferes destructively in the far field. The propeller is found to be acoustically compact for frequencies up to 100 times the rotation rate. The overall far-field acoustic signature is broadband. Locations of maximum sound of the propeller occur along the axis of rotation both up and downstream. The propeller hub is found to be a source of significant sound to observers in the rotor plane, due to flow separation and interaction with the blade-root wakes. The majority of the propeller sound is generated by localized unsteadiness at the blade tip, which is caused by shedding of the tip vortex. Tonal blade sound is found to be caused by the periodic motion of the loaded blades. Turbulence created in the blade boundary layer is convected past the blade trailing edge leading to generation of broadband noise along the blade. Acoustic energy is distributed among higher frequencies as local Reynolds number increases radially along the blades. Sound source correlation and spectra are examined in the context of noise modeling.

A method for environmental acoustic analysis improvement based on individual evaluation of common sources in urban areas.

PubMed

López-Pacheco, María G; Sánchez-Fernández, Luis P; Molina-Lozano, Herón

2014-01-15

Noise levels of common sources such as vehicles, whistles, sirens, car horns and crowd sounds are mixed in urban soundscapes. Nowadays, environmental acoustic analysis is performed based on mixture signals recorded by monitoring systems. These mixed signals make it difficult for individual analysis which is useful in taking actions to reduce and control environmental noise. This paper aims at separating, individually, the noise source from recorded mixtures in order to evaluate the noise level of each estimated source. A method based on blind deconvolution and blind source separation in the wavelet domain is proposed. This approach provides a basis to improve results obtained in monitoring and analysis of common noise sources in urban areas. The method validation is through experiments based on knowledge of the predominant noise sources in urban soundscapes. Actual recordings of common noise sources are used to acquire mixture signals using a microphone array in semi-controlled environments. The developed method has demonstrated great performance improvements in identification, analysis and evaluation of common urban sources. © 2013 Elsevier B.V. All rights reserved.

Mathematically trivial control of sound using a parametric beam focusing source.

PubMed

Tanaka, Nobuo; Tanaka, Motoki

2011-01-01

By exploiting a case regarded as trivial, this paper presents global active noise control using a parametric beam focusing source (PBFS). As with a dipole model, one is used for a primary sound source and the other for a control sound source, the control effect for minimizing a total acoustic power depends on the distance between the two. When the distance becomes zero, the total acoustic power becomes null, hence nothing less than a trivial case. Because of the constraints in practice, there exist difficulties in placing a control source close enough to a primary source. However, by projecting a sound beam of a parametric array loudspeaker onto the target sound source (primary source), a virtual sound source may be created on the target sound source, thereby enabling the collocation of the sources. In order to further ensure feasibility of the trivial case, a PBFS is then introduced in an effort to meet the size of the two sources. Reflected sound wave of the PBFS, which is tantamount to the virtual sound source output, aims to suppress the primary sound. Finally, a numerical analysis as well as an experiment is conducted, verifying the validity of the proposed methodology.

A Flexible 360-Degree Thermal Sound Source Based on Laser Induced Graphene

PubMed Central

Tao, Lu-Qi; Liu, Ying; Ju, Zhen-Yi; Tian, He; Xie, Qian-Yi; Yang, Yi; Ren, Tian-Ling

2016-01-01

A flexible sound source is essential in a whole flexible system. It’s hard to integrate a conventional sound source based on a piezoelectric part into a whole flexible system. Moreover, the sound pressure from the back side of a sound source is usually weaker than that from the front side. With the help of direct laser writing (DLW) technology, the fabrication of a flexible 360-degree thermal sound source becomes possible. A 650-nm low-power laser was used to reduce the graphene oxide (GO). The stripped laser induced graphene thermal sound source was then attached to the surface of a cylindrical bottle so that it could emit sound in a 360-degree direction. The sound pressure level and directivity of the sound source were tested, and the results were in good agreement with the theoretical results. Because of its 360-degree sound field, high flexibility, high efficiency, low cost, and good reliability, the 360-degree thermal acoustic sound source will be widely applied in consumer electronics, multi-media systems, and ultrasonic detection and imaging. PMID:28335239

Neural Networks for Segregation of Multiple Objects: Visual Figure-Ground Separation and Auditory Pitch Perception.

NASA Astrophysics Data System (ADS)

Wyse, Lonce

An important component of perceptual object recognition is the segmentation into coherent perceptual units of the "blooming buzzing confusion" that bombards the senses. The work presented herein develops neural network models of some key processes of pre-attentive vision and audition that serve this goal. A neural network model, called an FBF (Feature -Boundary-Feature) network, is proposed for automatic parallel separation of multiple figures from each other and their backgrounds in noisy images. Figure-ground separation is accomplished by iterating operations of a Boundary Contour System (BCS) that generates a boundary segmentation of a scene, and a Feature Contour System (FCS) that compensates for variable illumination and fills-in surface properties using boundary signals. A key new feature is the use of the FBF filling-in process for the figure-ground separation of connected regions, which are subsequently more easily recognized. The new CORT-X 2 model is a feed-forward version of the BCS that is designed to detect, regularize, and complete boundaries in up to 50 percent noise. It also exploits the complementary properties of on-cells and off -cells to generate boundary segmentations and to compensate for boundary gaps during filling-in. In the realm of audition, many sounds are dominated by energy at integer multiples, or "harmonics", of a fundamental frequency. For such sounds (e.g., vowels in speech), the individual frequency components fuse, so that they are perceived as one sound source with a pitch at the fundamental frequency. Pitch is integral to separating auditory sources, as well as to speaker identification and speech understanding. A neural network model of pitch perception called SPINET (SPatial PItch NETwork) is developed and used to simulate a broader range of perceptual data than previous spectral models. The model employs a bank of narrowband filters as a simple model of basilar membrane mechanics, spectral on-center off-surround competitive interactions, and a "harmonic sieve" mechanism whereby the strength of a pitch depends only on spectral regions near harmonics. The model is evaluated using data involving mistuned components, shifted harmonics, complex tones with varying phase relationships, and continuous spectra such as rippled noise and narrow noise bands.

Promoting the perception of two and three concurrent sound objects: An event-related potential study.

PubMed

Kocsis, Zsuzsanna; Winkler, István; Bendixen, Alexandra; Alain, Claude

2016-09-01

The auditory environment typically comprises several simultaneously active sound sources. In contrast to the perceptual segregation of two concurrent sounds, the perception of three simultaneous sound objects has not yet been studied systematically. We conducted two experiments in which participants were presented with complex sounds containing sound segregation cues (mistuning, onset asynchrony, differences in frequency or amplitude modulation or in sound location), which were set up to promote the perceptual organization of the tonal elements into one, two, or three concurrent sounds. In Experiment 1, listeners indicated whether they heard one, two, or three concurrent sounds. In Experiment 2, participants watched a silent subtitled movie while EEG was recorded to extract the object-related negativity (ORN) component of the event-related potential. Listeners predominantly reported hearing two sounds when the segregation promoting manipulations were applied to the same tonal element. When two different tonal elements received manipulations promoting them to be heard as separate auditory objects, participants reported hearing two and three concurrent sounds objects with equal probability. The ORN was elicited in most conditions; sounds that included the amplitude- or the frequency-modulation cue generated the smallest ORN amplitudes. Manipulating two different tonal elements yielded numerically and often significantly smaller ORNs than the sum of the ORNs elicited when the same cues were applied on a single tonal element. These results suggest that ORN reflects the presence of multiple concurrent sounds, but not their number. The ORN results are compatible with the horse-race principle of combining different cues of concurrent sound segregation. Copyright © 2016 Elsevier B.V. All rights reserved.

Characterizing, synthesizing, and/or canceling out acoustic signals from sound sources

DOEpatents

Holzrichter, John F [Berkeley, CA; Ng, Lawrence C [Danville, CA

2007-03-13

A system for characterizing, synthesizing, and/or canceling out acoustic signals from inanimate and animate sound sources. Electromagnetic sensors monitor excitation sources in sound producing systems, such as animate sound sources such as the human voice, or from machines, musical instruments, and various other structures. Acoustical output from these sound producing systems is also monitored. From such information, a transfer function characterizing the sound producing system is generated. From the transfer function, acoustical output from the sound producing system may be synthesized or canceled. The systems disclosed enable accurate calculation of transfer functions relating specific excitations to specific acoustical outputs. Knowledge of such signals and functions can be used to effect various sound replication, sound source identification, and sound cancellation applications.

On the Computation of Sound by Large-Eddy Simulations

NASA Technical Reports Server (NTRS)

Piomelli, Ugo; Streett, Craig L.; Sarkar, Sutanu

1997-01-01

The effect of the small scales on the source term in Lighthill's acoustic analogy is investigated, with the objective of determining the accuracy of large-eddy simulations when applied to studies of flow-generated sound. The distribution of the turbulent quadrupole is predicted accurately, if models that take into account the trace of the SGS stresses are used. Its spatial distribution is also correct, indicating that the low-wave-number (or frequency) part of the sound spectrum can be predicted well by LES. Filtering, however, removes the small-scale fluctuations that contribute significantly to the higher derivatives in space and time of Lighthill's stress tensor T(sub ij). The rms fluctuations of the filtered derivatives are substantially lower than those of the unfiltered quantities. The small scales, however, are not strongly correlated, and are not expected to contribute significantly to the far-field sound; separate modeling of the subgrid-scale density fluctuations might, however, be required in some configurations.

Directional Hearing and Sound Source Localization in Fishes.

PubMed

Sisneros, Joseph A; Rogers, Peter H

2016-01-01

Evidence suggests that the capacity for sound source localization is common to mammals, birds, reptiles, and amphibians, but surprisingly it is not known whether fish locate sound sources in the same manner (e.g., combining binaural and monaural cues) or what computational strategies they use for successful source localization. Directional hearing and sound source localization in fishes continues to be important topics in neuroethology and in the hearing sciences, but the empirical and theoretical work on these topics have been contradictory and obscure for decades. This chapter reviews the previous behavioral work on directional hearing and sound source localization in fishes including the most recent experiments on sound source localization by the plainfin midshipman fish (Porichthys notatus), which has proven to be an exceptional species for fish studies of sound localization. In addition, the theoretical models of directional hearing and sound source localization for fishes are reviewed including a new model that uses a time-averaged intensity approach for source localization that has wide applicability with regard to source type, acoustic environment, and time waveform.

Performance of an open-source heart sound segmentation algorithm on eight independent databases.

PubMed

Liu, Chengyu; Springer, David; Clifford, Gari D

2017-08-01

Heart sound segmentation is a prerequisite step for the automatic analysis of heart sound signals, facilitating the subsequent identification and classification of pathological events. Recently, hidden Markov model-based algorithms have received increased interest due to their robustness in processing noisy recordings. In this study we aim to evaluate the performance of the recently published logistic regression based hidden semi-Markov model (HSMM) heart sound segmentation method, by using a wider variety of independently acquired data of varying quality. Firstly, we constructed a systematic evaluation scheme based on a new collection of heart sound databases, which we assembled for the PhysioNet/CinC Challenge 2016. This collection includes a total of more than 120 000 s of heart sounds recorded from 1297 subjects (including both healthy subjects and cardiovascular patients) and comprises eight independent heart sound databases sourced from multiple independent research groups around the world. Then, the HSMM-based segmentation method was evaluated using the assembled eight databases. The common evaluation metrics of sensitivity, specificity, accuracy, as well as the [Formula: see text] measure were used. In addition, the effect of varying the tolerance window for determining a correct segmentation was evaluated. The results confirm the high accuracy of the HSMM-based algorithm on a separate test dataset comprised of 102 306 heart sounds. An average [Formula: see text] score of 98.5% for segmenting S1 and systole intervals and 97.2% for segmenting S2 and diastole intervals were observed. The [Formula: see text] score was shown to increases with an increases in the tolerance window size, as expected. The high segmentation accuracy of the HSMM-based algorithm on a large database confirmed the algorithm's effectiveness. The described evaluation framework, combined with the largest collection of open access heart sound data, provides essential resources for evaluators who need to test their algorithms with realistic data and share reproducible results.

Spectral analysis methods for vehicle interior vibro-acoustics identification

NASA Astrophysics Data System (ADS)

Hosseini Fouladi, Mohammad; Nor, Mohd. Jailani Mohd.; Ariffin, Ahmad Kamal

2009-02-01

Noise has various effects on comfort, performance and health of human. Sound are analysed by human brain based on the frequencies and amplitudes. In a dynamic system, transmission of sound and vibrations depend on frequency and direction of the input motion and characteristics of the output. It is imperative that automotive manufacturers invest a lot of effort and money to improve and enhance the vibro-acoustics performance of their products. The enhancement effort may be very difficult and time-consuming if one relies only on 'trial and error' method without prior knowledge about the sources itself. Complex noise inside a vehicle cabin originated from various sources and travel through many pathways. First stage of sound quality refinement is to find the source. It is vital for automotive engineers to identify the dominant noise sources such as engine noise, exhaust noise and noise due to vibration transmission inside of vehicle. The purpose of this paper is to find the vibro-acoustical sources of noise in a passenger vehicle compartment. The implementation of spectral analysis method is much faster than the 'trial and error' methods in which, parts should be separated to measure the transfer functions. Also by using spectral analysis method, signals can be recorded in real operational conditions which conduce to more consistent results. A multi-channel analyser is utilised to measure and record the vibro-acoustical signals. Computational algorithms are also employed to identify contribution of various sources towards the measured interior signal. These achievements can be utilised to detect, control and optimise interior noise performance of road transport vehicles.

Source and listener directivity for interactive wave-based sound propagation.

PubMed

Mehra, Ravish; Antani, Lakulish; Kim, Sujeong; Manocha, Dinesh

2014-04-01

We present an approach to model dynamic, data-driven source and listener directivity for interactive wave-based sound propagation in virtual environments and computer games. Our directional source representation is expressed as a linear combination of elementary spherical harmonic (SH) sources. In the preprocessing stage, we precompute and encode the propagated sound fields due to each SH source. At runtime, we perform the SH decomposition of the varying source directivity interactively and compute the total sound field at the listener position as a weighted sum of precomputed SH sound fields. We propose a novel plane-wave decomposition approach based on higher-order derivatives of the sound field that enables dynamic HRTF-based listener directivity at runtime. We provide a generic framework to incorporate our source and listener directivity in any offline or online frequency-domain wave-based sound propagation algorithm. We have integrated our sound propagation system in Valve's Source game engine and use it to demonstrate realistic acoustic effects such as sound amplification, diffraction low-passing, scattering, localization, externalization, and spatial sound, generated by wave-based propagation of directional sources and listener in complex scenarios. We also present results from our preliminary user study.

Sound source identification and sound radiation modeling in a moving medium using the time-domain equivalent source method.

PubMed

Zhang, Xiao-Zheng; Bi, Chuan-Xing; Zhang, Yong-Bin; Xu, Liang

2015-05-01

Planar near-field acoustic holography has been successfully extended to reconstruct the sound field in a moving medium, however, the reconstructed field still contains the convection effect that might lead to the wrong identification of sound sources. In order to accurately identify sound sources in a moving medium, a time-domain equivalent source method is developed. In the method, the real source is replaced by a series of time-domain equivalent sources whose strengths are solved iteratively by utilizing the measured pressure and the known convective time-domain Green's function, and time averaging is used to reduce the instability in the iterative solving process. Since these solved equivalent source strengths are independent of the convection effect, they can be used not only to identify sound sources but also to model sound radiations in both moving and static media. Numerical simulations are performed to investigate the influence of noise on the solved equivalent source strengths and the effect of time averaging on reducing the instability, and to demonstrate the advantages of the proposed method on the source identification and sound radiation modeling.

Method and apparatus for separating mixtures of gases using an acoustic wave

DOEpatents

Geller, Drew A.; Swift, Gregory W.; Backhaus, Scott N.

2004-05-11

A thermoacoustic device separates a mixture of gases. An elongated duct is provided with first and second ends and has a length that is greater than the wavelength of sound in the mixture of gases at a selected frequency, and a diameter that is greater than a thermal penetration depth in the mixture of gases. A first acoustic source is located at the first end of the duct to generate acoustic power at the selected frequency. A plurality of side branch acoustic sources are spaced along the length of the duct and are configured to introduce acoustic power into the mixture of gases so that a first gas is concentrated at the first end of the duct and a second gas is concentrated at the second end of the duct.

Multisensory object perception in infancy: 4-month-olds perceive a mistuned harmonic as a separate auditory and visual object

PubMed Central

A. Smith, Nicholas; A. Folland, Nicholas; Martinez, Diana M.; Trainor, Laurel J.

2017-01-01

Infants learn to use auditory and visual information to organize the sensory world into identifiable objects with particular locations. Here we use a behavioural method to examine infants' use of harmonicity cues to auditory object perception in a multisensory context. Sounds emitted by different objects sum in the air and the auditory system must figure out which parts of the complex waveform belong to different sources (auditory objects). One important cue to this source separation is that complex tones with pitch typically contain a fundamental frequency and harmonics at integer multiples of the fundamental. Consequently, adults hear a mistuned harmonic in a complex sound as a distinct auditory object (Alain et al., 2003). Previous work by our group demonstrated that 4-month-old infants are also sensitive to this cue. They behaviourally discriminate a complex tone with a mistuned harmonic from the same complex with in-tune harmonics, and show an object-related event-related potential (ERP) electrophysiological (EEG) response to the stimulus with mistuned harmonics. In the present study we use an audiovisual procedure to investigate whether infants perceive a complex tone with an 8% mistuned harmonic as emanating from two objects, rather than merely detecting the mistuned cue. We paired in-tune and mistuned complex tones with visual displays that contained either one or two bouncing balls. Four-month-old infants showed surprise at the incongruous pairings, looking longer at the display of two balls when paired with the in-tune complex and at the display of one ball when paired with the mistuned harmonic complex. We conclude that infants use harmonicity as a cue for source separation when integrating auditory and visual information in object perception. PMID:28346869

Activity in Human Auditory Cortex Represents Spatial Separation Between Concurrent Sounds.

PubMed

Shiell, Martha M; Hausfeld, Lars; Formisano, Elia

2018-05-23

The primary and posterior auditory cortex (AC) are known for their sensitivity to spatial information, but how this information is processed is not yet understood. AC that is sensitive to spatial manipulations is also modulated by the number of auditory streams present in a scene (Smith et al., 2010), suggesting that spatial and nonspatial cues are integrated for stream segregation. We reasoned that, if this is the case, then it is the distance between sounds rather than their absolute positions that is essential. To test this hypothesis, we measured human brain activity in response to spatially separated concurrent sounds with fMRI at 7 tesla in five men and five women. Stimuli were spatialized amplitude-modulated broadband noises recorded for each participant via in-ear microphones before scanning. Using a linear support vector machine classifier, we investigated whether sound location and/or location plus spatial separation between sounds could be decoded from the activity in Heschl's gyrus and the planum temporale. The classifier was successful only when comparing patterns associated with the conditions that had the largest difference in perceptual spatial separation. Our pattern of results suggests that the representation of spatial separation is not merely the combination of single locations, but rather is an independent feature of the auditory scene. SIGNIFICANCE STATEMENT Often, when we think of auditory spatial information, we think of where sounds are coming from-that is, the process of localization. However, this information can also be used in scene analysis, the process of grouping and segregating features of a soundwave into objects. Essentially, when sounds are further apart, they are more likely to be segregated into separate streams. Here, we provide evidence that activity in the human auditory cortex represents the spatial separation between sounds rather than their absolute locations, indicating that scene analysis and localization processes may be independent. Copyright © 2018 the authors 0270-6474/18/384977-08$15.00/0.

Multistability in auditory stream segregation: a predictive coding view

PubMed Central

Winkler, István; Denham, Susan; Mill, Robert; Bőhm, Tamás M.; Bendixen, Alexandra

2012-01-01

Auditory stream segregation involves linking temporally separate acoustic events into one or more coherent sequences. For any non-trivial sequence of sounds, many alternative descriptions can be formed, only one or very few of which emerge in awareness at any time. Evidence from studies showing bi-/multistability in auditory streaming suggest that some, perhaps many of the alternative descriptions are represented in the brain in parallel and that they continuously vie for conscious perception. Here, based on a predictive coding view, we consider the nature of these sound representations and how they compete with each other. Predictive processing helps to maintain perceptual stability by signalling the continuation of previously established patterns as well as the emergence of new sound sources. It also provides a measure of how well each of the competing representations describes the current acoustic scene. This account of auditory stream segregation has been tested on perceptual data obtained in the auditory streaming paradigm. PMID:22371621

Neuromimetic Sound Representation for Percept Detection and Manipulation

NASA Astrophysics Data System (ADS)

Zotkin, Dmitry N.; Chi, Taishih; Shamma, Shihab A.; Duraiswami, Ramani

2005-12-01

The acoustic wave received at the ears is processed by the human auditory system to separate different sounds along the intensity, pitch, and timbre dimensions. Conventional Fourier-based signal processing, while endowed with fast algorithms, is unable to easily represent a signal along these attributes. In this paper, we discuss the creation of maximally separable sounds in auditory user interfaces and use a recently proposed cortical sound representation, which performs a biomimetic decomposition of an acoustic signal, to represent and manipulate sound for this purpose. We briefly overview algorithms for obtaining, manipulating, and inverting a cortical representation of a sound and describe algorithms for manipulating signal pitch and timbre separately. The algorithms are also used to create sound of an instrument between a "guitar" and a "trumpet." Excellent sound quality can be achieved if processing time is not a concern, and intelligible signals can be reconstructed in reasonable processing time (about ten seconds of computational time for a one-second signal sampled at [InlineEquation not available: see fulltext.]). Work on bringing the algorithms into the real-time processing domain is ongoing.

Experimental investigation of sound generation by a protuberance in a laminar boundary layer

DOE Office of Scientific and Technical Information (OSTI.GOV)

Kobayashi, M.; Asai, M.; Inasawa, A.

2014-08-15

Sound radiation from a two-dimensional protuberance glued on the wall in a laminar boundary layer was investigated experimentally at low Mach numbers. When the protuberance was as high as the boundary-layer thickness, a feedback-loop mechanism set in between protuberance-generated sound and Tollmien-Schlichting (T-S) waves generated by the leading-edge receptivity to the upstream-propagating sound. Although occurrence of a separation bubble immediately upstream of the protuberance played important roles in the evolution of instability waves into vortices interacting with the protuberance, the frequency of tonal vortex sound was determined by the selective amplification of T-S waves in the linear instability stage upstreammore » of the separation bubble and was not affected by the instability of the separation bubble.« less

Analysis of temporal decay of diffuse broadband sound fields in enclosures by decomposition in powers of an absorption parameter

NASA Astrophysics Data System (ADS)

Bliss, Donald; Franzoni, Linda; Rouse, Jerry; Manning, Ben

2005-09-01

An analysis method for time-dependent broadband diffuse sound fields in enclosures is described. Beginning with a formulation utilizing time-dependent broadband intensity boundary sources, the strength of these wall sources is expanded in a series in powers of an absorption parameter, thereby giving a separate boundary integral problem for each power. The temporal behavior is characterized by a Taylor expansion in the delay time for a source to influence an evaluation point. The lowest-order problem has a uniform interior field proportional to the reciprocal of the absorption parameter, as expected, and exhibits relatively slow exponential decay. The next-order problem gives a mean-square pressure distribution that is independent of the absorption parameter and is primarily responsible for the spatial variation of the reverberant field. This problem, which is driven by input sources and the lowest-order reverberant field, depends on source location and the spatial distribution of absorption. Additional problems proceed at integer powers of the absorption parameter, but are essentially higher-order corrections to the spatial variation. Temporal behavior is expressed in terms of an eigenvalue problem, with boundary source strength distributions expressed as eigenmodes. Solutions exhibit rapid short-time spatial redistribution followed by long-time decay of a predominant spatial mode.

Sound source localization identification accuracy: Envelope dependencies.

PubMed

Yost, William A

2017-07-01

Sound source localization accuracy as measured in an identification procedure in a front azimuth sound field was studied for click trains, modulated noises, and a modulated tonal carrier. Sound source localization accuracy was determined as a function of the number of clicks in a 64 Hz click train and click rate for a 500 ms duration click train. The clicks were either broadband or high-pass filtered. Sound source localization accuracy was also measured for a single broadband filtered click and compared to a similar broadband filtered, short-duration noise. Sound source localization accuracy was determined as a function of sinusoidal amplitude modulation and the "transposed" process of modulation of filtered noises and a 4 kHz tone. Different rates (16 to 512 Hz) of modulation (including unmodulated conditions) were used. Providing modulation for filtered click stimuli, filtered noises, and the 4 kHz tone had, at most, a very small effect on sound source localization accuracy. These data suggest that amplitude modulation, while providing information about interaural time differences in headphone studies, does not have much influence on sound source localization accuracy in a sound field.

System and method for characterizing synthesizing and/or canceling out acoustic signals from inanimate sound sources

DOEpatents

Holzrichter, John F.; Burnett, Greg C.; Ng, Lawrence C.

2003-01-01

A system and method for characterizing, synthesizing, and/or canceling out acoustic signals from inanimate sound sources is disclosed. Propagating wave electromagnetic sensors monitor excitation sources in sound producing systems, such as machines, musical instruments, and various other structures. Acoustical output from these sound producing systems is also monitored. From such information, a transfer function characterizing the sound producing system is generated. From the transfer function, acoustical output from the sound producing system may be synthesized or canceled. The methods disclosed enable accurate calculation of matched transfer functions relating specific excitations to specific acoustical outputs. Knowledge of such signals and functions can be used to effect various sound replication, sound source identification, and sound cancellation applications.

System and method for characterizing, synthesizing, and/or canceling out acoustic signals from inanimate sound sources

DOEpatents

Holzrichter, John F; Burnett, Greg C; Ng, Lawrence C

2013-05-21

A system and method for characterizing, synthesizing, and/or canceling out acoustic signals from inanimate sound sources is disclosed. Propagating wave electromagnetic sensors monitor excitation sources in sound producing systems, such as machines, musical instruments, and various other structures. Acoustical output from these sound producing systems is also monitored. From such information, a transfer function characterizing the sound producing system is generated. From the transfer function, acoustical output from the sound producing system may be synthesized or canceled. The methods disclosed enable accurate calculation of matched transfer functions relating specific excitations to specific acoustical outputs. Knowledge of such signals and functions can be used to effect various sound replication, sound source identification, and sound cancellation applications.

System and method for characterizing, synthesizing, and/or canceling out acoustic signals from inanimate sound sources

DOEpatents

Holzrichter, John F.; Burnett, Greg C.; Ng, Lawrence C.

2007-10-16

A system and method for characterizing, synthesizing, and/or canceling out acoustic signals from inanimate sound sources is disclosed. Propagating wave electromagnetic sensors monitor excitation sources in sound producing systems, such as machines, musical instruments, and various other structures. Acoustical output from these sound producing systems is also monitored. From such information, a transfer function characterizing the sound producing system is generated. From the transfer function, acoustical output from the sound producing system may be synthesized or canceled. The methods disclosed enable accurate calculation of matched transfer functions relating specific excitations to specific acoustical outputs. Knowledge of such signals and functions can be used to effect various sound replication, sound source identification, and sound cancellation applications.

SoundCompass: A Distributed MEMS Microphone Array-Based Sensor for Sound Source Localization

PubMed Central

Tiete, Jelmer; Domínguez, Federico; da Silva, Bruno; Segers, Laurent; Steenhaut, Kris; Touhafi, Abdellah

2014-01-01

Sound source localization is a well-researched subject with applications ranging from localizing sniper fire in urban battlefields to cataloging wildlife in rural areas. One critical application is the localization of noise pollution sources in urban environments, due to an increasing body of evidence linking noise pollution to adverse effects on human health. Current noise mapping techniques often fail to accurately identify noise pollution sources, because they rely on the interpolation of a limited number of scattered sound sensors. Aiming to produce accurate noise pollution maps, we developed the SoundCompass, a low-cost sound sensor capable of measuring local noise levels and sound field directionality. Our first prototype is composed of a sensor array of 52 Microelectromechanical systems (MEMS) microphones, an inertial measuring unit and a low-power field-programmable gate array (FPGA). This article presents the SoundCompass’s hardware and firmware design together with a data fusion technique that exploits the sensing capabilities of the SoundCompass in a wireless sensor network to localize noise pollution sources. Live tests produced a sound source localization accuracy of a few centimeters in a 25-m2 anechoic chamber, while simulation results accurately located up to five broadband sound sources in a 10,000-m2 open field. PMID:24463431

Infrasound, Its Sources and Its Effects on Man

DTIC Science & Technology

1976-05-01

modulated by an infra - Annoyance has been broken out as a separate sonic frequency. For instance, the amplified topic because I believe that the greatest...importance is the nigh frequency response of quency sound. In general, infrasound does not the measurement system. Measurement of infra - often occur at levels...esuential for detailed analysis and changes in barometric pressure would be con- from these recordings a narrow band spectral sidered infrasonic . The

76 FR 23191 - Traffic Separation Schemes: In the Strait of Juan de Fuca and Its Approaches; in Puget Sound and...

Federal Register 2010, 2011, 2012, 2013, 2014

2011-04-26

...-AA48 Traffic Separation Schemes: In the Strait of Juan de Fuca and Its Approaches; in Puget Sound and... rule codifying traffic separation schemes in the Strait of Juan de Fuca and its Approaches; in Puget... established these traffic separation schemes under authority of the Ports and Waterways Safety Act. DATES...

75 FR 70818 - Traffic Separation Schemes: In the Strait of Juan de Fuca and Its Approaches; in Puget Sound and...

Federal Register 2010, 2011, 2012, 2013, 2014

2010-11-19

...-AA48 Traffic Separation Schemes: In the Strait of Juan de Fuca and Its Approaches; in Puget Sound and..., the Coast Guard codifies traffic separation schemes in the Strait of Juan de Fuca and its approaches.... These traffic separation schemes (TSSs) were validated by a Port Access Route Study (PARS) conducted...

Is 9 louder than 1? Audiovisual cross-modal interactions between number magnitude and judged sound loudness.

PubMed

Alards-Tomalin, Doug; Walker, Alexander C; Shaw, Joshua D M; Leboe-McGowan, Launa C

2015-09-01

The cross-modal impact of number magnitude (i.e. Arabic digits) on perceived sound loudness was examined. Participants compared a target sound's intensity level against a previously heard reference sound (which they judged as quieter or louder). Paired with each target sound was a task irrelevant Arabic digit that varied in magnitude, being either small (1, 2, 3) or large (7, 8, 9). The degree to which the sound and the digit were synchronized was manipulated, with the digit and sound occurring simultaneously in Experiment 1, and the digit preceding the sound in Experiment 2. Firstly, when target sounds and digits occurred simultaneously, sounds paired with large digits were categorized as loud more frequently than sounds paired with small digits. Secondly, when the events were separated, number magnitude ceased to bias sound intensity judgments. In Experiment 3, the events were still separated, however the participants held the number in short-term memory. In this instance the bias returned. Copyright © 2015 Elsevier B.V. All rights reserved.

33 CFR 167.1701 - In Prince William Sound: Precautionary areas.

Code of Federal Regulations, 2010 CFR

2010-07-01

... HOMELAND SECURITY (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1701 In Prince William Sound...

Separation of acoustic waves in isentropic flow perturbations

DOE Office of Scientific and Technical Information (OSTI.GOV)

Henke, Christian, E-mail: christian.henke@atlas-elektronik.com

2015-04-15

The present contribution investigates the mechanisms of sound generation and propagation in the case of highly-unsteady flows. Based on the linearisation of the isentropic Navier–Stokes equation around a new pathline-averaged base flow, it is demonstrated for the first time that flow perturbations of a non-uniform flow can be split into acoustic and vorticity modes, with the acoustic modes being independent of the vorticity modes. Therefore, we can propose this acoustic perturbation as a general definition of sound. As a consequence of the splitting result, we conclude that the present acoustic perturbation is propagated by the convective wave equation and fulfilsmore » Lighthill’s acoustic analogy. Moreover, we can define the deviations of the Navier–Stokes equation from the convective wave equation as “true” sound sources. In contrast to other authors, no assumptions on a slowly varying or irrotational flow are necessary. Using a symmetry argument for the conservation laws, an energy conservation result and a generalisation of the sound intensity are provided. - Highlights: • First splitting of non-uniform flows in acoustic and non-acoustic components. • These result leads to a generalisation of sound which is compatible with Lighthill’s acoustic analogy. • A closed equation for the generation and propagation of sound is given.« less

Effects of Different Spectral Shapes and Amplitude Modulation of Broadband Noise on Annoyance Reactions in a Controlled Listening Experiment.

PubMed

Schäffer, Beat; Pieren, Reto; Schlittmeier, Sabine J; Brink, Mark

2018-05-19

Environmental noise from transportation or industrial infrastructure typically has a broad frequency range. Different sources may have disparate acoustical characteristics, which may in turn affect noise annoyance. However, knowledge of the relative contribution of the different acoustical characteristics of broadband noise to annoyance is still scarce. In this study, the subjectively perceived short-term (acute) annoyance reactions to different broadband sounds (namely, realistic outdoor wind turbine and artificial, generic sounds) at 40 dBA were investigated in a controlled laboratory listening experiment. Combined with the factorial design of the experiment, the sounds allowed for separation of the effects of three acoustical characteristics on annoyance, namely, spectral shape, depth of periodic amplitude modulation (AM), and occurrence (or absence) of random AM. Fifty-two participants rated their annoyance with the sounds. Annoyance increased with increasing energy content in the low-frequency range as well as with depth of periodic AM, and was higher in situations with random AM than without. Similar annoyance changes would be evoked by sound pressure level changes of up to 8 dB. The results suggest that besides standard sound pressure level metrics, other acoustical characteristics of (broadband) noise should also be considered in environmental impact assessments, e.g., in the context of wind turbine installations.

Working Towards Deep-Ocean Temperature Monitoring by Studying the Acoustic Ambient Noise Field in the South Pacific Ocean

NASA Astrophysics Data System (ADS)

Sambell, K.; Evers, L. G.; Snellen, M.

2017-12-01

Deriving the deep-ocean temperature is a challenge. In-situ observations and satellite observations are hardly applicable. However, knowledge about changes in the deep ocean temperature is important in relation to climate change. Oceans are filled with low-frequency sound waves created by sources such as underwater volcanoes, earthquakes and seismic surveys. The propagation of these sound waves is temperature dependent and therefore carries valuable information that can be used for temperature monitoring. This phenomenon is investigated by applying interferometry to hydroacoustic data measured in the South Pacific Ocean. The data is measured at hydrophone station H03 which is part of the International Monitoring System (IMS). This network consists of several stations around the world and is in place for the verification of the Comprehensive Nuclear-Test-Ban Treaty (CTBT). The station consists of two arrays located north and south of Robinson Crusoe Island separated by 50 km. Both arrays consist of three hydrophones with an intersensor distance of 2 km located at a depth of 1200 m. This depth is in range of the SOFAR channel. Hydroacoustic data measured at the south station is cross-correlated for the time period 2014-2017. The results are improved by applying one-bit normalization as a preprocessing step. Furthermore, beamforming is applied to the hydroacoustic data in order to characterize ambient noise sources around the array. This shows the presence of a continuous source at a backazimuth between 180 and 200 degrees throughout the whole time period, which is in agreement with the results obtained by cross-correlation. Studies on source strength show a seasonal dependence. This is an indication that the sound is related to acoustic activity in Antarctica. Results on this are supported by acoustic propagation modeling. The normal mode technique is used to study the sound propagation from possible source locations towards station H03.

The effect of spatial distribution on the annoyance caused by simultaneous sounds

NASA Astrophysics Data System (ADS)

Vos, Joos; Bronkhorst, Adelbert W.; Fedtke, Thomas

2004-05-01

A considerable part of the population is exposed to simultaneous and/or successive environmental sounds from different sources. In many cases, these sources are different with respect to their locations also. In a laboratory study, it was investigated whether the annoyance caused by the multiple sounds is affected by the spatial distribution of the sources. There were four independent variables: (1) sound category (stationary or moving), (2) sound type (stationary: lawn-mower, leaf-blower, and chain saw; moving: road traffic, railway, and motorbike), (3) spatial location (left, right, and combinations), and (4) A-weighted sound exposure level (ASEL of single sources equal to 50, 60, or 70 dB). In addition to the individual sounds in isolation, various combinations of two or three different sources within each sound category and sound level were presented for rating. The annoyance was mainly determined by sound level and sound source type. In most cases there were neither significant main effects of spatial distribution nor significant interaction effects between spatial distribution and the other variables. It was concluded that for rating the spatially distrib- uted sounds investigated, the noise dose can simply be determined by a summation of the levels for the left and right channels. [Work supported by CEU.

Spherical loudspeaker array for local active control of sound.

PubMed

Rafaely, Boaz

2009-05-01

Active control of sound has been employed to reduce noise levels around listeners' head using destructive interference from noise-canceling sound sources. Recently, spherical loudspeaker arrays have been studied as multiple-channel sound sources, capable of generating sound fields with high complexity. In this paper, the potential use of a spherical loudspeaker array for local active control of sound is investigated. A theoretical analysis of the primary and secondary sound fields around a spherical sound source reveals that the natural quiet zones for the spherical source have a shell-shape. Using numerical optimization, quiet zones with other shapes are designed, showing potential for quiet zones with extents that are significantly larger than the well-known limit of a tenth of a wavelength for monopole sources. The paper presents several simulation examples showing quiet zones in various configurations.

Localizing the sources of two independent noises: Role of time varying amplitude differences

PubMed Central

Yost, William A.; Brown, Christopher A.

2013-01-01

Listeners localized the free-field sources of either one or two simultaneous and independently generated noise bursts. Listeners' localization performance was better when localizing one rather than two sound sources. With two sound sources, localization performance was better when the listener was provided prior information about the location of one of them. Listeners also localized two simultaneous noise bursts that had sinusoidal amplitude modulation (AM) applied, in which the modulation envelope was in-phase across the two source locations or was 180° out-of-phase. The AM was employed to investigate a hypothesis as to what process listeners might use to localize multiple sound sources. The results supported the hypothesis that localization of two sound sources might be based on temporal-spectral regions of the combined waveform in which the sound from one source was more intense than that from the other source. The interaural information extracted from such temporal-spectral regions might provide reliable estimates of the sound source location that produced the more intense sound in that temporal-spectral region. PMID:23556597

Localizing the sources of two independent noises: role of time varying amplitude differences.

PubMed

Yost, William A; Brown, Christopher A

2013-04-01

Listeners localized the free-field sources of either one or two simultaneous and independently generated noise bursts. Listeners' localization performance was better when localizing one rather than two sound sources. With two sound sources, localization performance was better when the listener was provided prior information about the location of one of them. Listeners also localized two simultaneous noise bursts that had sinusoidal amplitude modulation (AM) applied, in which the modulation envelope was in-phase across the two source locations or was 180° out-of-phase. The AM was employed to investigate a hypothesis as to what process listeners might use to localize multiple sound sources. The results supported the hypothesis that localization of two sound sources might be based on temporal-spectral regions of the combined waveform in which the sound from one source was more intense than that from the other source. The interaural information extracted from such temporal-spectral regions might provide reliable estimates of the sound source location that produced the more intense sound in that temporal-spectral region.

Broadband Processing in a Noisy Shallow Ocean Environment: A Particle Filtering Approach

DOE PAGES

Candy, J. V.

2016-04-14

Here we report that when a broadband source propagates sound in a shallow ocean the received data can become quite complicated due to temperature-related sound-speed variations and therefore a highly dispersive environment. Noise and uncertainties disrupt this already chaotic environment even further because disturbances propagate through the same inherent acoustic channel. The broadband (signal) estimation/detection problem can be decomposed into a set of narrowband solutions that are processed separately and then combined to achieve more enhancement of signal levels than that available from a single frequency, thereby allowing more information to be extracted leading to a more reliable source detection.more » A Bayesian solution to the broadband modal function tracking, pressure-field enhancement, and source detection problem is developed that leads to nonparametric estimates of desired posterior distributions enabling the estimation of useful statistics and an improved processor/detector. In conclusion, to investigate the processor capabilities, we synthesize an ensemble of noisy, broadband, shallow-ocean measurements to evaluate its overall performance using an information theoretical metric for the preprocessor and the receiver operating characteristic curve for the detector.« less

Modelling Nonlinear Ultrasound Propagation in Bone

NASA Astrophysics Data System (ADS)

Cleveland, Robin O.; Johnson, Paul A.; Muller, Marie; Talmant, Maryline; Padilla, Frederic; Laugier, Pascal

2006-05-01

Simulations have been carried out to assess the possibility for detecting the nonlinear properties of bone in vivo. We employed a time domain solution to the KZK equation to determine the nonlinear field generated by an unfocussed circular transducer in both cancellous and cortical bone. The results indicate that determining nonlinear properties from the generation of higher harmonics is challenging in both bone types (for propagation distances and source amplitudes appropriate in the body). In cancellous bone this is because the attenuation length scale is very short (about 5 mm) and in cortical bone because the high sound speed and density result in long nonlinear length scales (hundreds of millimeters). An alternative approach to determine the nonlinear properties was considered using self-demodulation of sound. For cancellous bone this may result in a detectable signal although the predicted amplitude of the self-demodulation signal was almost 90 dB below the source level (1 MPa). In cortical bone the self-demodulated signal was even weaker that in cancellous bone (˜110 dB down) and, for a practical length signal, was not easy to separate from the components associated with the source.

Treefrogs as Animal Models for Research on Auditory Scene Analysis and the Cocktail Party Problem

PubMed Central

Bee, Mark A.

2014-01-01

The perceptual analysis of acoustic scenes involves binding together sounds from the same source and separating them from other sounds in the environment. In large social groups, listeners experience increased difficulty performing these tasks due to high noise levels and interference from the concurrent signals of multiple individuals. While a substantial body of literature on these issues pertains to human hearing and speech communication, few studies have investigated how nonhuman animals may be evolutionarily adapted to solve biologically analogous communication problems. Here, I review recent and ongoing work aimed at testing hypotheses about perceptual mechanisms that enable treefrogs in the genus Hyla to communicate vocally in noisy, multi-source social environments. After briefly introducing the genus and the methods used to study hearing in frogs, I outline several functional constraints on communication posed by the acoustic environment of breeding “choruses”. Then, I review studies of sound source perception aimed at uncovering how treefrog listeners may be adapted to cope with these constraints. Specifically, this review covers research on the acoustic cues used in sequential and simultaneous auditory grouping, spatial release from masking, and dip listening. Throughout the paper, I attempt to illustrate how broad-scale, comparative studies of carefully considered animal models may ultimately reveal an evolutionary diversity of underlying mechanisms for solving cocktail-party-like problems in communication. PMID:24424243

Stage separation study of Nike-Black Brant V Sounding Rocket System

NASA Technical Reports Server (NTRS)

Ferragut, N. J.

1976-01-01

A new Sounding Rocket System has been developed. It consists of a Nike Booster and a Black Brant V Sustainer with slanted fins which extend beyond its nozzle exit plane. A cursory look was taken at different factors which must be considered when studying a passive separation system. That is, one separation system without mechanical constraints in the axial direction and which will allow separation due to drag differential accelerations between the Booster and the Sustainer. The equations of motion were derived for rigid body motions and exact solutions were obtained. The analysis developed could be applied to any other staging problem of a Sounding Rocket System.

A Binaural Cochlear Implant Sound Coding Strategy Inspired by the Contralateral Medial Olivocochlear Reflex

PubMed Central

Eustaquio-Martín, Almudena; Stohl, Joshua S.; Wolford, Robert D.; Schatzer, Reinhold; Wilson, Blake S.

2016-01-01

Objectives: In natural hearing, cochlear mechanical compression is dynamically adjusted via the efferent medial olivocochlear reflex (MOCR). These adjustments probably help understanding speech in noisy environments and are not available to the users of current cochlear implants (CIs). The aims of the present study are to: (1) present a binaural CI sound processing strategy inspired by the control of cochlear compression provided by the contralateral MOCR in natural hearing; and (2) assess the benefits of the new strategy for understanding speech presented in competition with steady noise with a speech-like spectrum in various spatial configurations of the speech and noise sources. Design: Pairs of CI sound processors (one per ear) were constructed to mimic or not mimic the effects of the contralateral MOCR on compression. For the nonmimicking condition (standard strategy or STD), the two processors in a pair functioned similarly to standard clinical processors (i.e., with fixed back-end compression and independently of each other). When configured to mimic the effects of the MOCR (MOC strategy), the two processors communicated with each other and the amount of back-end compression in a given frequency channel of each processor in the pair decreased/increased dynamically (so that output levels dropped/increased) with increases/decreases in the output energy from the corresponding frequency channel in the contralateral processor. Speech reception thresholds in speech-shaped noise were measured for 3 bilateral CI users and 2 single-sided deaf unilateral CI users. Thresholds were compared for the STD and MOC strategies in unilateral and bilateral listening conditions and for three spatial configurations of the speech and noise sources in simulated free-field conditions: speech and noise sources colocated in front of the listener, speech on the left ear with noise in front of the listener, and speech on the left ear with noise on the right ear. In both bilateral and unilateral listening, the electrical stimulus delivered to the test ear(s) was always calculated as if the listeners were wearing bilateral processors. Results: In both unilateral and bilateral listening conditions, mean speech reception thresholds were comparable with the two strategies for colocated speech and noise sources, but were at least 2 dB lower (better) with the MOC than with the STD strategy for spatially separated speech and noise sources. In unilateral listening conditions, mean thresholds improved with increasing the spatial separation between the speech and noise sources regardless of the strategy but the improvement was significantly greater with the MOC strategy. In bilateral listening conditions, thresholds improved significantly with increasing the speech-noise spatial separation only with the MOC strategy. Conclusions: The MOC strategy (1) significantly improved the intelligibility of speech presented in competition with a spatially separated noise source, both in unilateral and bilateral listening conditions; (2) produced significant spatial release from masking in bilateral listening conditions, something that did not occur with fixed compression; and (3) enhanced spatial release from masking in unilateral listening conditions. The MOC strategy as implemented here, or a modified version of it, may be usefully applied in CIs and in hearing aids. PMID:26862711

A Binaural Cochlear Implant Sound Coding Strategy Inspired by the Contralateral Medial Olivocochlear Reflex.

PubMed

Lopez-Poveda, Enrique A; Eustaquio-Martín, Almudena; Stohl, Joshua S; Wolford, Robert D; Schatzer, Reinhold; Wilson, Blake S

2016-01-01

In natural hearing, cochlear mechanical compression is dynamically adjusted via the efferent medial olivocochlear reflex (MOCR). These adjustments probably help understanding speech in noisy environments and are not available to the users of current cochlear implants (CIs). The aims of the present study are to: (1) present a binaural CI sound processing strategy inspired by the control of cochlear compression provided by the contralateral MOCR in natural hearing; and (2) assess the benefits of the new strategy for understanding speech presented in competition with steady noise with a speech-like spectrum in various spatial configurations of the speech and noise sources. Pairs of CI sound processors (one per ear) were constructed to mimic or not mimic the effects of the contralateral MOCR on compression. For the nonmimicking condition (standard strategy or STD), the two processors in a pair functioned similarly to standard clinical processors (i.e., with fixed back-end compression and independently of each other). When configured to mimic the effects of the MOCR (MOC strategy), the two processors communicated with each other and the amount of back-end compression in a given frequency channel of each processor in the pair decreased/increased dynamically (so that output levels dropped/increased) with increases/decreases in the output energy from the corresponding frequency channel in the contralateral processor. Speech reception thresholds in speech-shaped noise were measured for 3 bilateral CI users and 2 single-sided deaf unilateral CI users. Thresholds were compared for the STD and MOC strategies in unilateral and bilateral listening conditions and for three spatial configurations of the speech and noise sources in simulated free-field conditions: speech and noise sources colocated in front of the listener, speech on the left ear with noise in front of the listener, and speech on the left ear with noise on the right ear. In both bilateral and unilateral listening, the electrical stimulus delivered to the test ear(s) was always calculated as if the listeners were wearing bilateral processors. In both unilateral and bilateral listening conditions, mean speech reception thresholds were comparable with the two strategies for colocated speech and noise sources, but were at least 2 dB lower (better) with the MOC than with the STD strategy for spatially separated speech and noise sources. In unilateral listening conditions, mean thresholds improved with increasing the spatial separation between the speech and noise sources regardless of the strategy but the improvement was significantly greater with the MOC strategy. In bilateral listening conditions, thresholds improved significantly with increasing the speech-noise spatial separation only with the MOC strategy. The MOC strategy (1) significantly improved the intelligibility of speech presented in competition with a spatially separated noise source, both in unilateral and bilateral listening conditions; (2) produced significant spatial release from masking in bilateral listening conditions, something that did not occur with fixed compression; and (3) enhanced spatial release from masking in unilateral listening conditions. The MOC strategy as implemented here, or a modified version of it, may be usefully applied in CIs and in hearing aids.

Effects of multiple congruent cues on concurrent sound segregation during passive and active listening: an event-related potential (ERP) study.

PubMed

Kocsis, Zsuzsanna; Winkler, István; Szalárdy, Orsolya; Bendixen, Alexandra

2014-07-01

In two experiments, we assessed the effects of combining different cues of concurrent sound segregation on the object-related negativity (ORN) and the P400 event-related potential components. Participants were presented with sequences of complex tones, half of which contained some manipulation: one or two harmonic partials were mistuned, delayed, or presented from a different location than the rest. In separate conditions, one, two, or three of these manipulations were combined. Participants watched a silent movie (passive listening) or reported after each tone whether they perceived one or two concurrent sounds (active listening). ORN was found in almost all conditions except for location difference alone during passive listening. Combining several cues or manipulating more than one partial consistently led to sub-additive effects on the ORN amplitude. These results support the view that ORN reflects a combined, feature-unspecific assessment of the auditory system regarding the contribution of two sources to the incoming sound. Copyright © 2014 Elsevier B.V. All rights reserved.

Geologic sources of asbestos in Seattle's tolt reservoir

USGS Publications Warehouse

Reid, M.E.; Craven, G.

1996-01-01

Water from Seattle's South Fork Tolt Reservoir contains chrysotile and amphibole asbestos fibers, derived from natural sources. Using optical petrographic techniques, X-ray diffraction, and scanning electron microscopy, we identified the geologic source of these asbestiform minerals within the watershed. No asbestos was found in the bedrock underlying the watershed, while both chrysotile and amphibole fibers were found in sediments transported by Puget-lobe glacial processes. These materials, widely distributed throughout the lower watershed, would be difficult to separate from the reservoir sediments. The probable source of this asbestos is in pods of ultramafic rock occurring north of the watershed. Because asbestos is contained in widespread Pugetlobe glacial materials, it may be naturally distributed in other watersheds in the Puget Sound area.

Issues in Humanoid Audition and Sound Source Localization by Active Audition

NASA Astrophysics Data System (ADS)

Nakadai, Kazuhiro; Okuno, Hiroshi G.; Kitano, Hiroaki

In this paper, we present an active audition system which is implemented on the humanoid robot "SIG the humanoid". The audition system for highly intelligent humanoids localizes sound sources and recognizes auditory events in the auditory scene. Active audition reported in this paper enables SIG to track sources by integrating audition, vision, and motor movements. Given the multiple sound sources in the auditory scene, SIG actively moves its head to improve localization by aligning microphones orthogonal to the sound source and by capturing the possible sound sources by vision. However, such an active head movement inevitably creates motor noises.The system adaptively cancels motor noises using motor control signals and the cover acoustics. The experimental result demonstrates that active audition by integration of audition, vision, and motor control attains sound source tracking in variety of conditions.onditions.

Sound source localization method in an environment with flow based on Amiet-IMACS

NASA Astrophysics Data System (ADS)

Wei, Long; Li, Min; Qin, Sheng; Fu, Qiang; Yang, Debin

2017-05-01

A sound source localization method is proposed to localize and analyze the sound source in an environment with airflow. It combines the improved mapping of acoustic correlated sources (IMACS) method and Amiet's method, and is called Amiet-IMACS. It can localize uncorrelated and correlated sound sources with airflow. To implement this approach, Amiet's method is used to correct the sound propagation path in 3D, which improves the accuracy of the array manifold matrix and decreases the position error of the localized source. Then, the mapping of acoustic correlated sources (MACS) method, which is as a high-resolution sound source localization algorithm, is improved by self-adjusting the constraint parameter at each irritation process to increase convergence speed. A sound source localization experiment using a pair of loud speakers in an anechoic wind tunnel under different flow speeds is conducted. The experiment exhibits the advantage of Amiet-IMACS in localizing a more accurate sound source position compared with implementing IMACS alone in an environment with flow. Moreover, the aerodynamic noise produced by a NASA EPPLER 862 STRUT airfoil model in airflow with a velocity of 80 m/s is localized using the proposed method, which further proves its effectiveness in a flow environment. Finally, the relationship between the source position of this airfoil model and its frequency, along with its generation mechanism, is determined and interpreted.

A stethoscope with wavelet separation of cardiac and respiratory sounds for real time telemedicine implemented on field-programmable gate array

NASA Astrophysics Data System (ADS)

Castro, Víctor M.; Muñoz, Nestor A.; Salazar, Antonio J.

2015-01-01

Auscultation is one of the most utilized physical examination procedures for listening to lung, heart and intestinal sounds during routine consults and emergencies. Heart and lung sounds overlap in the thorax. An algorithm was used to separate them based on the discrete wavelet transform with multi-resolution analysis, which decomposes the signal into approximations and details. The algorithm was implemented in software and in hardware to achieve real-time signal separation. The heart signal was found in detail eight and the lung signal in approximation six. The hardware was used to separate the signals with a delay of 256 ms. Sending wavelet decomposition data - instead of the separated full signa - allows telemedicine applications to function in real time over low-bandwidth communication channels.

The role of diffusive architectural surfaces on auditory spatial discrimination in performance venues.

PubMed

Robinson, Philip W; Pätynen, Jukka; Lokki, Tapio; Jang, Hyung Suk; Jeon, Jin Yong; Xiang, Ning

2013-06-01

In musical or theatrical performance, some venues allow listeners to individually localize and segregate individual performers, while others produce a well blended ensemble sound. The room acoustic conditions that make this possible, and the psycho-acoustic effects at work are not fully understood. This research utilizes auralizations from measured and simulated performance venues to investigate spatial discrimination of multiple acoustic sources in rooms. Signals were generated from measurements taken in a small theater, and listeners in the audience area were asked to distinguish pairs of speech sources on stage with various spatial separations. This experiment was repeated with the proscenium splay walls treated to be flat, diffusive, or absorptive. Similar experiments were conducted in a simulated hall, utilizing 11 early reflections with various characteristics, and measured late reverberation. The experiments reveal that discriminating the lateral arrangement of two sources is possible at narrower separation angles when reflections come from flat or absorptive rather than diffusive surfaces.

An evaluation of talker localization based on direction of arrival estimation and statistical sound source identification

NASA Astrophysics Data System (ADS)

Nishiura, Takanobu; Nakamura, Satoshi

2002-11-01

It is very important to capture distant-talking speech for a hands-free speech interface with high quality. A microphone array is an ideal candidate for this purpose. However, this approach requires localizing the target talker. Conventional talker localization algorithms in multiple sound source environments not only have difficulty localizing the multiple sound sources accurately, but also have difficulty localizing the target talker among known multiple sound source positions. To cope with these problems, we propose a new talker localization algorithm consisting of two algorithms. One is DOA (direction of arrival) estimation algorithm for multiple sound source localization based on CSP (cross-power spectrum phase) coefficient addition method. The other is statistical sound source identification algorithm based on GMM (Gaussian mixture model) for localizing the target talker position among localized multiple sound sources. In this paper, we particularly focus on the talker localization performance based on the combination of these two algorithms with a microphone array. We conducted evaluation experiments in real noisy reverberant environments. As a result, we confirmed that multiple sound signals can be identified accurately between ''speech'' or ''non-speech'' by the proposed algorithm. [Work supported by ATR, and MEXT of Japan.

Sound source localization and segregation with internally coupled ears: the treefrog model

PubMed Central

Christensen-Dalsgaard, Jakob

2016-01-01

Acoustic signaling plays key roles in mediating many of the reproductive and social behaviors of anurans (frogs and toads). Moreover, acoustic signaling often occurs at night, in structurally complex habitats, such as densely vegetated ponds, and in dense breeding choruses characterized by high levels of background noise and acoustic clutter. Fundamental to anuran behavior is the ability of the auditory system to determine accurately the location from where sounds originate in space (sound source localization) and to assign specific sounds in the complex acoustic milieu of a chorus to their correct sources (sound source segregation). Here, we review anatomical, biophysical, neurophysiological, and behavioral studies aimed at identifying how the internally coupled ears of frogs contribute to sound source localization and segregation. Our review focuses on treefrogs in the genus Hyla, as they are the most thoroughly studied frogs in terms of sound source localization and segregation. They also represent promising model systems for future work aimed at understanding better how internally coupled ears contribute to sound source localization and segregation. We conclude our review by enumerating directions for future research on these animals that will require the collaborative efforts of biologists, physicists, and roboticists. PMID:27730384

Application of acoustic radiosity methods to noise propagation within buildings

NASA Astrophysics Data System (ADS)

Muehleisen, Ralph T.; Beamer, C. Walter

2005-09-01

The prediction of sound pressure levels in rooms from transmitted sound is a difficult problem. The sound energy in the source room incident on the common wall must be accurately predicted. In the receiving room, the propagation of sound from the planar wall source must also be accurately predicted. The radiosity method naturally computes the spatial distribution of sound energy incident on a wall and also naturally predicts the propagation of sound from a planar area source. In this paper, the application of the radiosity method to sound transmission problems is introduced and explained.

Ejectable underwater sound source recovery assembly

NASA Technical Reports Server (NTRS)

Irick, S. C. (Inventor)

1974-01-01

An underwater sound source is described that may be ejectably mounted on any mobile device that travels over water, to facilitate in the location and recovery of the device when submerged. A length of flexible line maintains a connection between the mobile device and the sound source. During recovery, the sound source is located be particularly useful in the recovery of spent rocket motors that bury in the ocean floor upon impact.

Visual Presentation Effects on Identification of Multiple Environmental Sounds

PubMed Central

Masakura, Yuko; Ichikawa, Makoto; Shimono, Koichi; Nakatsuka, Reio

2016-01-01

This study examined how the contents and timing of a visual stimulus affect the identification of mixed sounds recorded in a daily life environment. For experiments, we presented four environment sounds as auditory stimuli for 5 s along with a picture or a written word as a visual stimulus that might or might not denote the source of one of the four sounds. Three conditions of temporal relations between the visual stimuli and sounds were used. The visual stimulus was presented either: (a) for 5 s simultaneously with the sound; (b) for 5 s, 1 s before the sound (SOA between the audio and visual stimuli was 6 s); or (c) for 33 ms, 1 s before the sound (SOA was 1033 ms). Participants reported all identifiable sounds for those audio–visual stimuli. To characterize the effects of visual stimuli on sound identification, the following were used: the identification rates of sounds for which the visual stimulus denoted its sound source, the rates of other sounds for which the visual stimulus did not denote the sound source, and the frequency of false hearing of a sound that was not presented for each sound set. Results of the four experiments demonstrated that a picture or a written word promoted identification of the sound when it was related to the sound, particularly when the visual stimulus was presented for 5 s simultaneously with the sounds. However, a visual stimulus preceding the sounds had a benefit only for the picture, not for the written word. Furthermore, presentation with a picture denoting a sound simultaneously with the sound reduced the frequency of false hearing. These results suggest three ways that presenting a visual stimulus affects identification of the auditory stimulus. First, activation of the visual representation extracted directly from the picture promotes identification of the denoted sound and suppresses the processing of sounds for which the visual stimulus did not denote the sound source. Second, effects based on processing of the conceptual information promote identification of the denoted sound and suppress the processing of sounds for which the visual stimulus did not denote the sound source. Third, processing of the concurrent visual representation suppresses false hearing. PMID:26973478

The effect of brain lesions on sound localization in complex acoustic environments.

PubMed

Zündorf, Ida C; Karnath, Hans-Otto; Lewald, Jörg

2014-05-01

Localizing sound sources of interest in cluttered acoustic environments--as in the 'cocktail-party' situation--is one of the most demanding challenges to the human auditory system in everyday life. In this study, stroke patients' ability to localize acoustic targets in a single-source and in a multi-source setup in the free sound field were directly compared. Subsequent voxel-based lesion-behaviour mapping analyses were computed to uncover the brain areas associated with a deficit in localization in the presence of multiple distracter sound sources rather than localization of individually presented sound sources. Analyses revealed a fundamental role of the right planum temporale in this task. The results from the left hemisphere were less straightforward, but suggested an involvement of inferior frontal and pre- and postcentral areas. These areas appear to be particularly involved in the spectrotemporal analyses crucial for effective segregation of multiple sound streams from various locations, beyond the currently known network for localization of isolated sound sources in otherwise silent surroundings.

Temporal and Spatial Comparisons of Underwater Sound Signatures of Different Reef Habitats in Moorea Island, French Polynesia.

PubMed

Bertucci, Frédéric; Parmentier, Eric; Berten, Laëtitia; Brooker, Rohan M; Lecchini, David

2015-01-01

As environmental sounds are used by larval fish and crustaceans to locate and orientate towards habitat during settlement, variations in the acoustic signature produced by habitats could provide valuable information about habitat quality, helping larvae to differentiate between potential settlement sites. However, very little is known about how acoustic signatures differ between proximate habitats. This study described within- and between-site differences in the sound spectra of five contiguous habitats at Moorea Island, French Polynesia: the inner reef crest, the barrier reef, the fringing reef, a pass and a coastal mangrove forest. Habitats with coral (inner, barrier and fringing reefs) were characterized by a similar sound spectrum with average intensities ranging from 70 to 78 dB re 1 μPa.Hz(-1). The mangrove forest had a lower sound intensity of 70 dB re 1 μPa.Hz(-1) while the pass was characterized by a higher sound level with an average intensity of 91 dB re 1 μPa.Hz(-1). Habitats showed significantly different intensities for most frequencies, and a decreasing intensity gradient was observed from the reef to the shore. While habitats close to the shore showed no significant diel variation in sound intensities, sound levels increased at the pass during the night and barrier reef during the day. These two habitats also appeared to be louder in the North than in the West. These findings suggest that daily variations in sound intensity and across-reef sound gradients could be a valuable source of information for settling larvae. They also provide further evidence that closely related habitats, separated by less than 1 km, can differ significantly in their spectral composition and that these signatures might be typical and conserved along the coast of Moorea.

Using nonlocal means to separate cardiac and respiration sounds

NASA Astrophysics Data System (ADS)

Rudnitskii, A. G.

2014-11-01

The paper presents the results of applying nonlocal means (NLMs) approach in the problem of separating respiration and cardiac sounds in a signal recorded on a human chest wall. The performance of the algorithm was tested both by simulated and real signals. As a quantitative efficiency measure of NLM filtration, the angle of divergence between isolated and reference signal was used. It is shown that for a wide range of signal-to-noise ratios, the algorithm makes it possible to efficiently solve this problem of separating cardiac and respiration sounds in the sum signal recorded on a human chest wall.

Source levels of social sounds in migrating humpback whales (Megaptera novaeangliae).

PubMed

Dunlop, Rebecca A; Cato, Douglas H; Noad, Michael J; Stokes, Dale M

2013-07-01

The source level of an animal sound is important in communication, since it affects the distance over which the sound is audible. Several measurements of source levels of whale sounds have been reported, but the accuracy of many is limited because the distance to the source and the acoustic transmission loss were estimated rather than measured. This paper presents measurements of source levels of social sounds (surface-generated and vocal sounds) of humpback whales from a sample of 998 sounds recorded from 49 migrating humpback whale groups. Sources were localized using a wide baseline five hydrophone array and transmission loss was measured for the site. Social vocalization source levels were found to range from 123 to 183 dB re 1 μPa @ 1 m with a median of 158 dB re 1 μPa @ 1 m. Source levels of surface-generated social sounds ("breaches" and "slaps") were narrower in range (133 to 171 dB re 1 μPa @ 1 m) but slightly higher in level (median of 162 dB re 1 μPa @ 1 m) compared to vocalizations. The data suggest that group composition has an effect on group vocalization source levels in that singletons and mother-calf-singing escort groups tend to vocalize at higher levels compared to other group compositions.

Interdependent encoding of pitch, timbre and spatial location in auditory cortex

PubMed Central

Bizley, Jennifer K.; Walker, Kerry M. M.; Silverman, Bernard W.; King, Andrew J.; Schnupp, Jan W. H.

2009-01-01

Because we can perceive the pitch, timbre and spatial location of a sound source independently, it seems natural to suppose that cortical processing of sounds might separate out spatial from non-spatial attributes. Indeed, recent studies support the existence of anatomically segregated ‘what’ and ‘where’ cortical processing streams. However, few attempts have been made to measure the responses of individual neurons in different cortical fields to sounds that vary simultaneously across spatial and non-spatial dimensions. We recorded responses to artificial vowels presented in virtual acoustic space to investigate the representations of pitch, timbre and sound source azimuth in both core and belt areas of ferret auditory cortex. A variance decomposition technique was used to quantify the way in which altering each parameter changed neural responses. Most units were sensitive to two or more of these stimulus attributes. Whilst indicating that neural encoding of pitch, location and timbre cues is distributed across auditory cortex, significant differences in average neuronal sensitivity were observed across cortical areas and depths, which could form the basis for the segregation of spatial and non-spatial cues at higher cortical levels. Some units exhibited significant non-linear interactions between particular combinations of pitch, timbre and azimuth. These interactions were most pronounced for pitch and timbre and were less commonly observed between spatial and non-spatial attributes. Such non-linearities were most prevalent in primary auditory cortex, although they tended to be small compared with stimulus main effects. PMID:19228960

Wave field synthesis of moving virtual sound sources with complex radiation properties.

PubMed

Ahrens, Jens; Spors, Sascha

2011-11-01

An approach to the synthesis of moving virtual sound sources with complex radiation properties in wave field synthesis is presented. The approach exploits the fact that any stationary sound source of finite spatial extent radiates spherical waves at sufficient distance. The angular dependency of the radiation properties of the source under consideration is reflected by the amplitude and phase distribution on the spherical wave fronts. The sound field emitted by a uniformly moving monopole source is derived and the far-field radiation properties of the complex virtual source under consideration are incorporated in order to derive a closed-form expression for the loudspeaker driving signal. The results are illustrated via numerical simulations of the synthesis of the sound field of a sample moving complex virtual source.

Some factors influencing radiation of sound from flow interaction with edges of finite surfaces

NASA Technical Reports Server (NTRS)

Hayden, R. E.; Fox, H. L.; Chanaud, R. C.

1976-01-01

Edges of surfaces which are exposed to unsteady flow cause both strictly acoustic effects and hydrodynamic effects, in the form of generation of new hydrodynamic sources in the immediate vicinity of the edge. An analytical model is presented which develops the explicit sound-generation role of the velocity and Mach number of the eddy convection past the edge, and the importance of relative scale lengths of the turbulence, as well as the relative intensity of pressure fluctuations. The Mach number (velocity) effects show that the important paramater is the convection Mach number of the eddies. The effects of turbulence scale lengths, isotropy, and spatial density (separation) are shown to be important in determining the level and spectrum of edge sound radiated for the edge dipole mechanism. Experimental data is presented which provides support for the dipole edge noise model in terms of Mach number (velocity) scaling, parametric dependence on flow field parameter, directivity, and edge diffraction effects.

Perceptual Grouping Affects Pitch Judgments Across Time and Frequency

PubMed Central

Borchert, Elizabeth M. O.; Micheyl, Christophe; Oxenham, Andrew J.

2010-01-01

Pitch, the perceptual correlate of fundamental frequency (F0), plays an important role in speech, music and animal vocalizations. Changes in F0 over time help define musical melodies and speech prosody, while comparisons of simultaneous F0 are important for musical harmony, and for segregating competing sound sources. This study compared listeners’ ability to detect differences in F0 between pairs of sequential or simultaneous tones that were filtered into separate, non-overlapping spectral regions. The timbre differences induced by filtering led to poor F0 discrimination in the sequential, but not the simultaneous, conditions. Temporal overlap of the two tones was not sufficient to produce good performance; instead performance appeared to depend on the two tones being integrated into the same perceptual object. The results confirm the difficulty of comparing the pitches of sequential sounds with different timbres and suggest that, for simultaneous sounds, pitch differences may be detected through a decrease in perceptual fusion rather than an explicit coding and comparison of the underlying F0s. PMID:21077719

Depth dependence of wind-driven, broadband ambient noise in the Philippine Sea.

PubMed

Barclay, David R; Buckingham, Michael J

2013-01-01

In 2009, as part of PhilSea09, the instrument platform known as Deep Sound was deployed in the Philippine Sea, descending under gravity to a depth of 6000 m, where it released a drop weight, allowing buoyancy to return it to the surface. On the descent and ascent, at a speed of 0.6 m/s, Deep Sound continuously recorded broadband ambient noise on two vertically aligned hydrophones separated by 0.5 m. For frequencies between 1 and 10 kHz, essentially all the noise was found to be downward traveling, exhibiting a depth-independent directional density function having the simple form cos θ, where θ ≤ 90° is the polar angle measured from the zenith. The spatial coherence and cross-spectral density of the noise show no change in character in the vicinity of the critical depth, consistent with a local, wind-driven surface-source distribution. The coherence function accurately matches that predicted by a simple model of deep-water, wind-generated noise, provided that the theoretical coherence is evaluated using the local sound speed. A straightforward inversion procedure is introduced for recovering the sound speed profile from the cross-correlation function of the noise, returning sound speeds with a root-mean-square error relative to an independently measured profile of 8.2 m/s.

Forced sound transmission through a finite-sized single leaf panel subject to a point source excitation.

PubMed

Wang, Chong

2018-03-01

In the case of a point source in front of a panel, the wavefront of the incident wave is spherical. This paper discusses spherical sound waves transmitting through a finite sized panel. The forced sound transmission performance that predominates in the frequency range below the coincidence frequency is the focus. Given the point source located along the centerline of the panel, forced sound transmission coefficient is derived through introducing the sound radiation impedance for spherical incident waves. It is found that in addition to the panel mass, forced sound transmission loss also depends on the distance from the source to the panel as determined by the radiation impedance. Unlike the case of plane incident waves, sound transmission performance of a finite sized panel does not necessarily converge to that of an infinite panel, especially when the source is away from the panel. For practical applications, the normal incidence sound transmission loss expression of plane incident waves can be used if the distance between the source and panel d and the panel surface area S satisfy d/S>0.5. When d/S ≈0.1, the diffuse field sound transmission loss expression may be a good approximation. An empirical expression for d/S=0  is also given.

Numerical and Experimental Determination of the Geometric Far Field for Round Jets

NASA Technical Reports Server (NTRS)

Koch, L. Danielle; Bridges, James; Brown, Cliff; Khavaran, Abbas

2003-01-01

To reduce ambiguity in the reporting of far field jet noise, three round jets operating at subsonic conditions have recently been studied at the NASA Glenn Research Center. The goal of the investigation was to determine the location of the geometric far field both numerically and experimentally. The combination of the WIND Reynolds-Averaged Navier-Stokes solver and the MGBK jet noise prediction code was used for the computations, and the experimental data was collected in the Aeroacoustic Propulsion Laboratory. While noise sources are distributed throughout the jet plume, at great distances from the nozzle the noise will appear to be emanating from a point source and the assumption of linear propagation is valid. Closer to the jet, nonlinear propagation may be a problem, along with the known geometric issues. By comparing sound spectra at different distances from the jet, both from computational methods that assume linear propagation, and from experiments, the contributions of geometry and nonlinearity can be separately ascertained and the required measurement distance for valid experiments can be established. It is found that while the shortest arc considered here (approx. 8D) was already in the geometric far field for the high frequency sound (St greater than 2.0), the low frequency noise due to its extended source distribution reached the geometric far field at or about 50D. It is also found that sound spectra at far downstream angles does not strictly scale on Strouhal number, an observation that current modeling does not capture.

Orthodontic elastic separator-induced periodontal abscess: a case report.

PubMed

Becker, Talia; Neronov, Alex

2012-01-01

Aim. Orthodontic elastic bands were proposed as being the source of gingival abscesses that can rapidly lead to bone loss and teeth exfoliation. We report an adolescent, otherwise, healthy patient whose periodontal status was sound. Shortly after undergoing preparations for orthodontic treatment consisting of orthodontic separators, he presented with a periodontal abscess for which there was no apparent etiology. A non-orthoradial X-ray was inconclusive, but an appropriate one revealed a subgingival orthodontic separator as the cause of the abscess. Removal of the separator and thorough scaling led to complete resolution of the abscess, but there was already residual mild damage to the alveolar bone. Summary. Failure to use appropriate imaging to reveal the cause of gingival abscesses can result in the delay of implementing treatment and halting irreversible alveolar bone loss. An inflammatory process restricted to the gingiva and refractive to conventional therapy should raise the possibility of a foreign body etiology.

Orthodontic Elastic Separator-Induced Periodontal Abscess: A Case Report

PubMed Central

Becker, Talia; Neronov, Alex

2012-01-01

Aim. Orthodontic elastic bands were proposed as being the source of gingival abscesses that can rapidly lead to bone loss and teeth exfoliation. We report an adolescent, otherwise, healthy patient whose periodontal status was sound. Shortly after undergoing preparations for orthodontic treatment consisting of orthodontic separators, he presented with a periodontal abscess for which there was no apparent etiology. A non-orthoradial X-ray was inconclusive, but an appropriate one revealed a subgingival orthodontic separator as the cause of the abscess. Removal of the separator and thorough scaling led to complete resolution of the abscess, but there was already residual mild damage to the alveolar bone. Summary. Failure to use appropriate imaging to reveal the cause of gingival abscesses can result in the delay of implementing treatment and halting irreversible alveolar bone loss. An inflammatory process restricted to the gingiva and refractive to conventional therapy should raise the possibility of a foreign body etiology. PMID:22567456

Sensitivity of the mouse to changes in azimuthal sound location: Angular separation, spectral composition, and sound level

PubMed Central

Allen, Paul D.; Ison, James R.

2010-01-01

Auditory spatial acuity was measured in mice using prepulse inhibition (PPI) of the acoustic startle reflex (ASR) as the indicator response for stimulus detection. The prepulse was a “speaker swap” (SSwap), shifting a noise between two speakers located along the azimuth. Their angular separation, and the spectral composition and sound level of the noise were varied, as was the interstimulus interval (ISI) between SSwap and ASR elicitation. In Experiment 1 a 180° SSwap of wide band noise (WBN) was compared with WBN Onset and Offset. SSwap and WBN Onset had near equal effects, but less than Offset. In Experiment 2 WBN SSwap was measured with speaker separations of 15°, 22.5°, 45°, and 90°. Asymptotic level and the growth rate of PPI increased with increased separation from 15° to 90°, but even the 15° SSwap provided significant PPI for the mean performance of the group. SSwap in Experiment 3 used octave band noise (2–4, 4–8, 8–16, or 16–32 kHz) and separations of 7.5° to 180°. SSwap was most effective for the highest frequencies, with no significant PPI for SSwap below 8–16 kHz, or for separations of 7.5°. In Experiment 4 SSwap had WBN sound levels from 40 to 78 dB SPL, and separations of 22.5°, 45°, 90° and 180°: PPI increased with level, this effect varying with ISI and angular separation. These experiments extend the prior findings on sound localization in mice, and the dependence of PPI on ISI adds a reaction-time-like dimension to this behavioral analysis. PMID:20364886

33 CFR 167.1323 - In Puget Sound and its approaches: Puget Sound.

Code of Federal Regulations, 2014 CFR

2014-07-01

... 33 Navigation and Navigable Waters 2 2014-07-01 2014-07-01 false In Puget Sound and its approaches: Puget Sound. 167.1323 Section 167.1323 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1323 In Puget Sound and its...

33 CFR 167.1323 - In Puget Sound and its approaches: Puget Sound.

Code of Federal Regulations, 2013 CFR

2013-07-01

... 33 Navigation and Navigable Waters 2 2013-07-01 2013-07-01 false In Puget Sound and its approaches: Puget Sound. 167.1323 Section 167.1323 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1323 In Puget Sound and its...

Noise Source Identification in a Reverberant Field Using Spherical Beamforming

NASA Astrophysics Data System (ADS)

Choi, Young-Chul; Park, Jin-Ho; Yoon, Doo-Byung; Kwon, Hyu-Sang

Identification of noise sources, their locations and strengths, has been taken great attention. The method that can identify noise sources normally assumes that noise sources are located at a free field. However, the sound in a reverberant field consists of that coming directly from the source plus sound reflected or scattered by the walls or objects in the field. In contrast to the exterior sound field, reflections are added to sound field. Therefore, the source location estimated by the conventional methods may give unacceptable error. In this paper, we explain the effects of reverberant field on interior source identification process and propose the method that can identify noise sources in the reverberant field.

Auditory stream segregation in children with Asperger syndrome

PubMed Central

Lepistö, T.; Kuitunen, A.; Sussman, E.; Saalasti, S.; Jansson-Verkasalo, E.; Nieminen-von Wendt, T.; Kujala, T.

2009-01-01

Individuals with Asperger syndrome (AS) often have difficulties in perceiving speech in noisy environments. The present study investigated whether this might be explained by deficient auditory stream segregation ability, that is, by a more basic difficulty in separating simultaneous sound sources from each other. To this end, auditory event-related brain potentials were recorded from a group of school-aged children with AS and a group of age-matched controls using a paradigm specifically developed for studying stream segregation. Differences in the amplitudes of ERP components were found between groups only in the stream segregation conditions and not for simple feature discrimination. The results indicated that children with AS have difficulties in segregating concurrent sound streams, which ultimately may contribute to the difficulties in speech-in-noise perception. PMID:19751798

Spatial release from masking based on binaural processing for up to six maskers

PubMed Central

Yost, William A.

2017-01-01

Spatial Release from Masking (SRM) was measured for identification of a female target word spoken in the presence of male masker words. Target words from a single loudspeaker located at midline were presented when two, four, or six masker words were presented either from the same source as the target or from spatially separated masker sources. All masker words were presented from loudspeakers located symmetrically around the centered target source in the front azimuth hemifield. Three masking conditions were employed: speech-in-speech masking (involving both informational and energetic masking), speech-in-noise masking (involving energetic masking), and filtered speech-in-filtered speech masking (involving informational masking). Psychophysical results were summarized as three-point psychometric functions relating proportion of correct word identification to target-to-masker ratio (in decibels) for both the co-located and spatially separated target and masker sources cases. SRM was then calculated by comparing the slopes and intercepts of these functions. SRM decreased as the number of symmetrically placed masker sources increased from two to six. This decrease was independent of the type of masking, with almost no SRM measured for six masker sources. These results suggest that when SRM is dependent primarily on binaural processing, SRM is effectively limited to fewer than six sound sources. PMID:28372135

Aerodynamic sound of flow past an airfoil

NASA Technical Reports Server (NTRS)

Wang, Meng

1995-01-01

The long term objective of this project is to develop a computational method for predicting the noise of turbulence-airfoil interactions, particularly at the trailing edge. We seek to obtain the energy-containing features of the turbulent boundary layers and the near-wake using Navier-Stokes Simulation (LES or DNS), and then to calculate the far-field acoustic characteristics by means of acoustic analogy theories, using the simulation data as acoustic source functions. Two distinct types of noise can be emitted from airfoil trailing edges. The first, a tonal or narrowband sound caused by vortex shedding, is normally associated with blunt trailing edges, high angles of attack, or laminar flow airfoils. The second source is of broadband nature arising from the aeroacoustic scattering of turbulent eddies by the trailing edge. Due to its importance to airframe noise, rotor and propeller noise, etc., trailing edge noise has been the subject of extensive theoretical (e.g. Crighton & Leppington 1971; Howe 1978) as well as experimental investigations (e.g. Brooks & Hodgson 1981; Blake & Gershfeld 1988). A number of challenges exist concerning acoustic analogy based noise computations. These include the elimination of spurious sound caused by vortices crossing permeable computational boundaries in the wake, the treatment of noncompact source regions, and the accurate description of wave reflection by the solid surface and scattering near the edge. In addition, accurate turbulence statistics in the flow field are required for the evaluation of acoustic source functions. Major efforts to date have been focused on the first two challenges. To this end, a paradigm problem of laminar vortex shedding, generated by a two dimensional, uniform stream past a NACA0012 airfoil, is used to address the relevant numerical issues. Under the low Mach number approximation, the near-field flow quantities are obtained by solving the incompressible Navier-Stokes equations numerically at chord Reynolds number of 104. The far-field noise is computed using Curle's extension to the Lighthill analogy (Curle 1955). An effective method for separating the physical noise source from spurious boundary contributions is developed. This allows an accurate evaluation of the Reynolds stress volume quadrupoles, in addition to the more readily computable surface dipoles due to the unsteady lift and drag. The effect of noncompact source distribution on the far-field sound is assessed using an efficient integration scheme for the Curle integral, with full account of retarded-time variations. The numerical results confirm in quantitative terms that the far-field sound is dominated by the surface pressure dipoles at low Mach number. The techniques developed are applicable to a wide range of flows, including jets and mixing layers, where the Reynolds stress quadrupoles play a prominent or even dominant role in the overall sound generation.

A Numerical Experiment on the Role of Surface Shear Stress in the Generation of Sound

NASA Technical Reports Server (NTRS)

Shariff, Karim; Wang, Meng; Merriam, Marshal (Technical Monitor)

1996-01-01

The sound generated due to a localized flow over an infinite flat surface is considered. It is known that the unsteady surface pressure, while appearing in a formal solution to the Lighthill equation, does not constitute a source of sound but rather represents the effect of image quadrupoles. The question of whether a similar surface shear stress term constitutes a true source of dipole sound is less settled. Some have boldly assumed it is a true source while others have argued that, like the surface pressure, it depends on the sound field (via an acoustic boundary layer) and is therefore not a true source. A numerical experiment based on the viscous, compressible Navier-Stokes equations was undertaken to investigate the issue. A small region of a wall was oscillated tangentially. The directly computed sound field was found to to agree with an acoustic analogy based calculation which regards the surface shear as an acoustically compact dipole source of sound.

CLIVAR Mode Water Dynamics Experiment (CLIMODE), Fall 2006 R/V Oceanus Voyage 434, November 16, 2006-December 3, 2006

DTIC Science & Technology

2007-12-01

except for the dive zero time which needed to be programmed during the cruise when the deployment schedule dates were confirmed. _ ACM - Aanderaa ACM...guards bolted on to complete the frame prior to deployment. Sound Source - Sound sources were scheduled to be redeployed. Sound sources were originally...battery voltages and a vacuum. A +27 second time drift was noted and the time was reset. The sound source was scheduled to go to full power on November

Acoustic Green's function extraction in the ocean

NASA Astrophysics Data System (ADS)

Zang, Xiaoqin

The acoustic Green's function (GF) is the key to understanding the acoustic properties of ocean environments. With knowledge of the acoustic GF, the physics of sound propagation, such as dispersion, can be analyzed; underwater communication over thousands of miles can be understood; physical properties of the ocean, including ocean temperature, ocean current speed, as well as seafloor bathymetry, can be investigated. Experimental methods of acoustic GF extraction can be categorized as active methods and passive methods. Active methods are based on employment of man-made sound sources. These active methods require less computational complexity and time, but may cause harm to marine mammals. Passive methods cost much less and do not harm marine mammals, but require more theoretical and computational work. Both methods have advantages and disadvantages that should be carefully tailored to fit the need of each specific environment and application. In this dissertation, we study one passive method, the noise interferometry method, and one active method, the inverse filter processing method, to achieve acoustic GF extraction in the ocean. The passive method of noise interferometry makes use of ambient noise to extract an approximation to the acoustic GF. In an environment with a diffusive distribution of sound sources, sound waves that pass through two hydrophones at two locations carry the information of the acoustic GF between these two locations; by listening to the long-term ambient noise signals and cross-correlating the noise data recorded at two locations, the acoustic GF emerges from the noise cross-correlation function (NCF); a coherent stack of many realizations of NCFs yields a good approximation to the acoustic GF between these two locations, with all the deterministic structures clearly exhibited in the waveform. To test the performance of noise interferometry in different types of ocean environments, two field experiments were performed and ambient noise data were collected in a 100-meter deep coastal ocean environment and a 600-meter deep ocean environment. In the coastal ocean environment, the collected noise data were processed by coherently stacking five days of cross-correlation functions between pairs of hydrophones separated by 5 km, 10 km and 15 km, respectively. NCF waveforms were modeled using the KRAKEN normal mode model, with the difference between the NCFs and the acoustic GFs quantified by a weighting function. Through waveform inversion of NCFs, an optimal geoacoustic model was obtained by minimizing the two-norm misfit between the simulation and the measurement. Using a simulated time-reversal mirror, the extracted GF was back propagated from the receiver location to the virtual source, and a strong focus was found in the vicinity of the source, which provides additional support for the optimality of the aforementioned geoacoustic model. With the extracted GF, dispersion in experimental shallow water environment was visualized in the time-frequency representation. Normal modes of GFs were separated using the time-warping transformation. By separating the modes in the frequency domain of the time-warped signal, we isolated modal arrivals and reconstructed the NCF by summing up the isolated modes, thereby significantly improving the signal-to-noise ratio of NCFs. Finally, these reconstructed NCFs were employed to estimate the depth-averaged current speed in the Florida Straits, based on an effective sound speed approximation. In the mid-deep ocean environment, the noise data were processed using the same noise interferometry method, but the obtained NCFs were not as good as those in the coastal ocean environment. Several highly possible reasons of the difference in the noise interferometry performance were investigated and discussed. The first one is the noise source composition, which is different in the spectrograms of noise records in two environments. The second is strong ocean current variability that can result in coherence loss and undermine the utility of coherent stacking. The third one is the downward refracting sound speed profile, which impedes strong coupling between near surface noise sources and the near-bottom instruments. The active method of inverse filter processing was tested in a long-range deep-ocean environment. The high-power sound source, which was located near the sound channel axis, transmitted a pre-designed signal that was composed of a precursor signal and a communication signal. After traveling 1428.5 km distance in the north Pacific Ocean, the transmitted signal was detected by the receiver and was processed using the inverse filter. The probe signal, which was composed of M sequences and was known at the receiver, was utilized for the GF extraction in the inverse filter; the communication signal was then interpreted with the extracted GF. With a glitch in the length of communication signal, the inverse filter processing method was shown to be effective for long-range low-frequency deep ocean acoustic communication. (Abstract shortened by ProQuest.).

An integrated system for dynamic control of auditory perspective in a multichannel sound field

NASA Astrophysics Data System (ADS)

Corey, Jason Andrew

An integrated system providing dynamic control of sound source azimuth, distance and proximity to a room boundary within a simulated acoustic space is proposed for use in multichannel music and film sound production. The system has been investigated, implemented, and psychoacoustically tested within the ITU-R BS.775 recommended five-channel (3/2) loudspeaker layout. The work brings together physical and perceptual models of room simulation to allow dynamic placement of virtual sound sources at any location of a simulated space within the horizontal plane. The control system incorporates a number of modules including simulated room modes, "fuzzy" sources, and tracking early reflections, whose parameters are dynamically changed according to sound source location within the simulated space. The control functions of the basic elements, derived from theories of perception of a source in a real room, have been carefully tuned to provide efficient, effective, and intuitive control of a sound source's perceived location. Seven formal listening tests were conducted to evaluate the effectiveness of the algorithm design choices. The tests evaluated: (1) loudness calibration of multichannel sound images; (2) the effectiveness of distance control; (3) the resolution of distance control provided by the system; (4) the effectiveness of the proposed system when compared to a commercially available multichannel room simulation system in terms of control of source distance and proximity to a room boundary; (5) the role of tracking early reflection patterns on the perception of sound source distance; (6) the role of tracking early reflection patterns on the perception of lateral phantom images. The listening tests confirm the effectiveness of the system for control of perceived sound source distance, proximity to room boundaries, and azimuth, through fine, dynamic adjustment of parameters according to source location. All of the parameters are grouped and controlled together to create a perceptually strong impression of source location and movement within a simulated space.

Sound localization and auditory response capabilities in round goby (Neogobius melanostomus)

NASA Astrophysics Data System (ADS)

Rollo, Audrey K.; Higgs, Dennis M.

2005-04-01

A fundamental role in vertebrate auditory systems is determining the direction of a sound source. While fish show directional responses to sound, sound localization remains in dispute. The species used in the current study, Neogobius melanostomus (round goby) uses sound in reproductive contexts, with both male and female gobies showing directed movement towards a calling male. The two-choice laboratory experiment was used (active versus quiet speaker) to analyze behavior of gobies in response to sound stimuli. When conspecific male spawning sounds were played, gobies moved in a direct path to the active speaker, suggesting true localization to sound. Of the animals that responded to conspecific sounds, 85% of the females and 66% of the males moved directly to the sound source. Auditory playback of natural and synthetic sounds showed differential behavioral specificity. Of gobies that responded, 89% were attracted to the speaker playing Padogobius martensii sounds, 87% to 100 Hz tone, 62% to white noise, and 56% to Gobius niger sounds. Swimming speed, as well as mean path angle to the speaker, will be presented during the presentation. Results suggest a strong localization of the round goby to a sound source, with some differential sound specificity.

Experiments of multichannel least-square methods for sound field reproduction inside aircraft mock-up: Objective evaluations

NASA Astrophysics Data System (ADS)

Gauthier, P.-A.; Camier, C.; Lebel, F.-A.; Pasco, Y.; Berry, A.; Langlois, J.; Verron, C.; Guastavino, C.

2016-08-01

Sound environment reproduction of various flight conditions in aircraft mock-ups is a valuable tool for the study, prediction, demonstration and jury testing of interior aircraft sound quality and annoyance. To provide a faithful reproduced sound environment, time, frequency and spatial characteristics should be preserved. Physical sound field reproduction methods for spatial sound reproduction are mandatory to immerse the listener's body in the proper sound fields so that localization cues are recreated at the listener's ears. Vehicle mock-ups pose specific problems for sound field reproduction. Confined spaces, needs for invisible sound sources and very specific acoustical environment make the use of open-loop sound field reproduction technologies such as wave field synthesis (based on free-field models of monopole sources) not ideal. In this paper, experiments in an aircraft mock-up with multichannel least-square methods and equalization are reported. The novelty is the actual implementation of sound field reproduction with 3180 transfer paths and trim panel reproduction sources in laboratory conditions with a synthetic target sound field. The paper presents objective evaluations of reproduced sound fields using various metrics as well as sound field extrapolation and sound field characterization.

A Comparative Study on Fetal Heart Rates Estimated from Fetal Phonography and Cardiotocography

PubMed Central

Ibrahim, Emad A.; Al Awar, Shamsa; Balayah, Zuhur H.; Hadjileontiadis, Leontios J.; Khandoker, Ahsan H.

2017-01-01

The aim of this study is to investigate that fetal heart rates (fHR) extracted from fetal phonocardiography (fPCG) could convey similar information of fHR from cardiotocography (CTG). Four-channel fPCG sensors made of low cost (<$1) ceramic piezo vibration sensor within 3D-printed casings were used to collect abdominal phonogram signals from 20 pregnant mothers (>34 weeks of gestation). A novel multi-lag covariance matrix-based eigenvalue decomposition technique was used to separate maternal breathing, fetal heart sounds (fHS) and maternal heart sounds (mHS) from abdominal phonogram signals. Prior to the fHR estimation, the fPCG signals were denoised using a multi-resolution wavelet-based filter. The proposed source separation technique was first tested in separating sources from synthetically mixed signals and then on raw abdominal phonogram signals. fHR signals extracted from fPCG signals were validated using simultaneous recorded CTG-based fHR recordings.The experimental results have shown that the fHR derived from the acquired fPCG can be used to detect periods of acceleration and deceleration, which are critical indication of the fetus' well-being. Moreover, a comparative analysis demonstrated that fHRs from CTG and fPCG signals were in good agreement (Bland Altman plot has mean = −0.21 BPM and ±2 SD = ±3) with statistical significance (p < 0.001 and Spearman correlation coefficient ρ = 0.95). The study findings show that fHR estimated from fPCG could be a reliable substitute for fHR from the CTG, opening up the possibility of a low cost monitoring tool for fetal well-being. PMID:29089896

Modeling of reverberant room responses for two-dimensional spatial sound field analysis and synthesis.

PubMed

Bai, Mingsian R; Li, Yi; Chiang, Yi-Hao

2017-10-01

A unified framework is proposed for analysis and synthesis of two-dimensional spatial sound field in reverberant environments. In the sound field analysis (SFA) phase, an unbaffled 24-element circular microphone array is utilized to encode the sound field based on the plane-wave decomposition. Depending on the sparsity of the sound sources, the SFA stage can be implemented in two manners. For sparse-source scenarios, a one-stage algorithm based on compressive sensing algorithm is utilized. Alternatively, a two-stage algorithm can be used, where the minimum power distortionless response beamformer is used to localize the sources and Tikhonov regularization algorithm is used to extract the source amplitudes. In the sound field synthesis (SFS), a 32-element rectangular loudspeaker array is employed to decode the target sound field using pressure matching technique. To establish the room response model, as required in the pressure matching step of the SFS phase, an SFA technique for nonsparse-source scenarios is utilized. Choice of regularization parameters is vital to the reproduced sound field. In the SFS phase, three SFS approaches are compared in terms of localization performance and voice reproduction quality. Experimental results obtained in a reverberant room are presented and reveal that an accurate room response model is vital to immersive rendering of the reproduced sound field.

Sound Source Localization and Speech Understanding in Complex Listening Environments by Single-sided Deaf Listeners After Cochlear Implantation.

PubMed

Zeitler, Daniel M; Dorman, Michael F; Natale, Sarah J; Loiselle, Louise; Yost, William A; Gifford, Rene H

2015-09-01

To assess improvements in sound source localization and speech understanding in complex listening environments after unilateral cochlear implantation for single-sided deafness (SSD). Nonrandomized, open, prospective case series. Tertiary referral center. Nine subjects with a unilateral cochlear implant (CI) for SSD (SSD-CI) were tested. Reference groups for the task of sound source localization included young (n = 45) and older (n = 12) normal-hearing (NH) subjects and 27 bilateral CI (BCI) subjects. Unilateral cochlear implantation. Sound source localization was tested with 13 loudspeakers in a 180 arc in front of the subject. Speech understanding was tested with the subject seated in an 8-loudspeaker sound system arrayed in a 360-degree pattern. Directionally appropriate noise, originally recorded in a restaurant, was played from each loudspeaker. Speech understanding in noise was tested using the Azbio sentence test and sound source localization quantified using root mean square error. All CI subjects showed poorer-than-normal sound source localization. SSD-CI subjects showed a bimodal distribution of scores: six subjects had scores near the mean of those obtained by BCI subjects, whereas three had scores just outside the 95th percentile of NH listeners. Speech understanding improved significantly in the restaurant environment when the signal was presented to the side of the CI. Cochlear implantation for SSD can offer improved speech understanding in complex listening environments and improved sound source localization in both children and adults. On tasks of sound source localization, SSD-CI patients typically perform as well as BCI patients and, in some cases, achieve scores at the upper boundary of normal performance.

Real-Time Visualization of Joint Cavitation

PubMed Central

Rowe, Lindsay

2015-01-01

Cracking sounds emitted from human synovial joints have been attributed historically to the sudden collapse of a cavitation bubble formed as articular surfaces are separated. Unfortunately, bubble collapse as the source of joint cracking is inconsistent with many physical phenomena that define the joint cracking phenomenon. Here we present direct evidence from real-time magnetic resonance imaging that the mechanism of joint cracking is related to cavity formation rather than bubble collapse. In this study, ten metacarpophalangeal joints were studied by inserting the finger of interest into a flexible tube tightened around a length of cable used to provide long-axis traction. Before and after traction, static 3D T1-weighted magnetic resonance images were acquired. During traction, rapid cine magnetic resonance images were obtained from the joint midline at a rate of 3.2 frames per second until the cracking event occurred. As traction forces increased, real-time cine magnetic resonance imaging demonstrated rapid cavity inception at the time of joint separation and sound production after which the resulting cavity remained visible. Our results offer direct experimental evidence that joint cracking is associated with cavity inception rather than collapse of a pre-existing bubble. These observations are consistent with tribonucleation, a known process where opposing surfaces resist separation until a critical point where they then separate rapidly creating sustained gas cavities. Observed previously in vitro, this is the first in-vivo macroscopic demonstration of tribonucleation and as such, provides a new theoretical framework to investigate health outcomes associated with joint cracking. PMID:25875374

Position-dependent hearing in three species of bushcrickets (Tettigoniidae, Orthoptera)

PubMed Central

Lakes-Harlan, Reinhard; Scherberich, Jan

2015-01-01

A primary task of auditory systems is the localization of sound sources in space. Sound source localization in azimuth is usually based on temporal or intensity differences of sounds between the bilaterally arranged ears. In mammals, localization in elevation is possible by transfer functions at the ear, especially the pinnae. Although insects are able to locate sound sources, little attention is given to the mechanisms of acoustic orientation to elevated positions. Here we comparatively analyse the peripheral hearing thresholds of three species of bushcrickets in respect to sound source positions in space. The hearing thresholds across frequencies depend on the location of a sound source in the three-dimensional hearing space in front of the animal. Thresholds differ for different azimuthal positions and for different positions in elevation. This position-dependent frequency tuning is species specific. Largest differences in thresholds between positions are found in Ancylecha fenestrata. Correspondingly, A. fenestrata has a rather complex ear morphology including cuticular folds covering the anterior tympanal membrane. The position-dependent tuning might contribute to sound source localization in the habitats. Acoustic orientation might be a selective factor for the evolution of morphological structures at the bushcricket ear and, speculatively, even for frequency fractioning in the ear. PMID:26543574

Position-dependent hearing in three species of bushcrickets (Tettigoniidae, Orthoptera).

PubMed

Lakes-Harlan, Reinhard; Scherberich, Jan

2015-06-01

A primary task of auditory systems is the localization of sound sources in space. Sound source localization in azimuth is usually based on temporal or intensity differences of sounds between the bilaterally arranged ears. In mammals, localization in elevation is possible by transfer functions at the ear, especially the pinnae. Although insects are able to locate sound sources, little attention is given to the mechanisms of acoustic orientation to elevated positions. Here we comparatively analyse the peripheral hearing thresholds of three species of bushcrickets in respect to sound source positions in space. The hearing thresholds across frequencies depend on the location of a sound source in the three-dimensional hearing space in front of the animal. Thresholds differ for different azimuthal positions and for different positions in elevation. This position-dependent frequency tuning is species specific. Largest differences in thresholds between positions are found in Ancylecha fenestrata. Correspondingly, A. fenestrata has a rather complex ear morphology including cuticular folds covering the anterior tympanal membrane. The position-dependent tuning might contribute to sound source localization in the habitats. Acoustic orientation might be a selective factor for the evolution of morphological structures at the bushcricket ear and, speculatively, even for frequency fractioning in the ear.

Computational study of the interaction between a shock and a near-wall vortex using a weighted compact nonlinear scheme

NASA Astrophysics Data System (ADS)

Zuo, Zhifeng; Maekawa, Hiroshi

2014-02-01

The interaction between a moderate-strength shock wave and a near-wall vortex is studied numerically by solving the two-dimensional, unsteady compressible Navier-Stokes equations using a weighted compact nonlinear scheme with a simple low-dissipation advection upstream splitting method for flux splitting. Our main purpose is to clarify the development of the flow field and the generation of sound waves resulting from the interaction. The effects of the vortex-wall distance on the sound generation associated with variations in the flow structures are also examined. The computational results show that three sound sources are involved in this problem: (i) a quadrupolar sound source due to the shock-vortex interaction; (ii) a dipolar sound source due to the vortex-wall interaction; and (iii) a dipolar sound source due to unsteady wall shear stress. The sound field is the combination of the sound waves produced by all three sound sources. In addition to the interaction of the incident shock with the vortex, a secondary shock-vortex interaction is caused by the reflection of the reflected shock (MR2) from the wall. The flow field is dominated by the primary and secondary shock-vortex interactions. The generation mechanism of the third sound, which is newly discovered, due to the MR2-vortex interaction is presented. The pressure variations generated by (ii) become significant with decreasing vortex-wall distance. The sound waves caused by (iii) are extremely weak compared with those caused by (i) and (ii) and are negligible in the computed sound field.

Virtual environment display for a 3D audio room simulation

NASA Astrophysics Data System (ADS)

Chapin, William L.; Foster, Scott

1992-06-01

Recent developments in virtual 3D audio and synthetic aural environments have produced a complex acoustical room simulation. The acoustical simulation models a room with walls, ceiling, and floor of selected sound reflecting/absorbing characteristics and unlimited independent localizable sound sources. This non-visual acoustic simulation, implemented with 4 audio ConvolvotronsTM by Crystal River Engineering and coupled to the listener with a Poihemus IsotrakTM, tracking the listener's head position and orientation, and stereo headphones returning binaural sound, is quite compelling to most listeners with eyes closed. This immersive effect should be reinforced when properly integrated into a full, multi-sensory virtual environment presentation. This paper discusses the design of an interactive, visual virtual environment, complementing the acoustic model and specified to: 1) allow the listener to freely move about the space, a room of manipulable size, shape, and audio character, while interactively relocating the sound sources; 2) reinforce the listener's feeling of telepresence into the acoustical environment with visual and proprioceptive sensations; 3) enhance the audio with the graphic and interactive components, rather than overwhelm or reduce it; and 4) serve as a research testbed and technology transfer demonstration. The hardware/software design of two demonstration systems, one installed and one portable, are discussed through the development of four iterative configurations. The installed system implements a head-coupled, wide-angle, stereo-optic tracker/viewer and multi-computer simulation control. The portable demonstration system implements a head-mounted wide-angle, stereo-optic display, separate head and pointer electro-magnetic position trackers, a heterogeneous parallel graphics processing system, and object oriented C++ program code.

Reconstruction of sound source signal by analytical passive TR in the environment with airflow

NASA Astrophysics Data System (ADS)

Wei, Long; Li, Min; Yang, Debin; Niu, Feng; Zeng, Wu

2017-03-01

In the acoustic design of air vehicles, the time-domain signals of noise sources on the surface of air vehicles can serve as data support to reveal the noise source generation mechanism, analyze acoustic fatigue, and take measures for noise insulation and reduction. To rapidly reconstruct the time-domain sound source signals in an environment with flow, a method combining the analytical passive time reversal mirror (AP-TR) with a shear flow correction is proposed. In this method, the negative influence of flow on sound wave propagation is suppressed by the shear flow correction, obtaining the corrected acoustic propagation time delay and path. Those corrected time delay and path together with the microphone array signals are then submitted to the AP-TR, reconstructing more accurate sound source signals in the environment with airflow. As an analytical method, AP-TR offers a supplementary way in 3D space to reconstruct the signal of sound source in the environment with airflow instead of the numerical TR. Experiments on the reconstruction of the sound source signals of a pair of loud speakers are conducted in an anechoic wind tunnel with subsonic airflow to validate the effectiveness and priorities of the proposed method. Moreover the comparison by theorem and experiment result between the AP-TR and the time-domain beamforming in reconstructing the sound source signal is also discussed.

Acoustic Localization with Infrasonic Signals

NASA Astrophysics Data System (ADS)

Threatt, Arnesha; Elbing, Brian

2015-11-01

Numerous geophysical and anthropogenic events emit infrasonic frequencies (<20 Hz), including volcanoes, hurricanes, wind turbines and tornadoes. These sounds, which cannot be heard by the human ear, can be detected from large distances (in excess of 100 miles) due to low frequency acoustic signals having a very low decay rate in the atmosphere. Thus infrasound could be used for long-range, passive monitoring and detection of these events. An array of microphones separated by known distances can be used to locate a given source, which is known as acoustic localization. However, acoustic localization with infrasound is particularly challenging due to contamination from other signals, sensitivity to wind noise and producing a trusted source for system development. The objective of the current work is to create an infrasonic source using a propane torch wand or a subwoofer and locate the source using multiple infrasonic microphones. This presentation will present preliminary results from various microphone configurations used to locate the source.

Localization of sound sources in a room with one microphone

NASA Astrophysics Data System (ADS)

Peić Tukuljac, Helena; Lissek, Hervé; Vandergheynst, Pierre

2017-08-01

Estimation of the location of sound sources is usually done using microphone arrays. Such settings provide an environment where we know the difference between the received signals among different microphones in the terms of phase or attenuation, which enables localization of the sound sources. In our solution we exploit the properties of the room transfer function in order to localize a sound source inside a room with only one microphone. The shape of the room and the position of the microphone are assumed to be known. The design guidelines and limitations of the sensing matrix are given. Implementation is based on the sparsity in the terms of voxels in a room that are occupied by a source. What is especially interesting about our solution is that we provide localization of the sound sources not only in the horizontal plane, but in the terms of the 3D coordinates inside the room.

The auditory and non-auditory brain areas involved in tinnitus. An emergent property of multiple parallel overlapping subnetworks

PubMed Central

Vanneste, Sven; De Ridder, Dirk

2012-01-01

Tinnitus is the perception of a sound in the absence of an external sound source. It is characterized by sensory components such as the perceived loudness, the lateralization, the tinnitus type (pure tone, noise-like) and associated emotional components, such as distress and mood changes. Source localization of quantitative electroencephalography (qEEG) data demonstrate the involvement of auditory brain areas as well as several non-auditory brain areas such as the anterior cingulate cortex (dorsal and subgenual), auditory cortex (primary and secondary), dorsal lateral prefrontal cortex, insula, supplementary motor area, orbitofrontal cortex (including the inferior frontal gyrus), parahippocampus, posterior cingulate cortex and the precuneus, in different aspects of tinnitus. Explaining these non-auditory brain areas as constituents of separable subnetworks, each reflecting a specific aspect of the tinnitus percept increases the explanatory power of the non-auditory brain areas involvement in tinnitus. Thus, the unified percept of tinnitus can be considered an emergent property of multiple parallel dynamically changing and partially overlapping subnetworks, each with a specific spontaneous oscillatory pattern and functional connectivity signature. PMID:22586375

Converting a Monopole Emission into a Dipole Using a Subwavelength Structure

NASA Astrophysics Data System (ADS)

Fan, Xu-Dong; Zhu, Yi-Fan; Liang, Bin; Cheng, Jian-chun; Zhang, Likun

2018-03-01

High-efficiency emission of multipoles is unachievable by a source much smaller than the wavelength, preventing compact acoustic devices for generating directional sound beams. Here, we present a primary scheme towards solving this problem by numerically and experimentally enclosing a monopole sound source in a structure with a dimension of around 1 /10 sound wavelength to emit a dipolar field. The radiated sound power is found to be more than twice that of a bare dipole. Our study of efficient emission of directional low-frequency sound from a monopole source in a subwavelength space may have applications such as focused ultrasound for imaging, directional underwater sound beams, miniaturized sonar, etc.

Effect of the spectrum of a high-intensity sound source on the sound-absorbing properties of a resonance-type acoustic lining

NASA Astrophysics Data System (ADS)

Ipatov, M. S.; Ostroumov, M. N.; Sobolev, A. F.

2012-07-01

Experimental results are presented on the effect of both the sound pressure level and the type of spectrum of a sound source on the impedance of an acoustic lining. The spectra under study include those of white noise, a narrow-band signal, and a signal with a preset waveform. It is found that, to obtain reliable data on the impedance of an acoustic lining from the results of interferometric measurements, the total sound pressure level of white noise or the maximal sound pressure level of a pure tone (at every oscillation frequency) needs to be identical to the total sound pressure level of the actual source at the site of acoustic lining on the channel wall.

3D Sound Techniques for Sound Source Elevation in a Loudspeaker Listening Environment

NASA Astrophysics Data System (ADS)

Kim, Yong Guk; Jo, Sungdong; Kim, Hong Kook; Jang, Sei-Jin; Lee, Seok-Pil

In this paper, we propose several 3D sound techniques for sound source elevation in stereo loudspeaker listening environments. The proposed method integrates a head-related transfer function (HRTF) for sound positioning and early reflection for adding reverberant circumstance. In addition, spectral notch filtering and directional band boosting techniques are also included for increasing elevation perception capability. In order to evaluate the elevation performance of the proposed method, subjective listening tests are conducted using several kinds of sound sources such as white noise, sound effects, speech, and music samples. It is shown from the tests that the degrees of perceived elevation by the proposed method are around the 17º to 21º when the stereo loudspeakers are located on the horizontal plane.

Cortical contributions to the auditory frequency-following response revealed by MEG

PubMed Central

Coffey, Emily B. J.; Herholz, Sibylle C.; Chepesiuk, Alexander M. P.; Baillet, Sylvain; Zatorre, Robert J.

2016-01-01

The auditory frequency-following response (FFR) to complex periodic sounds is used to study the subcortical auditory system, and has been proposed as a biomarker for disorders that feature abnormal sound processing. Despite its value in fundamental and clinical research, the neural origins of the FFR are unclear. Using magnetoencephalography, we observe a strong, right-asymmetric contribution to the FFR from the human auditory cortex at the fundamental frequency of the stimulus, in addition to signal from cochlear nucleus, inferior colliculus and medial geniculate. This finding is highly relevant for our understanding of plasticity and pathology in the auditory system, as well as higher-level cognition such as speech and music processing. It suggests that previous interpretations of the FFR may need re-examination using methods that allow for source separation. PMID:27009409

Perceptual constancy in auditory perception of distance to railway tracks.

PubMed

De Coensel, Bert; Nilsson, Mats E; Berglund, Birgitta; Brown, A L

2013-07-01

Distance to a sound source can be accurately estimated solely from auditory information. With a sound source such as a train that is passing by at a relatively large distance, the most important auditory information for the listener for estimating its distance consists of the intensity of the sound, spectral changes in the sound caused by air absorption, and the motion-induced rate of change of intensity. However, these cues are relative because prior information/experience of the sound source-its source power, its spectrum and the typical speed at which it moves-is required for such distance estimates. This paper describes two listening experiments that allow investigation of further prior contextual information taken into account by listeners-viz., whether they are indoors or outdoors. Asked to estimate the distance to the track of a railway, it is shown that listeners assessing sounds heard inside the dwelling based their distance estimates on the expected train passby sound level outdoors rather than on the passby sound level actually experienced indoors. This form of perceptual constancy may have consequences for the assessment of annoyance caused by railway noise.

Recent paleoseismicity record in Prince William Sound, Alaska, USA

NASA Astrophysics Data System (ADS)

Kuehl, Steven A.; Miller, Eric J.; Marshall, Nicole R.; Dellapenna, Timothy M.

2017-12-01

Sedimentological and geochemical investigation of sediment cores collected in the deep (>400 m) central basin of Prince William Sound, along with geochemical fingerprinting of sediment source areas, are used to identify earthquake-generated sediment gravity flows. Prince William Sound receives sediment from two distinct sources: from offshore (primarily Copper River) through Hinchinbrook Inlet, and from sources within the Sound (primarily Columbia Glacier). These sources are found to have diagnostic elemental ratios indicative of provenance; Copper River Basin sediments were significantly higher in Sr/Pb and Cu/Pb, whereas Prince William Sound sediments were significantly higher in K/Ca and Rb/Sr. Within the past century, sediment gravity flows deposited within the deep central channel of Prince William Sound have robust geochemical (provenance) signatures that can be correlated with known moderate to large earthquakes in the region. Given the thick Holocene sequence in the Sound ( 200 m) and correspondingly high sedimentation rates (>1 cm year-1), this relationship suggests that sediments within the central basin of Prince William Sound may contain an extraordinary high-resolution record of paleoseismicity in the region.

A social survey on the noise impact in open-plan working environments in China.

PubMed

Zhang, Mei; Kang, Jian; Jiao, Fenglei

2012-11-01

The aim of this study is to reveal noise impact in open-plan working environments in China, through a series of questionnaire surveys and acoustic measurements in typical open-plan working environments. It has been found that compared to other physical environmental factors in open-plan working environments, people are much less satisfied with the acoustic environment. The noise impact in the surveyed working environments is rather significant, in terms of sound level inside the office, understanding of colleagues' conversation, and the use of background music such as music players. About 30-50% of the interviewees think that various noise sources inside and outside offices are 'very disturbing' and 'disturbing', and the most annoying sounds include noises from outside, ventilation systems, office equipment, and keyboard typing. Using higher panels to separate work space, or working in enclosed offices, are regarded as effective improvement measures, whereas introducing natural sounds to mask unwanted sounds seems to be not preferable. There are significant correlations between the evaluation of acoustic environment and office symptoms, including hypersensitivity to loud sounds, easily getting tired and depression. There are also significant correlations between evaluation of various acoustics-related factors and certain statements relating to job satisfaction, including sensitivity to noise, as well as whether conversations could be heard by colleagues. Copyright © 2012 Elsevier B.V. All rights reserved.

Multiple sound source localization using gammatone auditory filtering and direct sound componence detection

NASA Astrophysics Data System (ADS)

Chen, Huaiyu; Cao, Li

2017-06-01

In order to research multiple sound source localization with room reverberation and background noise, we analyze the shortcomings of traditional broadband MUSIC and ordinary auditory filtering based broadband MUSIC method, then a new broadband MUSIC algorithm with gammatone auditory filtering of frequency component selection control and detection of ascending segment of direct sound componence is proposed. The proposed algorithm controls frequency component within the interested frequency band in multichannel bandpass filter stage. Detecting the direct sound componence of the sound source for suppressing room reverberation interference is also proposed, whose merits are fast calculation and avoiding using more complex de-reverberation processing algorithm. Besides, the pseudo-spectrum of different frequency channels is weighted by their maximum amplitude for every speech frame. Through the simulation and real room reverberation environment experiments, the proposed method has good performance. Dynamic multiple sound source localization experimental results indicate that the average absolute error of azimuth estimated by the proposed algorithm is less and the histogram result has higher angle resolution.

Series expansions of rotating two and three dimensional sound fields.

PubMed

Poletti, M A

2010-12-01

The cylindrical and spherical harmonic expansions of oscillating sound fields rotating at a constant rate are derived. These expansions are a generalized form of the stationary sound field expansions. The derivations are based on the representation of interior and exterior sound fields using the simple source approach and determination of the simple source solutions with uniform rotation. Numerical simulations of rotating sound fields are presented to verify the theory.

Approaches to the study of neural coding of sound source location and sound envelope in real environments

PubMed Central

Kuwada, Shigeyuki; Bishop, Brian; Kim, Duck O.

2012-01-01

The major functions of the auditory system are recognition (what is the sound) and localization (where is the sound). Although each of these has received considerable attention, rarely are they studied in combination. Furthermore, the stimuli used in the bulk of studies did not represent sound location in real environments and ignored the effects of reverberation. Another ignored dimension is the distance of a sound source. Finally, there is a scarcity of studies conducted in unanesthetized animals. We illustrate a set of efficient methods that overcome these shortcomings. We use the virtual auditory space method (VAS) to efficiently present sounds at different azimuths, different distances and in different environments. Additionally, this method allows for efficient switching between binaural and monaural stimulation and alteration of acoustic cues singly or in combination to elucidate neural mechanisms underlying localization and recognition. Such procedures cannot be performed with real sound field stimulation. Our research is designed to address the following questions: Are IC neurons specialized to process what and where auditory information? How does reverberation and distance of the sound source affect this processing? How do IC neurons represent sound source distance? Are neural mechanisms underlying envelope processing binaural or monaural? PMID:22754505

Characterizing large river sounds: Providing context for understanding the environmental effects of noise produced by hydrokinetic turbines.

PubMed

Bevelhimer, Mark S; Deng, Z Daniel; Scherelis, Constantin

2016-01-01

Underwater noise associated with the installation and operation of hydrokinetic turbines in rivers and tidal zones presents a potential environmental concern for fish and marine mammals. Comparing the spectral quality of sounds emitted by hydrokinetic turbines to natural and other anthropogenic sound sources is an initial step at understanding potential environmental impacts. Underwater recordings were obtained from passing vessels and natural underwater sound sources in static and flowing waters. Static water measurements were taken in a lake with minimal background noise. Flowing water measurements were taken at a previously proposed deployment site for hydrokinetic turbines on the Mississippi River, where sounds created by flowing water are part of all measurements, both natural ambient and anthropogenic sources. Vessel sizes ranged from a small fishing boat with 60 hp outboard motor to an 18-unit barge train being pushed upstream by tugboat. As expected, large vessels with large engines created the highest sound levels, which were, on average, 40 dB greater than the sound created by an operating hydrokinetic turbine. A comparison of sound levels from the same sources at different distances using both spherical and cylindrical sound attenuation functions suggests that spherical model results more closely approximate observed sound attenuation.

Spatial hearing ability of the pigmented Guinea pig (Cavia porcellus): Minimum audible angle and spatial release from masking in azimuth.

PubMed

Greene, Nathaniel T; Anbuhl, Kelsey L; Ferber, Alexander T; DeGuzman, Marisa; Allen, Paul D; Tollin, Daniel J

2018-08-01

Despite the common use of guinea pigs in investigations of the neural mechanisms of binaural and spatial hearing, their behavioral capabilities in spatial hearing tasks have surprisingly not been thoroughly investigated. To begin to fill this void, we tested the spatial hearing of adult male guinea pigs in several experiments using a paradigm based on the prepulse inhibition (PPI) of the acoustic startle response. In the first experiment, we presented continuous broadband noise from one speaker location and switched to a second speaker location (the "prepulse") along the azimuth prior to presenting a brief, ∼110 dB SPL startle-eliciting stimulus. We found that the startle response amplitude was systematically reduced for larger changes in speaker swap angle (i.e., greater PPI), indicating that using the speaker "swap" paradigm is sufficient to assess stimulus detection of spatially separated sounds. In a second set of experiments, we swapped low- and high-pass noise across the midline to estimate their ability to utilize interaural time- and level-difference cues, respectively. The results reveal that guinea pigs can utilize both binaural cues to discriminate azimuthal sound sources. A third set of experiments examined spatial release from masking using a continuous broadband noise masker and a broadband chirp signal, both presented concurrently at various speaker locations. In general, animals displayed an increase in startle amplitude (i.e., lower PPI) when the masker was presented at speaker locations near that of the chirp signal, and reduced startle amplitudes (increased PPI) indicating lower detection thresholds when the noise was presented from more distant speaker locations. In summary, these results indicate that guinea pigs can: 1) discriminate changes in source location within a hemifield as well as across the midline, 2) discriminate sources of low- and high-pass sounds, demonstrating that they can effectively utilize both low-frequency interaural time and high-frequency level difference sound localization cues, and 3) utilize spatial release from masking to discriminate sound sources. This report confirms the guinea pig as a suitable spatial hearing model and reinforces prior estimates of guinea pig hearing ability from acoustical and physiological measurements. Copyright © 2018 Elsevier B.V. All rights reserved.

Sound Source Localization Using Non-Conformal Surface Sound Field Transformation Based on Spherical Harmonic Wave Decomposition

PubMed Central

Zhang, Lanyue; Ding, Dandan; Yang, Desen; Wang, Jia; Shi, Jie

2017-01-01

Spherical microphone arrays have been paid increasing attention for their ability to locate a sound source with arbitrary incident angle in three-dimensional space. Low-frequency sound sources are usually located by using spherical near-field acoustic holography. The reconstruction surface and holography surface are conformal surfaces in the conventional sound field transformation based on generalized Fourier transform. When the sound source is on the cylindrical surface, it is difficult to locate by using spherical surface conformal transform. The non-conformal sound field transformation by making a transfer matrix based on spherical harmonic wave decomposition is proposed in this paper, which can achieve the transformation of a spherical surface into a cylindrical surface by using spherical array data. The theoretical expressions of the proposed method are deduced, and the performance of the method is simulated. Moreover, the experiment of sound source localization by using a spherical array with randomly and uniformly distributed elements is carried out. Results show that the non-conformal surface sound field transformation from a spherical surface to a cylindrical surface is realized by using the proposed method. The localization deviation is around 0.01 m, and the resolution is around 0.3 m. The application of the spherical array is extended, and the localization ability of the spherical array is improved. PMID:28489065

Sound reduction by metamaterial-based acoustic enclosure

DOE Office of Scientific and Technical Information (OSTI.GOV)

Yao, Shanshan; Li, Pei; Zhou, Xiaoming

In many practical systems, acoustic radiation control on noise sources contained within a finite volume by an acoustic enclosure is of great importance, but difficult to be accomplished at low frequencies due to the enhanced acoustic-structure interaction. In this work, we propose to use acoustic metamaterials as the enclosure to efficiently reduce sound radiation at their negative-mass frequencies. Based on a circularly-shaped metamaterial model, sound radiation properties by either central or eccentric sources are analyzed by numerical simulations for structured metamaterials. The parametric analyses demonstrate that the barrier thickness, the cavity size, the source type, and the eccentricity of themore » source have a profound effect on the sound reduction. It is found that increasing the thickness of the metamaterial barrier is an efficient approach to achieve large sound reduction over the negative-mass frequencies. These results are helpful in designing highly efficient acoustic enclosures for blockage of sound in low frequencies.« less

Cross-correlation, triangulation, and curved-wavefront focusing of coral reef sound using a bi-linear hydrophone array.

PubMed

Freeman, Simon E; Buckingham, Michael J; Freeman, Lauren A; Lammers, Marc O; D'Spain, Gerald L

2015-01-01

A seven element, bi-linear hydrophone array was deployed over a coral reef in the Papahãnaumokuãkea Marine National Monument, Northwest Hawaiian Islands, in order to investigate the spatial, temporal, and spectral properties of biological sound in an environment free of anthropogenic influences. Local biological sound sources, including snapping shrimp and other organisms, produced curved-wavefront acoustic arrivals at the array, allowing source location via focusing to be performed over an area of 1600 m(2). Initially, however, a rough estimate of source location was obtained from triangulation of pair-wise cross-correlations of the sound. Refinements to these initial source locations, and source frequency information, were then obtained using two techniques, conventional and adaptive focusing. It was found that most of the sources were situated on or inside the reef structure itself, rather than over adjacent sandy areas. Snapping-shrimp-like sounds, all with similar spectral characteristics, originated from individual sources predominantly in one area to the east of the array. To the west, the spectral and spatial distributions of the sources were more varied, suggesting the presence of a multitude of heterogeneous biological processes. In addition to the biological sounds, some low-frequency noise due to distant breaking waves was received from end-fire north of the array.

Changes in acoustic features and their conjunctions are processed by separate neuronal populations.

PubMed

Takegata, R; Huotilainen, M; Rinne, T; Näätänen, R; Winkler, I

2001-03-05

We investigated the relationship between the neuronal populations involved in detecting change in two acoustic features and their conjunction. Equivalent current dipole (ECD) models of the magnetic mismatch negativity (MMNm) generators were calculated for infrequent changes in pitch, perceived sound source location, and the conjunction of these two features. All of these three changes elicited MMNms that were generated in the vicinity of auditory cortex. The location of the ECD best describing the MMNm to the conjunction deviant was anterior to those for the MMNm responses elicited by either one of the constituent features. The present data thus suggest that at least partially separate neuronal populations are involved in detecting change in acoustic features and feature conjunctions.

Underwater Acoustic Source Localisation Among Blind and Sighted Scuba Divers: Comparative study.

PubMed

Cambi, Jacopo; Livi, Ludovica; Livi, Walter

2017-05-01

Many blind individuals demonstrate enhanced auditory spatial discrimination or localisation of sound sources in comparison to sighted subjects. However, this hypothesis has not yet been confirmed with regards to underwater spatial localisation. This study therefore aimed to investigate underwater acoustic source localisation among blind and sighted scuba divers. This study took place between February and June 2015 in Elba, Italy, and involved two experimental groups of divers with either acquired (n = 20) or congenital (n = 10) blindness and a control group of 30 sighted divers. Each subject took part in five attempts at an under-water acoustic source localisation task, in which the divers were requested to swim to the source of a sound originating from one of 24 potential locations. The control group had their sight obscured during the task. The congenitally blind divers demonstrated significantly better underwater sound localisation compared to the control group or those with acquired blindness ( P = 0.0007). In addition, there was a significant correlation between years of blindness and underwater sound localisation ( P <0.0001). Congenital blindness was found to positively affect the ability of a diver to recognise the source of a sound in an underwater environment. As the correct localisation of sounds underwater may help individuals to avoid imminent danger, divers should perform sound localisation tests during training sessions.

Combination sound and vibration isolation curb for rooftop air-handling systems

NASA Astrophysics Data System (ADS)

Paige, Thomas S.

2005-09-01

This paper introduces the new Model ESSR Sound and Vibration Isolation Curb manufactured by Kinetics Noise Control, Inc. This product was specially designed to address all of the common transmission paths associated with noise and vibration sources from roof-mounted air-handling equipment. These include: reduction of airborne fan noise in supply and return air ductwork, reduction of duct rumble and breakout noise, reduction of direct airborne sound transmission through the roof deck, and reduction of vibration and structure-borne noise transmission to the building structure. Upgrade options are available for increased seismic restraint and wind-load protection. The advantages of this new system over the conventional approach of installing separate duct silencers in the room ceiling space below the rooftop unit are discussed. Several case studies are presented with the emphasis on completed projects pertaining to classrooms and school auditorium applications. Some success has also been achieved by adding active noise control components to improve low-frequency attenuation. This is an innovative product designed for conformance with the new classroom acoustics standard ANSI S12.60.

The role of envelope shape in the localization of multiple sound sources and echoes in the barn owl.

PubMed

Baxter, Caitlin S; Nelson, Brian S; Takahashi, Terry T

2013-02-01

Echoes and sounds of independent origin often obscure sounds of interest, but echoes can go undetected under natural listening conditions, a perception called the precedence effect. How does the auditory system distinguish between echoes and independent sources? To investigate, we presented two broadband noises to barn owls (Tyto alba) while varying the similarity of the sounds' envelopes. The carriers of the noises were identical except for a 2- or 3-ms delay. Their onsets and offsets were also synchronized. In owls, sound localization is guided by neural activity on a topographic map of auditory space. When there are two sources concomitantly emitting sounds with overlapping amplitude spectra, space map neurons discharge when the stimulus in their receptive field is louder than the one outside it and when the averaged amplitudes of both sounds are rising. A model incorporating these features calculated the strengths of the two sources' representations on the map (B. S. Nelson and T. T. Takahashi; Neuron 67: 643-655, 2010). The target localized by the owls could be predicted from the model's output. The model also explained why the echo is not localized at short delays: when envelopes are similar, peaks in the leading sound mask corresponding peaks in the echo, weakening the echo's space map representation. When the envelopes are dissimilar, there are few or no corresponding peaks, and the owl localizes whichever source is predicted by the model to be less masked. Thus the precedence effect in the owl is a by-product of a mechanism for representing multiple sound sources on its map.

Sensitivity to Angular and Radial Source Movements as a Function of Acoustic Complexity in Normal and Impaired Hearing

PubMed Central

Grimm, Giso; Hohmann, Volker; Laugesen, Søren; Neher, Tobias

2017-01-01

In contrast to static sounds, spatially dynamic sounds have received little attention in psychoacoustic research so far. This holds true especially for acoustically complex (reverberant, multisource) conditions and impaired hearing. The current study therefore investigated the influence of reverberation and the number of concurrent sound sources on source movement detection in young normal-hearing (YNH) and elderly hearing-impaired (EHI) listeners. A listening environment based on natural environmental sounds was simulated using virtual acoustics and rendered over headphones. Both near-far (‘radial’) and left-right (‘angular’) movements of a frontal target source were considered. The acoustic complexity was varied by adding static lateral distractor sound sources as well as reverberation. Acoustic analyses confirmed the expected changes in stimulus features that are thought to underlie radial and angular source movements under anechoic conditions and suggested a special role of monaural spectral changes under reverberant conditions. Analyses of the detection thresholds showed that, with the exception of the single-source scenarios, the EHI group was less sensitive to source movements than the YNH group, despite adequate stimulus audibility. Adding static sound sources clearly impaired the detectability of angular source movements for the EHI (but not the YNH) group. Reverberation, on the other hand, clearly impaired radial source movement detection for the EHI (but not the YNH) listeners. These results illustrate the feasibility of studying factors related to auditory movement perception with the help of the developed test setup. PMID:28675088

Sensitivity to Angular and Radial Source Movements as a Function of Acoustic Complexity in Normal and Impaired Hearing.

PubMed

Lundbeck, Micha; Grimm, Giso; Hohmann, Volker; Laugesen, Søren; Neher, Tobias

2017-01-01

In contrast to static sounds, spatially dynamic sounds have received little attention in psychoacoustic research so far. This holds true especially for acoustically complex (reverberant, multisource) conditions and impaired hearing. The current study therefore investigated the influence of reverberation and the number of concurrent sound sources on source movement detection in young normal-hearing (YNH) and elderly hearing-impaired (EHI) listeners. A listening environment based on natural environmental sounds was simulated using virtual acoustics and rendered over headphones. Both near-far ('radial') and left-right ('angular') movements of a frontal target source were considered. The acoustic complexity was varied by adding static lateral distractor sound sources as well as reverberation. Acoustic analyses confirmed the expected changes in stimulus features that are thought to underlie radial and angular source movements under anechoic conditions and suggested a special role of monaural spectral changes under reverberant conditions. Analyses of the detection thresholds showed that, with the exception of the single-source scenarios, the EHI group was less sensitive to source movements than the YNH group, despite adequate stimulus audibility. Adding static sound sources clearly impaired the detectability of angular source movements for the EHI (but not the YNH) group. Reverberation, on the other hand, clearly impaired radial source movement detection for the EHI (but not the YNH) listeners. These results illustrate the feasibility of studying factors related to auditory movement perception with the help of the developed test setup.

Auditory stream segregation in monkey auditory cortex: effects of frequency separation, presentation rate, and tone duration

NASA Astrophysics Data System (ADS)

Fishman, Yonatan I.; Arezzo, Joseph C.; Steinschneider, Mitchell

2004-09-01

Auditory stream segregation refers to the organization of sequential sounds into ``perceptual streams'' reflecting individual environmental sound sources. In the present study, sequences of alternating high and low tones, ``...ABAB...,'' similar to those used in psychoacoustic experiments on stream segregation, were presented to awake monkeys while neural activity was recorded in primary auditory cortex (A1). Tone frequency separation (ΔF), tone presentation rate (PR), and tone duration (TD) were systematically varied to examine whether neural responses correlate with effects of these variables on perceptual stream segregation. ``A'' tones were fixed at the best frequency of the recording site, while ``B'' tones were displaced in frequency from ``A'' tones by an amount=ΔF. As PR increased, ``B'' tone responses decreased in amplitude to a greater extent than ``A'' tone responses, yielding neural response patterns dominated by ``A'' tone responses occurring at half the alternation rate. Increasing TD facilitated the differential attenuation of ``B'' tone responses. These findings parallel psychoacoustic data and suggest a physiological model of stream segregation whereby increasing ΔF, PR, or TD enhances spatial differentiation of ``A'' tone and ``B'' tone responses along the tonotopic map in A1.

An experimental investigation of velocity fields in divergent glottal models of the human vocal tract

NASA Astrophysics Data System (ADS)

Erath, Byron D.; Plesniak, Michael W.

2005-09-01

In speech, sound production arises from fluid-structure interactions within the larynx as well as viscous flow phenomena that is most likely to occur during the divergent orientation of the vocal folds. Of particular interest are the flow mechanisms that influence the location of flow separation points on the vocal folds walls. Physiologically scaled pulsatile flow fields in 7.5 times real size static divergent glottal models were investigated. Three divergence angles were investigated using phase-averaged particle image velocimetry (PIV). The pulsatile glottal jet exhibited a bi-modal stability toward both glottal walls, although there was a significant amount of variance in the angle the jet deflected from the midline. The attachment of the Coanda effect to the glottal model walls occurred when the pulsatile velocity was a maximum, and the acceleration of the waveform was zero. The location of the separation and reattachment points of the flow from the glottal models was a function of the velocity waveform and divergence angle. Acoustic analogies show that a dipole sound source contribution arising from the fluid interaction (Coanda jet) with the vocal fold walls is expected. [Work funded by NIH Grant RO1 DC03577.

Acoustophoretic separation of airborne millimeter-size particles by a Fresnel lens.

PubMed

Cicek, Ahmet; Korozlu, Nurettin; Adem Kaya, Olgun; Ulug, Bulent

2017-03-02

We numerically demonstrate acoustophoretic separation of spherical solid particles in air by means of an acoustic Fresnel lens. Beside gravitational and drag forces, freely-falling millimeter-size particles experience large acoustic radiation forces around the focus of the lens, where interplay of forces lead to differentiation of particle trajectories with respect to either size or material properties. Due to the strong acoustic field at the focus, radiation force can divert particles with source intensities significantly smaller than those required for acoustic levitation in a standing field. When the lens is designed to have a focal length of 100 mm at 25 kHz, finite-element method simulations reveal a sharp focus with a full-width at half-maximum of 0.5 wavelenghts and a field enhancement of 18 dB. Through numerical calculation of forces and simulation of particle trajectories, we demonstrate size-based separation of acrylic particles at a source sound pressure level of 153 dB such that particles with diameters larger than 0.5 mm are admitted into the central hole, whereas smaller particles are rejected. Besides, efficient separation of particles with similar acoustic properties such as polyethylene, polystyrene and acrylic particles of the same size is also demonstrated.

Acoustophoretic separation of airborne millimeter-size particles by a Fresnel lens

NASA Astrophysics Data System (ADS)

Cicek, Ahmet; Korozlu, Nurettin; Adem Kaya, Olgun; Ulug, Bulent

2017-03-01

We numerically demonstrate acoustophoretic separation of spherical solid particles in air by means of an acoustic Fresnel lens. Beside gravitational and drag forces, freely-falling millimeter-size particles experience large acoustic radiation forces around the focus of the lens, where interplay of forces lead to differentiation of particle trajectories with respect to either size or material properties. Due to the strong acoustic field at the focus, radiation force can divert particles with source intensities significantly smaller than those required for acoustic levitation in a standing field. When the lens is designed to have a focal length of 100 mm at 25 kHz, finite-element method simulations reveal a sharp focus with a full-width at half-maximum of 0.5 wavelenghts and a field enhancement of 18 dB. Through numerical calculation of forces and simulation of particle trajectories, we demonstrate size-based separation of acrylic particles at a source sound pressure level of 153 dB such that particles with diameters larger than 0.5 mm are admitted into the central hole, whereas smaller particles are rejected. Besides, efficient separation of particles with similar acoustic properties such as polyethylene, polystyrene and acrylic particles of the same size is also demonstrated.

Acoustophoretic separation of airborne millimeter-size particles by a Fresnel lens

PubMed Central

Cicek, Ahmet; Korozlu, Nurettin; Adem Kaya, Olgun; Ulug, Bulent

2017-01-01

We numerically demonstrate acoustophoretic separation of spherical solid particles in air by means of an acoustic Fresnel lens. Beside gravitational and drag forces, freely-falling millimeter-size particles experience large acoustic radiation forces around the focus of the lens, where interplay of forces lead to differentiation of particle trajectories with respect to either size or material properties. Due to the strong acoustic field at the focus, radiation force can divert particles with source intensities significantly smaller than those required for acoustic levitation in a standing field. When the lens is designed to have a focal length of 100 mm at 25 kHz, finite-element method simulations reveal a sharp focus with a full-width at half-maximum of 0.5 wavelenghts and a field enhancement of 18 dB. Through numerical calculation of forces and simulation of particle trajectories, we demonstrate size-based separation of acrylic particles at a source sound pressure level of 153 dB such that particles with diameters larger than 0.5 mm are admitted into the central hole, whereas smaller particles are rejected. Besides, efficient separation of particles with similar acoustic properties such as polyethylene, polystyrene and acrylic particles of the same size is also demonstrated. PMID:28252033

Achieving perceptually-accurate aural telepresence

NASA Astrophysics Data System (ADS)

Henderson, Paul D.

Immersive multimedia requires not only realistic visual imagery but also a perceptually-accurate aural experience. A sound field may be presented simultaneously to a listener via a loudspeaker rendering system using the direct sound from acoustic sources as well as a simulation or "auralization" of room acoustics. Beginning with classical Wave-Field Synthesis (WFS), improvements are made to correct for asymmetries in loudspeaker array geometry. Presented is a new Spatially-Equalized WFS (SE-WFS) technique to maintain the energy-time balance of a simulated room by equalizing the reproduced spectrum at the listener for a distribution of possible source angles. Each reproduced source or reflection is filtered according to its incidence angle to the listener. An SE-WFS loudspeaker array of arbitrary geometry reproduces the sound field of a room with correct spectral and temporal balance, compared with classically-processed WFS systems. Localization accuracy of human listeners in SE-WFS sound fields is quantified by psychoacoustical testing. At a loudspeaker spacing of 0.17 m (equivalent to an aliasing cutoff frequency of 1 kHz), SE-WFS exhibits a localization blur of 3 degrees, nearly equal to real point sources. Increasing the loudspeaker spacing to 0.68 m (for a cutoff frequency of 170 Hz) results in a blur of less than 5 degrees. In contrast, stereophonic reproduction is less accurate with a blur of 7 degrees. The ventriloquist effect is psychometrically investigated to determine the effect of an intentional directional incongruence between audio and video stimuli. Subjects were presented with prerecorded full-spectrum speech and motion video of a talker's head as well as broadband noise bursts with a static image. The video image was displaced from the audio stimulus in azimuth by varying amounts, and the perceived auditory location measured. A strong bias was detectable for small angular discrepancies between audio and video stimuli for separations of less than 8 degrees for speech and less than 4 degrees with a pink noise burst. The results allow for the density of WFS systems to be selected from the required localization accuracy. Also, by exploiting the ventriloquist effect, the angular resolution of an audio rendering may be reduced when combined with spatially-accurate video.

A training system of orientation and mobility for blind people using acoustic virtual reality.

PubMed

Seki, Yoshikazu; Sato, Tetsuji

2011-02-01

A new auditory orientation training system was developed for blind people using acoustic virtual reality (VR) based on a head-related transfer function (HRTF) simulation. The present training system can reproduce a virtual training environment for orientation and mobility (O&M) instruction, and the trainee can walk through the virtual training environment safely by listening to sounds such as vehicles, stores, ambient noise, etc., three-dimensionally through headphones. The system can reproduce not only sound sources but also sound reflection and insulation, so that the trainee can learn both sound location and obstacle perception skills. The virtual training environment is described in extensible markup language (XML), and the O&M instructor can edit it easily according to the training curriculum. Evaluation experiments were conducted to test the efficiency of some features of the system. Thirty subjects who had not acquired O&M skills attended the experiments. The subjects were separated into three groups: a no-training group, a virtual-training group using the present system, and a real-training group in real environments. The results suggested that virtual-training can reduce "veering" more than real-training and also can reduce stress as much as real training. The subjective technical and anxiety scores also improved.

Research on Hydrodynamic Interference Suppression of Bottom-Mounted Monitoring Platform with Fairing Structure

NASA Astrophysics Data System (ADS)

Wang, Zhen; Zheng, Yi; Mao, Yu-feng; Wang, Ya-zhou; Yu, Yan-ting; Liu, Hong-ning

2018-03-01

In the disturbance of unsteady flow field under the sea, the monitoring accuracy and precision of the bottom-mounted acoustic monitoring platform will decrease. In order to reduce the hydrodynamic interference, the platform wrapped with fairing structure and separated from the retrieval unit is described. The suppression effect evaluation based on the correlation theory of sound pressure and particle velocity for spherical wave in infinite homogeneous medium is proposed and the difference value between them is used to evaluate the hydrodynamic restraining performance of the bottom-mounted platform under far field condition. Through the sea test, it is indicated that the platform with sparse layers fairing structure (there are two layers for the fairing, in which the inside layer is 6-layers sparse metal net, and the outside layer is 1-layer polyester cloth, and then it takes sparse layers for short) has no attenuation in the sound pressure response to the sound source signal, but obvious suppression in the velocity response to the hydrodynamic noise. The effective frequency of the fairing structure is decreased below 10 Hz, and the noise magnitude is reduced by 10 dB. With the comparison of different fairing structures, it is concluded that the tighter fairing structure can enhance the performance of sound transmission and flow restraining.

Characterizing large river sounds: Providing context for understanding the environmental effects of noise produced by hydrokinetic turbines

DOE PAGES

Bevelhimer, Mark S.; Deng, Z. Daniel; Scherelis, Constantin C.

2016-01-06

Underwaternoise associated with the installation and operation of hydrokinetic turbines in rivers and tidal zones presents a potential environmental concern for fish and marine mammals. Comparing the spectral quality of sounds emitted by hydrokinetic turbines to natural and other anthropogenic sound sources is an initial step at understanding potential environmental impacts. Underwater recordings were obtained from passing vessels and natural underwater sound sources in static and flowing waters. Static water measurements were taken in a lake with minimal background noise. Flowing water measurements were taken at a previously proposed deployment site for hydrokinetic turbines on the Mississippi River, where soundsmore » created by flowing water are part of all measurements, both natural ambient and anthropogenic sources. Vessel sizes ranged from a small fishing boat with 60 hp outboard motor to an 18-unit barge train being pushed upstream by tugboat. As expected, large vessels with large engines created the highest sound levels, which were, on average, 40 dB greater than the sound created by an operating hydrokinetic turbine. As a result, a comparison of sound levels from the same sources at different distances using both spherical and cylindrical sound attenuation functions suggests that spherical model results more closely approximate observed sound attenuation.« less

Personal sound zone reproduction with room reflections

NASA Astrophysics Data System (ADS)

Olik, Marek

Loudspeaker-based sound systems, capable of a convincing reproduction of different audio streams to listeners in the same acoustic enclosure, are a convenient alternative to headphones. Such systems aim to generate "sound zones" in which target sound programmes are to be reproduced with minimum interference from any alternative programmes. This can be achieved with appropriate filtering of the source (loudspeaker) signals, so that the target sound's energy is directed to the chosen zone while being attenuated elsewhere. The existing methods are unable to produce the required sound energy ratio (acoustic contrast) between the zones with a small number of sources when strong room reflections are present. Optimization of parameters is therefore required for systems with practical limitations to improve their performance in reflective acoustic environments. One important parameter is positioning of sources with respect to the zones and room boundaries. The first contribution of this thesis is a comparison of the key sound zoning methods implemented on compact and distributed geometrical source arrangements. The study presents previously unpublished detailed evaluation and ranking of such arrangements for systems with a limited number of sources in a reflective acoustic environment similar to a domestic room. Motivated by the requirement to investigate the relationship between source positioning and performance in detail, the central contribution of this thesis is a study on optimizing source arrangements when strong individual room reflections occur. Small sound zone systems are studied analytically and numerically to reveal relationships between the geometry of source arrays and performance in terms of acoustic contrast and array effort (related to system efficiency). Three novel source position optimization techniques are proposed to increase the contrast, and geometrical means of reducing the effort are determined. Contrary to previously published case studies, this work presents a systematic examination of the key problem of first order reflections and proposes general optimization techniques, thus forming an important contribution. The remaining contribution considers evaluation and comparison of the proposed techniques with two alternative approaches to sound zone generation under reflective conditions: acoustic contrast control (ACC) combined with anechoic source optimization and sound power minimization (SPM). The study provides a ranking of the examined approaches which could serve as a guideline for method selection for rooms with strong individual reflections.

Marine mammal audibility of selected shallow-water survey sources.

PubMed

MacGillivray, Alexander O; Racca, Roberto; Li, Zizheng

2014-01-01

Most attention about the acoustic effects of marine survey sound sources on marine mammals has focused on airgun arrays, with other common sources receiving less scrutiny. Sound levels above hearing threshold (sensation levels) were modeled for six marine mammal species and seven different survey sources in shallow water. The model indicated that odontocetes were most likely to hear sounds from mid-frequency sources (fishery, communication, and hydrographic systems), mysticetes from low-frequency sources (sub-bottom profiler and airguns), and pinnipeds from both mid- and low-frequency sources. High-frequency sources (side-scan and multibeam) generated the lowest estimated sensation levels for all marine mammal species groups.

Performance of active feedforward control systems in non-ideal, synthesized diffuse sound fields.

PubMed

Misol, Malte; Bloch, Christian; Monner, Hans Peter; Sinapius, Michael

2014-04-01

The acoustic performance of passive or active panel structures is usually tested in sound transmission loss facilities. A reverberant sending room, equipped with one or a number of independent sound sources, is used to generate a diffuse sound field excitation which acts as a disturbance source on the structure under investigation. The spatial correlation and coherence of such a synthesized non-ideal diffuse-sound-field excitation, however, might deviate significantly from the ideal case. This has consequences for the operation of an active feedforward control system which heavily relies on the acquisition of coherent disturbance source information. This work, therefore, evaluates the spatial correlation and coherence of ideal and non-ideal diffuse sound fields and considers the implications on the performance of a feedforward control system. The system under consideration is an aircraft-typical double panel system, equipped with an active sidewall panel (lining), which is realized in a transmission loss facility. Experimental results for different numbers of sound sources in the reverberation room are compared to simulation results of a comparable generic double panel system excited by an ideal diffuse sound field. It is shown that the number of statistically independent noise sources acting on the primary structure of the double panel system depends not only on the type of diffuse sound field but also on the sample lengths of the processed signals. The experimental results show that the number of reference sensors required for a defined control performance exhibits an inverse relationship to control filter length.

The influence of underwater data transmission sounds on the displacement behaviour of captive harbour seals (Phoca vitulina).

PubMed

Kastelein, Ronald A; van der Heul, Sander; Verboom, Willem C; Triesscheijn, Rob J V; Jennings, Nancy V

2006-02-01

To prevent grounding of ships and collisions between ships in shallow coastal waters, an underwater data collection and communication network (ACME) using underwater sounds to encode and transmit data is currently under development. Marine mammals might be affected by ACME sounds since they may use sound of a similar frequency (around 12 kHz) for communication, orientation, and prey location. If marine mammals tend to avoid the vicinity of the acoustic transmitters, they may be kept away from ecologically important areas by ACME sounds. One marine mammal species that may be affected in the North Sea is the harbour seal (Phoca vitulina). No information is available on the effects of ACME-like sounds on harbour seals, so this study was carried out as part of an environmental impact assessment program. Nine captive harbour seals were subjected to four sound types, three of which may be used in the underwater acoustic data communication network. The effect of each sound was judged by comparing the animals' location in a pool during test periods to that during baseline periods, during which no sound was produced. Each of the four sounds could be made into a deterrent by increasing its amplitude. The seals reacted by swimming away from the sound source. The sound pressure level (SPL) at the acoustic discomfort threshold was established for each of the four sounds. The acoustic discomfort threshold is defined as the boundary between the areas that the animals generally occupied during the transmission of the sounds and the areas that they generally did not enter during transmission. The SPLs at the acoustic discomfort thresholds were similar for each of the sounds (107 dB re 1 microPa). Based on this discomfort threshold SPL, discomfort zones at sea for several source levels (130-180 dB re 1 microPa) of the sounds were calculated, using a guideline sound propagation model for shallow water. The discomfort zone is defined as the area around a sound source that harbour seals are expected to avoid. The definition of the discomfort zone is based on behavioural discomfort, and does not necessarily coincide with the physical discomfort zone. Based on these results, source levels can be selected that have an acceptable effect on harbour seals in particular areas. The discomfort zone of a communication sound depends on the sound, the source level, and the propagation characteristics of the area in which the sound system is operational. The source level of the communication system should be adapted to each area (taking into account the width of a sea arm, the local sound propagation, and the importance of an area to the affected species). The discomfort zone should not coincide with ecologically important areas (for instance resting, breeding, suckling, and feeding areas), or routes between these areas.

Silent oceans: ocean acidification impoverishes natural soundscapes by altering sound production of the world's noisiest marine invertebrate.

PubMed

Rossi, Tullio; Connell, Sean D; Nagelkerken, Ivan

2016-03-16

Soundscapes are multidimensional spaces that carry meaningful information for many species about the location and quality of nearby and distant resources. Because soundscapes are the sum of the acoustic signals produced by individual organisms and their interactions, they can be used as a proxy for the condition of whole ecosystems and their occupants. Ocean acidification resulting from anthropogenic CO2 emissions is known to have profound effects on marine life. However, despite the increasingly recognized ecological importance of soundscapes, there is no empirical test of whether ocean acidification can affect biological sound production. Using field recordings obtained from three geographically separated natural CO2 vents, we show that forecasted end-of-century ocean acidification conditions can profoundly reduce the biological sound level and frequency of snapping shrimp snaps. Snapping shrimp were among the noisiest marine organisms and the suppression of their sound production at vents was responsible for the vast majority of the soundscape alteration observed. To assess mechanisms that could account for these observations, we tested whether long-term exposure (two to three months) to elevated CO2 induced a similar reduction in the snapping behaviour (loudness and frequency) of snapping shrimp. The results indicated that the soniferous behaviour of these animals was substantially reduced in both frequency (snaps per minute) and sound level of snaps produced. As coastal marine soundscapes are dominated by biological sounds produced by snapping shrimp, the observed suppression of this component of soundscapes could have important and possibly pervasive ecological consequences for organisms that use soundscapes as a source of information. This trend towards silence could be of particular importance for those species whose larval stages use sound for orientation towards settlement habitats. © 2016 The Author(s).

Silent oceans: ocean acidification impoverishes natural soundscapes by altering sound production of the world's noisiest marine invertebrate

PubMed Central

Rossi, Tullio; Connell, Sean D.; Nagelkerken, Ivan

2016-01-01

Soundscapes are multidimensional spaces that carry meaningful information for many species about the location and quality of nearby and distant resources. Because soundscapes are the sum of the acoustic signals produced by individual organisms and their interactions, they can be used as a proxy for the condition of whole ecosystems and their occupants. Ocean acidification resulting from anthropogenic CO2 emissions is known to have profound effects on marine life. However, despite the increasingly recognized ecological importance of soundscapes, there is no empirical test of whether ocean acidification can affect biological sound production. Using field recordings obtained from three geographically separated natural CO2 vents, we show that forecasted end-of-century ocean acidification conditions can profoundly reduce the biological sound level and frequency of snapping shrimp snaps. Snapping shrimp were among the noisiest marine organisms and the suppression of their sound production at vents was responsible for the vast majority of the soundscape alteration observed. To assess mechanisms that could account for these observations, we tested whether long-term exposure (two to three months) to elevated CO2 induced a similar reduction in the snapping behaviour (loudness and frequency) of snapping shrimp. The results indicated that the soniferous behaviour of these animals was substantially reduced in both frequency (snaps per minute) and sound level of snaps produced. As coastal marine soundscapes are dominated by biological sounds produced by snapping shrimp, the observed suppression of this component of soundscapes could have important and possibly pervasive ecological consequences for organisms that use soundscapes as a source of information. This trend towards silence could be of particular importance for those species whose larval stages use sound for orientation towards settlement habitats. PMID:26984624

The Emergence of the Allophonic Perception of Unfamiliar Speech Sounds: The Effects of Contextual Distribution and Phonetic Naturalness

ERIC Educational Resources Information Center

Noguchi, Masaki; Hudson Kam, Carla L.

2018-01-01

In human languages, different speech sounds can be contextual variants of a single phoneme, called allophones. Learning which sounds are allophones is an integral part of the acquisition of phonemes. Whether given sounds are separate phonemes or allophones in a listener's language affects speech perception. Listeners tend to be less sensitive to…

Monitoring Anthropogenic Ocean Sound from Shipping Using an Acoustic Sensor Network and a Compressive Sensing Approach.

PubMed

Harris, Peter; Philip, Rachel; Robinson, Stephen; Wang, Lian

2016-03-22

Monitoring ocean acoustic noise has been the subject of considerable recent study, motivated by the desire to assess the impact of anthropogenic noise on marine life. A combination of measuring ocean sound using an acoustic sensor network and modelling sources of sound and sound propagation has been proposed as an approach to estimating the acoustic noise map within a region of interest. However, strategies for developing a monitoring network are not well established. In this paper, considerations for designing a network are investigated using a simulated scenario based on the measurement of sound from ships in a shipping lane. Using models for the sources of the sound and for sound propagation, a noise map is calculated and measurements of the noise map by a sensor network within the region of interest are simulated. A compressive sensing algorithm, which exploits the sparsity of the representation of the noise map in terms of the sources, is used to estimate the locations and levels of the sources and thence the entire noise map within the region of interest. It is shown that although the spatial resolution to which the sound sources can be identified is generally limited, estimates of aggregated measures of the noise map can be obtained that are more reliable compared with those provided by other approaches.

Monitoring Anthropogenic Ocean Sound from Shipping Using an Acoustic Sensor Network and a Compressive Sensing Approach †

PubMed Central

Harris, Peter; Philip, Rachel; Robinson, Stephen; Wang, Lian

2016-01-01

Monitoring ocean acoustic noise has been the subject of considerable recent study, motivated by the desire to assess the impact of anthropogenic noise on marine life. A combination of measuring ocean sound using an acoustic sensor network and modelling sources of sound and sound propagation has been proposed as an approach to estimating the acoustic noise map within a region of interest. However, strategies for developing a monitoring network are not well established. In this paper, considerations for designing a network are investigated using a simulated scenario based on the measurement of sound from ships in a shipping lane. Using models for the sources of the sound and for sound propagation, a noise map is calculated and measurements of the noise map by a sensor network within the region of interest are simulated. A compressive sensing algorithm, which exploits the sparsity of the representation of the noise map in terms of the sources, is used to estimate the locations and levels of the sources and thence the entire noise map within the region of interest. It is shown that although the spatial resolution to which the sound sources can be identified is generally limited, estimates of aggregated measures of the noise map can be obtained that are more reliable compared with those provided by other approaches. PMID:27011187

Underwater Acoustic Source Localisation Among Blind and Sighted Scuba Divers

PubMed Central

Cambi, Jacopo; Livi, Ludovica; Livi, Walter

2017-01-01

Objectives Many blind individuals demonstrate enhanced auditory spatial discrimination or localisation of sound sources in comparison to sighted subjects. However, this hypothesis has not yet been confirmed with regards to underwater spatial localisation. This study therefore aimed to investigate underwater acoustic source localisation among blind and sighted scuba divers. Methods This study took place between February and June 2015 in Elba, Italy, and involved two experimental groups of divers with either acquired (n = 20) or congenital (n = 10) blindness and a control group of 30 sighted divers. Each subject took part in five attempts at an under-water acoustic source localisation task, in which the divers were requested to swim to the source of a sound originating from one of 24 potential locations. The control group had their sight obscured during the task. Results The congenitally blind divers demonstrated significantly better underwater sound localisation compared to the control group or those with acquired blindness (P = 0.0007). In addition, there was a significant correlation between years of blindness and underwater sound localisation (P <0.0001). Conclusion Congenital blindness was found to positively affect the ability of a diver to recognise the source of a sound in an underwater environment. As the correct localisation of sounds underwater may help individuals to avoid imminent danger, divers should perform sound localisation tests during training sessions. PMID:28690888

The role of envelope shape in the localization of multiple sound sources and echoes in the barn owl

PubMed Central

Baxter, Caitlin S.; Takahashi, Terry T.

2013-01-01

Echoes and sounds of independent origin often obscure sounds of interest, but echoes can go undetected under natural listening conditions, a perception called the precedence effect. How does the auditory system distinguish between echoes and independent sources? To investigate, we presented two broadband noises to barn owls (Tyto alba) while varying the similarity of the sounds' envelopes. The carriers of the noises were identical except for a 2- or 3-ms delay. Their onsets and offsets were also synchronized. In owls, sound localization is guided by neural activity on a topographic map of auditory space. When there are two sources concomitantly emitting sounds with overlapping amplitude spectra, space map neurons discharge when the stimulus in their receptive field is louder than the one outside it and when the averaged amplitudes of both sounds are rising. A model incorporating these features calculated the strengths of the two sources' representations on the map (B. S. Nelson and T. T. Takahashi; Neuron 67: 643–655, 2010). The target localized by the owls could be predicted from the model's output. The model also explained why the echo is not localized at short delays: when envelopes are similar, peaks in the leading sound mask corresponding peaks in the echo, weakening the echo's space map representation. When the envelopes are dissimilar, there are few or no corresponding peaks, and the owl localizes whichever source is predicted by the model to be less masked. Thus the precedence effect in the owl is a by-product of a mechanism for representing multiple sound sources on its map. PMID:23175801

Comparison of sound reproduction using higher order loudspeakers and equivalent line arrays in free-field conditions.

PubMed

Poletti, Mark A; Betlehem, Terence; Abhayapala, Thushara D

2014-07-01

Higher order sound sources of Nth order can radiate sound with 2N + 1 orthogonal radiation patterns, which can be represented as phase modes or, equivalently, amplitude modes. This paper shows that each phase mode response produces a spiral wave front with a different spiral rate, and therefore a different direction of arrival of sound. Hence, for a given receiver position a higher order source is equivalent to a linear array of 2N + 1 monopole sources. This interpretation suggests performance similar to a circular array of higher order sources can be produced by an array of sources, each of which consists of a line array having monopoles at the apparent source locations of the corresponding phase modes. Simulations of higher order arrays and arrays of equivalent line sources are presented. It is shown that the interior fields produced by the two arrays are essentially the same, but that the exterior fields differ because the higher order sources produces different equivalent source locations for field positions outside the array. This work provides an explanation of the fact that an array of L Nth order sources can reproduce sound fields whose accuracy approaches the performance of (2N + 1)L monopoles.

Structure of supersonic jet flow and its radiated sound

NASA Technical Reports Server (NTRS)

Mankbadi, Reda R.; Hayer, M. Ehtesham; Povinelli, Louis A.

1994-01-01

The present paper explores the use of large-eddy simulations as a tool for predicting noise from first principles. A high-order numerical scheme is used to perform large-eddy simulations of a supersonic jet flow with emphasis on capturing the time-dependent flow structure representating the sound source. The wavelike nature of this structure under random inflow disturbances is demonstrated. This wavelike structure is then enhanced by taking the inflow disturbances to be purely harmonic. Application of Lighthill's theory to calculate the far-field noise, with the sound source obtained from the calculated time-dependent near field, is demonstrated. Alternative approaches to coupling the near-field sound source to the far-field sound are discussed.

Spacecraft Internal Acoustic Environment Modeling

NASA Technical Reports Server (NTRS)

Chu, SShao-sheng R.; Allen, Christopher S.

2009-01-01

Acoustic modeling can be used to identify key noise sources, determine/analyze sub-allocated requirements, keep track of the accumulation of minor noise sources, and to predict vehicle noise levels at various stages in vehicle development, first with estimates of noise sources, later with experimental data. In FY09, the physical mockup developed in FY08, with interior geometric shape similar to Orion CM (Crew Module) IML (Interior Mode Line), was used to validate SEA (Statistical Energy Analysis) acoustic model development with realistic ventilation fan sources. The sound power levels of these sources were unknown a priori, as opposed to previous studies that RSS (Reference Sound Source) with known sound power level was used. The modeling results were evaluated based on comparisons to measurements of sound pressure levels over a wide frequency range, including the frequency range where SEA gives good results. Sound intensity measurement was performed over a rectangular-shaped grid system enclosing the ventilation fan source. Sound intensities were measured at the top, front, back, right, and left surfaces of the and system. Sound intensity at the bottom surface was not measured, but sound blocking material was placed tinder the bottom surface to reflect most of the incident sound energy back to the remaining measured surfaces. Integrating measured sound intensities over measured surfaces renders estimated sound power of the source. The reverberation time T6o of the mockup interior had been modified to match reverberation levels of ISS US Lab interior for speech frequency bands, i.e., 0.5k, 1k, 2k, 4 kHz, by attaching appropriately sized Thinsulate sound absorption material to the interior wall of the mockup. Sound absorption of Thinsulate was modeled in three methods: Sabine equation with measured mockup interior reverberation time T60, layup model based on past impedance tube testing, and layup model plus air absorption correction. The evaluation/validation was carried out by acquiring octave band microphone data simultaneously at ten fixed locations throughout the mockup. SPLs (Sound Pressure Levels) predicted by our SEA model match well with measurements for our CM mockup, with a more complicated shape. Additionally in FY09, background NC noise (Noise Criterion) simulation and MRT (Modified Rhyme Test) were developed and performed in the mockup to determine the maximum noise level in CM habitable volume for fair crew voice communications. Numerous demonstrations of simulated noise environment in the mockup and associated SIL (Speech Interference Level) via MRT were performed for various communities, including members from NASA and Orion prime-/sub-contractors. Also, a new HSIR (Human-Systems Integration Requirement) for limiting pre- and post-landing SIL was proposed.

Environmental Sound Training in Cochlear Implant Users

PubMed Central

Sheft, Stanley; Kuvadia, Sejal; Gygi, Brian

2015-01-01

Purpose The study investigated the effect of a short computer-based environmental sound training regimen on the perception of environmental sounds and speech in experienced cochlear implant (CI) patients. Method Fourteen CI patients with the average of 5 years of CI experience participated. The protocol consisted of 2 pretests, 1 week apart, followed by 4 environmental sound training sessions conducted on separate days in 1 week, and concluded with 2 posttest sessions, separated by another week without training. Each testing session included an environmental sound test, which consisted of 40 familiar everyday sounds, each represented by 4 different tokens, as well as the Consonant Nucleus Consonant (CNC) word test, and Revised Speech Perception in Noise (SPIN-R) sentence test. Results Environmental sounds scores were lower than for either of the speech tests. Following training, there was a significant average improvement of 15.8 points in environmental sound perception, which persisted 1 week later after training was discontinued. No significant improvements were observed for either speech test. Conclusions The findings demonstrate that environmental sound perception, which remains problematic even for experienced CI patients, can be improved with a home-based computer training regimen. Such computer-based training may thus provide an effective low-cost approach to rehabilitation for CI users, and potentially, other hearing impaired populations. PMID:25633579

Perception of environmental sounds by experienced cochlear implant patients.

PubMed

Shafiro, Valeriy; Gygi, Brian; Cheng, Min-Yu; Vachhani, Jay; Mulvey, Megan

2011-01-01

Environmental sound perception serves an important ecological function by providing listeners with information about objects and events in their immediate environment. Environmental sounds such as car horns, baby cries, or chirping birds can alert listeners to imminent dangers as well as contribute to one's sense of awareness and well being. Perception of environmental sounds as acoustically and semantically complex stimuli may also involve some factors common to the processing of speech. However, very limited research has investigated the abilities of cochlear implant (CI) patients to identify common environmental sounds, despite patients' general enthusiasm about them. This project (1) investigated the ability of patients with modern-day CIs to perceive environmental sounds, (2) explored associations among speech, environmental sounds, and basic auditory abilities, and (3) examined acoustic factors that might be involved in environmental sound perception. Seventeen experienced postlingually deafened CI patients participated in the study. Environmental sound perception was assessed with a large-item test composed of 40 sound sources, each represented by four different tokens. The relationship between speech and environmental sound perception and the role of working memory and some basic auditory abilities were examined based on patient performance on a battery of speech tests (HINT, CNC, and individual consonant and vowel tests), tests of basic auditory abilities (audiometric thresholds, gap detection, temporal pattern, and temporal order for tones tests), and a backward digit recall test. The results indicated substantially reduced ability to identify common environmental sounds in CI patients (45.3%). Except for vowels, all speech test scores significantly correlated with the environmental sound test scores: r = 0.73 for HINT in quiet, r = 0.69 for HINT in noise, r = 0.70 for CNC, r = 0.64 for consonants, and r = 0.48 for vowels. HINT and CNC scores in quiet moderately correlated with the temporal order for tones. However, the correlation between speech and environmental sounds changed little after partialling out the variance due to other variables. Present findings indicate that environmental sound identification is difficult for CI patients. They further suggest that speech and environmental sounds may overlap considerably in their perceptual processing. Certain spectrotemproral processing abilities are separately associated with speech and environmental sound performance. However, they do not appear to mediate the relationship between speech and environmental sounds in CI patients. Environmental sound rehabilitation may be beneficial to some patients. Environmental sound testing may have potential diagnostic applications, especially with difficult-to-test populations and might also be predictive of speech performance for prelingually deafened patients with cochlear implants.

Underwater sound of rigid-hulled inflatable boats.

PubMed

Erbe, Christine; Liong, Syafrin; Koessler, Matthew Walter; Duncan, Alec J; Gourlay, Tim

2016-06-01

Underwater sound of rigid-hulled inflatable boats was recorded 142 times in total, over 3 sites: 2 in southern British Columbia, Canada, and 1 off Western Australia. Underwater sound peaked between 70 and 400 Hz, exhibiting strong tones in this frequency range related to engine and propeller rotation. Sound propagation models were applied to compute monopole source levels, with the source assumed 1 m below the sea surface. Broadband source levels (10-48 000 Hz) increased from 134 to 171 dB re 1 μPa @ 1 m with speed from 3 to 16 m/s (10-56 km/h). Source power spectral density percentile levels and 1/3 octave band levels are given for use in predictive modeling of underwater sound of these boats as part of environmental impact assessments.

Control of Toxic Chemicals in Puget Sound, Phase 3: Study Of Atmospheric Deposition of Air Toxics to the Surface of Puget Sound

DTIC Science & Technology

2007-01-01

deposition directly to Puget Sound was an important source of PAHs, polybrominated diphenyl ethers (PBDEs), and heavy metals . In most cases, atmospheric...versus Atmospheric Fluxes ........................................................................66  PAH Source Apportionment ...temperature inversions) on air quality during the wet season. A semi-quantitative apportionment study permitted a first-order characterization of source

Binaural Processing of Multiple Sound Sources

DTIC Science & Technology

2016-08-18

Sound Source Localization Identification, and Sound Source Localization When Listeners Move. The CI research was also supported by an NIH grant...8217Cochlear Implant Performance in Realistic Listening Environments,’ Dr. Michael Dorman, Principal Investigator, Dr. William Yost unpaid advisor. The other... Listeners Move. The CI research was also supported by an NIH grant (“Cochlear Implant Performance in Realistic Listening Environments,” Dr. Michael Dorman

New Families of Skewed Higher-Order Kernel Estimators to Solve the BSS/ICA Problem for Multimodal Sources Mixtures.

PubMed

Jabbar, Ahmed Najah

2018-04-13

This letter suggests two new types of asymmetrical higher-order kernels (HOK) that are generated using the orthogonal polynomials Laguerre (positive or right skew) and Bessel (negative or left skew). These skewed HOK are implemented in the blind source separation/independent component analysis (BSS/ICA) algorithm. The tests for these proposed HOK are accomplished using three scenarios to simulate a real environment using actual sound sources, an environment of mixtures of multimodal fast-changing probability density function (pdf) sources that represent a challenge to the symmetrical HOK, and an environment of an adverse case (near gaussian). The separation is performed by minimizing the mutual information (MI) among the mixed sources. The performance of the skewed kernels is compared to the performance of the standard kernels such as Epanechnikov, bisquare, trisquare, and gaussian and the performance of the symmetrical HOK generated using the polynomials Chebyshev1, Chebyshev2, Gegenbauer, Jacobi, and Legendre to the tenth order. The gaussian HOK are generated using the Hermite polynomial and the Wand and Schucany procedure. The comparison among the 96 kernels is based on the average intersymbol interference ratio (AISIR) and the time needed to complete the separation. In terms of AISIR, the skewed kernels' performance is better than that of the standard kernels and rivals most of the symmetrical kernels' performance. The importance of these new skewed HOK is manifested in the environment of the multimodal pdf mixtures. In such an environment, the skewed HOK come in first place compared with the symmetrical HOK. These new families can substitute for symmetrical HOKs in such applications.

Acoustic signatures of sound source-tract coupling.

PubMed

Arneodo, Ezequiel M; Perl, Yonatan Sanz; Mindlin, Gabriel B

2011-04-01

Birdsong is a complex behavior, which results from the interaction between a nervous system and a biomechanical peripheral device. While much has been learned about how complex sounds are generated in the vocal organ, little has been learned about the signature on the vocalizations of the nonlinear effects introduced by the acoustic interactions between a sound source and the vocal tract. The variety of morphologies among bird species makes birdsong a most suitable model to study phenomena associated to the production of complex vocalizations. Inspired by the sound production mechanisms of songbirds, in this work we study a mathematical model of a vocal organ, in which a simple sound source interacts with a tract, leading to a delay differential equation. We explore the system numerically, and by taking it to the weakly nonlinear limit, we are able to examine its periodic solutions analytically. By these means we are able to explore the dynamics of oscillatory solutions of a sound source-tract coupled system, which are qualitatively different from those of a sound source-filter model of a vocal organ. Nonlinear features of the solutions are proposed as the underlying mechanisms of observed phenomena in birdsong, such as unilaterally produced "frequency jumps," enhancement of resonances, and the shift of the fundamental frequency observed in heliox experiments. ©2011 American Physical Society

Acoustic signatures of sound source-tract coupling

PubMed Central

Arneodo, Ezequiel M.; Perl, Yonatan Sanz; Mindlin, Gabriel B.

2014-01-01

Birdsong is a complex behavior, which results from the interaction between a nervous system and a biomechanical peripheral device. While much has been learned about how complex sounds are generated in the vocal organ, little has been learned about the signature on the vocalizations of the nonlinear effects introduced by the acoustic interactions between a sound source and the vocal tract. The variety of morphologies among bird species makes birdsong a most suitable model to study phenomena associated to the production of complex vocalizations. Inspired by the sound production mechanisms of songbirds, in this work we study a mathematical model of a vocal organ, in which a simple sound source interacts with a tract, leading to a delay differential equation. We explore the system numerically, and by taking it to the weakly nonlinear limit, we are able to examine its periodic solutions analytically. By these means we are able to explore the dynamics of oscillatory solutions of a sound source-tract coupled system, which are qualitatively different from those of a sound source-filter model of a vocal organ. Nonlinear features of the solutions are proposed as the underlying mechanisms of observed phenomena in birdsong, such as unilaterally produced “frequency jumps,” enhancement of resonances, and the shift of the fundamental frequency observed in heliox experiments. PMID:21599213

Method for chemically analyzing a solution by acoustic means

DOEpatents

Beller, Laurence S.

1997-01-01

A method and apparatus for determining a type of solution and the concention of that solution by acoustic means. Generally stated, the method consists of: immersing a sound focusing transducer within a first liquid filled container; locating a separately contained specimen solution at a sound focal point within the first container; locating a sound probe adjacent to the specimen, generating a variable intensity sound signal from the transducer; measuring fundamental and multiple harmonic sound signal amplitudes; and then comparing a plot of a specimen sound response with a known solution sound response, thereby determining the solution type and concentration.

Intensity-invariant coding in the auditory system.

PubMed

Barbour, Dennis L

2011-11-01

The auditory system faithfully represents sufficient details from sound sources such that downstream cognitive processes are capable of acting upon this information effectively even in the face of signal uncertainty, degradation or interference. This robust sound source representation leads to an invariance in perception vital for animals to interact effectively with their environment. Due to unique nonlinearities in the cochlea, sound representations early in the auditory system exhibit a large amount of variability as a function of stimulus intensity. In other words, changes in stimulus intensity, such as for sound sources at differing distances, create a unique challenge for the auditory system to encode sounds invariantly across the intensity dimension. This challenge and some strategies available to sensory systems to eliminate intensity as an encoding variable are discussed, with a special emphasis upon sound encoding. Copyright © 2011 Elsevier Ltd. All rights reserved.

A new diagnostic method for separating airborne and structureborne noise radiated by plates with applications for propeller driven aircraft

NASA Technical Reports Server (NTRS)

Mcgary, Michael C.

1988-01-01

The anticipated application of advanced turboprop propulsion systems is expected to increase the interior noise of future aircraft to unacceptably high levels. The absence of technically and economically feasible noise source-path diagnostic tools has been a prime obstacle in the development of efficient noise control treatments for propeller-driven aircraft. A new diagnostic method that permits the separation and prediction of the fully coherent airborne and structureborne components of the sound radiated by plates or thin shells has been developed. Analytical and experimental studies of the proposed method were performed on an aluminum plate. The results of the study indicate that the proposed method could be used in flight, and has fewer encumbrances than the other diagnostic tools currently available.

Numerical Models for Sound Propagation in Long Spaces

NASA Astrophysics Data System (ADS)

Lai, Chenly Yuen Cheung

Both reverberation time and steady-state sound field are the key elements for assessing the acoustic condition in an enclosed space. They affect the noise propagation, speech intelligibility, clarity index, and definition. Since the sound field in a long space is non diffuse, classical room acoustics theory does not apply in this situation. The ray tracing technique and the image source methods are two common models to fathom both reverberation time and steady-state sound field in long enclosures nowadays. Although both models can give an accurate estimate of reverberation times and steady-state sound field directly or indirectly, they often involve time-consuming calculations. In order to simplify the acoustic consideration, a theoretical formulation has been developed for predicting both steady-state sound fields and reverberation times in street canyons. The prediction model is further developed to predict the steady-state sound field in a long enclosure. Apart from the straight long enclosure, there are other variations such as a cross junction, a long enclosure with a T-intersection, an U-turn long enclosure. In the present study, an theoretical and experimental investigations were conducted to develop formulae for predicting reverberation times and steady-state sound fields in a junction of a street canyon and in a long enclosure with T-intersection. The theoretical models are validated by comparing the numerical predictions with published experimental results. The theoretical results are also compared with precise indoor measurements and large-scale outdoor experimental results. In all of previous acoustical studies related to long enclosure, most of the studies are focused on the monopole sound source. Besides non-directional noise source, many noise sources in long enclosure are dipole like, such as train noise and fan noise. In order to study the characteristics of directional noise sources, a review of available dipole source was conducted. A dipole was constructed which was subsequent used for experimental studies. In additional, a theoretical model was developed for predicting dipole sound fields. The theoretical model can be used to study the effect of a dipole source on the speech intelligibility in long enclosures.

Sound source tracking device for telematic spatial sound field reproduction

NASA Astrophysics Data System (ADS)

Cardenas, Bruno

This research describes an algorithm that localizes sound sources for use in telematic applications. The localization algorithm is based on amplitude differences between various channels of a microphone array of directional shotgun microphones. The amplitude differences will be used to locate multiple performers and reproduce their voices, which were recorded at close distance with lavalier microphones, spatially corrected using a loudspeaker rendering system. In order to track multiple sound sources in parallel the information gained from the lavalier microphones will be utilized to estimate the signal-to-noise ratio between each performer and the concurrent performers.

A Corticothalamic Circuit Model for Sound Identification in Complex Scenes

PubMed Central

Otazu, Gonzalo H.; Leibold, Christian

2011-01-01

The identification of the sound sources present in the environment is essential for the survival of many animals. However, these sounds are not presented in isolation, as natural scenes consist of a superposition of sounds originating from multiple sources. The identification of a source under these circumstances is a complex computational problem that is readily solved by most animals. We present a model of the thalamocortical circuit that performs level-invariant recognition of auditory objects in complex auditory scenes. The circuit identifies the objects present from a large dictionary of possible elements and operates reliably for real sound signals with multiple concurrently active sources. The key model assumption is that the activities of some cortical neurons encode the difference between the observed signal and an internal estimate. Reanalysis of awake auditory cortex recordings revealed neurons with patterns of activity corresponding to such an error signal. PMID:21931668

Quantitative measurement of pass-by noise radiated by vehicles running at high speeds

NASA Astrophysics Data System (ADS)

Yang, Diange; Wang, Ziteng; Li, Bing; Luo, Yugong; Lian, Xiaomin

2011-03-01

It has been a challenge in the past to accurately locate and quantify the pass-by noise source radiated by the running vehicles. A system composed of a microphone array is developed in our current work to do this work. An acoustic-holography method for moving sound sources is designed to handle the Doppler effect effectively in the time domain. The effective sound pressure distribution is reconstructed on the surface of a running vehicle. The method has achieved a high calculation efficiency and is able to quantitatively measure the sound pressure at the sound source and identify the location of the main sound source. The method is also validated by the simulation experiments and the measurement tests with known moving speakers. Finally, the engine noise, tire noise, exhaust noise and wind noise of the vehicle running at different speeds are successfully identified by this method.

Auditory performance in an open sound field

NASA Astrophysics Data System (ADS)

Fluitt, Kim F.; Letowski, Tomasz; Mermagen, Timothy

2003-04-01

Detection and recognition of acoustic sources in an open field are important elements of situational awareness on the battlefield. They are affected by many technical and environmental conditions such as type of sound, distance to a sound source, terrain configuration, meteorological conditions, hearing capabilities of the listener, level of background noise, and the listener's familiarity with the sound source. A limited body of knowledge about auditory perception of sources located over long distances makes it difficult to develop models predicting auditory behavior on the battlefield. The purpose of the present study was to determine the listener's abilities to detect, recognize, localize, and estimate distances to sound sources from 25 to 800 m from the listing position. Data were also collected for meteorological conditions (wind direction and strength, temperature, atmospheric pressure, humidity) and background noise level for each experimental trial. Forty subjects (men and women, ages 18 to 25) participated in the study. Nine types of sounds were presented from six loudspeakers in random order; each series was presented four times. Partial results indicate that both detection and recognition declined at distances greater than approximately 200 m and distance estimation was grossly underestimated by listeners. Specific results will be presented.

Evolutionary trends in directional hearing

PubMed Central

Carr, Catherine E.; Christensen-Dalsgaard, Jakob

2016-01-01

Tympanic hearing is a true evolutionary novelty that arose in parallel within early tetrapods. We propose that in these tetrapods, selection for sound localization in air acted upon pre-existing directionally sensitive brainstem circuits, similar to those in fishes. Auditory circuits in birds and lizards resemble this ancestral, directionally sensitive framework. Despite this anatomically similarity, coding of sound source location differs between birds and lizards. In birds, brainstem circuits compute sound location from interaural cues. Lizards, however, have coupled ears, and do not need to compute source location in the brain. Thus their neural processing of sound direction differs, although all show mechanisms for enhancing sound source directionality. Comparisons with mammals reveal similarly complex interactions between coding strategies and evolutionary history. PMID:27448850

Enhancing Auditory Selective Attention Using a Visually Guided Hearing Aid.

PubMed

Kidd, Gerald

2017-10-17

Listeners with hearing loss, as well as many listeners with clinically normal hearing, often experience great difficulty segregating talkers in a multiple-talker sound field and selectively attending to the desired "target" talker while ignoring the speech from unwanted "masker" talkers and other sources of sound. This listening situation forms the classic "cocktail party problem" described by Cherry (1953) that has received a great deal of study over the past few decades. In this article, a new approach to improving sound source segregation and enhancing auditory selective attention is described. The conceptual design, current implementation, and results obtained to date are reviewed and discussed in this article. This approach, embodied in a prototype "visually guided hearing aid" (VGHA) currently used for research, employs acoustic beamforming steered by eye gaze as a means for improving the ability of listeners to segregate and attend to one sound source in the presence of competing sound sources. The results from several studies demonstrate that listeners with normal hearing are able to use an attention-based "spatial filter" operating primarily on binaural cues to selectively attend to one source among competing spatially distributed sources. Furthermore, listeners with sensorineural hearing loss generally are less able to use this spatial filter as effectively as are listeners with normal hearing especially in conditions high in "informational masking." The VGHA enhances auditory spatial attention for speech-on-speech masking and improves signal-to-noise ratio for conditions high in "energetic masking." Visual steering of the beamformer supports the coordinated actions of vision and audition in selective attention and facilitates following sound source transitions in complex listening situations. Both listeners with normal hearing and with sensorineural hearing loss may benefit from the acoustic beamforming implemented by the VGHA, especially for nearby sources in less reverberant sound fields. Moreover, guiding the beam using eye gaze can be an effective means of sound source enhancement for listening conditions where the target source changes frequently over time as often occurs during turn-taking in a conversation. http://cred.pubs.asha.org/article.aspx?articleid=2601621.

Enhancing Auditory Selective Attention Using a Visually Guided Hearing Aid

PubMed Central

2017-01-01

Purpose Listeners with hearing loss, as well as many listeners with clinically normal hearing, often experience great difficulty segregating talkers in a multiple-talker sound field and selectively attending to the desired “target” talker while ignoring the speech from unwanted “masker” talkers and other sources of sound. This listening situation forms the classic “cocktail party problem” described by Cherry (1953) that has received a great deal of study over the past few decades. In this article, a new approach to improving sound source segregation and enhancing auditory selective attention is described. The conceptual design, current implementation, and results obtained to date are reviewed and discussed in this article. Method This approach, embodied in a prototype “visually guided hearing aid” (VGHA) currently used for research, employs acoustic beamforming steered by eye gaze as a means for improving the ability of listeners to segregate and attend to one sound source in the presence of competing sound sources. Results The results from several studies demonstrate that listeners with normal hearing are able to use an attention-based “spatial filter” operating primarily on binaural cues to selectively attend to one source among competing spatially distributed sources. Furthermore, listeners with sensorineural hearing loss generally are less able to use this spatial filter as effectively as are listeners with normal hearing especially in conditions high in “informational masking.” The VGHA enhances auditory spatial attention for speech-on-speech masking and improves signal-to-noise ratio for conditions high in “energetic masking.” Visual steering of the beamformer supports the coordinated actions of vision and audition in selective attention and facilitates following sound source transitions in complex listening situations. Conclusions Both listeners with normal hearing and with sensorineural hearing loss may benefit from the acoustic beamforming implemented by the VGHA, especially for nearby sources in less reverberant sound fields. Moreover, guiding the beam using eye gaze can be an effective means of sound source enhancement for listening conditions where the target source changes frequently over time as often occurs during turn-taking in a conversation. Presentation Video http://cred.pubs.asha.org/article.aspx?articleid=2601621 PMID:29049603

Localization and separation of acoustic sources by using a 2.5-dimensional circular microphone array.

PubMed

Bai, Mingsian R; Lai, Chang-Sheng; Wu, Po-Chen

2017-07-01

Circular microphone arrays (CMAs) are sufficient in many immersive audio applications because azimuthal angles of sources are considered more important than the elevation angles in those occasions. However, the fact that CMAs do not resolve the elevation angle well can be a limitation for some applications which involves three-dimensional sound images. This paper proposes a 2.5-dimensional (2.5-D) CMA comprised of a CMA and a vertical logarithmic-spacing linear array (LLA) on the top. In the localization stage, two delay-and-sum beamformers are applied to the CMA and the LLA, respectively. The direction of arrival (DOA) is estimated from the product of two array output signals. In the separation stage, Tikhonov regularization and convex optimization are employed to extract the source amplitudes on the basis of the estimated DOA. The extracted signals from two arrays are further processed by the normalized least-mean-square algorithm with the internal iteration to yield the source signal with improved quality. To validate the 2.5-D CMA experimentally, a three-dimensionally printed circular array comprised of a 24-element CMA and an eight-element LLA is constructed. Objective perceptual evaluation of speech quality test and a subjective listening test are also undertaken.

Relation of sound intensity and accuracy of localization.

PubMed

Farrimond, T

1989-08-01

Tests were carried out on 17 subjects to determine the accuracy of monaural sound localization when the head is not free to turn toward the sound source. Maximum accuracy of localization for a constant-volume sound source coincided with the position for maximum perceived intensity of the sound in the front quadrant. There was a tendency for sounds to be perceived more often as coming from a position directly toward the ear. That is, for sounds in the front quadrant, errors of localization tended to be predominantly clockwise (i.e., biased toward a line directly facing the ear). Errors for sounds occurring in the rear quadrant tended to be anticlockwise. The pinna's differential effect on sound intensity between front and rear quadrants would assist in identifying the direction of movement of objects, for example an insect, passing the ear.

Source sparsity control of sound field reproduction using the elastic-net and the lasso minimizers.

PubMed

Gauthier, P-A; Lecomte, P; Berry, A

2017-04-01

Sound field reproduction is aimed at the reconstruction of a sound pressure field in an extended area using dense loudspeaker arrays. In some circumstances, sound field reproduction is targeted at the reproduction of a sound field captured using microphone arrays. Although methods and algorithms already exist to convert microphone array recordings to loudspeaker array signals, one remaining research question is how to control the spatial sparsity in the resulting loudspeaker array signals and what would be the resulting practical advantages. Sparsity is an interesting feature for spatial audio since it can drastically reduce the number of concurrently active reproduction sources and, therefore, increase the spatial contrast of the solution at the expense of a difference between the target and reproduced sound fields. In this paper, the application of the elastic-net cost function to sound field reproduction is compared to the lasso cost function. It is shown that the elastic-net can induce solution sparsity and overcomes limitations of the lasso: The elastic-net solves the non-uniqueness of the lasso solution, induces source clustering in the sparse solution, and provides a smoother solution within the activated source clusters.

An improved method for the calculation of Near-Field Acoustic Radiation Modes

NASA Astrophysics Data System (ADS)

Liu, Zu-Bin; Maury, Cédric

2016-02-01

Sensing and controlling Acoustic Radiation Modes (ARMs) in the near-field of vibrating structures is of great interest for broadband noise reduction or enhancement, as ARMs are velocity distributions defined over a vibrating surface, that independently and optimally contribute to the acoustic power in the acoustic field. But present methods only provide far-field ARMs (FFARMs) that are inadequate for the acoustic near-field problem. The Near-Field Acoustic Radiation Modes (NFARMs) are firstly studied with an improved numerical method, the Pressure-Velocity method, which rely on the eigen decomposition of the acoustic transfers between the vibrating source and a conformal observation surface, including sound pressure and velocity transfer matrices. The active and reactive parts of the sound power are separated and lead to the active and reactive ARMs. NFARMs are studied for a 2D baffled beam and for a 3D baffled plate, and so as differences between the NFARMS and the classical FFARMs. Comparisons of the NFARMs are analyzed when varying frequency and observation distance to the source. It is found that the efficiencies and shapes of the optimal active ARMs are independent on the distance while that of the reactive ones are distinctly related on.

Sound quality indicators for urban places in Paris cross-validated by Milan data.

PubMed

Ricciardi, Paola; Delaitre, Pauline; Lavandier, Catherine; Torchia, Francesca; Aumond, Pierre

2015-10-01

A specific smartphone application was developed to collect perceptive and acoustic data in Paris. About 3400 questionnaires were analyzed, regarding the global sound environment characterization, the perceived loudness of some emergent sources and the presence time ratio of sources that do not emerge from the background. Sound pressure level was recorded each second from the mobile phone's microphone during a 10-min period. The aim of this study is to propose indicators of urban sound quality based on linear regressions with perceptive variables. A cross validation of the quality models extracted from Paris data was carried out by conducting the same survey in Milan. The proposed sound quality general model is correlated with the real perceived sound quality (72%). Another model without visual amenity and familiarity is 58% correlated with perceived sound quality. In order to improve the sound quality indicator, a site classification was performed by Kohonen's Artificial Neural Network algorithm, and seven specific class models were developed. These specific models attribute more importance on source events and are slightly closer to the individual data than the global model. In general, the Parisian models underestimate the sound quality of Milan environments assessed by Italian people.

Loss and persistence of implicit memory for sound: evidence from auditory stream segregation context effects.

PubMed

Snyder, Joel S; Weintraub, David M

2013-07-01

An important question is the extent to which declines in memory over time are due to passive loss or active interference from other stimuli. The purpose of the present study was to determine the extent to which implicit memory effects in the perceptual organization of sound sequences are subject to loss and interference. Toward this aim, we took advantage of two recently discovered context effects in the perceptual judgments of sound patterns, one that depends on stimulus features of previous sounds and one that depends on the previous perceptual organization of these sounds. The experiments measured how listeners' perceptual organization of a tone sequence (test) was influenced by the frequency separation, or the perceptual organization, of the two preceding sequences (context1 and context2). The results demonstrated clear evidence for loss of context effects over time but little evidence for interference. However, they also revealed that context effects can be surprisingly persistent. The robust effects of loss, followed by persistence, were similar for the two types of context effects. We discuss whether the same auditory memories might contain information about basic stimulus features of sounds (i.e., frequency separation), as well as the perceptual organization of these sounds.

On the acoustic signature of tandem airfoils: The sound of an elastic airfoil in the wake of a vortex generator

DOE Office of Scientific and Technical Information (OSTI.GOV)

Manela, A.

The acoustic signature of an acoustically compact tandem airfoil setup in uniform high-Reynolds number flow is investigated. The upstream airfoil is considered rigid and is actuated at its leading edge with small-amplitude harmonic pitching motion. The downstream airfoil is taken passive and elastic, with its motion forced by the vortex-street excitation of the upstream airfoil. The non-linear near-field description is obtained via potential thin-airfoil theory. It is then applied as a source term into the Powell-Howe acoustic analogy to yield the far-field dipole radiation of the system. To assess the effect of downstream-airfoil elasticity, results are compared with counterpart calculationsmore » for a non-elastic setup, where the downstream airfoil is rigid and stationary. Depending on the separation distance between airfoils, airfoil-motion and airfoil-wake dynamics shift between in-phase (synchronized) and counter-phase behaviors. Consequently, downstream airfoil elasticity may act to amplify or suppress sound through the direct contribution of elastic-airfoil motion to the total signal. Resonance-type motion of the elastic airfoil is found when the upstream airfoil is actuated at the least stable eigenfrequency of the downstream structure. This, again, results in system sound amplification or suppression, depending on the separation distance between airfoils. With increasing actuation frequency, the acoustic signal becomes dominated by the direct contribution of the upstream airfoil motion, whereas the relative contribution of the elastic airfoil to the total signature turns negligible.« less

33 CFR 167.1321 - In Puget Sound and its approaches: Rosario Strait.

Code of Federal Regulations, 2012 CFR

2012-07-01

... 33 Navigation and Navigable Waters 2 2012-07-01 2012-07-01 false In Puget Sound and its approaches: Rosario Strait. 167.1321 Section 167.1321 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1321 In Puget Sound and its...

33 CFR 167.1321 - In Puget Sound and its approaches: Rosario Strait.

Code of Federal Regulations, 2013 CFR

2013-07-01

... 33 Navigation and Navigable Waters 2 2013-07-01 2013-07-01 false In Puget Sound and its approaches: Rosario Strait. 167.1321 Section 167.1321 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1321 In Puget Sound and its...

33 CFR 167.1321 - In Puget Sound and its approaches: Rosario Strait.

Code of Federal Regulations, 2014 CFR

2014-07-01

... 33 Navigation and Navigable Waters 2 2014-07-01 2014-07-01 false In Puget Sound and its approaches: Rosario Strait. 167.1321 Section 167.1321 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1321 In Puget Sound and its...

33 CFR 167.1321 - In Puget Sound and its approaches: Rosario Strait.

Code of Federal Regulations, 2011 CFR

2011-07-01

... 33 Navigation and Navigable Waters 2 2011-07-01 2011-07-01 false In Puget Sound and its approaches: Rosario Strait. 167.1321 Section 167.1321 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1321 In Puget Sound and its...

Development of an ICT-Based Air Column Resonance Learning Media

NASA Astrophysics Data System (ADS)

Purjiyanta, Eka; Handayani, Langlang; Marwoto, Putut

2016-08-01

Commonly, the sound source used in the air column resonance experiment is the tuning fork having disadvantage of unoptimal resonance results due to the sound produced which is getting weaker. In this study we made tones with varying frequency using the Audacity software which were, then, stored in a mobile phone as a source of sound. One advantage of this sound source is the stability of the resulting sound enabling it to produce the same powerful sound. The movement of water in a glass tube mounted on the tool resonance and the tone sound that comes out from the mobile phone were recorded by using a video camera. Sound resonances recorded were first, second, and third resonance, for each tone frequency mentioned. The resulting sound stays longer, so it can be used for the first, second, third and next resonance experiments. This study aimed to (1) explain how to create tones that can substitute tuning forks sound used in air column resonance experiments, (2) illustrate the sound wave that occurred in the first, second, and third resonance in the experiment, and (3) determine the speed of sound in the air. This study used an experimental method. It was concluded that; (1) substitute tones of a tuning fork sound can be made by using the Audacity software; (2) the form of sound waves that occured in the first, second, and third resonance in the air column resonance can be drawn based on the results of video recording of the air column resonance; and (3) based on the experiment result, the speed of sound in the air is 346.5 m/s, while based on the chart analysis with logger pro software, the speed of sound in the air is 343.9 ± 0.3171 m/s.

Method for chemically analyzing a solution by acoustic means

DOEpatents

Beller, L.S.

1997-04-22

A method and apparatus are disclosed for determining a type of solution and the concentration of that solution by acoustic means. Generally stated, the method consists of: immersing a sound focusing transducer within a first liquid filled container; locating a separately contained specimen solution at a sound focal point within the first container; locating a sound probe adjacent to the specimen, generating a variable intensity sound signal from the transducer; measuring fundamental and multiple harmonic sound signal amplitudes; and then comparing a plot of a specimen sound response with a known solution sound response, thereby determining the solution type and concentration. 10 figs.

75 FR 1540 - Treatment of Undeliverable Books and Sound Recordings

Federal Register 2010, 2011, 2012, 2013, 2014

2010-01-12

... and sound recordings that are undeliverable-as-addressed (UAA) in their original packaging. The... ``loose in the mail'' (contents separated from packaging and other address information), to the original...

Egocentric and allocentric representations in auditory cortex

PubMed Central

Brimijoin, W. Owen; Bizley, Jennifer K.

2017-01-01

A key function of the brain is to provide a stable representation of an object’s location in the world. In hearing, sound azimuth and elevation are encoded by neurons throughout the auditory system, and auditory cortex is necessary for sound localization. However, the coordinate frame in which neurons represent sound space remains undefined: classical spatial receptive fields in head-fixed subjects can be explained either by sensitivity to sound source location relative to the head (egocentric) or relative to the world (allocentric encoding). This coordinate frame ambiguity can be resolved by studying freely moving subjects; here we recorded spatial receptive fields in the auditory cortex of freely moving ferrets. We found that most spatially tuned neurons represented sound source location relative to the head across changes in head position and direction. In addition, we also recorded a small number of neurons in which sound location was represented in a world-centered coordinate frame. We used measurements of spatial tuning across changes in head position and direction to explore the influence of sound source distance and speed of head movement on auditory cortical activity and spatial tuning. Modulation depth of spatial tuning increased with distance for egocentric but not allocentric units, whereas, for both populations, modulation was stronger at faster movement speeds. Our findings suggest that early auditory cortex primarily represents sound source location relative to ourselves but that a minority of cells can represent sound location in the world independent of our own position. PMID:28617796

How the owl tracks its prey – II

PubMed Central

Takahashi, Terry T.

2010-01-01

Barn owls can capture prey in pitch darkness or by diving into snow, while homing in on the sounds made by their prey. First, the neural mechanisms by which the barn owl localizes a single sound source in an otherwise quiet environment will be explained. The ideas developed for the single source case will then be expanded to environments in which there are multiple sound sources and echoes – environments that are challenging for humans with impaired hearing. Recent controversies regarding the mechanisms of sound localization will be discussed. Finally, the case in which both visual and auditory information are available to the owl will be considered. PMID:20889819

Design of laser monitoring and sound localization system

NASA Astrophysics Data System (ADS)

Liu, Yu-long; Xu, Xi-ping; Dai, Yu-ming; Qiao, Yang

2013-08-01

In this paper, a novel design of laser monitoring and sound localization system is proposed. It utilizes laser to monitor and locate the position of the indoor conversation. In China most of the laser monitors no matter used in labor in an instrument uses photodiode or phototransistor as a detector at present. At the laser receivers of those facilities, light beams are adjusted to ensure that only part of the window in photodiodes or phototransistors received the beams. The reflection would deviate from its original path because of the vibration of the detected window, which would cause the changing of imaging spots in photodiode or phototransistor. However, such method is limited not only because it could bring in much stray light in receivers but also merely single output of photocurrent could be obtained. Therefore a new method based on quadrant detector is proposed. It utilizes the relation of the optical integral among quadrants to locate the position of imaging spots. This method could eliminate background disturbance and acquired two-dimensional spots vibrating data pacifically. The principle of this whole system could be described as follows. Collimated laser beams are reflected from vibrate-window caused by the vibration of sound source. Therefore reflected beams are modulated by vibration source. Such optical signals are collected by quadrant detectors and then are processed by photoelectric converters and corresponding circuits. Speech signals are eventually reconstructed. In addition, sound source localization is implemented by the means of detecting three different reflected light sources simultaneously. Indoor mathematical models based on the principle of Time Difference Of Arrival (TDOA) are established to calculate the twodimensional coordinate of sound source. Experiments showed that this system is able to monitor the indoor sound source beyond 15 meters with a high quality of speech reconstruction and to locate the sound source position accurately.

Measurement of sound speed vs. depth in South Pole ice for neutrino astronomy

NASA Astrophysics Data System (ADS)

Abbasi, R.; Abdou, Y.; Ackermann, M.; Adams, J.; Aguilar, J. A.; Ahlers, M.; Andeen, K.; Auffenberg, J.; Bai, X.; Baker, M.; Barwick, S. W.; Bay, R.; Bazo Alba, J. L.; Beattie, K.; Beatty, J. J.; Bechet, S.; Becker, J. K.; Becker, K.-H.; Benabderrahmane, M. L.; Berdermann, J.; Berghaus, P.; Berley, D.; Bernardini, E.; Bertrand, D.; Besson, D. Z.; Bissok, M.; Blaufuss, E.; Boersma, D. J.; Bohm, C.; Bolmont, J.; Böser, S.; Botner, O.; Bradley, L.; Braun, J.; Breder, D.; Castermans, T.; Chirkin, D.; Christy, B.; Clem, J.; Cohen, S.; Cowen, D. F.; D'Agostino, M. V.; Danninger, M.; Day, C. T.; De Clercq, C.; Demirörs, L.; Depaepe, O.; Descamps, F.; Desiati, P.; de Vries-Uiterweerd, G.; DeYoung, T.; Diaz-Velez, J. C.; Dreyer, J.; Dumm, J. P.; Duvoort, M. R.; Edwards, W. R.; Ehrlich, R.; Eisch, J.; Ellsworth, R. W.; Engdegård, O.; Euler, S.; Evenson, P. A.; Fadiran, O.; Fazely, A. R.; Feusels, T.; Filimonov, K.; Finley, C.; Foerster, M. M.; Fox, B. D.; Franckowiak, A.; Franke, R.; Gaisser, T. K.; Gallagher, J.; Ganugapati, R.; Gerhardt, L.; Gladstone, L.; Goldschmidt, A.; Goodman, J. A.; Gozzini, R.; Grant, D.; Griesel, T.; Groß, A.; Grullon, S.; Gunasingha, R. M.; Gurtner, M.; Ha, C.; Hallgren, A.; Halzen, F.; Han, K.; Hanson, K.; Hasegawa, Y.; Heise, J.; Helbing, K.; Herquet, P.; Hickford, S.; Hill, G. C.; Hoffman, K. D.; Hoshina, K.; Hubert, D.; Huelsnitz, W.; Hülß, J.-P.; Hulth, P. O.; Hultqvist, K.; Hussain, S.; Imlay, R. L.; Inaba, M.; Ishihara, A.; Jacobsen, J.; Japaridze, G. S.; Johansson, H.; Joseph, J. M.; Kampert, K.-H.; Kappes, A.; Karg, T.; Karle, A.; Kelley, J. L.; Kenny, P.; Kiryluk, J.; Kislat, F.; Klein, S. R.; Klepser, S.; Knops, S.; Kohnen, G.; Kolanoski, H.; Köpke, L.; Kowalski, M.; Kowarik, T.; Krasberg, M.; Kuehn, K.; Kuwabara, T.; Labare, M.; Lafebre, S.; Laihem, K.; Landsman, H.; Lauer, R.; Leich, H.; Lennarz, D.; Lucke, A.; Lundberg, J.; Lünemann, J.; Madsen, J.; Majumdar, P.; Maruyama, R.; Mase, K.; Matis, H. S.; McParland, C. P.; Meagher, K.; Merck, M.; Mészáros, P.; Middell, E.; Milke, N.; Miyamoto, H.; Mohr, A.; Montaruli, T.; Morse, R.; Movit, S. M.; Münich, K.; Nahnhauer, R.; Nam, J. W.; Nießen, P.; Nygren, D. R.; Odrowski, S.; Olivas, A.; Olivo, M.; Ono, M.; Panknin, S.; Patton, S.; Pérez de los Heros, C.; Petrovic, J.; Piegsa, A.; Pieloth, D.; Pohl, A. C.; Porrata, R.; Potthoff, N.; Price, P. B.; Prikockis, M.; Przybylski, G. T.; Rawlins, K.; Redl, P.; Resconi, E.; Rhode, W.; Ribordy, M.; Rizzo, A.; Rodrigues, J. P.; Roth, P.; Rothmaier, F.; Rott, C.; Roucelle, C.; Rutledge, D.; Ryckbosch, D.; Sander, H.-G.; Sarkar, S.; Satalecka, K.; Schlenstedt, S.; Schmidt, T.; Schneider, D.; Schukraft, A.; Schulz, O.; Schunck, M.; Seckel, D.; Semburg, B.; Seo, S. H.; Sestayo, Y.; Seunarine, S.; Silvestri, A.; Slipak, A.; Spiczak, G. M.; Spiering, C.; Stamatikos, M.; Stanev, T.; Stephens, G.; Stezelberger, T.; Stokstad, R. G.; Stoufer, M. C.; Stoyanov, S.; Strahler, E. A.; Straszheim, T.; Sulanke, K.-H.; Sullivan, G. W.; Swillens, Q.; Taboada, I.; Tarasova, O.; Tepe, A.; Ter-Antonyan, S.; Terranova, C.; Tilav, S.; Tluczykont, M.; Toale, P. A.; Tosi, D.; Turčan, D.; van Eijndhoven, N.; Vandenbroucke, J.; Van Overloop, A.; Vogt, C.; Voigt, B.; Walck, C.; Waldenmaier, T.; Walter, M.; Wendt, C.; Westerhoff, S.; Whitehorn, N.; Wiebusch, C. H.; Wiedemann, A.; Wikström, G.; Williams, D. R.; Wischnewski, R.; Wissing, H.; Woschnagg, K.; Xu, X. W.; Yodh, G.; Yoshida, S.; IceCube Collaboration

2010-06-01

We have measured the speed of both pressure waves and shear waves as a function of depth between 80 and 500 m depth in South Pole ice with better than 1% precision. The measurements were made using the South Pole Acoustic Test Setup (SPATS), an array of transmitters and sensors deployed in the ice at the South Pole in order to measure the acoustic properties relevant to acoustic detection of astrophysical neutrinos. The transmitters and sensors use piezoceramics operating at ˜5-25 kHz. Between 200 m and 500 m depth, the measured profile is consistent with zero variation of the sound speed with depth, resulting in zero refraction, for both pressure and shear waves. We also performed a complementary study featuring an explosive signal propagating vertically from 50 to 2250 m depth, from which we determined a value for the pressure wave speed consistent with that determined for shallower depths, higher frequencies, and horizontal propagation with the SPATS sensors. The sound speed profile presented here can be used to achieve good acoustic source position and emission time reconstruction in general, and neutrino direction and energy reconstruction in particular. The reconstructed quantities could also help separate neutrino signals from background.

Monaural Sound Localization Based on Reflective Structure and Homomorphic Deconvolution

PubMed Central

Park, Yeonseok; Choi, Anthony

2017-01-01

The asymmetric structure around the receiver provides a particular time delay for the specific incoming propagation. This paper designs a monaural sound localization system based on the reflective structure around the microphone. The reflective plates are placed to present the direction-wise time delay, which is naturally processed by convolutional operation with a sound source. The received signal is separated for estimating the dominant time delay by using homomorphic deconvolution, which utilizes the real cepstrum and inverse cepstrum sequentially to derive the propagation response’s autocorrelation. Once the localization system accurately estimates the information, the time delay model computes the corresponding reflection for localization. Because of the structure limitation, two stages of the localization process perform the estimation procedure as range and angle. The software toolchain from propagation physics and algorithm simulation realizes the optimal 3D-printed structure. The acoustic experiments in the anechoic chamber denote that 79.0% of the study range data from the isotropic signal is properly detected by the response value, and 87.5% of the specific direction data from the study range signal is properly estimated by the response time. The product of both rates shows the overall hit rate to be 69.1%. PMID:28946625

Assessment of Hydroacoustic Propagation Using Autonomous Hydrophones in the Scotia Sea

DTIC Science & Technology

2010-09-01

Award No. DE-AI52-08NA28654 Proposal No. BAA08-36 ABSTRACT The remote area of the Atlantic Ocean near the Antarctic Peninsula and the South...hydroacoustic blind spot. To investigate the sound propagation and interferences affected by these landmasses in the vicinity of the Antarctic polar...from large icebergs (near-surface sources) were utilized as natural sound sources. Surface sound sources, e.g., ice-related events, tend to suffer less

Active control of noise on the source side of a partition to increase its sound isolation

NASA Astrophysics Data System (ADS)

Tarabini, Marco; Roure, Alain; Pinhede, Cedric

2009-03-01

This paper describes a local active noise control system that virtually increases the sound isolation of a dividing wall by means of a secondary source array. With the proposed method, sound pressure on the source side of the partition is reduced using an array of loudspeakers that generates destructive interference on the wall surface, where an array of error microphones is placed. The reduction of sound pressure on the incident side of the wall is expected to decrease the sound radiated into the contiguous room. The method efficiency was experimentally verified by checking the insertion loss of the active noise control system; in order to investigate the possibility of using a large number of actuators, a decentralized FXLMS control algorithm was used. Active control performances and stability were tested with different array configurations, loudspeaker directivities and enclosure characteristics (sound source position and absorption coefficient). The influence of all these parameters was investigated with the factorial design of experiments. The main outcome of the experimental campaign was that the insertion loss produced by the secondary source array, in the 50-300 Hz frequency range, was close to 10 dB. In addition, the analysis of variance showed that the active noise control performance can be optimized with a proper choice of the directional characteristics of the secondary source and the distance between loudspeakers and error microphones.

The low-frequency sound power measuring technique for an underwater source in a non-anechoic tank

NASA Astrophysics Data System (ADS)

Zhang, Yi-Ming; Tang, Rui; Li, Qi; Shang, Da-Jing

2018-03-01

In order to determine the radiated sound power of an underwater source below the Schroeder cut-off frequency in a non-anechoic tank, a low-frequency extension measuring technique is proposed. This technique is based on a unique relationship between the transmission characteristics of the enclosed field and those of the free field, which can be obtained as a correction term based on previous measurements of a known simple source. The radiated sound power of an unknown underwater source in the free field can thereby be obtained accurately from measurements in a non-anechoic tank. To verify the validity of the proposed technique, a mathematical model of the enclosed field is established using normal-mode theory, and the relationship between the transmission characteristics of the enclosed and free fields is obtained. The radiated sound power of an underwater transducer source is tested in a glass tank using the proposed low-frequency extension measuring technique. Compared with the free field, the radiated sound power level of the narrowband spectrum deviation is found to be less than 3 dB, and the 1/3 octave spectrum deviation is found to be less than 1 dB. The proposed testing technique can be used not only to extend the low-frequency applications of non-anechoic tanks, but also for measurement of radiated sound power from complicated sources in non-anechoic tanks.

Modeling and analysis of secondary sources coupling for active sound field reduction in confined spaces

NASA Astrophysics Data System (ADS)

Montazeri, Allahyar; Taylor, C. James

2017-10-01

This article addresses the coupling of acoustic secondary sources in a confined space in a sound field reduction framework. By considering the coupling of sources in a rectangular enclosure, the set of coupled equations governing its acoustical behavior are solved. The model obtained in this way is used to analyze the behavior of multi-input multi-output (MIMO) active sound field control (ASC) systems, where the coupling of sources cannot be neglected. In particular, the article develops the analytical results to analyze the effect of coupling of an array of secondary sources on the sound pressure levels inside an enclosure, when an array of microphones is used to capture the acoustic characteristics of the enclosure. The results are supported by extensive numerical simulations showing how coupling of loudspeakers through acoustic modes of the enclosure will change the strength and hence the driving voltage signal applied to the secondary loudspeakers. The practical significance of this model is to provide a better insight on the performance of the sound reproduction/reduction systems in confined spaces when an array of loudspeakers and microphones are placed in a fraction of wavelength of the excitation signal to reduce/reproduce the sound field. This is of particular importance because the interaction of different sources affects their radiation impedance depending on the electromechanical properties of the loudspeakers.

Consistent modelling of wind turbine noise propagation from source to receiver.

PubMed

Barlas, Emre; Zhu, Wei Jun; Shen, Wen Zhong; Dag, Kaya O; Moriarty, Patrick

2017-11-01

The unsteady nature of wind turbine noise is a major reason for annoyance. The variation of far-field sound pressure levels is not only caused by the continuous change in wind turbine noise source levels but also by the unsteady flow field and the ground characteristics between the turbine and receiver. To take these phenomena into account, a consistent numerical technique that models the sound propagation from the source to receiver is developed. Large eddy simulation with an actuator line technique is employed for the flow modelling and the corresponding flow fields are used to simulate sound generation and propagation. The local blade relative velocity, angle of attack, and turbulence characteristics are input to the sound generation model. Time-dependent blade locations and the velocity between the noise source and receiver are considered within a quasi-3D propagation model. Long-range noise propagation of a 5 MW wind turbine is investigated. Sound pressure level time series evaluated at the source time are studied for varying wind speeds, surface roughness, and ground impedances within a 2000 m radius from the turbine.

Consistent modelling of wind turbine noise propagation from source to receiver

DOE PAGES

Barlas, Emre; Zhu, Wei Jun; Shen, Wen Zhong; ...

2017-11-28

The unsteady nature of wind turbine noise is a major reason for annoyance. The variation of far-field sound pressure levels is not only caused by the continuous change in wind turbine noise source levels but also by the unsteady flow field and the ground characteristics between the turbine and receiver. To take these phenomena into account, a consistent numerical technique that models the sound propagation from the source to receiver is developed. Large eddy simulation with an actuator line technique is employed for the flow modelling and the corresponding flow fields are used to simulate sound generation and propagation. Themore » local blade relative velocity, angle of attack, and turbulence characteristics are input to the sound generation model. Time-dependent blade locations and the velocity between the noise source and receiver are considered within a quasi-3D propagation model. Long-range noise propagation of a 5 MW wind turbine is investigated. Sound pressure level time series evaluated at the source time are studied for varying wind speeds, surface roughness, and ground impedances within a 2000 m radius from the turbine.« less

Consistent modelling of wind turbine noise propagation from source to receiver

DOE Office of Scientific and Technical Information (OSTI.GOV)

Barlas, Emre; Zhu, Wei Jun; Shen, Wen Zhong

The unsteady nature of wind turbine noise is a major reason for annoyance. The variation of far-field sound pressure levels is not only caused by the continuous change in wind turbine noise source levels but also by the unsteady flow field and the ground characteristics between the turbine and receiver. To take these phenomena into account, a consistent numerical technique that models the sound propagation from the source to receiver is developed. Large eddy simulation with an actuator line technique is employed for the flow modelling and the corresponding flow fields are used to simulate sound generation and propagation. Themore » local blade relative velocity, angle of attack, and turbulence characteristics are input to the sound generation model. Time-dependent blade locations and the velocity between the noise source and receiver are considered within a quasi-3D propagation model. Long-range noise propagation of a 5 MW wind turbine is investigated. Sound pressure level time series evaluated at the source time are studied for varying wind speeds, surface roughness, and ground impedances within a 2000 m radius from the turbine.« less

Non-Contact Ultrasonic Imaging

DTIC Science & Technology

2016-10-31

difficult to measure because of the amount of sound at the difference frequency still produced in the air. Nonlinear Reflection off of a Curved Surface...separate sound generated in air from sound generated in liquid. Two incoming rays incident upon a curved surface may reflect collinearly. At a different... sound reflecting off of the air-water interface from the air, the energy density of the incident and reflected waves are around 1000x that of the

Binaural room simulation

NASA Technical Reports Server (NTRS)

Lehnert, H.; Blauert, Jens; Pompetzki, W.

1991-01-01

In every-day listening the auditory event perceived by a listener is determined not only by the sound signal that a sound emits but also by a variety of environmental parameters. These parameters are the position, orientation and directional characteristics of the sound source, the listener's position and orientation, the geometrical and acoustical properties of surfaces which affect the sound field and the sound propagation properties of the surrounding fluid. A complete set of these parameters can be called an Acoustic Environment. If the auditory event perceived by a listener is manipulated in such a way that the listener is shifted acoustically into a different acoustic environment without moving himself physically, a Virtual Acoustic Environment has been created. Here, we deal with a special technique to set up nearly arbitrary Virtual Acoustic Environments, the Binaural Room Simulation. The purpose of the Binaural Room Simulation is to compute the binaural impulse response related to a virtual acoustic environment taking into account all parameters mentioned above. One possible way to describe a Virtual Acoustic Environment is the concept of the virtual sound sources. Each of the virtual sources emits a certain signal which is correlated but not necessarily identical with the signal emitted by the direct sound source. If source and receiver are non moving, the acoustic environment becomes a linear time-invariant system. Then, the Binaural Impulse Response from the source to a listener' s eardrums contains all relevant auditory information related to the Virtual Acoustic Environment. Listening into the simulated environment can easily be achieved by convolving the Binaural Impulse Response with dry signals and representing the results via headphones.

Broad band sound from wind turbine generators

NASA Technical Reports Server (NTRS)

Hubbard, H. H.; Shepherd, K. P.; Grosveld, F. W.

1981-01-01

Brief descriptions are given of the various types of large wind turbines and their sound characteristics. Candidate sources of broadband sound are identified and are rank ordered for a large upwind configuration wind turbine generator for which data are available. The rotor is noted to be the main source of broadband sound which arises from inflow turbulence and from the interactions of the turbulent boundary layer on the blade with its trailing edge. Sound is radiated about equally in all directions but the refraction effects of the wind produce an elongated contour pattern in the downwind direction.

Effects of sound source directivity on auralizations

NASA Astrophysics Data System (ADS)

Sheets, Nathan W.; Wang, Lily M.

2002-05-01

Auralization, the process of rendering audible the sound field in a simulated space, is a useful tool in the design of acoustically sensitive spaces. The auralization depends on the calculation of an impulse response between a source and a receiver which have certain directional behavior. Many auralizations created to date have used omnidirectional sources; the effects of source directivity on auralizations is a relatively unexplored area. To examine if and how the directivity of a sound source affects the acoustical results obtained from a room, we used directivity data for three sources in a room acoustic modeling program called Odeon. The three sources are: violin, piano, and human voice. The results from using directional data are compared to those obtained using omnidirectional source behavior, both through objective measure calculations and subjective listening tests.

Development of a directivity-controlled piezoelectric transducer for sound reproduction

NASA Astrophysics Data System (ADS)

Bédard, Magella; Berry, Alain

2008-04-01

Present sound reproduction systems do not attempt to simulate the spatial radiation of musical instruments, or sound sources in general, even though the spatial directivity has a strong impact on the psychoacoustic experience. A transducer consisting of 4 piezoelectric elemental sources made from curved PVDF films is used to generate a target directivity pattern in the horizontal plane, in the frequency range of 5-20 kHz. The vibratory and acoustical response of an elemental source is addressed, both theoretically and experimentally. Two approaches to synthesize the input signals to apply to each elemental source are developed in order to create a prescribed, frequency-dependent acoustic directivity. The circumferential Fourier decomposition of the target directivity provides a compromise between the magnitude and the phase reconstruction, whereas the minimization of a quadratic error criterion provides a best magnitude reconstruction. This transducer can improve sound reproduction by introducing the spatial radiation aspect of the original source at high frequency.

Callback response of dugongs to conspecific chirp playbacks.

PubMed

Ichikawa, Kotaro; Akamatsu, Tomonari; Shinke, Tomio; Adulyanukosol, Kanjana; Arai, Nobuaki

2011-06-01

Dugongs (Dugong dugon) produce bird-like calls such as chirps and trills. The vocal responses of dugongs to playbacks of several acoustic stimuli were investigated. Animals were exposed to four different playback stimuli: a recorded chirp from a wild dugong, a synthesized down-sweep sound, a synthesized constant-frequency sound, and silence. Wild dugongs vocalized more frequently after playback of broadcast chirps than that after constant-frequency sounds or silence. The down-sweep sound also elicited more vocal responses than did silence. No significant difference was found between the broadcast chirps and the down-sweep sound. The ratio of wild dugong chirps to all calls and the dominant frequencies of the wild dugong calls were significantly higher during playbacks of broadcast chirps, down-sweep sounds, and constant-frequency sounds than during those of silence. The source level and duration of dugong chirps increased significantly as signaling distance increased. No significant correlation was found between signaling distance and the source level of trills. These results show that dugongs vocalize to playbacks of frequency-modulated signals and suggest that the source level of dugong chirps may be manipulated to compensate for transmission loss between the source and receiver. This study provides the first behavioral observations revealing the function of dugong chirps. © 2011 Acoustical Society of America

Sound Recordings and the Library. Occasional Papers Number 179.

ERIC Educational Resources Information Center

Almquist, Sharon G.

The basic concept that sound waves could be traced or recorded on a solid object was developed separately by Leon Scott, Charles Cros, and Thomas Alva Edison between 1857 and 1877 and, by 1890, the foundation of the present-day commercial record industry was established. Although cylinders were the first sound recordings to be sold commercially,…

33 CFR 100.121 - Swim Across the Sound, Long Island Sound, Port Jefferson, NY to Captain's Cove Seaport...

Code of Federal Regulations, 2010 CFR

2010-07-01

... Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF HOMELAND SECURITY REGATTAS AND MARINE PARADES... 33 Navigation and Navigable Waters 1 2010-07-01 2010-07-01 false Swim Across the Sound, Long... the Federal Register, separate marine broadcasts and local notice to mariners. [USCG-2009-0395, 75 FR...

Atmospheric Propagation

NASA Technical Reports Server (NTRS)

Embleton, Tony F. W.; Daigle, Gilles A.

1991-01-01

Reviewed here is the current state of knowledge with respect to each basic mechanism of sound propagation in the atmosphere and how each mechanism changes the spectral or temporal characteristics of the sound received at a distance from the source. Some of the basic processes affecting sound wave propagation which are present in any situation are discussed. They are geometrical spreading, molecular absorption, and turbulent scattering. In geometrical spreading, sound levels decrease with increasing distance from the source; there is no frequency dependence. In molecular absorption, sound energy is converted into heat as the sound wave propagates through the air; there is a strong dependence on frequency. In turbulent scattering, local variations in wind velocity and temperature induce fluctuations in phase and amplitude of the sound waves as they propagate through an inhomogeneous medium; there is a moderate dependence on frequency.

Schema vs. primitive perceptual grouping: the relative weighting of sequential vs. spatial cues during an auditory grouping task in frogs.

PubMed

Farris, Hamilton E; Ryan, Michael J

2017-03-01

Perceptually, grouping sounds based on their sources is critical for communication. This is especially true in túngara frog breeding aggregations, where multiple males produce overlapping calls that consist of an FM 'whine' followed by harmonic bursts called 'chucks'. Phonotactic females use at least two cues to group whines and chucks: whine-chuck spatial separation and sequence. Spatial separation is a primitive cue, whereas sequence is schema-based, as chuck production is morphologically constrained to follow whines, meaning that males cannot produce the components simultaneously. When one cue is available, females perceptually group whines and chucks using relative comparisons: components with the smallest spatial separation or those closest to the natural sequence are more likely grouped. By simultaneously varying the temporal sequence and spatial separation of a single whine and two chucks, this study measured between-cue perceptual weighting during a specific grouping task. Results show that whine-chuck spatial separation is a stronger grouping cue than temporal sequence, as grouping is more likely for stimuli with smaller spatial separation and non-natural sequence than those with larger spatial separation and natural sequence. Compared to the schema-based whine-chuck sequence, we propose that spatial cues have less variance, potentially explaining their preferred use when grouping during directional behavioral responses.

How effectively do horizontal and vertical response strategies of long-finned pilot whales reduce sound exposure from naval sonar?

PubMed

Wensveen, Paul J; von Benda-Beckmann, Alexander M; Ainslie, Michael A; Lam, Frans-Peter A; Kvadsheim, Petter H; Tyack, Peter L; Miller, Patrick J O

2015-05-01

The behaviour of a marine mammal near a noise source can modulate the sound exposure it receives. We demonstrate that two long-finned pilot whales both surfaced in synchrony with consecutive arrivals of multiple sonar pulses. We then assess the effect of surfacing and other behavioural response strategies on the received cumulative sound exposure levels and maximum sound pressure levels (SPLs) by modelling realistic spatiotemporal interactions of a pilot whale with an approaching source. Under the propagation conditions of our model, some response strategies observed in the wild were effective in reducing received levels (e.g. movement perpendicular to the source's line of approach), but others were not (e.g. switching from deep to shallow diving; synchronous surfacing after maximum SPLs). Our study exemplifies how simulations of source-whale interactions guided by detailed observational data can improve our understanding about motivations behind behaviour responses observed in the wild (e.g., reducing sound exposure, prey movement). Copyright © 2015 Elsevier Ltd. All rights reserved.

Difference in precedence effect between children and adults signifies development of sound localization abilities in complex listening tasks

PubMed Central

Litovsky, Ruth Y.; Godar, Shelly P.

2010-01-01

The precedence effect refers to the fact that humans are able to localize sound in reverberant environments, because the auditory system assigns greater weight to the direct sound (lead) than the later-arriving sound (lag). In this study, absolute sound localization was studied for single source stimuli and for dual source lead-lag stimuli in 4–5 year old children and adults. Lead-lag delays ranged from 5–100 ms. Testing was conducted in free field, with pink noise bursts emitted from loudspeakers positioned on a horizontal arc in the frontal field. Listeners indicated how many sounds were heard and the perceived location of the first- and second-heard sounds. Results suggest that at short delays (up to 10 ms), the lead dominates sound localization strongly at both ages, and localization errors are similar to those with single-source stimuli. At longer delays errors can be large, stemming from over-integration of the lead and lag, interchanging of perceived locations of the first-heard and second-heard sounds due to temporal order confusion, and dominance of the lead over the lag. The errors are greater for children than adults. Results are discussed in the context of maturation of auditory and non-auditory factors. PMID:20968369

Simulated seal scarer sounds scare porpoises, but not seals: species-specific responses to 12 kHz deterrence sounds

PubMed Central

Hermannsen, Line; Beedholm, Kristian

2017-01-01

Acoustic harassment devices (AHD) or ‘seal scarers’ are used extensively, not only to deter seals from fisheries, but also as mitigation tools to deter marine mammals from potentially harmful sound sources, such as offshore pile driving. To test the effectiveness of AHDs, we conducted two studies with similar experimental set-ups on two key species: harbour porpoises and harbour seals. We exposed animals to 500 ms tone bursts at 12 kHz simulating that of an AHD (Lofitech), but with reduced output levels (source peak-to-peak level of 165 dB re 1 µPa). Animals were localized with a theodolite before, during and after sound exposures. In total, 12 sound exposures were conducted to porpoises and 13 exposures to seals. Porpoises were found to exhibit avoidance reactions out to ranges of 525 m from the sound source. Contrary to this, seal observations increased during sound exposure within 100 m of the loudspeaker. We thereby demonstrate that porpoises and seals respond very differently to AHD sounds. This has important implications for application of AHDs in multi-species habitats, as sound levels required to deter less sensitive species (seals) can lead to excessive and unwanted large deterrence ranges on more sensitive species (porpoises). PMID:28791155

Characterizing large river sounds: Providing context for understanding the environmental effects of noise produced by hydrokinetic turbines

DOE Office of Scientific and Technical Information (OSTI.GOV)

Bevelhimer, Mark S.; Deng, Z. Daniel; Scherelis, Constantin

2016-01-01

Underwater noise associated with the installation and operation of hydrokinetic turbines in rivers and tidal zones presents a potential environmental concern for fish and marine mammals. Comparing the spectral quality of sounds emitted by hydrokinetic turbines to natural and other anthropogenic sound sources is an initial step at understanding potential environmental impacts. Underwater recordings were obtained from passing vessels of different sizes and other underwater sound sources in both static and flowing waters. Static water measurements were taken in a lake with minimal background noise. Flowing water measurements were taken at a previously proposed deployment site for hydrokinetic turbines onmore » the Mississippi River, where the sound of flowing water is included in background measurements. The size of vessels measured ranged from a small fishing boat with a 60 HP outboard motor to an 18-unit barge train being pushed upstream by tugboat. As expected, large vessels with large engines created the highest sound levels, and when compared to the sound created by an operating HK turbine were many times greater. A comparison of sound levels from the same sources at different distances using both spherical and cylindrical sound attenuation functions suggests that spherical model results more closely approximate observed values.« less

Feasibility of making sound power measurements in the NASA Langley V/STOL tunnel test section

NASA Technical Reports Server (NTRS)

Brooks, T. F.; Scheiman, J.; Silcox, R. J.

1976-01-01

Based on exploratory acoustic measurements in Langley's V/STOL wind tunnel, recommendations are made on the methodology for making sound power measurements of aircraft components in the closed tunnel test section. During airflow, tunnel self-noise and microphone flow-induced noise place restrictions on the amplitude and spectrum of the sound source to be measured. Models of aircraft components with high sound level sources, such as thrust engines and powered lift systems, seem likely candidates for acoustic testing.

Varying sediment sources (Hudson Strait, Cumberland Sound, Baffin Bay) to the NW Labrador Sea slope between and during Heinrich events 0 to 4

USGS Publications Warehouse

Andrews, John T.; Barber, D.C.; Jennings, A.E.; Eberl, D.D.; Maclean, B.; Kirby, M.E.; Stoner, J.S.

2012-01-01

Core HU97048-007PC was recovered from the continental Labrador Sea slope at a water depth of 945 m, 250 km seaward from the mouth of Cumberland Sound, and 400 km north of Hudson Strait. Cumberland Sound is a structural trough partly floored by Cretaceous mudstones and Paleozoic carbonates. The record extends from ∼10 to 58 ka. On-board logging revealed a complex series of lithofacies, including buff-colored detrital carbonate-rich sediments [Heinrich (H)-events] frequently bracketed by black facies. We investigate the provenance of these facies using quantitative X-ray diffraction on drill-core samples from Paleozoic and Cretaceous bedrock from the SE Baffin Island Shelf, and on the < 2-mm sediment fraction in a transect of five cores from Cumberland Sound to the NW Labrador Sea. A sediment unmixing program was used to discriminate between sediment sources, which included dolomite-rich sediments from Baffin Bay, calcite-rich sediments from Hudson Strait and discrete sources from Cumberland Sound. Results indicated that the bulk of the sediment was derived from Cumberland Sound, but Baffin Bay contributed to sediments coeval with H-0 (Younger Dryas), whereas Hudson Strait was the source during H-events 1–4. Contributions from the Cretaceous outcrops within Cumberland Sound bracket H-events, thus both leading and lagging Hudson Strait-sourced H-events.

Peripheral mechanisms for vocal production in birds - differences and similarities to human speech and singing.

PubMed

Riede, Tobias; Goller, Franz

2010-10-01

Song production in songbirds is a model system for studying learned vocal behavior. As in humans, bird phonation involves three main motor systems (respiration, vocal organ and vocal tract). The avian respiratory mechanism uses pressure regulation in air sacs to ventilate a rigid lung. In songbirds sound is generated with two independently controlled sound sources, which reside in a uniquely avian vocal organ, the syrinx. However, the physical sound generation mechanism in the syrinx shows strong analogies to that in the human larynx, such that both can be characterized as myoelastic-aerodynamic sound sources. Similarities include active adduction and abduction, oscillating tissue masses which modulate flow rate through the organ and a layered structure of the oscillating tissue masses giving rise to complex viscoelastic properties. Differences in the functional morphology of the sound producing system between birds and humans require specific motor control patterns. The songbird vocal apparatus is adapted for high speed, suggesting that temporal patterns and fast modulation of sound features are important in acoustic communication. Rapid respiratory patterns determine the coarse temporal structure of song and maintain gas exchange even during very long songs. The respiratory system also contributes to the fine control of airflow. Muscular control of the vocal organ regulates airflow and acoustic features. The upper vocal tract of birds filters the sounds generated in the syrinx, and filter properties are actively adjusted. Nonlinear source-filter interactions may also play a role. The unique morphology and biomechanical system for sound production in birds presents an interesting model for exploring parallels in control mechanisms that give rise to highly convergent physical patterns of sound generation. More comparative work should provide a rich source for our understanding of the evolution of complex sound producing systems. Copyright © 2009 Elsevier Inc. All rights reserved.

Temporal-Spectral Characterization and Classification of Marine Mammal Vocalizations and Diesel-Electric Ships Radiated Sound over Continental Shelf Scale Regions with Coherent Hydrophone Array Measurements

NASA Astrophysics Data System (ADS)

Huang, Wei

The passive ocean acoustic waveguide remote sensing (POAWRS) technology is capable of monitoring a large variety of underwater sound sources over instantaneous wide areas spanning continental-shelf scale regions. POAWRS uses a large-aperture densely-sampled coherent hydrophone array to significantly enhance the signal-to-noise ratio via beamforming, enabling detection of sound sources roughly two-orders of magnitude more distant in range than that possible with a single hydrophone. The sound sources detected by POAWRS include ocean biology, geophysical processes, and man-made activities. POAWRS provides detection, bearing-time estimation, localization, and classification of underwater sound sources. The volume of underwater sounds detected by POAWRS is immense, typically exceeding a million unique signal detections per day, in the 10-4000 Hz frequency range, making it a tremendously challenging task to distinguish and categorize the various sound sources present in a given region. Here we develop various approaches for characterizing and clustering the signal detections for various subsets of data acquired using the POAWRS technology. The approaches include pitch tracking of the dominant signal detections, time-frequency feature extraction, clustering and categorization methods. These approaches are essential for automatic processing and enhancing the efficiency and accuracy of POAWRS data analysis. The results of the signal detection, clustering and classification analysis are required for further POAWRS processing, including localization and tracking of a large number of oceanic sound sources. Here the POAWRS detection, localization and clustering approaches are applied to analyze and elucidate the vocalization behavior of humpback, sperm and fin whales in the New England continental shelf and slope, including the Gulf of Maine from data acquired using coherent hydrophone arrays. The POAWRS technology can also be applied for monitoring ocean vehicles. Here the approach is calibrated by application to known ships present in the Gulf of Maine and in the Norwegian Sea from their underwater sounds received using a coherent hydrophone array. The vocalization behavior of humpback whales was monitored over vast areas of the Gulf of Maine using the POAWRS technique over multiple diel cycles in Fall 2006. The humpback vocalizations, received at a rate of roughly 1800+/-1100 calls per day, comprised of both song and non-song. The song vocalizations, composed of highly structured and repeatable set of phrases, are characterized by inter-pulse intervals of 3.5 +/- 1.8 s. Songs were detected throughout the diel cycle, occuring roughly 40% during the day and 60% during the night. The humpback non-song vocalizations, dominated by shorter duration (≤3 s) downsweep and bow-shaped moans, as well as a small fraction of longer duration (˜5 s) cries, have significantly larger mean and more variable inter-pulse intervals of 14.2 +/- 11 s. The non-song vocalizations were detected at night with negligible detections during the day, implying they probably function as nighttime communication signals. The humpback song and non-song vocalizations are separately localized using the moving array triangulation and array invariant techniques. The humpback song and non-song moan calls are both consistently localized to a dense area on northeastern Georges Bank and a less dense region extended from Franklin Basin to the Great South Channel. Humpback cries occur exclusively on northeastern Georges Bank and during nights with coincident dense Atlantic herring shoaling populations, implying the cries are feeding-related. Sperm whales in the New England continental shelf and slope were passively localized and classified from their vocalizations received using a single low-frequency (<2500 Hz) densely-sampled horizontal coherent hydrophone array deployed in Spring 2013 in Gulf of Maine. Whale bearings were estimated using time-domain beamforming that provided high coherent array gain in sperm whale click signal-to-noise ratio. Whale ranges from the receiver array center were estimated using the moving array triangulation technique from a sequence of whale bearing measurements. Multiple concurrently vocalizing sperm whales, in the far-field of the horizontal receiver array, were distinguished and classified based on their horizontal spatial locations and the inter-pulse intervals of their vocalized click signals. We provide detailed analysis of over 15,000 fin whale 20 Hz vocalizations received on Oct 1-3, 2006 in the Gulf of Maine. These vocalizations are separated into 16 clusters following the clustering approaches. Seven of these types are prominent, each acounting for between 8% to 16% and together comprise roughly 85% of all the analyzed vocalizations. The 7 prominent clusters are each more abundant during nighttime hours by a factor of roughly 2.5 times than that of the daytime. The diel-spatial correlation of the 7 prominent clusters to the simultaneously observed densities of their fish prey, the Atlantic herring in the Gulf of Maine, is provided which implies that the factor of roughly 2.5 increase in call rate during night-time hours can be attributed to increased fish-feeding activities. (Abstract shortened by ProQuest.).

A New Mechanism of Sound Generation in Songbirds

NASA Astrophysics Data System (ADS)

Goller, Franz; Larsen, Ole N.

1997-12-01

Our current understanding of the sound-generating mechanism in the songbird vocal organ, the syrinx, is based on indirect evidence and theoretical treatments. The classical avian model of sound production postulates that the medial tympaniform membranes (MTM) are the principal sound generators. We tested the role of the MTM in sound generation and studied the songbird syrinx more directly by filming it endoscopically. After we surgically incapacitated the MTM as a vibratory source, zebra finches and cardinals were not only able to vocalize, but sang nearly normal song. This result shows clearly that the MTM are not the principal sound source. The endoscopic images of the intact songbird syrinx during spontaneous and brain stimulation-induced vocalizations illustrate the dynamics of syringeal reconfiguration before phonation and suggest a different model for sound production. Phonation is initiated by rostrad movement and stretching of the syrinx. At the same time, the syrinx is closed through movement of two soft tissue masses, the medial and lateral labia, into the bronchial lumen. Sound production always is accompanied by vibratory motions of both labia, indicating that these vibrations may be the sound source. However, because of the low temporal resolution of the imaging system, the frequency and phase of labial vibrations could not be assessed in relation to that of the generated sound. Nevertheless, in contrast to the previous model, these observations show that both labia contribute to aperture control and strongly suggest that they play an important role as principal sound generators.

On the role of glottis-interior sources in the production of voiced sound.

PubMed

Howe, M S; McGowan, R S

2012-02-01

The voice source is dominated by aeroacoustic sources downstream of the glottis. In this paper an investigation is made of the contribution to voiced speech of secondary sources within the glottis. The acoustic waveform is ultimately determined by the volume velocity of air at the glottis, which is controlled by vocal fold vibration, pressure forcing from the lungs, and unsteady backreactions from the sound and from the supraglottal air jet. The theory of aerodynamic sound is applied to study the influence on the fine details of the acoustic waveform of "potential flow" added-mass-type glottal sources, glottis friction, and vorticity either in the glottis-wall boundary layer or in the portion of the free jet shear layer within the glottis. These sources govern predominantly the high frequency content of the sound when the glottis is near closure. A detailed analysis performed for a canonical, cylindrical glottis of rectangular cross section indicates that glottis-interior boundary/shear layer vortex sources and the surface frictional source are of comparable importance; the influence of the potential flow source is about an order of magnitude smaller. © 2012 Acoustical Society of America

Calibration of International Space Station (ISS) Node 1 Vibro-Acoustic Model-Report 2

NASA Technical Reports Server (NTRS)

Zhang, Weiguo; Raveendra, Ravi

2014-01-01

Reported here is the capability of the Energy Finite Element Method (E-FEM) to predict the vibro-acoustic sound fields within the International Space Station (ISS) Node 1 and to compare the results with simulated leak sounds. A series of electronically generated structural ultrasonic noise sources were created in the pressure wall to emulate leak signals at different locations of the Node 1 STA module during its period of storage at Stennis Space Center (SSC). The exact sound source profiles created within the pressure wall at the source were unknown, but were estimated from the closest sensor measurement. The E-FEM method represents a reverberant sound field calculation, and of importance to this application is the requirement to correctly handle the direct field effect of the sound generation. It was also important to be able to compute the sound energy fields in the ultrasonic frequency range. This report demonstrates the capability of this technology as applied to this type of application.

[Perception by teenagers and adults of the changed by amplitude sound sequences used in models of movement of the sound source].

PubMed

Andreeva, I G; Vartanian, I A

2012-01-01

The ability to evaluate direction of amplitude changes of sound stimuli was studied in adults and in the 11-12- and 15-16-year old teenagers. The stimuli representing sequences of fragments of the tone of 1 kHz, whose amplitude is changing with time, are used as model of approach and withdrawal of the sound sources. The 11-12-year old teenagers at estimation of direction of amplitude changes were shown to make the significantly higher number of errors as compared with two other examined groups, including those in repeated experiments. The structure of errors - the ratio of the portion of errors at estimation of increasing and decreasing by amplitude stimulus - turned out to be different in teenagers and in adults. The question is discussed about the effect of unspecific activation of the large hemisphere cortex in teenagers on processes if taking solution about the complex sound stimulus, including a possibility estimation of approach and withdrawal of the sound source.

Interior and exterior sound field control using general two-dimensional first-order sources.

PubMed

Poletti, M A; Abhayapala, T D

2011-01-01

Reproduction of a given sound field interior to a circular loudspeaker array without producing an undesirable exterior sound field is an unsolved problem over a broadband of frequencies. At low frequencies, by implementing the Kirchhoff-Helmholtz integral using a circular discrete array of line-source loudspeakers, a sound field can be recreated within the array and produce no exterior sound field, provided that the loudspeakers have azimuthal polar responses with variable first-order responses which are a combination of a two-dimensional (2D) monopole and a radially oriented 2D dipole. This paper examines the performance of circular discrete arrays of line-source loudspeakers which also include a tangential dipole, providing general variable-directivity responses in azimuth. It is shown that at low frequencies, the tangential dipoles are not required, but that near and above the Nyquist frequency, the tangential dipoles can both improve the interior accuracy and reduce the exterior sound field. The additional dipoles extend the useful range of the array by around an octave.

The silent base flow and the sound sources in a laminar jet.

PubMed

Sinayoko, Samuel; Agarwal, Anurag

2012-03-01

An algorithm to compute the silent base flow sources of sound in a jet is introduced. The algorithm is based on spatiotemporal filtering of the flow field and is applicable to multifrequency sources. It is applied to an axisymmetric laminar jet and the resulting sources are validated successfully. The sources are compared to those obtained from two classical acoustic analogies, based on quiescent and time-averaged base flows. The comparison demonstrates how the silent base flow sources shed light on the sound generation process. It is shown that the dominant source mechanism in the axisymmetric laminar jet is "shear-noise," which is a linear mechanism. The algorithm presented here could be applied to fully turbulent flows to understand the aerodynamic noise-generation mechanism. © 2012 Acoustical Society of America

Sound Radiated by a Wave-Like Structure in a Compressible Jet

NASA Technical Reports Server (NTRS)

Golubev, V. V.; Prieto, A. F.; Mankbadi, R. R.; Dahl, M. D.; Hixon, R.

2003-01-01

This paper extends the analysis of acoustic radiation from the source model representing spatially-growing instability waves in a round jet at high speeds. Compared to previous work, a modified approach to the sound source modeling is examined that employs a set of solutions to linearized Euler equations. The sound radiation is then calculated using an integral surface method.

The Distressed Brain: A Group Blind Source Separation Analysis on Tinnitus

PubMed Central

De Ridder, Dirk; Vanneste, Sven; Congedo, Marco

2011-01-01

Background Tinnitus, the perception of a sound without an external sound source, can lead to variable amounts of distress. Methodology In a group of tinnitus patients with variable amounts of tinnitus related distress, as measured by the Tinnitus Questionnaire (TQ), an electroencephalography (EEG) is performed, evaluating the patients' resting state electrical brain activity. This resting state electrical activity is compared with a control group and between patients with low (N = 30) and high distress (N = 25). The groups are homogeneous for tinnitus type, tinnitus duration or tinnitus laterality. A group blind source separation (BSS) analysis is performed using a large normative sample (N = 84), generating seven normative components to which high and low tinnitus patients are compared. A correlation analysis of the obtained normative components' relative power and distress is performed. Furthermore, the functional connectivity as reflected by lagged phase synchronization is analyzed between the brain areas defined by the components. Finally, a group BSS analysis on the Tinnitus group as a whole is performed. Conclusions Tinnitus can be characterized by at least four BSS components, two of which are posterior cingulate based, one based on the subgenual anterior cingulate and one based on the parahippocampus. Only the subgenual component correlates with distress. When performed on a normative sample, group BSS reveals that distress is characterized by two anterior cingulate based components. Spectral analysis of these components demonstrates that distress in tinnitus is related to alpha and beta changes in a network consisting of the subgenual anterior cingulate cortex extending to the pregenual and dorsal anterior cingulate cortex as well as the ventromedial prefrontal cortex/orbitofrontal cortex, insula, and parahippocampus. This network overlaps partially with brain areas implicated in distress in patients suffering from pain, functional somatic syndromes and posttraumatic stress disorder, and might therefore represent a specific distress network. PMID:21998628

Photoacoustic Effect Generated from an Expanding Spherical Source

NASA Astrophysics Data System (ADS)

Bai, Wenyu; Diebold, Gerald J.

2018-02-01

Although the photoacoustic effect is typically generated by amplitude-modulated continuous or pulsed radiation, the form of the wave equation for pressure that governs the generation of sound indicates that optical sources moving in an absorbing fluid can produce sound as well. Here, the characteristics of the acoustic wave produced by a radially symmetric Gaussian source expanding outwardly from the origin are found. The unique feature of the photoacoustic effect from the spherical source is a trailing compressive wave that arises from reflection of an inwardly propagating component of the wave. Similar to the one-dimensional geometry, an unbounded amplification effect is found for the Gaussian source expanding at the sound speed.

Measurement and Numerical Calculation of Force on a Particle in a Strong Acoustic Field Required for Levitation

NASA Astrophysics Data System (ADS)

Kozuka, Teruyuki; Yasui, Kyuichi; Tuziuti, Toru; Towata, Atsuya; Lee, Judy; Iida, Yasuo

2009-07-01

Using a standing-wave field generated between a sound source and a reflector, it is possible to trap small objects at nodes of the sound pressure distribution in air. In this study, a sound field generated under a flat or concave reflector was studied by both experimental measurement and numerical calculation. The calculated result agrees well with the experimental data. The maximum force generated between a sound source of 25.0 mm diameter and a concave reflector is 0.8 mN in the experiment. A steel ball of 2.0 mm in diameter was levitated in the sound field in air.

Bio-Inspired Micromechanical Directional Acoustic Sensor

NASA Astrophysics Data System (ADS)

Swan, William; Alves, Fabio; Karunasiri, Gamani

Conventional directional sound sensors employ an array of spatially separated microphones and the direction is determined using arrival times and amplitudes. In nature, insects such as the Ormia ochracea fly can determine the direction of sound using a hearing organ much smaller than the wavelength of sound it detects. The fly's eardrums are mechanically coupled, only separated by about 1 mm, and have remarkable directional sensitivity. A micromechanical sensor based on the fly's hearing system was designed and fabricated on a silicon on insulator (SOI) substrate using MEMS technology. The sensor consists of two 1 mm2 wings connected using a bridge and to the substrate using two torsional legs. The dimensions of the sensor and material stiffness determine the frequency response of the sensor. The vibration of the wings in response to incident sound at the bending resonance was measured using a laser vibrometer and found to be about 1 μm/Pa. The electronic response of the sensor to sound was measured using integrated comb finger capacitors and found to be about 25 V/Pa. The fabricated sensors showed good directional sensitivity. In this talk, the design, fabrication and characteristics of the directional sound sensor will be described. Supported by ONR and TDSI.

Fourth Computational Aeroacoustics (CAA) Workshop on Benchmark Problems

NASA Technical Reports Server (NTRS)

Dahl, Milo D. (Editor)

2004-01-01

This publication contains the proceedings of the Fourth Computational Aeroacoustics (CAA) Workshop on Benchmark Problems. In this workshop, as in previous workshops, the problems were devised to gauge the technological advancement of computational techniques to calculate all aspects of sound generation and propagation in air directly from the fundamental governing equations. A variety of benchmark problems have been previously solved ranging from simple geometries with idealized acoustic conditions to test the accuracy and effectiveness of computational algorithms and numerical boundary conditions; to sound radiation from a duct; to gust interaction with a cascade of airfoils; to the sound generated by a separating, turbulent viscous flow. By solving these and similar problems, workshop participants have shown the technical progress from the basic challenges to accurate CAA calculations to the solution of CAA problems of increasing complexity and difficulty. The fourth CAA workshop emphasized the application of CAA methods to the solution of realistic problems. The workshop was held at the Ohio Aerospace Institute in Cleveland, Ohio, on October 20 to 22, 2003. At that time, workshop participants presented their solutions to problems in one or more of five categories. Their solutions are presented in this proceedings along with the comparisons of their solutions to the benchmark solutions or experimental data. The five categories for the benchmark problems were as follows: Category 1:Basic Methods. The numerical computation of sound is affected by, among other issues, the choice of grid used and by the boundary conditions. Category 2:Complex Geometry. The ability to compute the sound in the presence of complex geometric surfaces is important in practical applications of CAA. Category 3:Sound Generation by Interacting With a Gust. The practical application of CAA for computing noise generated by turbomachinery involves the modeling of the noise source mechanism as a vortical gust interacting with an airfoil. Category 4:Sound Transmission and Radiation. Category 5:Sound Generation in Viscous Problems. Sound is generated under certain conditions by a viscous flow as the flow passes an object or a cavity.

Sound field reproduction as an equivalent acoustical scattering problem.

PubMed

Fazi, Filippo Maria; Nelson, Philip A

2013-11-01

Given a continuous distribution of acoustic sources, the determination of the source strength that ensures the synthesis of a desired sound field is shown to be identical to the solution of an equivalent acoustic scattering problem. The paper begins with the presentation of the general theory that underpins sound field reproduction with secondary sources continuously arranged on the boundary of the reproduction region. The process of reproduction by a continuous source distribution is modeled by means of an integral operator (the single layer potential). It is then shown how the solution of the sound reproduction problem corresponds to that of an equivalent scattering problem. Analytical solutions are computed for two specific instances of this problem, involving, respectively, the use of a secondary source distribution in spherical and planar geometries. The results are shown to be the same as those obtained with analyses based on High Order Ambisonics and Wave Field Synthesis, respectively, thus bringing to light a fundamental analogy between these two methods of sound reproduction. Finally, it is shown how the physical optics (Kirchhoff) approximation enables the derivation of a high-frequency simplification for the problem under consideration, this in turn being related to the secondary source selection criterion reported in the literature on Wave Field Synthesis.

Investigation of spherical loudspeaker arrays for local active control of sound.

PubMed

Peleg, Tomer; Rafaely, Boaz

2011-10-01

Active control of sound can be employed globally to reduce noise levels in an entire enclosure, or locally around a listener's head. Recently, spherical loudspeaker arrays have been studied as multiple-channel sources for local active control of sound, presenting the fundamental theory and several active control configurations. In this paper, important aspects of using a spherical loudspeaker array for local active control of sound are further investigated. First, the feasibility of creating sphere-shaped quiet zones away from the source is studied both theoretically and numerically, showing that these quiet zones are associated with sound amplification and poor system robustness. To mitigate the latter, the design of shell-shaped quiet zones around the source is investigated. A combination of two spherical sources is then studied with the aim of enlarging the quiet zone. The two sources are employed to generate quiet zones that surround a rigid sphere, investigating the application of active control around a listener's head. A significant improvement in performance is demonstrated in this case over a conventional headrest-type system that uses two monopole secondary sources. Finally, several simulations are presented to support the theoretical work and to demonstrate the performance and limitations of the system. © 2011 Acoustical Society of America

Efficient techniques for wave-based sound propagation in interactive applications

NASA Astrophysics Data System (ADS)

Mehra, Ravish

Sound propagation techniques model the effect of the environment on sound waves and predict their behavior from point of emission at the source to the final point of arrival at the listener. Sound is a pressure wave produced by mechanical vibration of a surface that propagates through a medium such as air or water, and the problem of sound propagation can be formulated mathematically as a second-order partial differential equation called the wave equation. Accurate techniques based on solving the wave equation, also called the wave-based techniques, are too expensive computationally and memory-wise. Therefore, these techniques face many challenges in terms of their applicability in interactive applications including sound propagation in large environments, time-varying source and listener directivity, and high simulation cost for mid-frequencies. In this dissertation, we propose a set of efficient wave-based sound propagation techniques that solve these three challenges and enable the use of wave-based sound propagation in interactive applications. Firstly, we propose a novel equivalent source technique for interactive wave-based sound propagation in large scenes spanning hundreds of meters. It is based on the equivalent source theory used for solving radiation and scattering problems in acoustics and electromagnetics. Instead of using a volumetric or surface-based approach, this technique takes an object-centric approach to sound propagation. The proposed equivalent source technique generates realistic acoustic effects and takes orders of magnitude less runtime memory compared to prior wave-based techniques. Secondly, we present an efficient framework for handling time-varying source and listener directivity for interactive wave-based sound propagation. The source directivity is represented as a linear combination of elementary spherical harmonic sources. This spherical harmonic-based representation of source directivity can support analytical, data-driven, rotating or time-varying directivity function at runtime. Unlike previous approaches, the listener directivity approach can be used to compute spatial audio (3D audio) for a moving, rotating listener at interactive rates. Lastly, we propose an efficient GPU-based time-domain solver for the wave equation that enables wave simulation up to the mid-frequency range in tens of minutes on a desktop computer. It is demonstrated that by carefully mapping all the components of the wave simulator to match the parallel processing capabilities of the graphics processors, significant improvement in performance can be achieved compared to the CPU-based simulators, while maintaining numerical accuracy. We validate these techniques with offline numerical simulations and measured data recorded in an outdoor scene. We present results of preliminary user evaluations conducted to study the impact of these techniques on user's immersion in virtual environment. We have integrated these techniques with the Half-Life 2 game engine, Oculus Rift head-mounted display, and Xbox game controller to enable users to experience high-quality acoustics effects and spatial audio in the virtual environment.

Pitch-informed solo and accompaniment separation towards its use in music education applications

NASA Astrophysics Data System (ADS)

Cano, Estefanía; Schuller, Gerald; Dittmar, Christian

2014-12-01

We present a system for the automatic separation of solo instruments and music accompaniment in polyphonic music recordings. Our approach is based on a pitch detection front-end and a tone-based spectral estimation. We assess the plausibility of using sound separation technologies to create practice material in a music education context. To better understand the sound separation quality requirements in music education, a listening test was conducted to determine the most perceptually relevant signal distortions that need to be improved. Results from the listening test show that solo and accompaniment tracks pose different quality requirements and should be optimized differently. We propose and evaluate algorithm modifications to better understand their effects on objective perceptual quality measures. Finally, we outline possible ways of optimizing our separation approach to better suit the requirements of music education applications.

Echolocation versus echo suppression in humans

PubMed Central

Wallmeier, Ludwig; Geßele, Nikodemus; Wiegrebe, Lutz

2013-01-01

Several studies have shown that blind humans can gather spatial information through echolocation. However, when localizing sound sources, the precedence effect suppresses spatial information of echoes, and thereby conflicts with effective echolocation. This study investigates the interaction of echolocation and echo suppression in terms of discrimination suppression in virtual acoustic space. In the ‘Listening’ experiment, sighted subjects discriminated between positions of a single sound source, the leading or the lagging of two sources, respectively. In the ‘Echolocation’ experiment, the sources were replaced by reflectors. Here, the same subjects evaluated echoes generated in real time from self-produced vocalizations and thereby discriminated between positions of a single reflector, the leading or the lagging of two reflectors, respectively. Two key results were observed. First, sighted subjects can learn to discriminate positions of reflective surfaces echo-acoustically with accuracy comparable to sound source discrimination. Second, in the Listening experiment, the presence of the leading source affected discrimination of lagging sources much more than vice versa. In the Echolocation experiment, however, the presence of both the lead and the lag strongly affected discrimination. These data show that the classically described asymmetry in the perception of leading and lagging sounds is strongly diminished in an echolocation task. Additional control experiments showed that the effect is owing to both the direct sound of the vocalization that precedes the echoes and owing to the fact that the subjects actively vocalize in the echolocation task. PMID:23986105

Two dimensional sound field reproduction using higher order sources to exploit room reflections.

PubMed

Betlehem, Terence; Poletti, Mark A

2014-04-01

In this paper, sound field reproduction is performed in a reverberant room using higher order sources (HOSs) and a calibrating microphone array. Previously a sound field was reproduced with fixed directivity sources and the reverberation compensated for using digital filters. However by virtue of their directive properties, HOSs may be driven to not only avoid the creation of excess reverberation but also to use room reflection to contribute constructively to the desired sound field. The manner by which the loudspeakers steer the sound around the room is determined by measuring the acoustic transfer functions. The requirements on the number and order N of HOSs for accurate reproduction in a reverberant room are derived, showing a 2N + 1-fold decrease in the number of loudspeakers in comparison to using monopole sources. HOSs are shown applicable to rooms with a rich variety of wall reflections while in an anechoic room their advantages may be lost. Performance is investigated in a room using extensions of both the diffuse field model and a more rigorous image-source simulation method, which account for the properties of the HOSs. The robustness of the proposed method is validated by introducing measurement errors.

Seismic and Biological Sources of Ambient Ocean Sound

NASA Astrophysics Data System (ADS)

Freeman, Simon Eric

Sound is the most efficient radiation in the ocean. Sounds of seismic and biological origin contain information regarding the underlying processes that created them. A single hydrophone records summary time-frequency information from the volume within acoustic range. Beamforming using a hydrophone array additionally produces azimuthal estimates of sound sources. A two-dimensional array and acoustic focusing produce an unambiguous two-dimensional `image' of sources. This dissertation describes the application of these techniques in three cases. The first utilizes hydrophone arrays to investigate T-phases (water-borne seismic waves) in the Philippine Sea. Ninety T-phases were recorded over a 12-day period, implying a greater number of seismic events occur than are detected by terrestrial seismic monitoring in the region. Observation of an azimuthally migrating T-phase suggests that reverberation of such sounds from bathymetric features can occur over megameter scales. In the second case, single hydrophone recordings from coral reefs in the Line Islands archipelago reveal that local ambient reef sound is spectrally similar to sounds produced by small, hard-shelled benthic invertebrates in captivity. Time-lapse photography of the reef reveals an increase in benthic invertebrate activity at sundown, consistent with an increase in sound level. The dominant acoustic phenomenon on these reefs may thus originate from the interaction between a large number of small invertebrates and the substrate. Such sounds could be used to take census of hard-shelled benthic invertebrates that are otherwise extremely difficult to survey. A two-dimensional `map' of sound production over a coral reef in the Hawaiian Islands was obtained using two-dimensional hydrophone array in the third case. Heterogeneously distributed bio-acoustic sources were generally co-located with rocky reef areas. Acoustically dominant snapping shrimp were largely restricted to one location within the area surveyed. This distribution of sources could reveal small-scale spatial ecological limitations, such as the availability of food and shelter. While array-based passive acoustic sensing is well established in seismoacoustics, the technique is little utilized in the study of ambient biological sound. With the continuance of Moore's law and advances in battery and memory technology, inferring biological processes from ambient sound may become a more accessible tool in underwater ecological evaluation and monitoring.

Calculating far-field radiated sound pressure levels from NASTRAN output

NASA Technical Reports Server (NTRS)

Lipman, R. R.

1986-01-01

FAFRAP is a computer program which calculates far field radiated sound pressure levels from quantities computed by a NASTRAN direct frequency response analysis of an arbitrarily shaped structure. Fluid loading on the structure can be computed directly by NASTRAN or an added-mass approximation to fluid loading on the structure can be used. Output from FAFRAP includes tables of radiated sound pressure levels and several types of graphic output. FAFRAP results for monopole and dipole sources compare closely with an explicit calculation of the radiated sound pressure level for those sources.

Doppler effect for sound emitted by a moving airborne source and received by acoustic sensors located above and below the sea surface.

PubMed

Ferguson, B G

1993-12-01

The acoustic emissions from a propeller-driven aircraft are received by a microphone mounted just above ground level and then by a hydrophone located below the sea surface. The dominant feature in the output spectrum of each acoustic sensor is the spectral line corresponding to the propeller blade rate. A frequency estimation technique is applied to the acoustic data from each sensor so that the Doppler shift in the blade rate can be observed at short time intervals during the aircraft's transit overhead. For each acoustic sensor, the observed variation with time of the Doppler-shifted blade rate is compared with the variation predicted by a simple ray-theory model that assumes the atmosphere and the sea are distinct isospeed sound propagation media separated by a plane boundary. The results of the comparison are shown for an aircraft flying with a speed of about 250 kn at altitudes of 500, 700, and 1000 ft.

Auditory Confrontation Naming in Alzheimer’s Disease

PubMed Central

Brandt, Jason; Bakker, Arnold; Maroof, David Aaron

2010-01-01

Naming is a fundamental aspect of language and is virtually always assessed with visual confrontation tests. Tests of the ability to name objects by their characteristic sounds would be particularly useful in the assessment of visually impaired patients, and may be particularly sensitive in Alzheimer’s disease (AD). We developed an Auditory Naming Task, requiring the identification of the source of environmental sounds (i.e., animal calls, musical instruments, vehicles) and multiple-choice recognition of those not identified. In two separate studies, mild-to-moderate AD patients performed more poorly than cognitively normal elderly on the Auditory Naming Task. This task was also more difficult than two versions of a comparable Visual Naming Task, and correlated more highly with Mini-Mental State Exam score. Internal consistency reliability was acceptable, although ROC analysis revealed auditory naming to be slightly less successful than visual confrontation naming in discriminating AD patients from normal subjects. Nonetheless, our Auditory Naming Test may prove useful in research and clinical practice, especially with visually-impaired patients. PMID:20981630

Modelling of human low frequency sound localization acuity demonstrates dominance of spatial variation of interaural time difference and suggests uniform just-noticeable differences in interaural time difference.

PubMed

Smith, Rosanna C G; Price, Stephen R

2014-01-01

Sound source localization is critical to animal survival and for identification of auditory objects. We investigated the acuity with which humans localize low frequency, pure tone sounds using timing differences between the ears. These small differences in time, known as interaural time differences or ITDs, are identified in a manner that allows localization acuity of around 1° at the midline. Acuity, a relative measure of localization ability, displays a non-linear variation as sound sources are positioned more laterally. All species studied localize sounds best at the midline and progressively worse as the sound is located out towards the side. To understand why sound localization displays this variation with azimuthal angle, we took a first-principles, systemic, analytical approach to model localization acuity. We calculated how ITDs vary with sound frequency, head size and sound source location for humans. This allowed us to model ITD variation for previously published experimental acuity data and determine the distribution of just-noticeable differences in ITD. Our results suggest that the best-fit model is one whereby just-noticeable differences in ITDs are identified with uniform or close to uniform sensitivity across the physiological range. We discuss how our results have several implications for neural ITD processing in different species as well as development of the auditory system.

Graphene-on-paper sound source devices.

PubMed

Tian, He; Ren, Tian-Ling; Xie, Dan; Wang, Yu-Feng; Zhou, Chang-Jian; Feng, Ting-Ting; Fu, Di; Yang, Yi; Peng, Ping-Gang; Wang, Li-Gang; Liu, Li-Tian

2011-06-28

We demonstrate an interesting phenomenon that graphene can emit sound. The application of graphene can be expanded in the acoustic field. Graphene-on-paper sound source devices are made by patterning graphene on paper substrates. Three graphene sheet samples with the thickness of 100, 60, and 20 nm were fabricated. Sound emission from graphene is measured as a function of power, distance, angle, and frequency in the far-field. The theoretical model of air/graphene/paper/PCB board multilayer structure is established to analyze the sound directivity, frequency response, and efficiency. Measured sound pressure level (SPL) and efficiency are in good agreement with theoretical results. It is found that graphene has a significant flat frequency response in the wide ultrasound range 20-50 kHz. In addition, the thinner graphene sheets can produce higher SPL due to its lower heat capacity per unit area (HCPUA). The infrared thermal images reveal that a thermoacoustic effect is the working principle. We find that the sound performance mainly depends on the HCPUA of the conductor and the thermal properties of the substrate. The paper-based graphene sound source devices have highly reliable, flexible, no mechanical vibration, simple structure and high performance characteristics. It could open wide applications in multimedia, consumer electronics, biological, medical, and many other areas.

Representation of Sound Objects within Early-Stage Auditory Areas: A Repetition Effect Study Using 7T fMRI

PubMed Central

Da Costa, Sandra; Bourquin, Nathalie M.-P.; Knebel, Jean-François; Saenz, Melissa; van der Zwaag, Wietske; Clarke, Stephanie

2015-01-01

Environmental sounds are highly complex stimuli whose recognition depends on the interaction of top-down and bottom-up processes in the brain. Their semantic representations were shown to yield repetition suppression effects, i. e. a decrease in activity during exposure to a sound that is perceived as belonging to the same source as a preceding sound. Making use of the high spatial resolution of 7T fMRI we have investigated the representations of sound objects within early-stage auditory areas on the supratemporal plane. The primary auditory cortex was identified by means of tonotopic mapping and the non-primary areas by comparison with previous histological studies. Repeated presentations of different exemplars of the same sound source, as compared to the presentation of different sound sources, yielded significant repetition suppression effects within a subset of early-stage areas. This effect was found within the right hemisphere in primary areas A1 and R as well as two non-primary areas on the antero-medial part of the planum temporale, and within the left hemisphere in A1 and a non-primary area on the medial part of Heschl’s gyrus. Thus, several, but not all early-stage auditory areas encode the meaning of environmental sounds. PMID:25938430

Conversion of environmental data to a digital-spatial database, Puget Sound area, Washington

USGS Publications Warehouse

Uhrich, M.A.; McGrath, T.S.

1997-01-01

Data and maps from the Puget Sound Environmental Atlas, compiled for the U.S. Environmental Protection Agency, the Puget Sound Water Quality Authority, and the U.S. Army Corps of Engineers, have been converted into a digital-spatial database using a geographic information system. Environmental data for the Puget Sound area,collected from sources other than the Puget SoundEnvironmental Atlas by different Federal, State, andlocal agencies, also have been converted into thisdigital-spatial database. Background on the geographic-information-system planning process, the design and implementation of the geographic information-system database, and the reasons for conversion to this digital-spatial database are included in this report. The Puget Sound Environmental Atlas data layers include information about seabird nesting areas, eelgrass and kelp habitat, marine mammal and fish areas, and shellfish resources and bed certification. Data layers, from sources other than the Puget Sound Environmental Atlas, include the Puget Sound shoreline, the water-body system, shellfish growing areas, recreational shellfish beaches, sewage-treatment outfalls, upland hydrography,watershed and political boundaries, and geographicnames. The sources of data, descriptions of the datalayers, and the steps and errors of processing associated with conversion to a digital-spatial database used in development of the Puget Sound Geographic Information System also are included in this report. The appendixes contain data dictionaries for each of the resource layers and error values for the conversion of Puget SoundEnvironmental Atlas data.

Integrating speech in time depends on temporal expectancies and attention.

PubMed

Scharinger, Mathias; Steinberg, Johanna; Tavano, Alessandro

2017-08-01

Sensory information that unfolds in time, such as in speech perception, relies on efficient chunking mechanisms in order to yield optimally-sized units for further processing. Whether or not two successive acoustic events receive a one-unit or a two-unit interpretation seems to depend on the fit between their temporal extent and a stipulated temporal window of integration. However, there is ongoing debate on how flexible this temporal window of integration should be, especially for the processing of speech sounds. Furthermore, there is no direct evidence of whether attention may modulate the temporal constraints on the integration window. For this reason, we here examine how different word durations, which lead to different temporal separations of sound onsets, interact with attention. In an Electroencephalography (EEG) study, participants actively and passively listened to words where word-final consonants were occasionally omitted. Words had either a natural duration or were artificially prolonged in order to increase the separation of speech sound onsets. Omission responses to incomplete speech input, originating in left temporal cortex, decreased when the critical speech sound was separated from previous sounds by more than 250 msec, i.e., when the separation was larger than the stipulated temporal window of integration (125-150 msec). Attention, on the other hand, only increased omission responses for stimuli with natural durations. We complemented the event-related potential (ERP) analyses by a frequency-domain analysis on the stimulus presentation rate. Notably, the power of stimulation frequency showed the same duration and attention effects than the omission responses. We interpret these findings on the background of existing research on temporal integration windows and further suggest that our findings may be accounted for within the framework of predictive coding. Copyright © 2017 Elsevier Ltd. All rights reserved.

Wave Field Synthesis of moving sources with arbitrary trajectory and velocity profile.

PubMed

Firtha, Gergely; Fiala, Péter

2017-08-01

The sound field synthesis of moving sound sources is of great importance when dynamic virtual sound scenes are to be reconstructed. Previous solutions considered only virtual sources moving uniformly along a straight trajectory, synthesized employing a linear loudspeaker array. This article presents the synthesis of point sources following an arbitrary trajectory. Under high-frequency assumptions 2.5D Wave Field Synthesis driving functions are derived for arbitrary shaped secondary source contours by adapting the stationary phase approximation to the dynamic description of sources in motion. It is explained how a referencing function should be chosen in order to optimize the amplitude of synthesis on an arbitrary receiver curve. Finally, a finite difference implementation scheme is considered, making the presented approach suitable for real-time applications.

Sound localization skills in children who use bilateral cochlear implants and in children with normal acoustic hearing

PubMed Central

Grieco-Calub, Tina M.; Litovsky, Ruth Y.

2010-01-01

Objectives To measure sound source localization in children who have sequential bilateral cochlear implants (BICIs); to determine if localization accuracy correlates with performance on a right-left discrimination task (i.e., spatial acuity); to determine if there is a measurable bilateral benefit on a sound source identification task (i.e., localization accuracy) by comparing performance under bilateral and unilateral listening conditions; to determine if sound source localization continues to improve with longer durations of bilateral experience. Design Two groups of children participated in this study: a group of 21 children who received BICIs in sequential procedures (5–14 years old) and a group of 7 typically-developing children with normal acoustic hearing (5 years old). Testing was conducted in a large sound-treated booth with loudspeakers positioned on a horizontal arc with a radius of 1.2 m. Children participated in two experiments that assessed spatial hearing skills. Spatial hearing acuity was assessed with a discrimination task in which listeners determined if a sound source was presented on the right or left side of center; the smallest angle at which performance on this task was reliably above chance is the minimum audible angle. Sound localization accuracy was assessed with a sound source identification task in which children identified the perceived position of the sound source from a multi-loudspeaker array (7 or 15); errors are quantified using the root-mean-square (RMS) error. Results Sound localization accuracy was highly variable among the children with BICIs, with RMS errors ranging from 19°–56°. Performance of the NH group, with RMS errors ranging from 9°–29° was significantly better. Within the BICI group, in 11/21 children RMS errors were smaller in the bilateral vs. unilateral listening condition, indicating bilateral benefit. There was a significant correlation between spatial acuity and sound localization accuracy (R2=0.68, p<0.01), suggesting that children who achieve small RMS errors tend to have the smallest MAAs. Although there was large intersubject variability, testing of 11 children in the BICI group at two sequential visits revealed a subset of children who show improvement in spatial hearing skills over time. Conclusions A subset of children who use sequential BICIs can acquire sound localization abilities, even after long intervals between activation of hearing in the first- and second-implanted ears. This suggests that children with activation of the second implant later in life may be capable of developing spatial hearing abilities. The large variability in performance among the children with BICIs suggests that maturation of sound localization abilities in children with BICIs may be dependent on various individual subject factors such as age of implantation and chronological age. PMID:20592615

33 CFR 167.1703 - In Prince William Sound: Valdez Arm Traffic Separation Scheme.

Code of Federal Regulations, 2012 CFR

2012-07-01

... Arm Traffic Separation Scheme. 167.1703 Section 167.1703 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF HOMELAND SECURITY (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1703 In Prince...

33 CFR 167.1703 - In Prince William Sound: Valdez Arm Traffic Separation Scheme.

Code of Federal Regulations, 2014 CFR

2014-07-01

... Arm Traffic Separation Scheme. 167.1703 Section 167.1703 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF HOMELAND SECURITY (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1703 In Prince...

33 CFR 167.1703 - In Prince William Sound: Valdez Arm Traffic Separation Scheme.

Code of Federal Regulations, 2013 CFR

2013-07-01

... Arm Traffic Separation Scheme. 167.1703 Section 167.1703 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF HOMELAND SECURITY (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1703 In Prince...

33 CFR 167.1703 - In Prince William Sound: Valdez Arm Traffic Separation Scheme.

Code of Federal Regulations, 2011 CFR

2011-07-01

... Arm Traffic Separation Scheme. 167.1703 Section 167.1703 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF HOMELAND SECURITY (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1703 In Prince...

33 CFR 167.1703 - In Prince William Sound: Valdez Arm Traffic Separation Scheme.

Code of Federal Regulations, 2010 CFR

2010-07-01

... Arm Traffic Separation Scheme. 167.1703 Section 167.1703 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF HOMELAND SECURITY (CONTINUED) PORTS AND WATERWAYS SAFETY OFFSHORE TRAFFIC SEPARATION SCHEMES Description of Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1703 In Prince...

Hardwall acoustical characteristics and measurement capabilities of the NASA Lewis 9 x 15 foot low speed wind tunnel

NASA Technical Reports Server (NTRS)

Rentz, P. E.

1976-01-01

Experimental evaluations of the acoustical characteristics and source sound power and directionality measurement capabilities of the NASA Lewis 9 x 15 foot low speed wind tunnel in the untreated or hardwall configuration were performed. The results indicate that source sound power estimates can be made using only settling chamber sound pressure measurements. The accuracy of these estimates, expressed as one standard deviation, can be improved from + or - 4 db to + or - 1 db if sound pressure measurements in the preparation room and diffuser are also used and source directivity information is utilized. A simple procedure is presented. Acceptably accurate measurements of source direct field acoustic radiation were found to be limited by the test section reverberant characteristics to 3.0 feet for omni-directional and highly directional sources. Wind-on noise measurements in the test section, settling chamber and preparation room were found to depend on the sixth power of tunnel velocity. The levels were compared with various analytic models. Results are presented and discussed.

Reduced order modeling of head related transfer functions for virtual acoustic displays

NASA Astrophysics Data System (ADS)

Willhite, Joel A.; Frampton, Kenneth D.; Grantham, D. Wesley

2003-04-01

The purpose of this work is to improve the computational efficiency in acoustic virtual applications by creating and testing reduced order models of the head related transfer functions used in localizing sound sources. State space models of varying order were generated from zero-elevation Head Related Impulse Responses (HRIRs) using Kungs Single Value Decomposition (SVD) technique. The inputs to the models are the desired azimuths of the virtual sound sources (from minus 90 deg to plus 90 deg, in 10 deg increments) and the outputs are the left and right ear impulse responses. Trials were conducted in an anechoic chamber in which subjects were exposed to real sounds that were emitted by individual speakers across a numbered speaker array, phantom sources generated from the original HRIRs, and phantom sound sources generated with the different reduced order state space models. The error in the perceived direction of the phantom sources generated from the reduced order models was compared to errors in localization using the original HRIRs.

Estimation of multiple sound sources with data and model uncertainties using the EM and evidential EM algorithms

NASA Astrophysics Data System (ADS)

Wang, Xun; Quost, Benjamin; Chazot, Jean-Daniel; Antoni, Jérôme

2016-01-01

This paper considers the problem of identifying multiple sound sources from acoustical measurements obtained by an array of microphones. The problem is solved via maximum likelihood. In particular, an expectation-maximization (EM) approach is used to estimate the sound source locations and strengths, the pressure measured by a microphone being interpreted as a mixture of latent signals emitted by the sources. This work also considers two kinds of uncertainties pervading the sound propagation and measurement process: uncertain microphone locations and uncertain wavenumber. These uncertainties are transposed to the data in the belief functions framework. Then, the source locations and strengths can be estimated using a variant of the EM algorithm, known as the Evidential EM (E2M) algorithm. Eventually, both simulation and real experiments are shown to illustrate the advantage of using the EM in the case without uncertainty and the E2M in the case of uncertain measurement.

Auditory Localization: An Annotated Bibliography

DTIC Science & Technology

1983-11-01

tranverse plane, natural sound localization in ,-- both horizontal and vertical planes can be performed with nearly the same accuracy as real sound sources...important for unscrambling the competing sounds which so often occur in natural environments. A workable sound sensor has been constructed and empirical

Speech Intelligibility in Various Noise Conditions with the Nucleus® 5 CP810 Sound Processor.

PubMed

Dillier, Norbert; Lai, Wai Kong

2015-06-11

The Nucleus(®) 5 System Sound Processor (CP810, Cochlear™, Macquarie University, NSW, Australia) contains two omnidirectional microphones. They can be configured as a fixed directional microphone combination (called Zoom) or as an adaptive beamformer (called Beam), which adjusts the directivity continuously to maximally reduce the interfering noise. Initial evaluation studies with the CP810 had compared performance and usability of the new processor in comparison with the Freedom™ Sound Processor (Cochlear™) for speech in quiet and noise for a subset of the processing options. This study compares the two processing options suggested to be used in noisy environments, Zoom and Beam, for various sound field conditions using a standardized speech in noise matrix test (Oldenburg sentences test). Nine German-speaking subjects who previously had been using the Freedom speech processor and subsequently were upgraded to the CP810 device participated in this series of additional evaluation tests. The speech reception threshold (SRT for 50% speech intelligibility in noise) was determined using sentences presented via loudspeaker at 65 dB SPL in front of the listener and noise presented either via the same loudspeaker (S0N0) or at 90 degrees at either the ear with the sound processor (S0NCI+) or the opposite unaided ear (S0NCI-). The fourth noise condition consisted of three uncorrelated noise sources placed at 90, 180 and 270 degrees. The noise level was adjusted through an adaptive procedure to yield a signal to noise ratio where 50% of the words in the sentences were correctly understood. In spatially separated speech and noise conditions both Zoom and Beam could improve the SRT significantly. For single noise sources, either ipsilateral or contralateral to the cochlear implant sound processor, average improvements with Beam of 12.9 and 7.9 dB in SRT were found. The average SRT of -8 dB for Beam in the diffuse noise condition (uncorrelated noise from both sides and back) is truly remarkable and comparable to the performance of normal hearing listeners in the same test environment. The static directivity (Zoom) option in the diffuse noise condition still provides a significant benefit of 5.9 dB in comparison with the standard omnidirectional microphone setting. These results indicate that CI recipients may improve their speech recognition in noisy environments significantly using these directional microphone-processing options.

Detection of Sound Image Movement During Horizontal Head Rotation

PubMed Central

Ohba, Kagesho; Iwaya, Yukio; Suzuki, Yôiti

2016-01-01

Movement detection for a virtual sound source was measured during the listener’s horizontal head rotation. Listeners were instructed to do head rotation at a given speed. A trial consisted of two intervals. During an interval, a virtual sound source was presented 60° to the right or left of the listener, who was instructed to rotate the head to face the sound image position. Then in one of a pair of intervals, the sound position was moved slightly in the middle of the rotation. Listeners were asked to judge the interval in a trial during which the sound stimuli moved. Results suggest that detection thresholds are higher when listeners do head rotation. Moreover, this effect was found to be independent of the rotation velocity. PMID:27698993

Rotorcraft Noise Model

NASA Technical Reports Server (NTRS)

Lucas, Michael J.; Marcolini, Michael A.

1997-01-01

The Rotorcraft Noise Model (RNM) is an aircraft noise impact modeling computer program being developed for NASA-Langley Research Center which calculates sound levels at receiver positions either on a uniform grid or at specific defined locations. The basic computational model calculates a variety of metria. Acoustic properties of the noise source are defined by two sets of sound pressure hemispheres, each hemisphere being centered on a noise source of the aircraft. One set of sound hemispheres provides the broadband data in the form of one-third octave band sound levels. The other set of sound hemispheres provides narrowband data in the form of pure-tone sound pressure levels and phase. Noise contours on the ground are output graphically or in tabular format, and are suitable for inclusion in Environmental Impact Statements or Environmental Assessments.

Differential presence of anthropogenic compounds dissolved in the marine waters of Puget Sound, WA and Barkley Sound, BC.

PubMed

Keil, Richard; Salemme, Keri; Forrest, Brittany; Neibauer, Jaqui; Logsdon, Miles

2011-11-01

Organic compounds were evaluated in March 2010 at 22 stations in Barkley Sound, Vancouver Island Canada and at 66 locations in Puget Sound. Of 37 compounds, 15 were xenobiotics, 8 were determined to have an anthropogenic imprint over natural sources, and 13 were presumed to be of natural or mixed origin. The three most frequently detected compounds were salicyclic acid, vanillin and thymol. The three most abundant compounds were diethylhexyl phthalate (DEHP), ethyl vanillin and benzaldehyde (∼600 n g L(-1) on average). Concentrations of xenobiotics were 10-100 times higher in Puget Sound relative to Barkley Sound. Three compound couplets are used to illustrate the influence of human activity on marine waters; vanillin and ethyl vanillin, salicylic acid and acetylsalicylic acid, and cinnamaldehyde and cinnamic acid. Ratios indicate that anthropogenic activities are the predominant source of these chemicals in Puget Sound. Published by Elsevier Ltd.

A SOUND SOURCE LOCALIZATION TECHNIQUE TO SUPPORT SEARCH AND RESCUE IN LOUD NOISE ENVIRONMENTS

NASA Astrophysics Data System (ADS)

Yoshinaga, Hiroshi; Mizutani, Koichi; Wakatsuki, Naoto

At some sites of earthquakes and other disasters, rescuers search for people buried under rubble by listening for the sounds which they make. Thus developing a technique to localize sound sources amidst loud noise will support such search and rescue operations. In this paper, we discuss an experiment performed to test an array signal processing technique which searches for unperceivable sound in loud noise environments. Two speakers simultaneously played a noise of a generator and a voice decreased by 20 dB (= 1/100 of power) from the generator noise at an outdoor space where cicadas were making noise. The sound signal was received by a horizontally set linear microphone array 1.05 m in length and consisting of 15 microphones. The direction and the distance of the voice were computed and the sound of the voice was extracted and played back as an audible sound by array signal processing.

Mapping the sound field of an erupting submarine volcano using an acoustic glider.

PubMed

Matsumoto, Haru; Haxel, Joseph H; Dziak, Robert P; Bohnenstiehl, Delwayne R; Embley, Robert W

2011-03-01

An underwater glider with an acoustic data logger flew toward a recently discovered erupting submarine volcano in the northern Lau basin. With the volcano providing a wide-band sound source, recordings from the two-day survey produced a two-dimensional sound level map spanning 1 km (depth) × 40 km(distance). The observed sound field shows depth- and range-dependence, with the first-order spatial pattern being consistent with the predictions of a range-dependent propagation model. The results allow constraining the acoustic source level of the volcanic activity and suggest that the glider provides an effective platform for monitoring natural and anthropogenic ocean sounds. © 2011 Acoustical Society of America

Cerebellar contribution to the prediction of self-initiated sounds.

PubMed

Knolle, Franziska; Schröger, Erich; Kotz, Sonja A

2013-10-01

In everyday life we frequently make the fundamental distinction between sensory input resulting from our own actions and sensory input that is externally-produced. It has been speculated that making this distinction involves the use of an internal forward-model, which enables the brain to adjust its response to self-produced sensory input. In the auditory domain, this idea has been supported by event-related potential and evoked-magnetic field studies revealing that self-initiated sounds elicit a suppressed N100/M100 brain response compared to externally-produced sounds. Moreover, a recent study reveals that patients with cerebellar lesions do not show a significant N100-suppression effect. This result supports the theory that the cerebellum is essential for generating internal forward predictions. However, all except one study compared self-initiated and externally-produced auditory stimuli in separate conditions. Such a setup prevents an unambiguous interpretation of the N100-suppression effect when distinguishing self- and externally-produced sensory stimuli: the N100-suppression can also be explained by differences in the allocation of attention in different conditions. In the current electroencephalography (EEG)-study we investigated the N100-suppression effect in an altered design comparing (i) self-initiated sounds to externally-produced sounds that occurred intermixed with these self-initiated sounds (i.e., both sound types occurred in the same condition) or (ii) self-initiated sounds to externally-produced sounds that occurred in separate conditions. Results reveal that the cerebellum generates selective predictions in response to self-initiated sounds independent of condition type: cerebellar patients, in contrast to healthy controls, do not display an N100-suppression effect in response to self-initiated sounds when intermixed with externally-produced sounds. Furthermore, the effect is not influenced by the temporal proximity of externally-produced sounds to self-produced sounds. Controls and patients showed a P200-reduction in response to self-initiated sounds. This suggests the existence of an additional and probably more conscious mechanism for identifying self-generated sounds that does not functionally depend on the cerebellum. Copyright © 2012 Elsevier Srl. All rights reserved.

Study of environmental sound source identification based on hidden Markov model for robust speech recognition

NASA Astrophysics Data System (ADS)

Nishiura, Takanobu; Nakamura, Satoshi

2003-10-01

Humans communicate with each other through speech by focusing on the target speech among environmental sounds in real acoustic environments. We can easily identify the target sound from other environmental sounds. For hands-free speech recognition, the identification of the target speech from environmental sounds is imperative. This mechanism may also be important for a self-moving robot to sense the acoustic environments and communicate with humans. Therefore, this paper first proposes hidden Markov model (HMM)-based environmental sound source identification. Environmental sounds are modeled by three states of HMMs and evaluated using 92 kinds of environmental sounds. The identification accuracy was 95.4%. This paper also proposes a new HMM composition method that composes speech HMMs and an HMM of categorized environmental sounds for robust environmental sound-added speech recognition. As a result of the evaluation experiments, we confirmed that the proposed HMM composition outperforms the conventional HMM composition with speech HMMs and a noise (environmental sound) HMM trained using noise periods prior to the target speech in a captured signal. [Work supported by Ministry of Public Management, Home Affairs, Posts and Telecommunications of Japan.

Development of Improved Surface Integral Methods for Jet Aeroacoustic Predictions

NASA Technical Reports Server (NTRS)

Pilon, Anthony R.; Lyrintzis, Anastasios S.

1997-01-01

The accurate prediction of aerodynamically generated noise has become an important goal over the past decade. Aeroacoustics must now be an integral part of the aircraft design process. The direct calculation of aerodynamically generated noise with CFD-like algorithms is plausible. However, large computer time and memory requirements often make these predictions impractical. It is therefore necessary to separate the aeroacoustics problem into two parts, one in which aerodynamic sound sources are determined, and another in which the propagating sound is calculated. This idea is applied in acoustic analogy methods. However, in the acoustic analogy, the determination of far-field sound requires the solution of a volume integral. This volume integration again leads to impractical computer requirements. An alternative to the volume integrations can be found in the Kirchhoff method. In this method, Green's theorem for the linear wave equation is used to determine sound propagation based on quantities on a surface surrounding the source region. The change from volume to surface integrals represents a tremendous savings in the computer resources required for an accurate prediction. This work is concerned with the development of enhancements of the Kirchhoff method for use in a wide variety of aeroacoustics problems. This enhanced method, the modified Kirchhoff method, is shown to be a Green's function solution of Lighthill's equation. It is also shown rigorously to be identical to the methods of Ffowcs Williams and Hawkings. This allows for development of versatile computer codes which can easily alternate between the different Kirchhoff and Ffowcs Williams-Hawkings formulations, using the most appropriate method for the problem at hand. The modified Kirchhoff method is developed primarily for use in jet aeroacoustics predictions. Applications of the method are shown for two dimensional and three dimensional jet flows. Additionally, the enhancements are generalized so that they may be used in any aeroacoustics problem.

Theoretical and Experimental Aspects of Acoustic Modelling of Engine Exhaust Systems with Applications to a Vacuum Pump

NASA Astrophysics Data System (ADS)

Sridhara, Basavapatna Sitaramaiah

In an internal combustion engine, the engine is the noise source and the exhaust pipe is the main transmitter of noise. Mufflers are often used to reduce engine noise level in the exhaust pipe. To optimize a muffler design, a series of experiments could be conducted using various mufflers installed in the exhaust pipe. For each configuration, the radiated sound pressure could be measured. However, this is not a very efficient method. A second approach would be to develop a scheme involving only a few measurements which can predict the radiated sound pressure at a specified distance from the open end of the exhaust pipe. In this work, the engine exhaust system was modelled as a lumped source-muffler-termination system. An expression for the predicted sound pressure level was derived in terms of the source and termination impedances, and the muffler geometry. The pressure source and monopole radiation models were used for the source and the open end of the exhaust pipe. The four pole parameters were used to relate the acoustic properties at two different cross sections of the muffler and the pipe. The developed formulation was verified through a series of experiments. Two loudspeakers and a reciprocating type vacuum pump were used as sound sources during the tests. The source impedance was measured using the direct, two-load and four-load methods. A simple expansion chamber and a side-branch resonator were used as mufflers. Sound pressure level measurements for the prediction scheme were made for several source-muffler and source-straight pipe combinations. The predicted and measured sound pressure levels were compared for all cases considered. In all cases, correlation of the experimental results and those predicted by the developed expressions was good. Predicted and measured values of the insertion loss of the mufflers were compared. The agreement between the two was good. Also, an error analysis of the four-load method was done.

Ambient Sound-Based Collaborative Localization of Indeterministic Devices

PubMed Central

Kamminga, Jacob; Le, Duc; Havinga, Paul

2016-01-01

Localization is essential in wireless sensor networks. To our knowledge, no prior work has utilized low-cost devices for collaborative localization based on only ambient sound, without the support of local infrastructure. The reason may be the fact that most low-cost devices are indeterministic and suffer from uncertain input latencies. This uncertainty makes accurate localization challenging. Therefore, we present a collaborative localization algorithm (Cooperative Localization on Android with ambient Sound Sources (CLASS)) that simultaneously localizes the position of indeterministic devices and ambient sound sources without local infrastructure. The CLASS algorithm deals with the uncertainty by splitting the devices into subsets so that outliers can be removed from the time difference of arrival values and localization results. Since Android is indeterministic, we select Android devices to evaluate our approach. The algorithm is evaluated with an outdoor experiment and achieves a mean Root Mean Square Error (RMSE) of 2.18 m with a standard deviation of 0.22 m. Estimated directions towards the sound sources have a mean RMSE of 17.5° and a standard deviation of 2.3°. These results show that it is feasible to simultaneously achieve a relative positioning of both devices and sound sources with sufficient accuracy, even when using non-deterministic devices and platforms, such as Android. PMID:27649176

Amplitude and Wavelength Measurement of Sound Waves in Free Space using a Sound Wave Phase Meter

NASA Astrophysics Data System (ADS)

Ham, Sounggil; Lee, Kiwon

2018-05-01

We developed a sound wave phase meter (SWPM) and measured the amplitude and wavelength of sound waves in free space. The SWPM consists of two parallel metal plates, where the front plate was operated as a diaphragm. An aluminum perforated plate was additionally installed in front of the diaphragm, and the same signal as that applied to the sound source was applied to the perforated plate. The SWPM measures both the sound wave signal due to the diaphragm vibration and the induction signal due to the electric field of the aluminum perforated plate. Therefore, the two measurement signals interfere with each other due to the phase difference according to the distance between the sound source and the SWPM, and the amplitude of the composite signal that is output as a result is periodically changed. We obtained the wavelength of the sound wave from this periodic amplitude change measured in the free space and compared it with the theoretically calculated values.

Investigation of the Statistics of Pure Tone Sound Power Injection from Low Frequency, Finite Sized Sources in a Reverberant Room

NASA Technical Reports Server (NTRS)

Smith, Wayne Farrior

1973-01-01

The effect of finite source size on the power statistics in a reverberant room for pure tone excitation was investigated. Theoretical results indicate that the standard deviation of low frequency, pure tone finite sources is always less than that predicted by point source theory and considerably less when the source dimension approaches one-half an acoustic wavelength or greater. A supporting experimental study was conducted utilizing an eight inch loudspeaker and a 30 inch loudspeaker at eleven source positions. The resulting standard deviation of sound power output of the smaller speaker is in excellent agreement with both the derived finite source theory and existing point source theory, if the theoretical data is adjusted to account for experimental incomplete spatial averaging. However, the standard deviation of sound power output of the larger speaker is measurably lower than point source theory indicates, but is in good agreement with the finite source theory.

Localizing nearby sound sources in a classroom: Binaural room impulse responses

NASA Astrophysics Data System (ADS)

Shinn-Cunningham, Barbara G.; Kopco, Norbert; Martin, Tara J.

2005-05-01

Binaural room impulse responses (BRIRs) were measured in a classroom for sources at different azimuths and distances (up to 1 m) relative to a manikin located in four positions in a classroom. When the listener is far from all walls, reverberant energy distorts signal magnitude and phase independently at each frequency, altering monaural spectral cues, interaural phase differences, and interaural level differences. For the tested conditions, systematic distortion (comb-filtering) from an early intense reflection is only evident when a listener is very close to a wall, and then only in the ear facing the wall. Especially for a nearby source, interaural cues grow less reliable with increasing source laterality and monaural spectral cues are less reliable in the ear farther from the sound source. Reverberation reduces the magnitude of interaural level differences at all frequencies; however, the direct-sound interaural time difference can still be recovered from the BRIRs measured in these experiments. Results suggest that bias and variability in sound localization behavior may vary systematically with listener location in a room as well as source location relative to the listener, even for nearby sources where there is relatively little reverberant energy. .

Localizing nearby sound sources in a classroom: binaural room impulse responses.

PubMed

Shinn-Cunningham, Barbara G; Kopco, Norbert; Martin, Tara J

2005-05-01

Binaural room impulse responses (BRIRs) were measured in a classroom for sources at different azimuths and distances (up to 1 m) relative to a manikin located in four positions in a classroom. When the listener is far from all walls, reverberant energy distorts signal magnitude and phase independently at each frequency, altering monaural spectral cues, interaural phase differences, and interaural level differences. For the tested conditions, systematic distortion (comb-filtering) from an early intense reflection is only evident when a listener is very close to a wall, and then only in the ear facing the wall. Especially for a nearby source, interaural cues grow less reliable with increasing source laterality and monaural spectral cues are less reliable in the ear farther from the sound source. Reverberation reduces the magnitude of interaural level differences at all frequencies; however, the direct-sound interaural time difference can still be recovered from the BRIRs measured in these experiments. Results suggest that bias and variability in sound localization behavior may vary systematically with listener location in a room as well as source location relative to the listener, even for nearby sources where there is relatively little reverberant energy.

Captive Bottlenose Dolphins Do Discriminate Human-Made Sounds Both Underwater and in the Air.

PubMed

Lima, Alice; Sébilleau, Mélissa; Boye, Martin; Durand, Candice; Hausberger, Martine; Lemasson, Alban

2018-01-01

Bottlenose dolphins ( Tursiops truncatus ) spontaneously emit individual acoustic signals that identify them to group members. We tested whether these cetaceans could learn artificial individual sound cues played underwater and whether they would generalize this learning to airborne sounds. Dolphins are thought to perceive only underwater sounds and their training depends largely on visual signals. We investigated the behavioral responses of seven dolphins in a group to learned human-made individual sound cues, played underwater and in the air. Dolphins recognized their own sound cue after hearing it underwater as they immediately moved toward the source, whereas when it was airborne they gazed more at the source of their own sound cue but did not approach it. We hypothesize that they perhaps detected modifications of the sound induced by air or were confused by the novelty of the situation, but nevertheless recognized they were being "targeted." They did not respond when hearing another group member's cue in either situation. This study provides further evidence that dolphins respond to individual-specific sounds and that these marine mammals possess some capacity for processing airborne acoustic signals.

Exploring positive hospital ward soundscape interventions.

PubMed

Mackrill, J; Jennings, P; Cain, R

2014-11-01

Sound is often considered as a negative aspect of an environment that needs mitigating, particularly in hospitals. It is worthwhile however, to consider how subjective responses to hospital sounds can be made more positive. The authors identified natural sound, steady state sound and written sound source information as having the potential to do this. Listening evaluations were conducted with 24 participants who rated their emotional (Relaxation) and cognitive (Interest and Understanding) response to a variety of hospital ward soundscape clips across these three interventions. A repeated measures ANOVA revealed that the 'Relaxation' response was significantly affected (n(2) = 0.05, p = 0.001) by the interventions with natural sound producing a 10.1% more positive response. Most interestingly, written sound source information produced a 4.7% positive change in response. The authors conclude that exploring different ways to improve the sounds of a hospital offers subjective benefits that move beyond sound level reduction. This is an area for future work to focus upon in an effort to achieve more positively experienced hospital soundscapes and environments. Copyright © 2014 Elsevier Ltd and The Ergonomics Society. All rights reserved.

Application of a finite-element model to low-frequency sound insulation in dwellings.

PubMed

Maluski, S P; Gibbs, B M

2000-10-01

The sound transmission between adjacent rooms has been modeled using a finite-element method. Predicted sound-level difference gave good agreement with experimental data using a full-scale and a quarter-scale model. Results show that the sound insulation characteristics of a party wall at low frequencies strongly depend on the modal characteristics of the sound field of both rooms and of the partition. The effect of three edge conditions of the separating wall on the sound-level difference at low frequencies was examined: simply supported, clamped, and a combination of clamped and simply supported. It is demonstrated that a clamped partition provides greater sound-level difference at low frequencies than a simply supported. It also is confirmed that the sound-pressure level difference is lower in equal room than in unequal room configurations.

Effect of Free Jet on Refraction and Noise

NASA Technical Reports Server (NTRS)

Khavaran, Abbas; Georgiadis, Nicholas J.; Bridges, James E.; Dippold, Vance F., III

2005-01-01

This article investigates the role of a free jet on the sound radiated from a jet. In particular, the role of an infinite wind tunnel, which simulates the forward flight condition, is compared to that of a finite wind tunnel. The second configuration is usually used in experiments, where the microphones are located in a static ambient medium far outside the free jet. To study the effect of the free jet on noise, both propagation and source strength need to be addressed. In this work, the exact Green's function in a locally parallel flow is derived for a simulated flight case. Numerical examples are presented that show a reduction in the magnitude of the Green's function in the aft arc and an increase in the forward arc for the simulated flight condition. The effect of finite wind tunnel on refraction is sensitive to the source location and is most pronounced in the aft arc. A Reynolds-averaged Navier-Stokes solution (RANS) yields the required mean flow and turbulence scales that are used in the jet mixing noise spectrum calculations. In addition to the sound/flow interaction, the separate effect of source strength and elongation of the noise-generating region of the jet in a forward flight is studied. Comparisons are made with experiments for the static and finite tunnel cases. Finally, the standard free-jet shear corrections that convert the finite wind tunnel measurements to an ideal wind tunnel arrangement are evaluated.

A precedence effect resolves phantom sound source illusions in the parasitoid fly Ormia ochracea

PubMed Central

Lee, Norman; Elias, Damian O.; Mason, Andrew C.

2009-01-01

Localizing individual sound sources under reverberant environmental conditions can be a challenge when the original source and its acoustic reflections arrive at the ears simultaneously from different paths that convey ambiguous directional information. The acoustic parasitoid fly Ormia ochracea (Diptera: Tachinidae) relies on a pair of ears exquisitely sensitive to sound direction to localize the 5-kHz tone pulsatile calling song of their host crickets. In nature, flies are expected to encounter a complex sound field with multiple sources and their reflections from acoustic clutter potentially masking temporal information relevant to source recognition and localization. In field experiments, O. ochracea were lured onto a test arena and subjected to small random acoustic asymmetries between 2 simultaneous sources. Most flies successfully localize a single source but some localize a ‘phantom’ source that is a summed effect of both source locations. Such misdirected phonotaxis can be elicited reliably in laboratory experiments that present symmetric acoustic stimulation. By varying onset delay between 2 sources, we test whether hyperacute directional hearing in O. ochracea can function to exploit small time differences to determine source location. Selective localization depends on both the relative timing and location of competing sources. Flies preferred phonotaxis to a forward source. With small onset disparities within a 10-ms temporal window of attention, flies selectively localize the leading source while the lagging source has minimal influence on orientation. These results demonstrate the precedence effect as a mechanism to overcome phantom source illusions that arise from acoustic reflections or competing sources. PMID:19332794

Short-term annoyance reactions to stationary and time-varying wind turbine and road traffic noise: A laboratory study.

PubMed

Schäffer, Beat; Schlittmeier, Sabine J; Pieren, Reto; Heutschi, Kurt; Brink, Mark; Graf, Ralf; Hellbrück, Jürgen

2016-05-01

Current literature suggests that wind turbine noise is more annoying than transportation noise. To date, however, it is not known which acoustic characteristics of wind turbines alone, i.e., without effect modifiers such as visibility, are associated with annoyance. The objective of this study was therefore to investigate and compare the short-term noise annoyance reactions to wind turbines and road traffic in controlled laboratory listening tests. A set of acoustic scenarios was created which, combined with the factorial design of the listening tests, allowed separating the individual associations of three acoustic characteristics with annoyance, namely, source type (wind turbine, road traffic), A-weighted sound pressure level, and amplitude modulation (without, periodic, random). Sixty participants rated their annoyance to the sounds. At the same A-weighted sound pressure level, wind turbine noise was found to be associated with higher annoyance than road traffic noise, particularly with amplitude modulation. The increased annoyance to amplitude modulation of wind turbines is not related to its periodicity, but seems to depend on the modulation frequency range. The study discloses a direct link of different acoustic characteristics to annoyance, yet the generalizability to long-term exposure in the field still needs to be verified.

Underwater passive acoustic localization of Pacific walruses in the northeastern Chukchi Sea.

PubMed

Rideout, Brendan P; Dosso, Stan E; Hannay, David E

2013-09-01

This paper develops and applies a linearized Bayesian localization algorithm based on acoustic arrival times of marine mammal vocalizations at spatially-separated receivers which provides three-dimensional (3D) location estimates with rigorous uncertainty analysis. To properly account for uncertainty in receiver parameters (3D hydrophone locations and synchronization times) and environmental parameters (water depth and sound-speed correction), these quantities are treated as unknowns constrained by prior estimates and prior uncertainties. Unknown scaling factors on both the prior and arrival-time uncertainties are estimated by minimizing Akaike's Bayesian information criterion (a maximum entropy condition). Maximum a posteriori estimates for sound source locations and times, receiver parameters, and environmental parameters are calculated simultaneously using measurements of arrival times for direct and interface-reflected acoustic paths. Posterior uncertainties for all unknowns incorporate both arrival time and prior uncertainties. Monte Carlo simulation results demonstrate that, for the cases considered here, linearization errors are small and the lack of an accurate sound-speed profile does not cause significant biases in the estimated locations. A sequence of Pacific walrus vocalizations, recorded in the Chukchi Sea northwest of Alaska, is localized using this technique, yielding a track estimate and uncertainties with an estimated speed comparable to normal walrus swim speeds.

Easily extensible unix software for spectral analysis, display, modification, and synthesis of musical sounds

NASA Astrophysics Data System (ADS)

Beauchamp, James W.

2002-11-01

Software has been developed which enables users to perform time-varying spectral analysis of individual musical tones or successions of them and to perform further processing of the data. The package, called sndan, is freely available in source code, uses EPS graphics for display, and is written in ansi c for ease of code modification and extension. Two analyzers, a fixed-filter-bank phase vocoder (''pvan'') and a frequency-tracking analyzer (''mqan'') constitute the analysis front end of the package. While pvan's output consists of continuous amplitudes and frequencies of harmonics, mqan produces disjoint ''tracks.'' However, another program extracts a fundamental frequency and separates harmonics from the tracks, resulting in a continuous harmonic output. ''monan'' is a program used to display harmonic data in a variety of formats, perform various spectral modifications, and perform additive resynthesis of the harmonic partials, including possible pitch-shifting and time-scaling. Sounds can also be synthesized according to a musical score using a companion synthesis language, Music 4C. Several other programs in the sndan suite can be used for specialized tasks, such as signal display and editing. Applications of the software include producing specialized sounds for music compositions or psychoacoustic experiments or as a basis for developing new synthesis algorithms.

Effects of End CAP and Aspect Ratio on Transmission of Sound across a Truss-Like Periodic Double Panel

NASA Astrophysics Data System (ADS)

EL-RAHEB, M.; WAGNER, P.

2002-02-01

Transmission of sound across 2-D truss-like periodic double panels separated by an air gap and in contact with an acoustic fluid on the external faces is analyzed. Each panel is made of repeated cells. Combining the transfer matrices of the unit cell forms a set of equations for the overall elastic frequency response. The acoustic pressure in the fluids is expressed using a source boundary element method. Adding rigid reflecting end caps confines the air in the gap between panels which influences sound transmission. Measured values of transmission loss differ from the 2-D model by the wide low-frequency dip of the mass-spring-mass or “msm” resonance also termed the “air gap resonance”. In this case, the panels act as rigid masses and the air gap acts as an adiabatic air spring. Results from the idealized 3-D and 2-D models, incorporating rigid cavities and elastic plates, reveal that the “msm” dip is absent in 2-D models radiating into a semi-infinite medium. The dip strengthens as aspect ratio approaches unity. Even when the dip disappears in 2-D, TL rises more steeply for frequencies above the “msm” frequency.

Study of double wall panels for use in propeller driven aircraft

NASA Technical Reports Server (NTRS)

Atwal, M.; Bernhard, R.

1984-01-01

Propeller driven aircraft have exhibited high levels of interior noise. Most absorption materials are not effective at low frequencies where maximum noise levels occur. Two panels separated by an air gap are suggested as an alternative means of noise attenuation. This design produces an impedance mismatch where a sound wave travels backwards to the source. The higher the impedance, the higher the reflected soundwave intensity. Two aluminum panels with helium in between and two panels with one being perforated were investigated. Helium increases the transmission loss because of a greater impedance mismatch than air. The transmission loss of the unperforated panel is higher throughout the frequency range tested.

Study of double wall panels for use in propeller driven aircraft

NASA Astrophysics Data System (ADS)

Atwal, M.; Bernhard, R.

1984-05-01

Propeller driven aircraft have exhibited high levels of interior noise. Most absorption materials are not effective at low frequencies where maximum noise levels occur. Two panels separated by an air gap are suggested as an alternative means of noise attenuation. This design produces an impedance mismatch where a sound wave travels backwards to the source. The higher the impedance, the higher the reflected soundwave intensity. Two aluminum panels with helium in between and two panels with one being perforated were investigated. Helium increases the transmission loss because of a greater impedance mismatch than air. The transmission loss of the unperforated panel is higher throughout the frequency range tested.

Shock waves and the Ffowcs Williams-Hawkings equation

NASA Technical Reports Server (NTRS)

Isom, Morris P.; Yu, Yung H.

1991-01-01

The expansion of the double divergence of the generalized Lighthill stress tensor, which is the basis of the concept of the role played by shock and contact discontinuities as sources of dipole and monopole sound, is presently applied to the simplest transonic flows: (1) a fixed wing in steady motion, for which there is no sound field, and (2) a hovering helicopter blade that produces a sound field. Attention is given to the contribution of the shock to sound from the viewpoint of energy conservation; the shock emerges as the source of only the quantity of entropy.

A study of sound absorption by street canyon boundaries and asphalt rubber concrete pavement

NASA Astrophysics Data System (ADS)

Drysdale, Graeme Robert

A sound field model, based on a classical diffusion equation, is extended to account for sound absorption in a diffusion parameter used to model sound energy in a narrow street canyon. The model accounts for a single sound absorption coefficient, separate accommodation coefficients and a combination of separate absorption and accommodation coefficients from parallel canyon walls. The new expressions are compared to the original formula through numerical simulations to reveal the effect of absorption on sound diffusion. The newly established analytical formulae demonstrate satisfactory agreement with their predecessor under perfect reflection. As well, the influence of the extended diffusion parameter on normalized sound pressure levels in a narrow street canyon is in agreement with experimental data. The diffusion parameters are used to model sound energy density in a street canyon as a function of the sound absorption coefficient of the street canyon walls. The acoustic and material properties of conventional and asphalt rubber concrete (ARC) pavement are also studied to assess how the crumb rubber content influences sound absorption in street canyons. The porosity and absolute permeability of compacted specimens of asphalt rubber concrete are measured and compared to their normal and random incidence sound absorption coefficients as a function of crumb rubber content in the modified binder. Nonlinear trends are found between the sound absorption coefficients, porosity and absolute permeability of the compacted specimens and the percentage of crumb rubber in the modified binders. The cross-sectional areas of the air voids on the surfaces of the compacted specimens are measured using digital image processing techniques and a linear relationship is obtained between the average void area and crumb rubber content. The measured material properties are used to construct an empirical formula relating the average porosity, normal incidence noise reduction coefficients and percentage of crumb rubber in the modified binder of the compacted specimens.

Hearing in three dimensions

NASA Astrophysics Data System (ADS)

Shinn-Cunningham, Barbara

2003-04-01

One of the key functions of hearing is to help us monitor and orient to events in our environment (including those outside the line of sight). The ability to compute the spatial location of a sound source is also important for detecting, identifying, and understanding the content of a sound source, especially in the presence of competing sources from other positions. Determining the spatial location of a sound source poses difficult computational challenges; however, we perform this complex task with proficiency, even in the presence of noise and reverberation. This tutorial will review the acoustic, psychoacoustic, and physiological processes underlying spatial auditory perception. First, the tutorial will examine how the many different features of the acoustic signals reaching a listener's ears provide cues for source direction and distance, both in anechoic and reverberant space. Then we will discuss psychophysical studies of three-dimensional sound localization in different environments and the basic neural mechanisms by which spatial auditory cues are extracted. Finally, ``virtual reality'' approaches for simulating sounds at different directions and distances under headphones will be reviewed. The tutorial will be structured to appeal to a diverse audience with interests in all fields of acoustics and will incorporate concepts from many areas, such as psychological and physiological acoustics, architectural acoustics, and signal processing.

Active control of sound radiation from a vibrating rectangular panel by sound sources and vibration inputs - An experimental comparison

NASA Technical Reports Server (NTRS)

Fuller, C. R.; Hansen, C. H.; Snyder, S. D.

1991-01-01

Active control of sound radiation from a rectangular panel by two different methods has been experimentally studied and compared. In the first method a single control force applied directly to the structure is used with a single error microphone located in the radiated acoustic field. Global attenuation of radiated sound was observed to occur by two main mechanisms. For 'on-resonance' excitation, the control force had the effect of increasing the total panel input impedance presented to the nosie source, thus reducing all radiated sound. For 'off-resonance' excitation, the control force tends not significantly to modify the panel total response amplitude but rather to restructure the relative phases of the modes leading to a more complex vibration pattern and a decrease in radiation efficiency. For acoustic control, the second method, the number of acoustic sources required for global reduction was seen to increase with panel modal order. The mechanism in this case was that the acoustic sources tended to create an inverse pressure distribution at the panel surface and thus 'unload' the panel by reducing the panel radiation impedance. In general, control by structural inputs appears more effective than control by acoustic sources for structurally radiated noise.

Late-Quaternary glaciation and postglacial emergence, southern Eureka Sound, high-Arctic Canada

NASA Astrophysics Data System (ADS)

O Cofaigh, Colm Seamus

Eureka Sound is the inter-island channel separating Ellesmere and Axel Heiberg islands, High Arctic Canada. This thesis reconstructs the glacial and sea level history of southern Eureka Sound through surficial geological mapping, studies of glacial sedimentology and geomorphology, surveying of raised marine shorelines, radiocarbon dating of marine shells and driftwood and surface exposure dating of erratics and bedrock. Granite dispersal trains, shelly till and ice-moulded bedrock record westerly-flow of warm-based, regional ice into Eureka Sound from a source on southeastern Ellesmere Island during the late Wisconsinan. Regional ice was coalescent with local ice domes over Raanes and northern Svendsen peninsulas. Marine limit (dating <=9.2 ka BP; <=9.9 ka cal BP) is inset into the dispersal trains and records early Holocene deglaciation of regional ice. Collectively these data indicate an extensive ice-cover in southern Eureka Sound during the Last Glacial Maximum. Ice-divides were located along the highlands of central Ellesmere and Axel Heiberg islands, from which ice converged on Eureka Sound, and subsequently flowed north and south along the channel. Deglaciation was characterised by a two-step retreat pattern, likely triggered by eustatic sea level rise and abrupt early Holocene warming. Initial break-up and radial retreat of ice in Eureka Sound and the larger fiords, preceded terrestrial stabilisation along coastlines and inner fiords. Location of deglacial depocentres was predominantly controlled by fiord bathymetry. Regionally, two-step deglaciation is reflected by prominent contrasts in glacial geomorphology between the inner and outer parts of many fiords. Glacial sedimentological and geomorphological evidence indicates spatial variation in basal thermal regime between retreating trunk glaciers. Holocene emergence of up to 150 m asl along southern Eureka Sound is recorded by raised marine deltas, beaches and washing limits. Emergence curves exhibit marked contrasts in the form and rate of initial unloading. Isobases drawn on the 8.5 ka shoreline for greater Eureka Sound demonstrate that a cell of highest emergence extends along the length of the channel, and closes in the vicinity of the entrance to Norwegian Bay. The isobase pattern indicates a distinct loading centre over the sound, and in conjunction with glacial geological evidence, suggests that the thickest late Wisconsinan ice lay over the channel.

Aerofoil broadband and tonal noise modelling using stochastic sound sources and incorporated large scale fluctuations

NASA Astrophysics Data System (ADS)

Proskurov, S.; Darbyshire, O. R.; Karabasov, S. A.

2017-12-01

The present work discusses modifications to the stochastic Fast Random Particle Mesh (FRPM) method featuring both tonal and broadband noise sources. The technique relies on the combination of incorporated vortex-shedding resolved flow available from Unsteady Reynolds-Averaged Navier-Stokes (URANS) simulation with the fine-scale turbulence FRPM solution generated via the stochastic velocity fluctuations in the context of vortex sound theory. In contrast to the existing literature, our method encompasses a unified treatment for broadband and tonal acoustic noise sources at the source level, thus, accounting for linear source interference as well as possible non-linear source interaction effects. When sound sources are determined, for the sound propagation, Acoustic Perturbation Equations (APE-4) are solved in the time-domain. Results of the method's application for two aerofoil benchmark cases, with both sharp and blunt trailing edges are presented. In each case, the importance of individual linear and non-linear noise sources was investigated. Several new key features related to the unsteady implementation of the method were tested and brought into the equation. Encouraging results have been obtained for benchmark test cases using the new technique which is believed to be potentially applicable to other airframe noise problems where both tonal and broadband parts are important.

Experimental localization of an acoustic sound source in a wind-tunnel flow by using a numerical time-reversal technique.

PubMed

Padois, Thomas; Prax, Christian; Valeau, Vincent; Marx, David

2012-10-01

The possibility of using the time-reversal technique to localize acoustic sources in a wind-tunnel flow is investigated. While the technique is widespread, it has scarcely been used in aeroacoustics up to now. The proposed method consists of two steps: in a first experimental step, the acoustic pressure fluctuations are recorded over a linear array of microphones; in a second numerical step, the experimental data are time-reversed and used as input data for a numerical code solving the linearized Euler equations. The simulation achieves the back-propagation of the waves from the array to the source and takes into account the effect of the mean flow on sound propagation. The ability of the method to localize a sound source in a typical wind-tunnel flow is first demonstrated using simulated data. A generic experiment is then set up in an anechoic wind tunnel to validate the proposed method with a flow at Mach number 0.11. Monopolar sources are first considered that are either monochromatic or have a narrow or wide-band frequency content. The source position estimation is well-achieved with an error inferior to the wavelength. An application to a dipolar sound source shows that this type of source is also very satisfactorily characterized.

Different categories of living and non-living sound-sources activate distinct cortical networks

PubMed Central

Engel, Lauren R.; Frum, Chris; Puce, Aina; Walker, Nathan A.; Lewis, James W.

2009-01-01

With regard to hearing perception, it remains unclear as to whether, or the extent to which, different conceptual categories of real-world sounds and related categorical knowledge are differentially represented in the brain. Semantic knowledge representations are reported to include the major divisions of living versus non-living things, plus more specific categories including animals, tools, biological motion, faces, and places—categories typically defined by their characteristic visual features. Here, we used functional magnetic resonance imaging (fMRI) to identify brain regions showing preferential activity to four categories of action sounds, which included non-vocal human and animal actions (living), plus mechanical and environmental sound-producing actions (non-living). The results showed a striking antero-posterior division in cortical representations for sounds produced by living versus non-living sources. Additionally, there were several significant differences by category, depending on whether the task was category-specific (e.g. human or not) versus non-specific (detect end-of-sound). In general, (1) human-produced sounds yielded robust activation in the bilateral posterior superior temporal sulci independent of task. Task demands modulated activation of left-lateralized fronto-parietal regions, bilateral insular cortices, and subcortical regions previously implicated in observation-execution matching, consistent with “embodied” and mirror-neuron network representations subserving recognition. (2) Animal action sounds preferentially activated the bilateral posterior insulae. (3) Mechanical sounds activated the anterior superior temporal gyri and parahippocampal cortices. (4) Environmental sounds preferentially activated dorsal occipital and medial parietal cortices. Overall, this multi-level dissociation of networks for preferentially representing distinct sound-source categories provides novel support for grounded cognition models that may underlie organizational principles for hearing perception. PMID:19465134

[Perception of approaching and withdrawing sound sources following exposure to broadband noise. The effect of spatial domain].

PubMed

Malinina, E S

2014-01-01

The spatial specificity of auditory aftereffect was studied after a short-time adaptation (5 s) to the broadband noise (20-20000 Hz). Adapting stimuli were sequences of noise impulses with the constant amplitude, test stimuli--with the constant and changing amplitude: an increase of amplitude of impulses in sequence was perceived by listeners as approach of the sound source, while a decrease of amplitude--as its withdrawal. The experiments were performed in an anechoic chamber. The auditory aftereffect was estimated under the following conditions: the adapting and test stimuli were presented from the loudspeaker located at a distance of 1.1 m from the listeners (the subjectively near spatial domain) or 4.5 m from the listeners (the subjectively near spatial domain) or 4.5 m from the listeners (the subjectively far spatial domain); the adapting and test stimuli were presented from different distances. The obtained data showed that perception of the imitated movement of the sound source in both spatial domains had the common characteristic peculiarities that manifested themselves both under control conditions without adaptation and after adaptation to noise. In the absence of adaptation for both distances, an asymmetry of psychophysical curves was observed: the listeners estimated the test stimuli more often as approaching. The overestimation by listeners of test stimuli as the approaching ones was more pronounced at their presentation from the distance of 1.1 m, i. e., from the subjectively near spatial domain. After adaptation to noise the aftereffects showed spatial specificity in both spatial domains: they were observed only at the spatial coincidence of adapting and test stimuli and were absent at their separation. The aftereffects observed in two spatial domains were similar in direction and value: the listeners estimated the test stimuli more often as withdrawing as compared to control. The result of such aftereffect was restoration of the symmetry of psychometric curves and of the equiprobable estimation of direction of movement of test signals.

The directivity of the sound radiation from panels and openings.

PubMed

Davy, John L

2009-06-01

This paper presents a method for calculating the directivity of the radiation of sound from a panel or opening, whose vibration is forced by the incidence of sound from the other side. The directivity of the radiation depends on the angular distribution of the incident sound energy in the room or duct in whose wall or end the panel or opening occurs. The angular distribution of the incident sound energy is predicted using a model which depends on the sound absorption coefficient of the room or duct surfaces. If the sound source is situated in the room or duct, the sound absorption coefficient model is used in conjunction with a model for the directivity of the sound source. For angles of radiation approaching 90 degrees to the normal to the panel or opening, the effect of the diffraction by the panel or opening, or by the finite baffle in which the panel or opening is mounted, is included. A simple empirical model is developed to predict the diffraction of sound into the shadow zone when the angle of radiation is greater than 90 degrees to the normal to the panel or opening. The method is compared with published experimental results.

A Tool for Low Noise Procedures Design and Community Noise Impact Assessment: The Rotorcraft Noise Model (RNM)

NASA Technical Reports Server (NTRS)

Conner, David A.; Page, Juliet A.

2002-01-01

To improve aircraft noise impact modeling capabilities and to provide a tool to aid in the development of low noise terminal area operations for rotorcraft and tiltrotors, the Rotorcraft Noise Model (RNM) was developed by the NASA Langley Research Center and Wyle Laboratories. RNM is a simulation program that predicts how sound will propagate through the atmosphere and accumulate at receiver locations located on flat ground or varying terrain, for single and multiple vehicle flight operations. At the core of RNM are the vehicle noise sources, input as sound hemispheres. As the vehicle "flies" along its prescribed flight trajectory, the source sound propagation is simulated and accumulated at the receiver locations (single points of interest or multiple grid points) in a systematic time-based manner. These sound signals at the receiver locations may then be analyzed to obtain single event footprints, integrated noise contours, time histories, or numerous other features. RNM may also be used to generate spectral time history data over a ground mesh for the creation of single event sound animation videos. Acoustic properties of the noise source(s) are defined in terms of sound hemispheres that may be obtained from theoretical predictions, wind tunnel experimental results, flight test measurements, or a combination of the three. The sound hemispheres may contain broadband data (source levels as a function of one-third octave band) and pure-tone data (in the form of specific frequency sound pressure levels and phase). A PC executable version of RNM is publicly available and has been adopted by a number of organizations for Environmental Impact Assessment studies of rotorcraft noise. This paper provides a review of the required input data, the theoretical framework of RNM's propagation model and the output results. Code validation results are provided from a NATO helicopter noise flight test as well as a tiltrotor flight test program that used the RNM as a tool to aid in the development of low noise approach profiles.

Discrete-frequency and broadband noise radiation from diesel engine cooling fans

NASA Astrophysics Data System (ADS)

Kim, Geon-Seok

This effort focuses on measuring and predicting the discrete-frequency and broadband noise radiated by diesel engine cooling fans. Unsteady forces developed by the interaction of the fan blade with inlet flow are the dominant source for both discrete-frequency and broadband noise of the subject propeller fan. In many cases, a primary source of discrepancy between fan noise prediction and measurement is due to incomplete description of the fan inflow. Particularly, in such engine cooling systems where space is very limited, it would be very difficult, if not, impossible to measure the fan inflow velocity field using the conventional, stationary hot-wire method. Instead, the fan inflow was measured with two-component x-type hot-film probes attached very close to the leading edge of a rotating blade. One of the advantages of the blade-mounted-probe measurement technique is that it measures velocities relative to the rotating probe, which enables the acquired data to be applied directly in many aerodynamic theories that have been developed for the airfoil fixed-coordinate system. However, the velocity time data measured by this technique contains the spatially non-uniform mean velocity field along with the temporal fluctuations. A phase-locked averaging technique was successfully employed to decompose the velocity data into time-invariant flow distortions and fluctuations due to turbulence. The angles of attack of the fan blades, obtained from inlet flow measurements, indicate that the blades are stalled. The fan's radiated noise was measured without contamination from the engine noise by driving the fan with an electric motor. The motor operated at a constant speed while a pair of speed controllable pulleys controlled the fan speed. Narrowband and 1/3-octave band sound power of the cooling fan was measured by using the comparison method with a reference sound source in a reverberant room. The spatially non-uniform mean velocity field was used in axial-flow fan noise theory to predict the discrete-frequency noise at the blade passing frequency (BPF) and harmonics. The unsteady lift was predicted by considering transverse and longitudinal velocity fluctuations. The influences due to an upstream finger guard were also investigated. The radiated sound power spectra that were measured for the fan are shown to be in excellent agreement with those predicted. The agreement between prediction and measurement is only fair at the fundamental BPF tone. Further experimental investigations revealed that the interaction noise between the fan blades and a shroud surrounding the fan might be the dominant source for the radiation at the first harmonic. The space-time correlation functions of the inflow velocity fluctuations were measured and utilized in stochastic lifting surface theory to calculate the unsteady blade lift and resulting broadband fan noise. The integral length scale of the inlet flow was found to be much smaller than the blade-to-blade separate distance of the fan. Therefore, contributions from blade-to-blade correlations of the various elements on different blades were found to be negligible and hence ignored; only the correlations between the strip elements on a given blade were considered. The cross-correlations measured between elements separated by more than the integral length scale were also found to be negligibly small. The predicted broadband sound power spectra agree well with those measured for frequencies greater than 400 Hz. There are deviations between the predictions and measurements at lower frequencies. These are likely due to fan blade stall, and possibly, anomalies in the noise measurement environment. In order to reduce the sound radiation at the blade rate tones, the baseline fan was replaced with a skewed fan. The backward skew angle of 30° was found to effectively reduce the 2nd and higher harmonics of the blade rate tone. The interaction of the shroud opening and the blade tips dominates the sound level at the fundamental tone. This tone was successfully reduced by incorporating a serrated shroud opening. Overall, a 2.8 dB sound power level reduction was achieved by applying the skewed fan and the serrated shroud opening simultaneously. Almost all blade rate tone levels were reduced without adversely affecting the flow performance of the system.

Spatial hearing in Cope’s gray treefrog: I. Open and closed loop experiments on sound localization in the presence and absence of noise

PubMed Central

Caldwell, Michael S.; Bee, Mark A.

2014-01-01

The ability to reliably locate sound sources is critical to anurans, which navigate acoustically complex breeding choruses when choosing mates. Yet, the factors influencing sound localization performance in frogs remain largely unexplored. We applied two complementary methodologies, open and closed loop playback trials, to identify influences on localization abilities in Cope’s gray treefrog, Hyla chrysoscelis. We examined localization acuity and phonotaxis behavior of females in response to advertisement calls presented from 12 azimuthal angles, at two signal levels, in the presence and absence of noise, and at two noise levels. Orientation responses were consistent with precise localization of sound sources, rather than binary discrimination between sources on either side of the body (lateralization). Frogs were unable to discriminate between sounds arriving from forward and rearward directions, and accurate localization was limited to forward sound presentation angles. Within this region, sound presentation angle had little effect on localization acuity. The presence of noise and low signal-to-noise ratios also did not strongly impair localization ability in open loop trials, but females exhibited reduced phonotaxis performance consistent with impaired localization during closed loop trials. We discuss these results in light of previous work on spatial hearing in anurans. PMID:24504182

Sex differences present in auditory looming perception, absent in auditory recession

NASA Astrophysics Data System (ADS)

Neuhoff, John G.; Seifritz, Erich

2005-04-01

When predicting the arrival time of an approaching sound source, listeners typically exhibit an anticipatory bias that affords a margin of safety in dealing with looming objects. The looming bias has been demonstrated behaviorally in the laboratory and in the field (Neuhoff 1998, 2001), neurally in fMRI studies (Seifritz et al., 2002), and comparatively in non-human primates (Ghazanfar, Neuhoff, and Logothetis, 2002). In the current work, male and female listeners were presented with three-dimensional looming sound sources and asked to press a button when the source was at the point of closest approach. Females exhibited a significantly greater anticipatory bias than males. Next, listeners were presented with sounds that either approached or receded and then stopped at three different terminal distances. Consistent with the time-to-arrival judgments, female terminal distance judgments for looming sources were significantly closer than male judgments. However, there was no difference between male and female terminal distance judgments for receding sounds. Taken together with the converging behavioral, neural, and comparative evidence, the current results illustrate the environmental salience of looming sounds and suggest that the anticipatory bias for auditory looming may have been shaped by evolution to provide a selective advantage in dealing with looming objects.

Understanding auditory distance estimation by humpback whales: a computational approach.

PubMed

Mercado, E; Green, S R; Schneider, J N

2008-02-01

Ranging, the ability to judge the distance to a sound source, depends on the presence of predictable patterns of attenuation. We measured long-range sound propagation in coastal waters to assess whether humpback whales might use frequency degradation cues to range singing whales. Two types of neural networks, a multi-layer and a single-layer perceptron, were trained to classify recorded sounds by distance traveled based on their frequency content. The multi-layer network successfully classified received sounds, demonstrating that the distorting effects of underwater propagation on frequency content provide sufficient cues to estimate source distance. Normalizing received sounds with respect to ambient noise levels increased the accuracy of distance estimates by single-layer perceptrons, indicating that familiarity with background noise can potentially improve a listening whale's ability to range. To assess whether frequency patterns predictive of source distance were likely to be perceived by whales, recordings were pre-processed using a computational model of the humpback whale's peripheral auditory system. Although signals processed with this model contained less information than the original recordings, neural networks trained with these physiologically based representations estimated source distance more accurately, suggesting that listening whales should be able to range singers using distance-dependent changes in frequency content.

Community Response to Multiple Sound Sources: Integrating Acoustic and Contextual Approaches in the Analysis

PubMed Central

Lercher, Peter; De Coensel, Bert; Dekonink, Luc; Botteldooren, Dick

2017-01-01

Sufficient data refer to the relevant prevalence of sound exposure by mixed traffic sources in many nations. Furthermore, consideration of the potential effects of combined sound exposure is required in legal procedures such as environmental health impact assessments. Nevertheless, current practice still uses single exposure response functions. It is silently assumed that those standard exposure-response curves accommodate also for mixed exposures—although some evidence from experimental and field studies casts doubt on this practice. The ALPNAP-study population (N = 1641) shows sufficient subgroups with combinations of rail-highway, highway-main road and rail-highway-main road sound exposure. In this paper we apply a few suggested approaches of the literature to investigate exposure-response curves and its major determinants in the case of exposure to multiple traffic sources. Highly/moderate annoyance and full scale mean annoyance served as outcome. The results show several limitations of the current approaches. Even facing the inherent methodological limitations (energy equivalent summation of sound, rating of overall annoyance) the consideration of main contextual factors jointly occurring with the sources (such as vibration, air pollution) or coping activities and judgments of the wider area soundscape increases the variance explanation from up to 8% (bivariate), up to 15% (base adjustments) up to 55% (full contextual model). The added predictors vary significantly, depending on the source combination. (e.g., significant vibration effects with main road/railway, not highway). Although no significant interactions were found, the observed additive effects are of public health importance. Especially in the case of a three source exposure situation the overall annoyance is already high at lower levels and the contribution of the acoustic indicators is small compared with the non-acoustic and contextual predictors. Noise mapping needs to go down to levels of 40 dBA,Lden to ensure the protection of quiet areas and prohibit the silent “filling up” of these areas with new sound sources. Eventually, to better predict the annoyance in the exposure range between 40 and 60 dBA and support the protection of quiet areas in city and rural areas in planning sound indicators need to be oriented at the noticeability of sound and consider other traffic related by-products (air quality, vibration, coping strain) in future studies and environmental impact assessments. PMID:28632198

A possible approach to optimization of parameters of sound-absorbing structures for multimode waveguides

NASA Astrophysics Data System (ADS)

Mironov, M. A.

2011-11-01

A method of allowing for the spatial sound field structure in designing the sound-absorbing structures for turbojet aircraft engine ducts is proposed. The acoustic impedance of a duct should be chosen so as to prevent the reflection of the primary sound field, which is generated by the sound source in the absence of the duct, from the duct walls.

Quantifying the influence of flow asymmetries on glottal sound sources in speech

NASA Astrophysics Data System (ADS)

Erath, Byron; Plesniak, Michael

2008-11-01

Human speech is made possible by the air flow interaction with the vocal folds. During phonation, asymmetries in the glottal flow field may arise from flow phenomena (e.g. the Coanda effect) as well as from pathological vocal fold motion (e.g. unilateral paralysis). In this study, the effects of flow asymmetries on glottal sound sources were investigated. Dynamically-programmable 7.5 times life-size vocal fold models with 2 degrees-of-freedom (linear and rotational) were constructed to provide a first-order approximation of vocal fold motion. Important parameters (Reynolds, Strouhal, and Euler numbers) were scaled to physiological values. Normal and abnormal vocal fold motions were synthesized, and the velocity field and instantaneous transglottal pressure drop were measured. Variability in the glottal jet trajectory necessitated sorting of the data according to the resulting flow configuration. The dipole sound source is related to the transglottal pressure drop via acoustic analogies. Variations in the transglottal pressure drop (and subsequently the dipole sound source) arising from flow asymmetries are discussed.

Psychophysical evidence for auditory motion parallax.

PubMed

Genzel, Daria; Schutte, Michael; Brimijoin, W Owen; MacNeilage, Paul R; Wiegrebe, Lutz

2018-04-17

Distance is important: From an ecological perspective, knowledge about the distance to either prey or predator is vital. However, the distance of an unknown sound source is particularly difficult to assess, especially in anechoic environments. In vision, changes in perspective resulting from observer motion produce a reliable, consistent, and unambiguous impression of depth known as motion parallax. Here we demonstrate with formal psychophysics that humans can exploit auditory motion parallax, i.e., the change in the dynamic binaural cues elicited by self-motion, to assess the relative depths of two sound sources. Our data show that sensitivity to relative depth is best when subjects move actively; performance deteriorates when subjects are moved by a motion platform or when the sound sources themselves move. This is true even though the dynamic binaural cues elicited by these three types of motion are identical. Our data demonstrate a perceptual strategy to segregate intermittent sound sources in depth and highlight the tight interaction between self-motion and binaural processing that allows assessment of the spatial layout of complex acoustic scenes.

Auditory event perception: the source-perception loop for posture in human gait.

PubMed

Pastore, Richard E; Flint, Jesse D; Gaston, Jeremy R; Solomon, Matthew J

2008-01-01

There is a small but growing literature on the perception of natural acoustic events, but few attempts have been made to investigate complex sounds not systematically controlled within a laboratory setting. The present study investigates listeners' ability to make judgments about the posture (upright-stooped) of the walker who generated acoustic stimuli contrasted on each trial. We use a comprehensive three-stage approach to event perception, in which we develop a solid understanding of the source event and its sound properties, as well as the relationships between these two event stages. Developing this understanding helps both to identify the limitations of common statistical procedures and to develop effective new procedures for investigating not only the two information stages above, but also the decision strategies employed by listeners in making source judgments from sound. The result is a comprehensive, ultimately logical, but not necessarily expected picture of both the source-sound-perception loop and the utility of alternative research tools.

Nonlinear theory of shocked sound propagation in a nearly choked duct flow

NASA Technical Reports Server (NTRS)

Myers, M. K.; Callegari, A. J.

1982-01-01

The development of shocks in the sound field propagating through a nearly choked duct flow is analyzed by extending a quasi-one dimensional theory. The theory is applied to the case in which sound is introduced into the flow by an acoustic source located in the vicinity of a near-sonic throat. Analytical solutions for the field are obtained which illustrate the essential features of the nonlinear interaction between sound and flow. Numerical results are presented covering ranges of variation of source strength, throat Mach number, and frequency. It is found that the development of shocks leads to appreciable attenuation of acoustic power transmitted upstream through the near-sonic flow. It is possible, for example, that the power loss in the fundamental harmonic can be as much as 90% of that introduced at the source.

Noise abatement in a pine plantation

Treesearch

R. E. Leonard; L. P. Herrington

1971-01-01

Observations on sound propagation were made in two red pine plantations. Measurements were taken of attenuation of prerecorded frequencies at various distances from the sound source. Sound absorption was strongly dependent on frequencies. Peak absorption was at 500 Hz.

Hearing in three dimensions: Sound localization

NASA Technical Reports Server (NTRS)

Wightman, Frederic L.; Kistler, Doris J.

1990-01-01

The ability to localize a source of sound in space is a fundamental component of the three dimensional character of the sound of audio. For over a century scientists have been trying to understand the physical and psychological processes and physiological mechanisms that subserve sound localization. This research has shown that important information about sound source position is provided by interaural differences in time of arrival, interaural differences in intensity and direction-dependent filtering provided by the pinnae. Progress has been slow, primarily because experiments on localization are technically demanding. Control of stimulus parameters and quantification of the subjective experience are quite difficult problems. Recent advances, such as the ability to simulate a three dimensional sound field over headphones, seem to offer potential for rapid progress. Research using the new techniques has already produced new information. It now seems that interaural time differences are a much more salient and dominant localization cue than previously believed.

Using sounds for making decisions: greater tube-nosed bats prefer antagonistic calls over non-communicative sounds when feeding

PubMed Central

Jiang, Tinglei; Long, Zhenyu; Ran, Xin; Zhao, Xue; Xu, Fei; Qiu, Fuyuan; Kanwal, Jagmeet S.

2016-01-01

ABSTRACT Bats vocalize extensively within different social contexts. The type and extent of information conveyed via their vocalizations and their perceptual significance, however, remains controversial and difficult to assess. Greater tube-nosed bats, Murina leucogaster, emit calls consisting of long rectangular broadband noise burst (rBNBl) syllables during aggression between males. To experimentally test the behavioral impact of these sounds for feeding, we deployed an approach and place-preference paradigm. Two food trays were placed on opposite sides and within different acoustic microenvironments, created by sound playback, within a specially constructed tent. Specifically, we tested whether the presence of rBNBl sounds at a food source effectively deters the approach of male bats in comparison to echolocation sounds and white noise. In each case, contrary to our expectation, males preferred to feed at a location where rBNBl sounds were present. We propose that the species-specific rBNBl provides contextual information, not present within non-communicative sounds, to facilitate approach towards a food source. PMID:27815241

What the Toadfish Ear Tells the Toadfish Brain About Sound.

PubMed

Edds-Walton, Peggy L

2016-01-01

Of the three, paired otolithic endorgans in the ear of teleost fishes, the saccule is the one most often demonstrated to have a major role in encoding frequencies of biologically relevant sounds. The toadfish saccule also encodes sound level and sound source direction in the phase-locked activity conveyed via auditory afferents to nuclei of the ipsilateral octaval column in the medulla. Although paired auditory receptors are present in teleost fishes, binaural processes were believed to be unimportant due to the speed of sound in water and the acoustic transparency of the tissues in water. In contrast, there are behavioral and anatomical data that support binaural processing in fishes. Studies in the toadfish combined anatomical tract-tracing and physiological recordings from identified sites along the ascending auditory pathway to document response characteristics at each level. Binaural computations in the medulla and midbrain sharpen the directional information provided by the saccule. Furthermore, physiological studies in the central nervous system indicated that encoding frequency, sound level, temporal pattern, and sound source direction are important components of what the toadfish ear tells the toadfish brain about sound.

Replacing the Orchestra? – The Discernibility of Sample Library and Live Orchestra Sounds

PubMed Central

Wolf, Anna; Platz, Friedrich; Mons, Jan

2016-01-01

Recently, musical sounds from pre-recorded orchestra sample libraries (OSL) have become indispensable in music production for the stage or popular charts. Surprisingly, it is unknown whether human listeners can identify sounds as stemming from real orchestras or OSLs. Thus, an internet-based experiment was conducted to investigate whether a classic orchestral work, produced with sounds from a state-of-the-art OSL, could be reliably discerned from a live orchestra recording of the piece. It could be shown that the entire sample of listeners (N = 602) on average identified the correct sound source at 72.5%. This rate slightly exceeded Alan Turing's well-known upper threshold of 70% for a convincing, simulated performance. However, while sound experts tended to correctly identify the sound source, participants with lower listening expertise, who resembled the majority of music consumers, only achieved 68.6%. As non-expert listeners in the experiment were virtually unable to tell the real-life and OSL sounds apart, it is assumed that OSLs will become more common in music production for economic reasons. PMID:27382932

The Coast Artillery Journal. Volume 65, Number 4, October 1926

DTIC Science & Technology

1926-10-01

sound. a. Sound location of airplanes by binaural observation in all antiaircraft regiments. b. Sound ranging on report of enemy guns, together with...Direction finding by binaural observation. [Subparagraphs 30 a and 30 c (l).J This applies to continuous sounds such as pro- pellor noises. b. Point...impacts. 32. The so-called binaural sense is our means of sensing the direc- tion of a sound source. When we hear a sound we judge the approxi- mate

Object localization using a biosonar beam: how opening your mouth improves localization.

PubMed

Arditi, G; Weiss, A J; Yovel, Y

2015-08-01

Determining the location of a sound source is crucial for survival. Both predators and prey usually produce sound while moving, revealing valuable information about their presence and location. Animals have thus evolved morphological and neural adaptations allowing precise sound localization. Mammals rely on the temporal and amplitude differences between the sound signals arriving at their two ears, as well as on the spectral cues available in the signal arriving at a single ear to localize a sound source. Most mammals rely on passive hearing and are thus limited by the acoustic characteristics of the emitted sound. Echolocating bats emit sound to perceive their environment. They can, therefore, affect the frequency spectrum of the echoes they must localize. The biosonar sound beam of a bat is directional, spreading different frequencies into different directions. Here, we analyse mathematically the spatial information that is provided by the beam and could be used to improve sound localization. We hypothesize how bats could improve sound localization by altering their echolocation signal design or by increasing their mouth gape (the size of the sound emitter) as they, indeed, do in nature. Finally, we also reveal a trade-off according to which increasing the echolocation signal's frequency improves the accuracy of sound localization but might result in undesired large localization errors under low signal-to-noise ratio conditions.

Object localization using a biosonar beam: how opening your mouth improves localization

PubMed Central

Arditi, G.; Weiss, A. J.; Yovel, Y.

2015-01-01

Determining the location of a sound source is crucial for survival. Both predators and prey usually produce sound while moving, revealing valuable information about their presence and location. Animals have thus evolved morphological and neural adaptations allowing precise sound localization. Mammals rely on the temporal and amplitude differences between the sound signals arriving at their two ears, as well as on the spectral cues available in the signal arriving at a single ear to localize a sound source. Most mammals rely on passive hearing and are thus limited by the acoustic characteristics of the emitted sound. Echolocating bats emit sound to perceive their environment. They can, therefore, affect the frequency spectrum of the echoes they must localize. The biosonar sound beam of a bat is directional, spreading different frequencies into different directions. Here, we analyse mathematically the spatial information that is provided by the beam and could be used to improve sound localization. We hypothesize how bats could improve sound localization by altering their echolocation signal design or by increasing their mouth gape (the size of the sound emitter) as they, indeed, do in nature. Finally, we also reveal a trade-off according to which increasing the echolocation signal's frequency improves the accuracy of sound localization but might result in undesired large localization errors under low signal-to-noise ratio conditions. PMID:26361552

Hemispherical breathing mode speaker using a dielectric elastomer actuator.

PubMed

Hosoya, Naoki; Baba, Shun; Maeda, Shingo

2015-10-01

Although indoor acoustic characteristics should ideally be assessed by measuring the reverberation time using a point sound source, a regular polyhedron loudspeaker, which has multiple loudspeakers on a chassis, is typically used. However, such a configuration is not a point sound source if the size of the loudspeaker is large relative to the target sound field. This study investigates a small lightweight loudspeaker using a dielectric elastomer actuator vibrating in the breathing mode (the pulsating mode such as the expansion and contraction of a balloon). Acoustic testing with regard to repeatability, sound pressure, vibration mode profiles, and acoustic radiation patterns indicate that dielectric elastomer loudspeakers may be feasible.

The role of reverberation-related binaural cues in the externalization of speech.

PubMed

Catic, Jasmina; Santurette, Sébastien; Dau, Torsten

2015-08-01

The perception of externalization of speech sounds was investigated with respect to the monaural and binaural cues available at the listeners' ears in a reverberant environment. Individualized binaural room impulse responses (BRIRs) were used to simulate externalized sound sources via headphones. The measured BRIRs were subsequently modified such that the proportion of the response containing binaural vs monaural information was varied. Normal-hearing listeners were presented with speech sounds convolved with such modified BRIRs. Monaural reverberation cues were found to be sufficient for the externalization of a lateral sound source. In contrast, for a frontal source, an increased amount of binaural cues from reflections was required in order to obtain well externalized sound images. It was demonstrated that the interaction between the interaural cues of the direct sound and the reverberation strongly affects the perception of externalization. An analysis of the short-term binaural cues showed that the amount of fluctuations of the binaural cues corresponded well to the externalization ratings obtained in the listening tests. The results further suggested that the precedence effect is involved in the auditory processing of the dynamic binaural cues that are utilized for externalization perception.

A stepped-plate bi-frequency source for generating a difference frequency sound with a parametric array.

PubMed

Je, Yub; Lee, Haksue; Park, Jongkyu; Moon, Wonkyu

2010-06-01

An ultrasonic radiator is developed to generate a difference frequency sound from two frequencies of ultrasound in air with a parametric array. A design method is proposed for an ultrasonic radiator capable of generating highly directive, high-amplitude ultrasonic sound beams at two different frequencies in air based on a modification of the stepped-plate ultrasonic radiator. The stepped-plate ultrasonic radiator was introduced by Gallego-Juarez et al. [Ultrasonics 16, 267-271 (1978)] in their previous study and can effectively generate highly directive, large-amplitude ultrasonic sounds in air, but only at a single frequency. Because parametric array sources must be able to generate sounds at more than one frequency, a design modification is crucial to the application of a stepped-plate ultrasonic radiator as a parametric array source in air. The aforementioned method was employed to design a parametric radiator for use in air. A prototype of this design was constructed and tested to determine whether it could successfully generate a difference frequency sound with a parametric array. The results confirmed that the proposed single small-area transducer was suitable as a parametric radiator in air.

An open access database for the evaluation of heart sound algorithms.

PubMed

Liu, Chengyu; Springer, David; Li, Qiao; Moody, Benjamin; Juan, Ricardo Abad; Chorro, Francisco J; Castells, Francisco; Roig, José Millet; Silva, Ikaro; Johnson, Alistair E W; Syed, Zeeshan; Schmidt, Samuel E; Papadaniil, Chrysa D; Hadjileontiadis, Leontios; Naseri, Hosein; Moukadem, Ali; Dieterlen, Alain; Brandt, Christian; Tang, Hong; Samieinasab, Maryam; Samieinasab, Mohammad Reza; Sameni, Reza; Mark, Roger G; Clifford, Gari D

2016-12-01

In the past few decades, analysis of heart sound signals (i.e. the phonocardiogram or PCG), especially for automated heart sound segmentation and classification, has been widely studied and has been reported to have the potential value to detect pathology accurately in clinical applications. However, comparative analyses of algorithms in the literature have been hindered by the lack of high-quality, rigorously validated, and standardized open databases of heart sound recordings. This paper describes a public heart sound database, assembled for an international competition, the PhysioNet/Computing in Cardiology (CinC) Challenge 2016. The archive comprises nine different heart sound databases sourced from multiple research groups around the world. It includes 2435 heart sound recordings in total collected from 1297 healthy subjects and patients with a variety of conditions, including heart valve disease and coronary artery disease. The recordings were collected from a variety of clinical or nonclinical (such as in-home visits) environments and equipment. The length of recording varied from several seconds to several minutes. This article reports detailed information about the subjects/patients including demographics (number, age, gender), recordings (number, location, state and time length), associated synchronously recorded signals, sampling frequency and sensor type used. We also provide a brief summary of the commonly used heart sound segmentation and classification methods, including open source code provided concurrently for the Challenge. A description of the PhysioNet/CinC Challenge 2016, including the main aims, the training and test sets, the hand corrected annotations for different heart sound states, the scoring mechanism, and associated open source code are provided. In addition, several potential benefits from the public heart sound database are discussed.

Neuromagnetic recordings reveal the temporal dynamics of auditory spatial processing in the human cortex.

PubMed

Tiitinen, Hannu; Salminen, Nelli H; Palomäki, Kalle J; Mäkinen, Ville T; Alku, Paavo; May, Patrick J C

2006-03-20

In an attempt to delineate the assumed 'what' and 'where' processing streams, we studied the processing of spatial sound in the human cortex by using magnetoencephalography in the passive and active recording conditions and two kinds of spatial stimuli: individually constructed, highly realistic spatial (3D) stimuli and stimuli containing interaural time difference (ITD) cues only. The auditory P1m, N1m, and P2m responses of the event-related field were found to be sensitive to the direction of sound source in the azimuthal plane. In general, the right-hemispheric responses to spatial sounds were more prominent than the left-hemispheric ones. The right-hemispheric P1m and N1m responses peaked earlier for sound sources in the contralateral than for sources in the ipsilateral hemifield and the peak amplitudes of all responses reached their maxima for contralateral sound sources. The amplitude of the right-hemispheric P2m response reflected the degree of spatiality of sound, being twice as large for the 3D than ITD stimuli. The results indicate that the right hemisphere is specialized in the processing of spatial cues in the passive recording condition. Minimum current estimate (MCE) localization revealed that temporal areas were activated both in the active and passive condition. This initial activation, taking place at around 100 ms, was followed by parietal and frontal activity at 180 and 200 ms, respectively. The latter activations, however, were specific to attentional engagement and motor responding. This suggests that parietal activation reflects active responding to a spatial sound rather than auditory spatial processing as such.

Sound Source Localization through 8 MEMS Microphones Array Using a Sand-Scorpion-Inspired Spiking Neural Network.

PubMed

Beck, Christoph; Garreau, Guillaume; Georgiou, Julius

2016-01-01

Sand-scorpions and many other arachnids perceive their environment by using their feet to sense ground waves. They are able to determine amplitudes the size of an atom and locate the acoustic stimuli with an accuracy of within 13° based on their neuronal anatomy. We present here a prototype sound source localization system, inspired from this impressive performance. The system presented utilizes custom-built hardware with eight MEMS microphones, one for each foot, to acquire the acoustic scene, and a spiking neural model to localize the sound source. The current implementation shows smaller localization error than those observed in nature.

Captive Bottlenose Dolphins Do Discriminate Human-Made Sounds Both Underwater and in the Air

PubMed Central

Lima, Alice; Sébilleau, Mélissa; Boye, Martin; Durand, Candice; Hausberger, Martine; Lemasson, Alban

2018-01-01

Bottlenose dolphins (Tursiops truncatus) spontaneously emit individual acoustic signals that identify them to group members. We tested whether these cetaceans could learn artificial individual sound cues played underwater and whether they would generalize this learning to airborne sounds. Dolphins are thought to perceive only underwater sounds and their training depends largely on visual signals. We investigated the behavioral responses of seven dolphins in a group to learned human-made individual sound cues, played underwater and in the air. Dolphins recognized their own sound cue after hearing it underwater as they immediately moved toward the source, whereas when it was airborne they gazed more at the source of their own sound cue but did not approach it. We hypothesize that they perhaps detected modifications of the sound induced by air or were confused by the novelty of the situation, but nevertheless recognized they were being “targeted.” They did not respond when hearing another group member’s cue in either situation. This study provides further evidence that dolphins respond to individual-specific sounds and that these marine mammals possess some capacity for processing airborne acoustic signals. PMID:29445350

The influence of crowd density on the sound environment of commercial pedestrian streets.

PubMed

Meng, Qi; Kang, Jian

2015-04-01

Commercial pedestrian streets are very common in China and Europe, with many situated in historic or cultural centres. The environments of these streets are important, including their sound environments. The objective of this study is to explore the relationships between the crowd density and the sound environments of commercial pedestrian streets. On-site measurements were performed at the case study site in Harbin, China, and a questionnaire was administered. The sound pressure measurements showed that the crowd density has an insignificant effect on sound pressure below 0.05 persons/m2, whereas when the crowd density is greater than 0.05 persons/m2, the sound pressure increases with crowd density. The sound sources were analysed, showing that several typical sound sources, such as traffic noise, can be masked by the sounds resulting from dense crowds. The acoustic analysis showed that crowd densities outside the range of 0.10 to 0.25 persons/m2 exhibited lower acoustic comfort evaluation scores. In terms of audiovisual characteristics, the subjective loudness increases with greater crowd density, while the acoustic comfort decreases. The results for an indoor underground shopping street are also presented for comparison. Copyright © 2014 Elsevier B.V. All rights reserved.

Soundscapes and the sense of hearing of fishes.

PubMed

Fay, Richard

2009-03-01

Underwater soundscapes have probably played an important role in the adaptation of ears and auditory systems of fishes throughout evolutionary time, and for all species. These sounds probably contain important information about the environment and about most objects and events that confront the receiving fish so that appropriate behavior is possible. For example, the sounds from reefs appear to be used by at least some fishes for their orientation and migration. These sorts of environmental sounds should be considered much like "acoustic daylight," that continuously bathes all environments and contain information that all organisms can potentially use to form a sort of image of the environment. At present, however, we are generally ignorant of the nature of ambient sound fields impinging on fishes, and the adaptive value of processing these fields to resolve the multiple sources of sound. Our field has focused almost exclusively on the adaptive value of processing species-specific communication sounds, and has not considered the informational value of ambient "noise." Since all fishes can detect and process acoustic particle motion, including the directional characteristics of this motion, underwater sound fields are potentially more complex and information-rich than terrestrial acoustic environments. The capacities of one fish species (goldfish) to receive and make use of such sound source information have been demonstrated (sound source segregation and auditory scene analysis), and it is suggested that all vertebrate species have this capacity. A call is made to better understand underwater soundscapes, and the associated behaviors they determine in fishes. © 2009 ISZS, Blackwell Publishing and IOZ/CAS.

Possibilities of psychoacoustics to determine sound quality

NASA Astrophysics Data System (ADS)

Genuit, Klaus

For some years, acoustic engineers have increasingly become aware of the importance of analyzing and minimizing noise problems not only with regard to the A-weighted sound pressure level, but to design sound quality. It is relatively easy to determine the A-weighted SPL according to international standards. However, the objective evaluation to describe subjectively perceived sound quality - taking into account psychoacoustic parameters such as loudness, roughness, fluctuation strength, sharpness and so forth - is more difficult. On the one hand, the psychoacoustic measurement procedures which are known so far have yet not been standardized. On the other hand, they have only been tested in laboratories by means of listening tests in the free-field and one sound source and simple signals. Therefore, the results achieved cannot be transferred to complex sound situations with several spatially distributed sound sources without difficulty. Due to the directional hearing and selectivity of human hearing, individual sound events can be selected among many. Already in the late seventies a new binaural Artificial Head Measurement System was developed which met the requirements of the automobile industry in terms of measurement technology. The first industrial application of the Artificial Head Measurement System was in 1981. Since that time the system was further developed, particularly by the cooperation between HEAD acoustics and Mercedes-Benz. In addition to a calibratable Artificial Head Measurement System which is compatible with standard measurement technologies and has transfer characteristics comparable to human hearing, a Binaural Analysis System is now also available. This system permits the analysis of binaural signals regarding physical and psychoacoustic procedures. Moreover, the signals to be analyzed can be simultaneously monitored through headphones and manipulated in the time and frequency domain so that those signal components being responsible for noise annoyance can be found. Especially in complex sound situations with several spatially distributed sound sources, standard, one-channel measurements methods cannot adequately determine sound quality, the acoustic comfort, or annoyance of sound events.

Using ILD or ITD Cues for Sound Source Localization and Speech Understanding in a Complex Listening Environment by Listeners With Bilateral and With Hearing-Preservation Cochlear Implants.

PubMed

Loiselle, Louise H; Dorman, Michael F; Yost, William A; Cook, Sarah J; Gifford, Rene H

2016-08-01

To assess the role of interaural time differences and interaural level differences in (a) sound-source localization, and (b) speech understanding in a cocktail party listening environment for listeners with bilateral cochlear implants (CIs) and for listeners with hearing-preservation CIs. Eleven bilateral listeners with MED-EL (Durham, NC) CIs and 8 listeners with hearing-preservation CIs with symmetrical low frequency, acoustic hearing using the MED-EL or Cochlear device were evaluated using 2 tests designed to task binaural hearing, localization, and a simulated cocktail party. Access to interaural cues for localization was constrained by the use of low-pass, high-pass, and wideband noise stimuli. Sound-source localization accuracy for listeners with bilateral CIs in response to the high-pass noise stimulus and sound-source localization accuracy for the listeners with hearing-preservation CIs in response to the low-pass noise stimulus did not differ significantly. Speech understanding in a cocktail party listening environment improved for all listeners when interaural cues, either interaural time difference or interaural level difference, were available. The findings of the current study indicate that similar degrees of benefit to sound-source localization and speech understanding in complex listening environments are possible with 2 very different rehabilitation strategies: the provision of bilateral CIs and the preservation of hearing.

Spherical harmonic analysis of the sound radiation from omnidirectional loudspeaker arrays

NASA Astrophysics Data System (ADS)

Pasqual, A. M.

2014-09-01

Omnidirectional sound sources are widely used in room acoustics. These devices are made up of loudspeakers mounted on a spherical or polyhedral cabinet, where the dodecahedral shape prevails. Although such electroacoustic sources have been made readily available to acousticians by many manufacturers, an in-depth investigation of their vibroacoustic behavior has not been provided yet. In order to fulfill this lack, this paper presents a theoretical study of the sound radiation from omnidirectional loudspeaker arrays, which is carried out by using a mathematical model based on the spherical harmonic analysis. Eight different loudspeaker arrangements on the sphere are considered: the well-known five Platonic solid layouts and three extremal system layouts. The latter possess useful properties for spherical loudspeaker arrays used as directivity controlled sound sources, so that these layouts are included here in order to investigate whether or not they could be of interest as omnidirectional sources as well. It is shown through a comparative analysis that the dodecahedral array leads to the lowest error in producing an omnidirectional sound field and to the highest acoustic power, which corroborates the prevalence of such a layout. In addition, if a source with less than 12 loudspeakers is required, it is shown that tetrahedra or hexahedra can be used alternatively, whereas the extremal system layouts are not interesting choices for omnidirectional loudspeaker arrays.

The use of an active controlled enclosure to attenuate sound radiation from a heavy radiator

NASA Astrophysics Data System (ADS)

Sun, Yao; Yang, Tiejun; Zhu, Minggang; Pan, Jie

2017-03-01

Active structural acoustical control usually experiences difficulty in the control of heavy sources or sources where direct applications of control forces are not practical. To overcome this difficulty, an active controlled enclosure, which forms a cavity with both flexible and open boundary, is employed. This configuration permits indirect implementation of active control in which the control inputs can be applied to subsidiary structures other than the sources. To determine the control effectiveness of the configuration, the vibro-acoustic behavior of the system, which consists of a top plate with an open, a sound cavity and a source panel, is investigated in this paper. A complete mathematical model of the system is formulated involving modified Fourier series formulations and the governing equations are solved using the Rayleigh-Ritz method. The coupling mechanisms of a partly opened cavity and a plate are analysed in terms of modal responses and directivity patterns. Furthermore, to attenuate sound power radiated from both the top panel and the open, two strategies are studied: minimizing the total radiated power and the cancellation of volume velocity. Moreover, three control configurations are compared, using a point force on the control panel (structural control), using a sound source in the cavity (acoustical control) and applying hybrid structural-acoustical control. In addition, the effects of boundary condition of the control panel on the sound radiation and control performance are discussed.

Material sound source localization through headphones

NASA Astrophysics Data System (ADS)

Dunai, Larisa; Peris-Fajarnes, Guillermo; Lengua, Ismael Lengua; Montaña, Ignacio Tortajada

2012-09-01

In the present paper a study of sound localization is carried out, considering two different sounds emitted from different hit materials (wood and bongo) as well as a Delta sound. The motivation of this research is to study how humans localize sounds coming from different materials, with the purpose of a future implementation of the acoustic sounds with better localization features in navigation aid systems or training audio-games suited for blind people. Wood and bongo sounds are recorded after hitting two objects made of these materials. Afterwards, they are analysed and processed. On the other hand, the Delta sound (click) is generated by using the Adobe Audition software, considering a frequency of 44.1 kHz. All sounds are analysed and convolved with previously measured non-individual Head-Related Transfer Functions both for an anechoic environment and for an environment with reverberation. The First Choice method is used in this experiment. Subjects are asked to localize the source position of the sound listened through the headphones, by using a graphic user interface. The analyses of the recorded data reveal that no significant differences are obtained either when considering the nature of the sounds (wood, bongo, Delta) or their environmental context (with or without reverberation). The localization accuracies for the anechoic sounds are: wood 90.19%, bongo 92.96% and Delta sound 89.59%, whereas for the sounds with reverberation the results are: wood 90.59%, bongo 92.63% and Delta sound 90.91%. According to these data, we can conclude that even when considering the reverberation effect, the localization accuracy does not significantly increase.

Differences in Talker Recognition by Preschoolers and Adults

ERIC Educational Resources Information Center

Creel, Sarah C.; Jimenez, Sofia R.

2012-01-01

Talker variability in speech influences language processing from infancy through adulthood and is inextricably embedded in the very cues that identify speech sounds. Yet little is known about developmental changes in the processing of talker information. On one account, children have not yet learned to separate speech sound variability from…

Evaluation of an inexpensive instrument to detect insects using multiple sound, vibration, and LED sensors

USDA-ARS?s Scientific Manuscript database

Instruments have been available for many years to detect insects using sound, vibration, or LED sensors separately. Most of these instruments are relatively expensive. An instrument was evaluated that incorporates all three types of sensors to improve the reliability of distinguishing different spec...

Blood pressure reprogramming adapter assists signal recording

NASA Technical Reports Server (NTRS)

Vick, H. A.

1967-01-01

Blood pressure reprogramming adapter separates the two components of a blood pressure signal, a dc pressure signal and an ac Korotkoff sounds signal, so that the Korotkoff sounds are recorded on one channel as received while the dc pressure signal is converted to FM and recorded on a second channel.

33 CFR 167.1701 - In Prince William Sound: Precautionary areas.

Code of Federal Regulations, 2011 CFR

2011-07-01

...: Precautionary areas. 167.1701 Section 167.1701 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1701 In Prince William Sound: Precautionary areas. (a) Cape Hinchinbrook. A precautionary area is established and is bounded by a line...

33 CFR 167.1701 - In Prince William Sound: Precautionary areas.

Code of Federal Regulations, 2013 CFR

2013-07-01

...: Precautionary areas. 167.1701 Section 167.1701 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1701 In Prince William Sound: Precautionary areas. (a) Cape Hinchinbrook. A precautionary area is established and is bounded by a line...

33 CFR 167.1701 - In Prince William Sound: Precautionary areas.

Code of Federal Regulations, 2014 CFR

2014-07-01

...: Precautionary areas. 167.1701 Section 167.1701 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1701 In Prince William Sound: Precautionary areas. (a) Cape Hinchinbrook. A precautionary area is established and is bounded by a line...

33 CFR 167.1701 - In Prince William Sound: Precautionary areas.

Code of Federal Regulations, 2012 CFR

2012-07-01

...: Precautionary areas. 167.1701 Section 167.1701 Navigation and Navigable Waters COAST GUARD, DEPARTMENT OF... Traffic Separation Schemes and Precautionary Areas Pacific West Coast § 167.1701 In Prince William Sound: Precautionary areas. (a) Cape Hinchinbrook. A precautionary area is established and is bounded by a line...

Handbook of Super 8 Production.

ERIC Educational Resources Information Center

Telzer, Ronnie, Ed.

This handbook is designed for anyone interested in producing super 8 films at any level of complexity and cost. Separate chapters present detailed discussions of the following topics: super 8 production systems and super 8 shooting and editing systems; budgeting; cinematography and sound recording; preparing to edit; editing; mixing sound tracks;…

Evaluation of an inexpensive instrument to detect insects using multiple sound, vibration, and LED sensors

USDA-ARS?s Scientific Manuscript database

Instruments have been available for many years to detect insects using sound, vibration, or LED sensors separately. Most of these instruments are relatively expensive. An instrument was evaluated that incorporates all three types of sensors to improve the reliability of distinguishing different sp...

Aerodynamics of Sounding-Rocket Geometries

NASA Technical Reports Server (NTRS)

Barrowman, J.

1982-01-01

Theoretical aerodynamics program TAD predicts aerodynamic characteristics of vehicles with sounding-rocket configurations. These slender, Axisymmetric finned vehicles have a wide range of aeronautical applications from rockets to high-speed armament. TAD calculates characteristics of separate portions of vehicle, calculates interference between portions, and combines results to form total vehicle solution.

Aeroacoustic analysis of the human phonation process based on a hybrid acoustic PIV approach

NASA Astrophysics Data System (ADS)

Lodermeyer, Alexander; Tautz, Matthias; Becker, Stefan; Döllinger, Michael; Birk, Veronika; Kniesburges, Stefan

2018-01-01

The detailed analysis of sound generation in human phonation is severely limited as the accessibility to the laryngeal flow region is highly restricted. Consequently, the physical basis of the underlying fluid-structure-acoustic interaction that describes the primary mechanism of sound production is not yet fully understood. Therefore, we propose the implementation of a hybrid acoustic PIV procedure to evaluate aeroacoustic sound generation during voice production within a synthetic larynx model. Focusing on the flow field downstream of synthetic, aerodynamically driven vocal folds, we calculated acoustic source terms based on the velocity fields obtained by time-resolved high-speed PIV applied to the mid-coronal plane. The radiation of these sources into the acoustic far field was numerically simulated and the resulting acoustic pressure was finally compared with experimental microphone measurements. We identified the tonal sound to be generated downstream in a small region close to the vocal folds. The simulation of the sound propagation underestimated the tonal components, whereas the broadband sound was well reproduced. Our results demonstrate the feasibility to locate aeroacoustic sound sources inside a synthetic larynx using a hybrid acoustic PIV approach. Although the technique employs a 2D-limited flow field, it accurately reproduces the basic characteristics of the aeroacoustic field in our larynx model. In future studies, not only the aeroacoustic mechanisms of normal phonation will be assessable, but also the sound generation of voice disorders can be investigated more profoundly.

Selective attention to sound location or pitch studied with event-related brain potentials and magnetic fields.

PubMed

Degerman, Alexander; Rinne, Teemu; Särkkä, Anna-Kaisa; Salmi, Juha; Alho, Kimmo

2008-06-01

Event-related brain potentials (ERPs) and magnetic fields (ERFs) were used to compare brain activity associated with selective attention to sound location or pitch in humans. Sixteen healthy adults participated in the ERP experiment, and 11 adults in the ERF experiment. In different conditions, the participants focused their attention on a designated sound location or pitch, or pictures presented on a screen, in order to detect target sounds or pictures among the attended stimuli. In the Attend Location condition, the location of sounds varied randomly (left or right), while their pitch (high or low) was kept constant. In the Attend Pitch condition, sounds of varying pitch (high or low) were presented at a constant location (left or right). Consistent with previous ERP results, selective attention to either sound feature produced a negative difference (Nd) between ERPs to attended and unattended sounds. In addition, ERPs showed a more posterior scalp distribution for the location-related Nd than for the pitch-related Nd, suggesting partially different generators for these Nds. The ERF source analyses found no source distribution differences between the pitch-related Ndm (the magnetic counterpart of the Nd) and location-related Ndm in the superior temporal cortex (STC), where the main sources of the Ndm effects are thought to be located. Thus, the ERP scalp distribution differences between the location-related and pitch-related Nd effects may have been caused by activity of areas outside the STC, perhaps in the inferior parietal regions.

Sound Explorations from the Ages of 10 to 37 Months: The Ontogenesis of Musical Conducts

ERIC Educational Resources Information Center

Delalande, Francois; Cornara, Silvia

2010-01-01

One of the forms of first musical conduct is the exploration of sound sources. When young children produce sounds with any object, these sounds may surprise them and so they make the sounds again--not exactly the same, but introducing some variation. A process of repetition with slight changes is set in motion which can be analysed, as did Piaget,…

Monitoring the Ocean Using High Frequency Ambient Sound

DTIC Science & Technology

2008-10-01

even identify specific groups within the resident killer whale type ( Puget Sound Southern Resident pods J, K and L) because these groups have...particular, the different populations of killer whales in the NE Pacific Ocean. This has been accomplished by detecting transient sounds with short...high sea state (the sound of spray), general shipping - close and distant, clanking and whale calls and clicking. These sound sources form the basis

Meteorological effects on long-range outdoor sound propagation

NASA Technical Reports Server (NTRS)

Klug, Helmut

1990-01-01

Measurements of sound propagation over distances up to 1000 m were carried out with an impulse sound source offering reproducible, short time signals. Temperature and wind speed at several heights were monitored simultaneously; the meteorological data are used to determine the sound speed gradients according to the Monin-Obukhov similarity theory. The sound speed profile is compared to a corresponding prediction, gained through the measured travel time difference between direct and ground reflected pulse (which depends on the sound speed gradient). Positive sound speed gradients cause bending of the sound rays towards the ground yielding enhanced sound pressure levels. The measured meteorological effects on sound propagation are discussed and illustrated by ray tracing methods.

The noise generated by a landing gear wheel with hub and rim cavities

NASA Astrophysics Data System (ADS)

Wang, Meng; Angland, David; Zhang, Xin

2017-03-01

Wheels are one of the major noise sources of landing gears. Accurate numerical predictions of wheel noise can provide an insight into the physical mechanism of landing gear noise generation and can aid in the design of noise control devices. The major noise sources of a 33% scaled isolated landing gear wheel are investigated by simulating three different wheel configurations using high-order numerical simulations to compute the flow field and the FW-H equation to obtain the far-field acoustic pressures. The baseline configuration is a wheel with a hub cavity and two rim cavities. Two additional simulations are performed; one with the hub cavity covered (NHC) and the other with both the hub cavity and rim cavities covered (NHCRC). These simulations isolate the effects of the hub cavity and rim cavities on the overall wheel noise. The surface flow patterns are visualised by shear stress lines and show that the flow separations and attachments on the side of the wheel, in both the baseline and the configuration with only the hub cavity covered, are significantly reduced by covering both the hub and rim cavities. A frequency-domain FW-H equation is used to identify the noise source regions on the surface of the wheel. The tyre is the main low frequency noise source and shows a lift dipole and side force dipole pattern depending on the frequency. The hub cavity is identified as the dominant middle frequency noise source and radiates in a frequency range centered around the first and second depth modes of the cylindrical hub cavity. The rim cavities are the main high-frequency noise sources. With the hub cavity and rim cavities covered, the largest reduction in Overall Sound Pressure Level (OASPL) is achieved in the hub side direction. In the other directivities, there is also a reduction in the radiated sound.

The Problems with "Noise Numbers" for Wind Farm Noise Assessment

ERIC Educational Resources Information Center

Thorne, Bob

2011-01-01

Human perception responds primarily to sound character rather than sound level. Wind farms are unique sound sources and exhibit special audible and inaudible characteristics that can be described as modulating sound or as a tonal complex. Wind farm compliance measures based on a specified noise number alone will fail to address problems with noise…

Feasibility Study for Habitat Development Using Dredged Material at Jetty Island, Everett, Washington.

DTIC Science & Technology

1984-07-01

I;.... • ° o- - .- .-.. .. ’f + .. , .+.: -.- 3XT "T v"ummy Jetty Island separates Port Gardner (an arm of Puget Sound ). Washington, from the lover...eelgrass habitats throughout Puget Sound . Dovever, the species diversity of benthic invertebrates was relatively low. The unvegetated mudflat areas...Tidal conditions are very similar to other areas of Puget Sound . Mean tidal range for fverett harbor is 7.4 feet, and the extreme range is estimated to

Geologic Map of the Yukon-Koyukuk Basin, Alaska

USGS Publications Warehouse

Patton, William W.; Wilson, Frederic H.; Labay, Keith A.; Shew, Nora B.

2009-01-01

This map and accompanying digital files represent part of a systematic effort to release geologic data for the United States in a uniform manner. All the geologic data in this series will be published as parts of the U.S. Geological Survey Data Series. The geologic data in this series have been compiled from a wide variety of sources, ranging from state and regional geologic maps to large-scale field mapping. The data are presented for use at a nominal scale of 1:500,000, although individual datasets may contain data suitable for use at larger scales. The metadata associated with each release will provide more detailed information on sources and appropriate scales for use. Associated attribute databases accompany the spatial database of the geology and are uniformly structured for ease in developing regional- and national-scale maps. The 1:500,000-scale geologic map of the Yukon-Koyukuk Basin, Alaska, covers more than 200,000 square kilometers of western Alaska or nearly 15 percent of the total land area of the state. It stretches from the Brooks Range on the north to the Kuskokwim River and lower reaches of the Yukon River on the south and from Kotzebue Sound, Seward Peninsula, and Norton Sound on the west to the Yukon-Tanana Uplands and Tanana-Kuskokwim Lowlands on the east. It includes not only the northern and central part of the basin, but also the lands that border the basin. The area is characterized by isolated clusters of hills and low mountain ranges separated by broad alluviated interior and coastal lowlands. Most of the lowlands, except those bordering Kotzebue Sound and Norton Sound, support a heavy vegetation cover. Exposures of bedrock are generally limited to rubble-strewn ridgetops and to cutbanks along the rivers. The map of the Yukon-Koyukuk Basin was prepared largely from geologic field data collected between 1953 and 1988 by the U.S. Geological Survey and published as 1:250,000-scale geologic quadrangle maps. Additional data for parts of the Wiseman, Ruby, Medfra, and Ophir quadrangles came from 1:63,360-scale quadrangle maps published by the Alaska Division of Geological and Geophysical Surveys. The map also incorporates some unpublished field data for the Ruby quadrangle collected by R.M. Chapman between 1944 and 1977 and for parts of the Tanana, Bettles, Norton Bay, and Candle quadrangles collected by W.W. Patton, Jr. and others between 1954 and 1985. Sources of geologic map data for each of the eighteen 1:250,000-scale quadrangles used in compiling this 1:500,000-scale map of the Yukon-Koyukuk Basin as well as sources of general geologic information pertaining to the entire map area are provided in the 'Sources of Information' section.

Change deafness for real spatialized environmental scenes.

PubMed

Gaston, Jeremy; Dickerson, Kelly; Hipp, Daniel; Gerhardstein, Peter

2017-01-01

The everyday auditory environment is complex and dynamic; often, multiple sounds co-occur and compete for a listener's cognitive resources. 'Change deafness', framed as the auditory analog to the well-documented phenomenon of 'change blindness', describes the finding that changes presented within complex environments are often missed. The present study examines a number of stimulus factors that may influence change deafness under real-world listening conditions. Specifically, an AX (same-different) discrimination task was used to examine the effects of both spatial separation over a loudspeaker array and the type of change (sound source additions and removals) on discrimination of changes embedded in complex backgrounds. Results using signal detection theory and accuracy analyses indicated that, under most conditions, errors were significantly reduced for spatially distributed relative to non-spatial scenes. A second goal of the present study was to evaluate a possible link between memory for scene contents and change discrimination. Memory was evaluated by presenting a cued recall test following each trial of the discrimination task. Results using signal detection theory and accuracy analyses indicated that recall ability was similar in terms of accuracy, but there were reductions in sensitivity compared to previous reports. Finally, the present study used a large and representative sample of outdoor, urban, and environmental sounds, presented in unique combinations of nearly 1000 trials per participant. This enabled the exploration of the relationship between change perception and the perceptual similarity between change targets and background scene sounds. These (post hoc) analyses suggest both a categorical and a stimulus-level relationship between scene similarity and the magnitude of change errors.

Spatial sound field synthesis and upmixing based on the equivalent source method.

PubMed

Bai, Mingsian R; Hsu, Hoshen; Wen, Jheng-Ciang

2014-01-01

Given scarce number of recorded signals, spatial sound field synthesis with an extended sweet spot is a challenging problem in acoustic array signal processing. To address the problem, a synthesis and upmixing approach inspired by the equivalent source method (ESM) is proposed. The synthesis procedure is based on the pressure signals recorded by a microphone array and requires no source model. The array geometry can also be arbitrary. Four upmixing strategies are adopted to enhance the resolution of the reproduced sound field when there are more channels of loudspeakers than the microphones. Multi-channel inverse filtering with regularization is exploited to deal with the ill-posedness in the reconstruction process. The distance between the microphone and loudspeaker arrays is optimized to achieve the best synthesis quality. To validate the proposed system, numerical simulations and subjective listening experiments are performed. The results demonstrated that all upmixing methods improved the quality of reproduced target sound field over the original reproduction. In particular, the underdetermined ESM interpolation method yielded the best spatial sound field synthesis in terms of the reproduction error, timbral quality, and spatial quality.

Effects of spectral complexity and sound duration on automatic complex-sound pitch processing in humans - a mismatch negativity study.

PubMed

Tervaniemi, M; Schröger, E; Saher, M; Näätänen, R

2000-08-18

The pitch of a spectrally rich sound is known to be more easily perceived than that of a sinusoidal tone. The present study compared the importance of spectral complexity and sound duration in facilitated pitch discrimination. The mismatch negativity (MMN), which reflects automatic neural discrimination, was recorded to a 2. 5% pitch change in pure tones with only one sinusoidal frequency component (500 Hz) and in spectrally rich tones with three (500-1500 Hz) and five (500-2500 Hz) harmonic partials. During the recordings, subjects concentrated on watching a silent movie. In separate blocks, stimuli were of 100 and 250 ms in duration. The MMN amplitude was enhanced with both spectrally rich sounds when compared with pure tones. The prolonged sound duration did not significantly enhance the MMN. This suggests that increased spectral rather than temporal information facilitates pitch processing of spectrally rich sounds.

Development and Testing of a High Level Axial Array Duct Sound Source for the NASA Flow Impedance Test Facility

NASA Technical Reports Server (NTRS)

Johnson, Marty E.; Fuller, Chris R.; Jones, Michael G. (Technical Monitor)

2000-01-01

In this report both a frequency domain method for creating high level harmonic excitation and a time domain inverse method for creating large pulses in a duct are developed. To create controllable, high level sound an axial array of six JBL-2485 compression drivers was used. The pressure downstream is considered as input voltages to the sources filtered by the natural dynamics of the sources and the duct. It is shown that this dynamic behavior can be compensated for by filtering the inputs such that both time delays and phase changes are taken into account. The methods developed maximize the sound output while (i) keeping within the power constraints of the sources and (ii) maintaining a suitable level of reproduction accuracy. Harmonic excitation pressure levels of over 155dB were created experimentally over a wide frequency range (1000-4000Hz). For pulse excitation there is a tradeoff between accuracy of reproduction and sound level achieved. However, the accurate reproduction of a pulse with a maximum pressure level over 6500Pa was achieved experimentally. It was also shown that the throat connecting the driver to the duct makes it difficult to inject sound just below the cut-on of each acoustic mode (pre cut-on loading effect).

Perceptual assessment of quality of urban soundscapes with combined noise sources and water sounds.

PubMed

Jeon, Jin Yong; Lee, Pyoung Jik; You, Jin; Kang, Jian

2010-03-01

In this study, urban soundscapes containing combined noise sources were evaluated through field surveys and laboratory experiments. The effect of water sounds on masking urban noises was then examined in order to enhance the soundscape perception. Field surveys in 16 urban spaces were conducted through soundwalking to evaluate the annoyance of combined noise sources. Synthesis curves were derived for the relationships between noise levels and the percentage of highly annoyed (%HA) and the percentage of annoyed (%A) for the combined noise sources. Qualitative analysis was also made using semantic scales for evaluating the quality of the soundscape, and it was shown that the perception of acoustic comfort and loudness was strongly related to the annoyance. A laboratory auditory experiment was then conducted in order to quantify the total annoyance caused by road traffic noise and four types of construction noise. It was shown that the annoyance ratings were related to the types of construction noise in combination with road traffic noise and the level of the road traffic noise. Finally, water sounds were determined to be the best sounds to use for enhancing the urban soundscape. The level of the water sounds should be similar to or not less than 3 dB below the level of the urban noises.

Development of a Finite-Difference Time Domain (FDTD) Model for Propagation of Transient Sounds in Very Shallow Water.

PubMed

Sprague, Mark W; Luczkovich, Joseph J

2016-01-01

This finite-difference time domain (FDTD) model for sound propagation in very shallow water uses pressure and velocity grids with both 3-dimensional Cartesian and 2-dimensional cylindrical implementations. Parameters, including water and sediment properties, can vary in each dimension. Steady-state and transient signals from discrete and distributed sources, such as the surface of a vibrating pile, can be used. The cylindrical implementation uses less computation but requires axial symmetry. The Cartesian implementation allows asymmetry. FDTD calculations compare well with those of a split-step parabolic equation. Applications include modeling the propagation of individual fish sounds, fish aggregation sounds, and distributed sources.

Adaptive near-field beamforming techniques for sound source imaging.

PubMed

Cho, Yong Thung; Roan, Michael J

2009-02-01

Phased array signal processing techniques such as beamforming have a long history in applications such as sonar for detection and localization of far-field sound sources. Two sometimes competing challenges arise in any type of spatial processing; these are to minimize contributions from directions other than the look direction and minimize the width of the main lobe. To tackle this problem a large body of work has been devoted to the development of adaptive procedures that attempt to minimize side lobe contributions to the spatial processor output. In this paper, two adaptive beamforming procedures-minimum variance distorsionless response and weight optimization to minimize maximum side lobes--are modified for use in source visualization applications to estimate beamforming pressure and intensity using near-field pressure measurements. These adaptive techniques are compared to a fixed near-field focusing technique (both techniques use near-field beamforming weightings focusing at source locations estimated based on spherical wave array manifold vectors with spatial windows). Sound source resolution accuracies of near-field imaging procedures with different weighting strategies are compared using numerical simulations both in anechoic and reverberant environments with random measurement noise. Also, experimental results are given for near-field sound pressure measurements of an enclosed loudspeaker.

A method for calculating strut and splitter plate noise in exit ducts: Theory and verification

NASA Technical Reports Server (NTRS)

Fink, M. R.

1978-01-01

Portions of a four-year analytical and experimental investigation relative to noise radiation from engine internal components in turbulent flow are summarized. Spectra measured for such airfoils over a range of chord, thickness ratio, flow velocity, and turbulence level were compared with predictions made by an available rigorous thin-airfoil analytical method. This analysis included the effects of flow compressibility and source noncompactness. Generally good agreement was obtained. This noise calculation method for isolated airfoils in turbulent flow was combined with a method for calculating transmission of sound through a subsonic exit duct and with an empirical far-field directivity shape. These three elements were checked separately and were individually shown to give close agreement with data. This combination provides a method for predicting engine internally generated aft-radiated noise from radial struts and stators, and annular splitter rings. Calculated sound power spectra, directivity, and acoustic pressure spectra were compared with the best available data. These data were for noise caused by a fan exit duct annular splitter ring, larger-chord stator blades, and turbine exit struts.

Physical processes in a coupled bay-estuary coastal system: Whitsand Bay and Plymouth Sound

NASA Astrophysics Data System (ADS)

Uncles, R. J.; Stephens, J. A.; Harris, C.

2015-09-01

Whitsand Bay and Plymouth Sound are located in the southwest of England. The Bay and Sound are separated by the ∼2-3 km-wide Rame Peninsula and connected by ∼10-20 m-deep English Channel waters. Results are presented from measurements of waves and currents, drogue tracking, surveys of salinity, temperature and turbidity during stratified and unstratified conditions, and bed sediment surveys. 2D and 3D hydrodynamic models are used to explore the generation of tidally- and wind-driven residual currents, flow separation and the formation of the Rame eddy, and the coupling between the Bay and the Sound. Tidal currents flow around the Rame Peninsula from the Sound to the Bay between approximately 3 h before to 2 h after low water and form a transport path between them that conveys lower salinity, higher turbidity waters from the Sound to the Bay. These waters are then transported into the Bay as part of the Bay-mouth limb of the Rame eddy and subsequently conveyed to the near-shore, east-going limb and re-circulated back towards Rame Head. The Simpson-Hunter stratification parameter indicates that much of the Sound and Bay are likely to stratify thermally during summer months. Temperature stratification in both is pronounced during summer and is largely determined by coastal, deeper-water stratification offshore. Small tidal stresses in the Bay are unable to move bed sediment of the observed sizes. However, the Bay and Sound are subjected to large waves that are capable of driving a substantial bed-load sediment transport. Measurements show relatively low levels of turbidity, but these respond rapidly to, and have a strong correlation with, wave height.

Perception of quality and trustworthiness of Internet resources by personal health information seekers.

PubMed

Stavri, P Zoë; Freeman, Donna J; Burroughs, Catherine M

2003-01-01

This paper focuses on one dimension of personal health information seeking: perception of quality and trustworthiness of information sources. Intensive interviews were conducted using a conversational, unstructured, exploratory interview style. Interviews were conducted at 3 publicly accessible library sites in Arizona, Hawaii and Nevada. Thirty-eight non-experts were interviewed. Three separate and distinct methods used to identify credible health information resources were identified. Consumers may have strong opinions about what they mistrust; use fairly rigorous evaluation protocols; or filter information based on intuition or common sense, eye appeal or an authoritative sounding sponsor or title. Many people use a mix of rational and/or intuitive criteria to assess the health information they use.

Aircraft laser sensing of sound velocity in water - Brillouin scattering

NASA Technical Reports Server (NTRS)

Hickman, G. D.; Harding, John M.; Carnes, Michael; Pressman, AL; Kattawar, George W.; Fry, Edward S.

1991-01-01

A real-time data source for sound speed in the upper 100 m has been proposed for exploratory development. This data source is planned to be generated via a ship- or aircraft-mounted optical pulsed laser using the spontaneous Brillouin scattering technique. The system should be capable (from a single 10 ns 500 mJ pulse) of yielding range resolved sound speed profiles in water to depths of 75-100 m to an accuracy of 1 m/s. The 100 m profiles will provide the capability of rapidly monitoring the upper-ocean vertical structure. They will also provide an extensive, subsurface-data source for existing real-time, operational ocean nowcast/forecast systems.

Reflection and Transmission of a Focused Finite Amplitude Sound Beam Incident on a Curved Interface

NASA Astrophysics Data System (ADS)

Makin, Inder Raj Singh

Reflection and transmission of a finite amplitude focused sound beam at a weakly curved interface separating two fluid-like media are investigated. The KZK parabolic wave equation, which accounts for thermoviscous absorption, diffraction, and nonlinearity, is used to describe the high intensity focused beam. The first part of the work deals with the quasilinear analysis of a weakly nonlinear beam after its reflection and transmission from a curved interface. A Green's function approach is used to define the field integrals describing the primary and the nonlinearly generated second harmonic beam. Closed-form solutions are obtained for the primary and second harmonic beams when a Gaussian amplitude distribution at the source is assumed. The second part of the research uses a numerical frequency domain solution of the KZK equation for a fully nonlinear analysis of the reflected and transmitted fields. Both piston and Gaussian sources are considered. Harmonic components generated in the medium due to propagation of the focused beam are evaluated, and formation of shocks in the reflected and transmitted beams is investigated. A finite amplitude focused beam is observed to be modified due to reflection and transmission from a curved interface in a manner distinct from that in the case of a small signal beam. Propagation curves, beam patterns, phase plots and time waveforms for various parameters defining the source and media pairs are presented, highlighting the effect of the interface curvature on the reflected and transmitted beams. Relevance of the current work to biomedical applications of ultrasound is discussed.

Investigation of hydraulic transmission noise sources

NASA Astrophysics Data System (ADS)

Klop, Richard J.

Advanced hydrostatic transmissions and hydraulic hybrids show potential in new market segments such as commercial vehicles and passenger cars. Such new applications regard low noise generation as a high priority, thus, demanding new quiet hydrostatic transmission designs. In this thesis, the aim is to investigate noise sources of hydrostatic transmissions to discover strategies for designing compact and quiet solutions. A model has been developed to capture the interaction of a pump and motor working in a hydrostatic transmission and to predict overall noise sources. This model allows a designer to compare noise sources for various configurations and to design compact and inherently quiet solutions. The model describes dynamics of the system by coupling lumped parameter pump and motor models with a one-dimensional unsteady compressible transmission line model. The model has been verified with dynamic pressure measurements in the line over a wide operating range for several system structures. Simulation studies were performed illustrating sensitivities of several design variables and the potential of the model to design transmissions with minimal noise sources. A semi-anechoic chamber has been designed and constructed suitable for sound intensity measurements that can be used to derive sound power. Measurements proved the potential to reduce audible noise by predicting and reducing both noise sources. Sound power measurements were conducted on a series hybrid transmission test bench to validate the model and compare predicted noise sources with sound power.

DOE Office of Scientific and Technical Information (OSTI.GOV)

Wolfe, D.A.

Three separate papers are represented in this final report; Toxicity of intertidal and subtidal sediments contaminated by the Exxon Valdez oil spill; Comparative toxicities of polar and non-polar organic fractions from sediments affected by the Exxon Valdez oil spill in Prince William Sound, Alaska; and Fate of the oil spilled from the T/V Exxon Valdez in Prince William Sound, Alaska.

On the Possible Detection of Lightning Storms by Elephants

PubMed Central

Kelley, Michael C.; Garstang, Michael

2013-01-01

Simple Summary We use data similar to that taken by the International Monitoring System for the detection of nuclear explosions, to determine whether elephants might be capable of detecting and locating the source of sounds generated by thunderstorms. Knowledge that elephants might be capable of responding to such storms, particularly at the end of the dry season when migrations are initiated, is of considerable interest to management and conservation. Abstract Theoretical calculations suggest that sounds produced by thunderstorms and detected by a system similar to the International Monitoring System (IMS) for the detection of nuclear explosions at distances ≥100 km, are at sound pressure levels equal to or greater than 6 × 10−3 Pa. Such sound pressure levels are well within the range of elephant hearing. Frequencies carrying these sounds might allow for interaural time delays such that adult elephants could not only hear but could also locate the source of these sounds. Determining whether it is possible for elephants to hear and locate thunderstorms contributes to the question of whether elephant movements are triggered or influenced by these abiotic sounds. PMID:26487406

Thin structured rigid body for acoustic absorption

NASA Astrophysics Data System (ADS)

Starkey, T. A.; Smith, J. D.; Hibbins, A. P.; Sambles, J. R.; Rance, H. J.

2017-01-01

We present a thin acoustic metamaterial absorber, comprised of only rigid metal and air, that gives rise to near unity absorption of airborne sound on resonance. This simple, easily fabricated, robust structure comprising a perforated metal plate separated from a rigid wall by a deeply subwavelength channel of air is an ideal candidate for a sound absorbing panel. The strong absorption in the system is attributed to the thermo-viscous losses arising from a sound wave guided between the plate and the wall, defining the subwavelength channel.

Shallow Scattering Layer (SSL): Emergence Behaviors of Coastal Macrofauna

DTIC Science & Technology

2003-09-30

group ascent and descent speeds were slower than those found in a deeper water column in Puget Sound by Kringel et al. ( 2003) despite the order...instruments separated by 50 m show high coherence, but they were collected at the same water depth. Our initial data record for West Sound , Orcas Island...West Sound , Orcas Island, Washington Volume backscattering strength at 265 kHz (dB) H ei g h t ab o v e T A P S ( m ) 0 0 10 20 midnight midnight

Investigation of the sound generation mechanisms for in-duct orifice plates.

PubMed

Tao, Fuyang; Joseph, Phillip; Zhang, Xin; Stalnov, Oksana; Siercke, Matthias; Scheel, Henning

2017-08-01

Sound generation due to an orifice plate in a hard-walled flow duct which is commonly used in air distribution systems (ADS) and flow meters is investigated. The aim is to provide an understanding of this noise generation mechanism based on measurements of the source pressure distribution over the orifice plate. A simple model based on Curle's acoustic analogy is described that relates the broadband in-duct sound field to the surface pressure cross spectrum on both sides of the orifice plate. This work describes careful measurements of the surface pressure cross spectrum over the orifice plate from which the surface pressure distribution and correlation length is deduced. This information is then used to predict the radiated in-duct sound field. Agreement within 3 dB between the predicted and directly measured sound fields is obtained, providing direct confirmation that the surface pressure fluctuations acting over the orifice plates are the main noise sources. Based on the developed model, the contributions to the sound field from different radial locations of the orifice plate are calculated. The surface pressure is shown to follow a U 3.9 velocity scaling law and the area over which the surface sources are correlated follows a U 1.8 velocity scaling law.

The Confirmation of the Inverse Square Law Using Diffraction Gratings

ERIC Educational Resources Information Center

Papacosta, Pangratios; Linscheid, Nathan

2014-01-01

Understanding the inverse square law, how for example the intensity of light or sound varies with distance, presents conceptual and mathematical challenges. Students know intuitively that intensity decreases with distance. A light source appears dimmer and sound gets fainter as the distance from the source increases. The difficulty is in…

The sound strength parameter G and its importance in evaluating and planning the acoustics of halls for music.

PubMed

Beranek, Leo

2011-05-01

The parameter, "Strength of Sound G" is closely related to loudness. Its magnitude is dependent, inversely, on the total sound absorption in a room. By comparison, the reverberation time (RT) is both inversely related to the total sound absorption in a hall and directly related to its cubic volume. Hence, G and RT in combination are vital in planning the acoustics of a concert hall. A newly proposed "Bass Index" is related to the loudness of the bass sound and equals the value of G at 125 Hz in decibels minus its value at mid-frequencies. Listener envelopment (LEV) is shown for most halls to be directly related to the mid-frequency value of G. The broadening of sound, i.e., apparent source width (ASW) is given by degree of source broadening (DSB) which is determined from the combined effect of early lateral reflections as measured by binaural quality index (BQI) and strength G. The optimum values and limits of these parameters are discussed.

The role of long-term familiarity and attentional maintenance in short-term memory for timbre.

PubMed

Siedenburg, Kai; McAdams, Stephen

2017-04-01

We study short-term recognition of timbre using familiar recorded tones from acoustic instruments and unfamiliar transformed tones that do not readily evoke sound-source categories. Participants indicated whether the timbre of a probe sound matched with one of three previously presented sounds (item recognition). In Exp. 1, musicians better recognised familiar acoustic compared to unfamiliar synthetic sounds, and this advantage was particularly large in the medial serial position. There was a strong correlation between correct rejection rate and the mean perceptual dissimilarity of the probe to the tones from the sequence. Exp. 2 compared musicians' and non-musicians' performance with concurrent articulatory suppression, visual interference, and with a silent control condition. Both suppression tasks disrupted performance by a similar margin, regardless of musical training of participants or type of sounds. Our results suggest that familiarity with sound source categories and attention play important roles in short-term memory for timbre, which rules out accounts solely based on sensory persistence.

Cardiovascular responses to noise: Effects of self-estimated sensitivity to noise, sex, and time of the day

NASA Astrophysics Data System (ADS)

di Nisi, J.; Muzet, A.; Weber, L. D.

1987-04-01

Eighty subjects of both sexes were selected according to their self-estimated high or low sensitivity to noise. Noise exposure took place during a mental task ("sound" condition) or during a video film illustrating the noises ("sound and video" condition). The experiments were conducted between 0900 and 1100 hours or between 1500 and 1700 hours. Heart rate response and finger pulse response amplitudes were averaged separately for "sound" and "sound and video" conditions. In the "sound" condition, the average amplitude of the heart rate response differed significantly between noise-sensitivity groups: the low sensitivity group showed a lower average amplitude of heart rate response than the high sensitivity group. A significant interaction between sex and time of the day (morning or afternoon) was observed in both "sound" and "sound and video" conditions. In the "sound" condition, the percentage of noises inducing a finger pulse response appeared higher in female than in male subjects.

Harmonic Hopping, and Both Punctuated and Gradual Evolution of Acoustic Characters in Selasphorus Hummingbird Tail-Feathers

PubMed Central

Clark, Christopher James

2014-01-01

Models of character evolution often assume a single mode of evolutionary change, such as continuous, or discrete. Here I provide an example in which a character exhibits both types of change. Hummingbirds in the genus Selasphorus produce sound with fluttering tail-feathers during courtship. The ancestral character state within Selasphorus is production of sound with an inner tail-feather, R2, in which the sound usually evolves gradually. Calliope and Allen's Hummingbirds have evolved autapomorphic acoustic mechanisms that involve feather-feather interactions. I develop a source-filter model of these interactions. The ‘source’ comprises feather(s) that are both necessary and sufficient for sound production, and are aerodynamically coupled to neighboring feathers, which act as filters. Filters are unnecessary or insufficient for sound production, but may evolve to become sources. Allen's Hummingbird has evolved to produce sound with two sources, one with feather R3, another frequency-modulated sound with R4, and their interaction frequencies. Allen's R2 retains the ancestral character state, a ∼1 kHz “ghost” fundamental frequency masked by R3, which is revealed when R3 is experimentally removed. In the ancestor to Allen's Hummingbird, the dominant frequency has ‘hopped’ to the second harmonic without passing through intermediate frequencies. This demonstrates that although the fundamental frequency of a communication sound may usually evolve gradually, occasional jumps from one character state to another can occur in a discrete fashion. Accordingly, mapping acoustic characters on a phylogeny may produce misleading results if the physical mechanism of production is not known. PMID:24722049

Sound produced by an oscillating arc in a high-pressure gas

NASA Astrophysics Data System (ADS)

Popov, Fedor K.; Shneider, Mikhail N.

2017-08-01

We suggest a simple theory to describe the sound generated by small periodic perturbations of a cylindrical arc in a dense gas. Theoretical analysis was done within the framework of the non-self-consistent channel arc model and supplemented with time-dependent gas dynamic equations. It is shown that an arc with power amplitude oscillations on the order of several percent is a source of sound whose intensity is comparable with external ultrasound sources used in experiments to increase the yield of nanoparticles in the high pressure arc systems for nanoparticle synthesis.

Focusing and directional beaming effects of airborne sound through a planar lens with zigzag slits

DOE Office of Scientific and Technical Information (OSTI.GOV)

Tang, Kun; Qiu, Chunyin, E-mail: cyqiu@whu.edu.cn; Lu, Jiuyang

2015-01-14

Based on the Huygens-Fresnel principle, we design a planar lens to efficiently realize the interconversion between the point-like sound source and Gaussian beam in ambient air. The lens is constructed by a planar plate perforated elaborately with a nonuniform array of zigzag slits, where the slit exits act as subwavelength-sized secondary sources carrying desired sound responses. The experiments operated at audible regime agree well with the theoretical predictions. This compact device could be useful in daily life applications, such as for medical and detection purposes.

Intelligibility in speech maskers with a binaural cochlear implant sound coding strategy inspired by the contralateral medial olivocochlear reflex.

PubMed

Lopez-Poveda, Enrique A; Eustaquio-Martín, Almudena; Stohl, Joshua S; Wolford, Robert D; Schatzer, Reinhold; Gorospe, José M; Ruiz, Santiago Santa Cruz; Benito, Fernando; Wilson, Blake S

2017-05-01

We have recently proposed a binaural cochlear implant (CI) sound processing strategy inspired by the contralateral medial olivocochlear reflex (the MOC strategy) and shown that it improves intelligibility in steady-state noise (Lopez-Poveda et al., 2016, Ear Hear 37:e138-e148). The aim here was to evaluate possible speech-reception benefits of the MOC strategy for speech maskers, a more natural type of interferer. Speech reception thresholds (SRTs) were measured in six bilateral and two single-sided deaf CI users with the MOC strategy and with a standard (STD) strategy. SRTs were measured in unilateral and bilateral listening conditions, and for target and masker stimuli located at azimuthal angles of (0°, 0°), (-15°, +15°), and (-90°, +90°). Mean SRTs were 2-5 dB better with the MOC than with the STD strategy for spatially separated target and masker sources. For bilateral CI users, the MOC strategy (1) facilitated the intelligibility of speech in competition with spatially separated speech maskers in both unilateral and bilateral listening conditions; and (2) led to an overall improvement in spatial release from masking in the two listening conditions. Insofar as speech is a more natural type of interferer than steady-state noise, the present results suggest that the MOC strategy holds potential for promising outcomes for CI users. Copyright © 2017. Published by Elsevier B.V.

Prediction of XV-15 tilt rotor discrete frequency aeroacoustic noise with WOPWOP

NASA Technical Reports Server (NTRS)

Coffen, Charles D.; George, Albert R.

1990-01-01

The results, methodology, and conclusions of noise prediction calculations carried out to study several possible discrete frequency harmonic noise mechanisms of the XV-15 Tilt Rotor Aircraft in hover and helicopter mode forward flight are presented. The mechanisms studied were thickness and loading noise. In particular, the loading noise caused by flow separation and the fountain/ground plane effect were predicted with calculations made using WOPWOP, a noise prediction program developed by NASA Langley. The methodology was to model the geometry and aerodynamics of the XV-15 rotor blades in hover and steady level flight and then create corresponding FORTRAN subroutines which were used an input for WOPWOP. The models are described and the simplifying assumptions made in creating them are evaluated, and the results of the computations are presented. The computations lead to the following conclusions: The fountain/ground plane effect is an important source of aerodynamic noise for the XV-15 in hover. Unsteady flow separation from the airfoil passing through the fountain at high angles of attack significantly affects the predicted sound spectra and may be an important noise mechanism for the XV-15 in hover mode. The various models developed did not predict the sound spectra in helicopter forward flight. The experimental spectra indicate the presence of blade vortex interactions which were not modeled in these calculations. A need for further study and development of more accurate aerodynamic models, including unsteady stall in hover and blade vortex interactions in forward flight.

Human middle-ear nonlinearity measurements using laser Doppler vibrometry

NASA Astrophysics Data System (ADS)

Gladiné, Kilian; Muyshondt, Pieter G. G.; Dirckx, Joris J. J.

2017-12-01

It has long been supposed that the middle-ear has near to perfect linear characteristics, and several attempts have been made to investigate this hypothesis. In conclusion, the middle-ear was regarded as a linear system at least up till sound pressure levels of 120 dB. Because of the linear relationship between Doppler shift of light and the vibration velocity of the object on which the light is reflected, laser Doppler vibrometry (LDV) is an intrinsically highly linear measurement technique. Therefore it allows straightforward detection of very small nonlinearities in a vibration response. In this paper, laser Doppler vibrometry and multisine stimulation are used to detect nonlinear distortions in the vibration response at the umbo of the tympanic membrane of seven human cadaver temporal bones. Nonlinear distortions were detected starting from sound pressure levels of 99 dB and measurements were performed up to 120 dB. These distortions can be subdivided into even degree (e.g. quadratic distortion tones) and odd degree nonlinear distortions (e.g. cubic distortion tones). We illustrate that with odd multisine stimulation the level of even and odd degree nonlinear distortions can be investigated separately. In conclusion, laser Doppler vibrometry is an adequate tool to detect nonlinear distortions in the middle-ear system and to quantify the level of such distortions even at 57 dB below the vibration response. The possibility to analyze even degree and odd degree nonlinear distortion levels separately can help in future work to pinpoint the source of the nonlinearity.

High-frequency monopole sound source for anechoic chamber qualification

NASA Astrophysics Data System (ADS)

Saussus, Patrick; Cunefare, Kenneth A.

2003-04-01

Anechoic chamber qualification procedures require the use of an omnidirectional monopole sound source. Required characteristics for these monopole sources are explicitly listed in ISO 3745. Building a high-frequency monopole source that meets these characteristics has proved difficult due to the size limitations imposed by small wavelengths at high frequency. A prototype design developed for use in hemianechoic chambers employs telescoping tubes, which act as an inverse horn. This same design can be used in anechoic chambers, with minor adaptations. A series of gradually decreasing brass telescoping tubes is attached to the throat of a well-insulated high-frequency compression driver. Therefore, all of the sound emitted from the driver travels through the horn and exits through an opening of approximately 2.5 mm. Directivity test data show that this design meets all of the requirements set forth by ISO 3745.

Animal Pitch Perception: Melodies and Harmonies

PubMed Central

Hoeschele, Marisa

2017-01-01

Pitch is a percept of sound that is based in part on fundamental frequency. Although pitch can be defined in a way that is clearly separable from other aspects of musical sounds, such as timbre, the perception of pitch is not a simple topic. Despite this, studying pitch separately from other aspects of sound has led to some interesting conclusions about how humans and other animals process acoustic signals. It turns out that pitch perception in humans is based on an assessment of pitch height, pitch chroma, relative pitch, and grouping principles. How pitch is broken down depends largely on the context. Most, if not all, of these principles appear to also be used by other species, but when and how accurately they are used varies across species and context. Studying how other animals compare to humans in their pitch abilities is partially a reevaluation of what we know about humans by considering ourselves in a biological context. PMID:28649291

Assessment of sound levels in a neonatal intensive care unit in tabriz, iran.

PubMed

Valizadeh, Sousan; Bagher Hosseini, Mohammad; Alavi, Nasrinsadat; Asadollahi, Malihe; Kashefimehr, Siamak

2013-03-01

High levels of sound have several negative effects, such as noise-induced hearing loss and delayed growth and development, on premature infants in neonatal intensive care units (NICUs). In order to reduce sound levels, they should first be measured. This study was performed to assess sound levels and determine sources of noise in the NICU of Alzahra Teaching Hospital (Tabriz, Iran). In a descriptive study, 24 hours in 4 workdays were randomly selected. Equivalent continuous sound level (Leq), sound level that is exceeded only 10% of the time (L10), maximum sound level (Lmax), and peak instantaneous sound pressure level (Lzpeak) were measured by CEL-440 sound level meter (SLM) at 6 fixed locations in the NICU. Data was collected using a questionnaire. SPSS13 was then used for data analysis. Mean values of Leq, L10, and Lmax were determined as 63.46 dBA, 65.81 dBA, and 71.30 dBA, respectively. They were all higher than standard levels (Leq < 45 dB, L10 ≤50 dB, and Lmax ≤65 dB). The highest Leq was measured at the time of nurse rounds. Leq was directly correlated with the number of staff members present in the ward. Finally, sources of noise were ordered based on their intensity. Considering that sound levels were higher than standard levels in our studied NICU, it is necessary to adopt policies to reduce sound.

Assessment of Sound Levels in a Neonatal Intensive Care Unit in Tabriz, Iran

PubMed Central

Valizadeh, Sousan; Bagher Hosseini, Mohammad; Alavi, Nasrinsadat; Asadollahi, Malihe; Kashefimehr, Siamak

2013-01-01

Introduction: High levels of sound have several negative effects, such as noise-induced hearing loss and delayed growth and development, on premature infants in neonatal intensive care units (NICUs). In order to reduce sound levels, they should first be measured. This study was performed to assess sound levels and determine sources of noise in the NICU of Alzahra Teaching Hospital (Tabriz, Iran). Methods: In a descriptive study, 24 hours in 4 workdays were randomly selected. Equivalent continuous sound level (Leq), sound level that is exceeded only 10% of the time (L10), maximum sound level (Lmax), and peak instantaneous sound pressure level (Lzpeak) were measured by CEL-440 sound level meter (SLM) at 6 fixed locations in the NICU. Data was collected using a questionnaire. SPSS13 was then used for data analysis. Results: Mean values of Leq, L10, and Lmax were determined as 63.46 dBA, 65.81 dBA, and 71.30 dBA, respectively. They were all higher than standard levels (Leq < 45 dB, L10 ≤50 dB, and Lmax ≤65 dB). The highest Leq was measured at the time of nurse rounds. Leq was directly correlated with the number of staff members present in the ward. Finally, sources of noise were ordered based on their intensity. Conclusion: Considering that sound levels were higher than standard levels in our studied NICU, it is necessary to adopt policies to reduce sound. PMID:25276706

Statistics of natural binaural sounds.

PubMed

Młynarski, Wiktor; Jost, Jürgen

2014-01-01

Binaural sound localization is usually considered a discrimination task, where interaural phase (IPD) and level (ILD) disparities at narrowly tuned frequency channels are utilized to identify a position of a sound source. In natural conditions however, binaural circuits are exposed to a stimulation by sound waves originating from multiple, often moving and overlapping sources. Therefore statistics of binaural cues depend on acoustic properties and the spatial configuration of the environment. Distribution of cues encountered naturally and their dependence on physical properties of an auditory scene have not been studied before. In the present work we analyzed statistics of naturally encountered binaural sounds. We performed binaural recordings of three auditory scenes with varying spatial configuration and analyzed empirical cue distributions from each scene. We have found that certain properties such as the spread of IPD distributions as well as an overall shape of ILD distributions do not vary strongly between different auditory scenes. Moreover, we found that ILD distributions vary much weaker across frequency channels and IPDs often attain much higher values, than can be predicted from head filtering properties. In order to understand the complexity of the binaural hearing task in the natural environment, sound waveforms were analyzed by performing Independent Component Analysis (ICA). Properties of learned basis functions indicate that in natural conditions soundwaves in each ear are predominantly generated by independent sources. This implies that the real-world sound localization must rely on mechanisms more complex than a mere cue extraction.

Statistics of Natural Binaural Sounds

PubMed Central

Młynarski, Wiktor; Jost, Jürgen

2014-01-01

Binaural sound localization is usually considered a discrimination task, where interaural phase (IPD) and level (ILD) disparities at narrowly tuned frequency channels are utilized to identify a position of a sound source. In natural conditions however, binaural circuits are exposed to a stimulation by sound waves originating from multiple, often moving and overlapping sources. Therefore statistics of binaural cues depend on acoustic properties and the spatial configuration of the environment. Distribution of cues encountered naturally and their dependence on physical properties of an auditory scene have not been studied before. In the present work we analyzed statistics of naturally encountered binaural sounds. We performed binaural recordings of three auditory scenes with varying spatial configuration and analyzed empirical cue distributions from each scene. We have found that certain properties such as the spread of IPD distributions as well as an overall shape of ILD distributions do not vary strongly between different auditory scenes. Moreover, we found that ILD distributions vary much weaker across frequency channels and IPDs often attain much higher values, than can be predicted from head filtering properties. In order to understand the complexity of the binaural hearing task in the natural environment, sound waveforms were analyzed by performing Independent Component Analysis (ICA). Properties of learned basis functions indicate that in natural conditions soundwaves in each ear are predominantly generated by independent sources. This implies that the real-world sound localization must rely on mechanisms more complex than a mere cue extraction. PMID:25285658

Electrophysiological correlates of cocktail-party listening.

PubMed

Lewald, Jörg; Getzmann, Stephan

2015-10-01

Detecting, localizing, and selectively attending to a particular sound source of interest in complex auditory scenes composed of multiple competing sources is a remarkable capacity of the human auditory system. The neural basis of this so-called "cocktail-party effect" has remained largely unknown. Here, we studied the cortical network engaged in solving the "cocktail-party" problem, using event-related potentials (ERPs) in combination with two tasks demanding horizontal localization of a naturalistic target sound presented either in silence or in the presence of multiple competing sound sources. Presentation of multiple sound sources, as compared to single sources, induced an increased P1 amplitude, a reduction in N1, and a strong N2 component, resulting in a pronounced negativity in the ERP difference waveform (N2d) around 260 ms after stimulus onset. About 100 ms later, the anterior contralateral N2 subcomponent (N2ac) occurred in the multiple-sources condition, as computed from the amplitude difference for targets in the left minus right hemispaces. Cortical source analyses of the ERP modulation, resulting from the contrast of multiple vs. single sources, generally revealed an initial enhancement of electrical activity in right temporo-parietal areas, including auditory cortex, by multiple sources (at P1) that is followed by a reduction, with the primary sources shifting from right inferior parietal lobule (at N1) to left dorso-frontal cortex (at N2d). Thus, cocktail-party listening, as compared to single-source localization, appears to be based on a complex chronology of successive electrical activities within a specific cortical network involved in spatial hearing in complex situations. Copyright © 2015 Elsevier B.V. All rights reserved.

Determination of equivalent sound speed profiles for ray tracing in near-ground sound propagation.

PubMed

Prospathopoulos, John M; Voutsinas, Spyros G

2007-09-01

The determination of appropriate sound speed profiles in the modeling of near-ground propagation using a ray tracing method is investigated using a ray tracing model which is capable of performing axisymmetric calculations of the sound field around an isolated source. Eigenrays are traced using an iterative procedure which integrates the trajectory equations for each ray launched from the source at a specific direction. The calculation of sound energy losses is made by introducing appropriate coefficients to the equations representing the effect of ground and atmospheric absorption and the interaction with the atmospheric turbulence. The model is validated against analytical and numerical predictions of other methodologies for simple cases, as well as against measurements for nonrefractive atmospheric environments. A systematic investigation for near-ground propagation in downward and upward refractive atmosphere is made using experimental data. Guidelines for the suitable simulation of the wind velocity profile are derived by correlating predictions with measurements.

Acoustic centering of sources measured by surrounding spherical microphone arrays.

PubMed

Hagai, Ilan Ben; Pollow, Martin; Vorländer, Michael; Rafaely, Boaz

2011-10-01

The radiation patterns of acoustic sources have great significance in a wide range of applications, such as measuring the directivity of loudspeakers and investigating the radiation of musical instruments for auralization. Recently, surrounding spherical microphone arrays have been studied for sound field analysis, facilitating measurement of the pressure around a sphere and the computation of the spherical harmonics spectrum of the sound source. However, the sound radiation pattern may be affected by the location of the source inside the microphone array, which is an undesirable property when aiming to characterize source radiation in a unique manner. This paper presents a theoretical analysis of the spherical harmonics spectrum of spatially translated sources and defines four measures for the misalignment of the acoustic center of a radiating source. Optimization is used to promote optimal alignment based on the proposed measures and the errors caused by numerical and array-order limitations are investigated. This methodology is examined using both simulated and experimental data in order to investigate the performance and limitations of the different alignment methods. © 2011 Acoustical Society of America

Propagation characteristics of audible noise generated by single corona source under positive DC voltage

NASA Astrophysics Data System (ADS)

Li, Xuebao; Cui, Xiang; Lu, Tiebing; Wang, Donglai

2017-10-01

The directivity and lateral profile of corona-generated audible noise (AN) from a single corona source are measured through experiments carried out in the semi-anechoic laboratory. The experimental results show that the waveform of corona-generated AN consists of a series of random sound pressure pulses whose pulse amplitudes decrease with the increase of measurement distance. A single corona source can be regarded as a non-directional AN source, and the A-weighted SPL (sound pressure level) decreases 6 dB(A) as doubling the measurement distance. Then, qualitative explanations for the rationality of treating the single corona source as a point source are given on the basis of the Ingard's theory for sound generation in corona discharge. Furthermore, we take into consideration of the ground reflection and the air attenuation to reconstruct the propagation features of AN from the single corona source. The calculated results agree with the measurement well, which validates the propagation model. Finally, the influence of the ground reflection on the SPL is presented in the paper.

Cognitive and Linguistic Sources of Variance in 2-Year-Olds' Speech-Sound Discrimination: A Preliminary Investigation

ERIC Educational Resources Information Center

Lalonde, Kaylah; Holt, Rachael Frush

2014-01-01

Purpose: This preliminary investigation explored potential cognitive and linguistic sources of variance in 2- year-olds' speech-sound discrimination by using the toddler change/no-change procedure and examined whether modifications would result in a procedure that can be used consistently with younger 2-year-olds. Method: Twenty typically…

The use of an intraoral electrolarynx for an edentulous patient: a clinical report.

PubMed

Wee, Alvin G; Wee, Lisa A; Cheng, Ansgar C; Cwynar, Roger B

2004-06-01

This clinical report describes the clinical requirements, treatment sequence, and use of a relatively new intraoral electrolarynx for a completely edentulous patient. This device consists of a sound source attached to the maxilla and a hand-held controller unit that controls the pitch and volume of the intraoral sound source via transmitted radio waves.

Spherical beamforming for spherical array with impedance surface

NASA Astrophysics Data System (ADS)

Tontiwattanakul, Khemapat

2018-01-01

Spherical microphone array beamforming has been a popular research topic for recent years. Due to their isotropic beam in three dimensional spaces as well as a certain frequency range, the arrays are widely used in many applications such as sound field recording, acoustic beamforming, and noise source localisation. The body of a spherical array is usually considered perfectly rigid. A sound field captured by the sensors on spherical array can be decomposed into a series of spherical harmonics. In noise source localisation, the amplitude density of sound sources is estimated and illustrated by mean of colour maps. In this work, a rigid spherical array covered by fibrous materials is studied via numerical simulation and the performance of the spherical beamforming is discussed.

Study on the Non-contact Acoustic Inspection Method for Concrete Structures by using Strong Ultrasonic Sound source

NASA Astrophysics Data System (ADS)

Sugimoto, Tsuneyoshi; Uechi, Itsuki; Sugimoto, Kazuko; Utagawa, Noriyuki; Katakura, Kageyoshi

Hammering test is widely used to inspect the defects in concrete structures. However, this method has a major difficulty in inspect at high-places, such as a tunnel ceiling or a bridge girder. Moreover, its detection accuracy is dependent on a tester's experience. Therefore, we study about the non-contact acoustic inspection method of the concrete structure using the air borne sound wave and a laser Doppler vibrometer. In this method, the concrete surface is excited by air-borne sound wave emitted with a long range acoustic device (LRAD), and the vibration velocity on the concrete surface is measured by a laser Doppler vibrometer. A defect part is detected by the same flexural resonance as the hammer method. It is already shown clearly that detection of a defect can be performed from a long distance of 5 m or more using a concrete test object. Moreover, it is shown that a real concrete structure can also be applied. However, when the conventional LRAD was used as a sound source, there were problems, such as restrictions of a measurement angle and the surrounding noise. In order to solve these problems, basic examination which used the strong ultrasonic wave sound source was carried out. In the experiment, the concrete test object which includes an imitation defect from 5-m distance was used. From the experimental result, when the ultrasonic sound source was used, restrictions of a measurement angle become less severe and it was shown that circumference noise also falls dramatically.

Sound for Film: Audio Education for Filmmakers.

ERIC Educational Resources Information Center

Lazar, Wanda

1998-01-01

Identifies the specific, unique, and important elements of audio education required by film professionals. Presents a model unit to be included in a film studies program, either as a separate course or as part of a film production or introduction to film course. Offers a model syllabus for such a course or unit on sound in film. (SR)

The Measurement of the Oral and Nasal Sound Pressure Levels of Speech

ERIC Educational Resources Information Center

Clarke, Wayne M.

1975-01-01

A nasal separator was used to measure the oral and nasal components in the speech of a normal adult Australian population. Results indicated no difference in oral and nasal sound pressure levels for read versus spontaneous speech samples; however, females tended to have a higher nasal component than did males. (Author/TL)

A Longitudinal Investigation of Morpho-Syntax in Children with Speech Sound Disorders

ERIC Educational Resources Information Center

Mortimer, Jennifer; Rvachew, Susan

2010-01-01

Purpose: The intent of this study was to examine the longitudinal morpho-syntactic progression of children with Speech Sound Disorders (SSD) grouped according to Mean Length of Utterance (MLU) scores. Methods: Thirty-seven children separated into four clusters were assessed in their pre-kindergarten and Grade 1 years. Cluster 1 were children with…

Polyphonic Music Information Retrieval Based on Multi-Label Cascade Classification System

ERIC Educational Resources Information Center

Jiang, Wenxin

2009-01-01

Recognition and separation of sounds played by various instruments is very useful in labeling audio files with semantic information. This is a non-trivial task requiring sound analysis, but the results can aid automatic indexing and browsing music data when searching for melodies played by user specified instruments. Melody match based on pitch…

75 FR 39915 - Marine Mammals; File No. 15483

Federal Register 2010, 2011, 2012, 2013, 2014

2010-07-13

... whales adjust their bearing to avoid received sound pressure levels greater than 120 dB, which would... marine mammals may be taken by Level B harassment as researchers attempt to provoke an avoidance response through sound transmission into their environment. The sound source consists of a transmitter and...

24 CFR 51.103 - Criteria and standards.

Code of Federal Regulations, 2011 CFR

2011-04-01

...-night average sound level produced as the result of the accumulation of noise from all sources contributing to the external noise environment at the site. Day-night average sound level, abbreviated as DNL and symbolized as Ldn, is the 24-hour average sound level, in decibels, obtained after addition of 10...

Characterisation of structure-borne sound source using reception plate method.

PubMed

Putra, A; Saari, N F; Bakri, H; Ramlan, R; Dan, R M

2013-01-01

A laboratory-based experiment procedure of reception plate method for structure-borne sound source characterisation is reported in this paper. The method uses the assumption that the input power from the source installed on the plate is equal to the power dissipated by the plate. In this experiment, rectangular plates having high and low mobility relative to that of the source were used as the reception plates and a small electric fan motor was acting as the structure-borne source. The data representing the source characteristics, namely, the free velocity and the source mobility, were obtained and compared with those from direct measurement. Assumptions and constraints employing this method are discussed.

Complete data listings for CSEM soundings on Kilauea Volcano, Hawaii

DOE Office of Scientific and Technical Information (OSTI.GOV)

Kauahikaua, J.; Jackson, D.B.; Zablocki, C.J.

1983-01-01

This document contains complete data from a controlled-source electromagnetic (CSEM) sounding/mapping project at Kilauea volcano, Hawaii. The data were obtained at 46 locations about a fixed-location, horizontal, polygonal loop source in the summit area of the volcano. The data consist of magnetic field amplitudes and phases at excitation frequencies between 0.04 and 8 Hz. The vector components were measured in a cylindrical coordinate system centered on the loop source. 5 references.

The Physiological Basis of Chinese Höömii Generation.

PubMed

Li, Gelin; Hou, Qian

2017-01-01

The study aimed to investigate the physiological basis of vibration mode of sound source of a variety of Mongolian höömii forms of singing in China. The participant is a Mongolian höömii performing artist who was recommended by the Chinese Medical Association of Art. He used three types of höömii, namely vibration höömii, whistle höömii, and overtone höömii, which were compared with general comfortable pronunciation of /i:/ as control. Phonation was observed during /i:/. A laryngostroboscope (Storz) was used to determine vibration source-mucosal wave in the throat. For vibration höömii, bilateral ventricular folds approximated to the midline and made contact at the midline during pronunciation. Ventricular and vocal folds oscillated together as a single unit to form a composite vibration (double oscillator) sound source. For whistle höömii, ventricular folds approximated to the midline to cover part of vocal folds, but did not contact each other. It did not produce mucosal wave. The vocal folds produced mucosal wave to form a single vibration sound source. For overtone höömii, the anterior two-thirds of ventricular folds touched each other during pronunciation. The last one-third produced the mucosal wave. The vocal folds produced mucosal wave at the same time, which was a composite vibration (double oscillator) sound source mode. The Höömii form of singing, including mixed voices and multivoice, was related to the presence of dual vibration sound sources. Its high overtone form of singing (whistle höömii) was related to stenosis at the resonance chambers' initiation site (ventricular folds level). Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Spacecraft Internal Acoustic Environment Modeling

NASA Technical Reports Server (NTRS)

Chu, Shao-Sheng R.; Allen Christopher S.

2010-01-01

Acoustic modeling can be used to identify key noise sources, determine/analyze sub-allocated requirements, keep track of the accumulation of minor noise sources, and to predict vehicle noise levels at various stages in vehicle development, first with estimates of noise sources, later with experimental data. This paper describes the implementation of acoustic modeling for design purposes by incrementally increasing model fidelity and validating the accuracy of the model while predicting the noise of sources under various conditions. During FY 07, a simple-geometry Statistical Energy Analysis (SEA) model was developed and validated using a physical mockup and acoustic measurements. A process for modeling the effects of absorptive wall treatments and the resulting reverberation environment were developed. During FY 08, a model with more complex and representative geometry of the Orion Crew Module (CM) interior was built, and noise predictions based on input noise sources were made. A corresponding physical mockup was also built. Measurements were made inside this mockup, and comparisons were made with the model and showed excellent agreement. During FY 09, the fidelity of the mockup and corresponding model were increased incrementally by including a simple ventilation system. The airborne noise contribution of the fans was measured using a sound intensity technique, since the sound power levels were not known beforehand. This is opposed to earlier studies where Reference Sound Sources (RSS) with known sound power level were used. Comparisons of the modeling result with the measurements in the mockup showed excellent results. During FY 10, the fidelity of the mockup and the model were further increased by including an ECLSS (Environmental Control and Life Support System) wall, associated closeout panels, and the gap between ECLSS wall and mockup wall. The effect of sealing the gap and adding sound absorptive treatment to ECLSS wall were also modeled and validated.

Modeling the utility of binaural cues for underwater sound localization.

PubMed

Schneider, Jennifer N; Lloyd, David R; Banks, Patchouly N; Mercado, Eduardo

2014-06-01

The binaural cues used by terrestrial animals for sound localization in azimuth may not always suffice for accurate sound localization underwater. The purpose of this research was to examine the theoretical limits of interaural timing and level differences available underwater using computational and physical models. A paired-hydrophone system was used to record sounds transmitted underwater and recordings were analyzed using neural networks calibrated to reflect the auditory capabilities of terrestrial mammals. Estimates of source direction based on temporal differences were most accurate for frequencies between 0.5 and 1.75 kHz, with greater resolution toward the midline (2°), and lower resolution toward the periphery (9°). Level cues also changed systematically with source azimuth, even at lower frequencies than expected from theoretical calculations, suggesting that binaural mechanical coupling (e.g., through bone conduction) might, in principle, facilitate underwater sound localization. Overall, the relatively limited ability of the model to estimate source position using temporal and level difference cues underwater suggests that animals such as whales may use additional cues to accurately localize conspecifics and predators at long distances. Copyright © 2014 Elsevier B.V. All rights reserved.

The meaning of city noises: Investigating sound quality in Paris (France)

NASA Astrophysics Data System (ADS)

Dubois, Daniele; Guastavino, Catherine; Maffiolo, Valerie; Guastavino, Catherine; Maffiolo, Valerie

2004-05-01

The sound quality of Paris (France) was investigated by using field inquiries in actual environments (open questionnaires) and using recordings under laboratory conditions (free-sorting tasks). Cognitive categories of soundscapes were inferred by means of psycholinguistic analyses of verbal data and of mathematical analyses of similarity judgments. Results show that auditory judgments mainly rely on source identification. The appraisal of urban noise therefore depends on the qualitative evaluation of noise sources. The salience of human sounds in public spaces has been demonstrated, in relation to pleasantness judgments: soundscapes with human presence tend to be perceived as more pleasant than soundscapes consisting solely of mechanical sounds. Furthermore, human sounds are qualitatively processed as indicators of human outdoor activities, such as open markets, pedestrian areas, and sidewalk cafe districts that reflect city life. In contrast, mechanical noises (mainly traffic noise) are commonly described in terms of physical properties (temporal structure, intensity) of a permanent background noise that also characterizes urban areas. This connotes considering both quantitative and qualitative descriptions to account for the diversity of cognitive interpretations of urban soundscapes, since subjective evaluations depend both on the meaning attributed to noise sources and on inherent properties of the acoustic signal.

Acoustic positioning for space processing experiments

NASA Technical Reports Server (NTRS)

Whymark, R. R.

1974-01-01

An acoustic positioning system is described that is adaptable to a range of processing chambers and furnace systems. Operation at temperatures exceeding 1000 C is demonstrated in experiments involving the levitation of liquid and solid glass materials up to several ounces in weight. The system consists of a single source of sound that is beamed at a reflecting surface placed a distance away. Stable levitation is achieved at a succession of discrete energy minima contained throughout the volume between the reflector and the sound source. Several specimens can be handled at one time. Metal discs up to 3 inches in diameter can be levitated, solid spheres of dense material up to 0.75 inches diameter, and liquids can be freely suspended in l-g in the form of near-spherical droplets up to 0.25 inch diameter, or flattened liquid discs up to 0.6 inches diameter. Larger specimens may be handled by increasing the size of the sound source or by reducing the sound frequency.

Sound source localization on an axial fan at different operating points

NASA Astrophysics Data System (ADS)

Zenger, Florian J.; Herold, Gert; Becker, Stefan; Sarradj, Ennes

2016-08-01

A generic fan with unskewed fan blades is investigated using a microphone array method. The relative motion of the fan with respect to the stationary microphone array is compensated by interpolating the microphone data to a virtual rotating array with the same rotational speed as the fan. Hence, beamforming algorithms with deconvolution, in this case CLEAN-SC, could be applied. Sound maps and integrated spectra of sub-components are evaluated for five operating points. At selected frequency bands, the presented method yields sound maps featuring a clear circular source pattern corresponding to the nine fan blades. Depending on the adjusted operating point, sound sources are located on the leading or trailing edges of the fan blades. Integrated spectra show that in most cases leading edge noise is dominant for the low-frequency part and trailing edge noise for the high-frequency part. The shift from leading to trailing edge noise is strongly dependent on the operating point and frequency range considered.

Study of the Acoustic Effects of Hydrokinetic Tidal Turbines in Admiralty Inlet, Puget Sound

DOE Office of Scientific and Technical Information (OSTI.GOV)

Brian Polagye; Jim Thomson; Chris Bassett

2012-03-30

Hydrokinetic turbines will be a source of noise in the marine environment - both during operation and during installation/removal. High intensity sound can cause injury or behavioral changes in marine mammals and may also affect fish and invertebrates. These noise effects are, however, highly dependent on the individual marine animals; the intensity, frequency, and duration of the sound; and context in which the sound is received. In other words, production of sound is a necessary, but not sufficient, condition for an environmental impact. At a workshop on the environmental effects of tidal energy development, experts identified sound produced by turbinesmore » as an area of potentially significant impact, but also high uncertainty. The overall objectives of this project are to improve our understanding of the potential acoustic effects of tidal turbines by: (1) Characterizing sources of existing underwater noise; (2) Assessing the effectiveness of monitoring technologies to characterize underwater noise and marine mammal responsiveness to noise; (3) Evaluating the sound profile of an operating tidal turbine; and (4) Studying the effect of turbine sound on surrogate species in a laboratory environment. This study focuses on a specific case study for tidal energy development in Admiralty Inlet, Puget Sound, Washington (USA), but the methodologies and results are applicable to other turbine technologies and geographic locations. The project succeeded in achieving the above objectives and, in doing so, substantially contributed to the body of knowledge around the acoustic effects of tidal energy development in several ways: (1) Through collection of data from Admiralty Inlet, established the sources of sound generated by strong currents (mobilizations of sediment and gravel) and determined that low-frequency sound recorded during periods of strong currents is non-propagating pseudo-sound. This helped to advance the debate within the marine and hydrokinetics acoustic community as to whether strong currents produce propagating sound. (2) Analyzed data collected from a tidal turbine operating at the European Marine Energy Center to develop a profile of turbine sound and developed a framework to evaluate the acoustic effects of deploying similar devices in other locations. This framework has been applied to Public Utility District No. 1 of Snohomish Country's demonstration project in Admiralty Inlet to inform postinstallation acoustic and marine mammal monitoring plans. (3) Demonstrated passive acoustic techniques to characterize the ambient noise environment at tidal energy sites (fixed, long-term observations recommended) and characterize the sound from anthropogenic sources (drifting, short-term observations recommended). (4) Demonstrated the utility and limitations of instrumentation, including bottom mounted instrumentation packages, infrared cameras, and vessel monitoring systems. In doing so, also demonstrated how this type of comprehensive information is needed to interpret observations from each instrument (e.g., hydrophone data can be combined with vessel tracking data to evaluate the contribution of vessel sound to ambient noise). (5) Conducted a study that suggests harbor porpoise in Admiralty Inlet may be habituated to high levels of ambient noise due to omnipresent vessel traffic. The inability to detect behavioral changes associated with a high intensity source of opportunity (passenger ferry) has informed the approach for post-installation marine mammal monitoring. (6) Conducted laboratory exposure experiments of juvenile Chinook salmon and showed that exposure to a worse than worst case acoustic dose of turbine sound does not result in changes to hearing thresholds or biologically significant tissue damage. Collectively, this means that Chinook salmon may be at a relatively low risk of injury from sound produced by tidal turbines located in or near their migration path. In achieving these accomplishments, the project has significantly advanced the District's goals of developing a demonstration-scale tidal energy project in Admiralty Inlet. Pilot demonstrations of this type are an essential step in the development of commercial-scale tidal energy in the United States. This is a renewable resource capable of producing electricity in a highly predictable manner.« less

Concerns of the Institute of Transport Study and Research for reducing the sound level inside completely repaired buses. [noise and vibration control

NASA Technical Reports Server (NTRS)

Groza, A.; Calciu, J.; Nicola, I.; Ionasek, A.

1974-01-01

Sound level measurements on noise sources on buses are used to observe the effects of attenuating acoustic pressure levels inside the bus by sound-proofing during complete repair. A spectral analysis of the sound level as a function of motor speed, bus speed along the road, and the category of the road is reported.

A sound budget for the southeastern Bering Sea: measuring wind, rainfall, shipping, and other sources of underwater sound.

PubMed

Nystuen, Jeffrey A; Moore, Sue E; Stabeno, Phyllis J

2010-07-01

Ambient sound in the ocean contains quantifiable information about the marine environment. A passive aquatic listener (PAL) was deployed at a long-term mooring site in the southeastern Bering Sea from 27 April through 28 September 2004. This was a chain mooring with lots of clanking. However, the sampling strategy of the PAL filtered through this noise and allowed the background sound field to be quantified for natural signals. Distinctive signals include the sound from wind, drizzle and rain. These sources dominate the sound budget and their intensity can be used to quantify wind speed and rainfall rate. The wind speed measurement has an accuracy of +/-0.4 m s(-1) when compared to a buoy-mounted anemometer. The rainfall rate measurement is consistent with a land-based measurement in the Aleutian chain at Cold Bay, AK (170 km south of the mooring location). Other identifiable sounds include ships and short transient tones. The PAL was designed to reject transients in the range important for quantification of wind speed and rainfall, but serendipitously recorded peaks in the sound spectrum between 200 Hz and 3 kHz. Some of these tones are consistent with whale calls, but most are apparently associated with mooring self-noise.

Functional morphology of the sound-generating labia in the syrinx of two songbird species.

PubMed

Riede, Tobias; Goller, Franz

2010-01-01

In songbirds, two sound sources inside the syrinx are used to produce the primary sound. Laterally positioned labia are passively set into vibration, thus interrupting a passing air stream. Together with subsyringeal pressure, the size and tension of the labia determine the spectral characteristics of the primary sound. Very little is known about how the histological composition and morphology of the labia affect their function as sound generators. Here we related the size and microstructure of the labia to their acoustic function in two songbird species with different acoustic characteristics, the white-crowned sparrow and zebra finch. Histological serial sections of the syrinx and different staining techniques were used to identify collagen, elastin and hyaluronan as extracellular matrix components. The distribution and orientation of elastic fibers indicated that the labia in white-crowned sparrows are multi-layered structures, whereas they are more uniformly structured in the zebra finch. Collagen and hyaluronan were evenly distributed in both species. A multi-layered composition could give rise to complex viscoelastic properties of each sound source. We also measured labia size. Variability was found along the dorso-ventral axis in both species. Lateral asymmetry was identified in some individuals but not consistently at the species level. Different size between the left and right sound sources could provide a morphological basis for the acoustic specialization of each sound generator, but only in some individuals. The inconsistency of its presence requires the investigation of alternative explanations, e.g. differences in viscoelastic properties of the labia of the left and right syrinx. Furthermore, we identified attachments of syringeal muscles to the labia as well as to bronchial half rings and suggest a mechanism for their biomechanical function.

Functional morphology of the sound-generating labia in the syrinx of two songbird species

PubMed Central

Riede, Tobias; Goller, Franz

2010-01-01

In songbirds, two sound sources inside the syrinx are used to produce the primary sound. Laterally positioned labia are passively set into vibration, thus interrupting a passing air stream. Together with subsyringeal pressure, the size and tension of the labia determine the spectral characteristics of the primary sound. Very little is known about how the histological composition and morphology of the labia affect their function as sound generators. Here we related the size and microstructure of the labia to their acoustic function in two songbird species with different acoustic characteristics, the white-crowned sparrow and zebra finch. Histological serial sections of the syrinx and different staining techniques were used to identify collagen, elastin and hyaluronan as extracellular matrix components. The distribution and orientation of elastic fibers indicated that the labia in white-crowned sparrows are multi-layered structures, whereas they are more uniformly structured in the zebra finch. Collagen and hyaluronan were evenly distributed in both species. A multi-layered composition could give rise to complex viscoelastic properties of each sound source. We also measured labia size. Variability was found along the dorso-ventral axis in both species. Lateral asymmetry was identified in some individuals but not consistently at the species level. Different size between the left and right sound sources could provide a morphological basis for the acoustic specialization of each sound generator, but only in some individuals. The inconsistency of its presence requires the investigation of alternative explanations, e.g. differences in viscoelastic properties of the labia of the left and right syrinx. Furthermore, we identified attachments of syringeal muscles to the labia as well as to bronchial half rings and suggest a mechanism for their biomechanical function. PMID:19900184

Theory of acoustic design of opera house and a design proposal

NASA Astrophysics Data System (ADS)

Ando, Yoichi

2004-05-01

First of all, the theory of subjective preference for sound fields based on the model of auditory-brain system is briefly mentioned. It consists of the temporal factors and spatial factors associated with the left and right cerebral hemispheres, respectively. The temporal criteria are the initial time delay gap between the direct sound and the first Reflection (Dt1) and the subsequent reverberation time (Tsub). These preferred conditions are related to the minimum value of effective duration of the running autocorrelation function of source signals (te)min. The spatial criteria are binaural listening level (LL) and the IACC, which may be extracted from the interaural crosscorrelation function. In the opera house, there are two different kind of sound sources, i.e., the vocal source of relatively short values of (te)min in the stage and the orchestra music of long values of (te)min in the pit. For these sources, a proposal is made here.

Diversity in sound pressure levels and estimated active space of resident killer whale vocalizations.

PubMed

Miller, Patrick J O

2006-05-01

Signal source intensity and detection range, which integrates source intensity with propagation loss, background noise and receiver hearing abilities, are important characteristics of communication signals. Apparent source levels were calculated for 819 pulsed calls and 24 whistles produced by free-ranging resident killer whales by triangulating the angles-of-arrival of sounds on two beamforming arrays towed in series. Levels in the 1-20 kHz band ranged from 131 to 168 dB re 1 microPa at 1 m, with differences in the means of different sound classes (whistles: 140.2+/-4.1 dB; variable calls: 146.6+/-6.6 dB; stereotyped calls: 152.6+/-5.9 dB), and among stereotyped call types. Repertoire diversity carried through to estimates of active space, with "long-range" stereotyped calls all containing overlapping, independently-modulated high-frequency components (mean estimated active space of 10-16 km in sea state zero) and "short-range" sounds (5-9 km) included all stereotyped calls without a high-frequency component, whistles, and variable calls. Short-range sounds are reported to be more common during social and resting behaviors, while long-range stereotyped calls predominate in dispersed travel and foraging behaviors. These results suggest that variability in sound pressure levels may reflect diverse social and ecological functions of the acoustic repertoire of killer whales.

Sounds and source levels from bowhead whales off Pt. Barrow, Alaska.

PubMed

Cummings, W C; Holliday, D V

1987-09-01

Sounds were recorded from bowhead whales migrating past Pt. Barrow, AK, to the Canadian Beaufort Sea. They mainly consisted of various low-frequency (25- to 900-Hz) moans and well-defined sound sequences organized into "song" (20-5000 Hz) recorded with our 2.46-km hydrophone array suspended from the ice. Songs were composed of up to 20 repeated phrases (mean, 10) which lasted up to 146 s (mean, 66.3). Several bowhead whales often were within acoustic range of the array at once, but usually only one sang at a time. Vocalizations exhibited diurnal peaks of occurrence (0600-0800, 1600-1800 h). Sounds which were located in the horizontal plane had peak source spectrum levels as follows--44 moans: 129-178 dB re: 1 microPa, 1 m (median, 159); 3 garglelike utterances: 152, 155, and 169 dB; 33 songs: 158-189 dB (median, 177), all presumably from different whales. Based on ambient noise levels, measured total propagation loss, and whale sound source levels, our detection of whale sounds was theoretically noise-limited beyond 2.5 km (moans) and beyond 10.7 km (songs), a model supported by actual localizations. This study showed that over much of the shallow Arctic and sub-Arctic waters, underwater communications of the bowhead whale would be limited to much shorter ranges than for other large whales in lower latitude, deep-water regions.

Neural population encoding and decoding of sound source location across sound level in the rabbit inferior colliculus

PubMed Central

Delgutte, Bertrand

2015-01-01

At lower levels of sensory processing, the representation of a stimulus feature in the response of a neural population can vary in complex ways across different stimulus intensities, potentially changing the amount of feature-relevant information in the response. How higher-level neural circuits could implement feature decoding computations that compensate for these intensity-dependent variations remains unclear. Here we focused on neurons in the inferior colliculus (IC) of unanesthetized rabbits, whose firing rates are sensitive to both the azimuthal position of a sound source and its sound level. We found that the azimuth tuning curves of an IC neuron at different sound levels tend to be linear transformations of each other. These transformations could either increase or decrease the mutual information between source azimuth and spike count with increasing level for individual neurons, yet population azimuthal information remained constant across the absolute sound levels tested (35, 50, and 65 dB SPL), as inferred from the performance of a maximum-likelihood neural population decoder. We harnessed evidence of level-dependent linear transformations to reduce the number of free parameters in the creation of an accurate cross-level population decoder of azimuth. Interestingly, this decoder predicts monotonic azimuth tuning curves, broadly sensitive to contralateral azimuths, in neurons at higher levels in the auditory pathway. PMID:26490292

Active noise control using a steerable parametric array loudspeaker.

PubMed

Tanaka, Nobuo; Tanaka, Motoki

2010-06-01

Arguably active noise control enables the sound suppression at the designated control points, while the sound pressure except the targeted locations is likely to augment. The reason is clear; a control source normally radiates the sound omnidirectionally. To cope with this problem, this paper introduces a parametric array loudspeaker (PAL) which produces a spatially focused sound beam due to the attribute of ultrasound used for carrier waves, thereby allowing one to suppress the sound pressure at the designated point without causing spillover in the whole sound field. First the fundamental characteristics of PAL are overviewed. The scattered pressure in the near field contributed by source strength of PAL is then described, which is needed for the design of an active noise control system. Furthermore, the optimal control law for minimizing the sound pressure at control points is derived, the control effect being investigated analytically and experimentally. With a view to tracking a moving target point, a steerable PAL based upon a phased array scheme is presented, with the result that the generation of a moving zone of quiet becomes possible without mechanically rotating the PAL. An experiment is finally conducted, demonstrating the validity of the proposed method.

A mechanism study of sound wave-trapping barriers.

PubMed

Yang, Cheng; Pan, Jie; Cheng, Li

2013-09-01

The performance of a sound barrier is usually degraded if a large reflecting surface is placed on the source side. A wave-trapping barrier (WTB), with its inner surface covered by wedge-shaped structures, has been proposed to confine waves within the area between the barrier and the reflecting surface, and thus improve the performance. In this paper, the deterioration in performance of a conventional sound barrier due to the reflecting surface is first explained in terms of the resonance effect of the trapped modes. At each resonance frequency, a strong and mode-controlled sound field is generated by the noise source both within and in the vicinity outside the region bounded by the sound barrier and the reflecting surface. It is found that the peak sound pressures in the barrier's shadow zone, which correspond to the minimum values in the barrier's insertion loss, are largely determined by the resonance frequencies and by the shapes and losses of the trapped modes. These peak pressures usually result in high sound intensity component impinging normal to the barrier surface near the top. The WTB can alter the sound wave diffraction at the top of the barrier if the wavelengths of the sound wave are comparable or smaller than the dimensions of the wedge. In this case, the modified barrier profile is capable of re-organizing the pressure distribution within the bounded domain and altering the acoustic properties near the top of the sound barrier.

Building Acoustics

NASA Astrophysics Data System (ADS)

Cowan, James

This chapter summarizes and explains key concepts of building acoustics. These issues include the behavior of sound waves in rooms, the most commonly used rating systems for sound and sound control in buildings, the most common noise sources found in buildings, practical noise control methods for these sources, and the specific topic of office acoustics. Common noise issues for multi-dwelling units can be derived from most of the sections of this chapter. Books can be and have been written on each of these topics, so the purpose of this chapter is to summarize this information and provide appropriate resources for further exploration of each topic.

Sound sensitivity of neurons in rat hippocampus during performance of a sound-guided task

PubMed Central

Vinnik, Ekaterina; Honey, Christian; Schnupp, Jan; Diamond, Mathew E.

2012-01-01

To investigate how hippocampal neurons encode sound stimuli, and the conjunction of sound stimuli with the animal's position in space, we recorded from neurons in the CA1 region of hippocampus in rats while they performed a sound discrimination task. Four different sounds were used, two associated with water reward on the right side of the animal and the other two with water reward on the left side. This allowed us to separate neuronal activity related to sound identity from activity related to response direction. To test the effect of spatial context on sound coding, we trained rats to carry out the task on two identical testing platforms at different locations in the same room. Twenty-one percent of the recorded neurons exhibited sensitivity to sound identity, as quantified by the difference in firing rate for the two sounds associated with the same response direction. Sensitivity to sound identity was often observed on only one of the two testing platforms, indicating an effect of spatial context on sensory responses. Forty-three percent of the neurons were sensitive to response direction, and the probability that any one neuron was sensitive to response direction was statistically independent from its sensitivity to sound identity. There was no significant coding for sound identity when the rats heard the same sounds outside the behavioral task. These results suggest that CA1 neurons encode sound stimuli, but only when those sounds are associated with actions. PMID:22219030

Effects of hydrokinetic turbine sound on the behavior of four species of fish within an experimental mesocosm

DOE PAGES

Schramm, Michael P.; Bevelhimer, Mark; Scherelis, Constantin

2017-02-04

The development of hydrokinetic energy technologies (e.g., tidal turbines) has raised concern over the potential impacts of underwater sound produced by hydrokinetic turbines on fish species likely to encounter these turbines. To assess the potential for behavioral impacts, we exposed four species of fish to varying intensities of recorded hydrokinetic turbine sound in a semi-natural environment. Although we tested freshwater species (redhorse suckers [Moxostoma spp], freshwater drum [Aplondinotus grunniens], largemouth bass [Micropterus salmoides], and rainbow trout [Oncorhynchus mykiss]), these species are also representative of the hearing physiology and sensitivity of estuarine species that would be affected at tidal energy sites.more » Here, we evaluated changes in fish position relative to different intensities of turbine sound as well as trends in location over time with linear mixed-effects and generalized additive mixed models. We also evaluated changes in the proportion of near-source detections relative to sound intensity and exposure time with generalized linear mixed models and generalized additive models. Models indicated that redhorse suckers may respond to sustained turbine sound by increasing distance from the sound source. Freshwater drum models suggested a mixed response to turbine sound, and largemouth bass and rainbow trout models did not indicate any likely responses to turbine sound. Lastly, findings highlight the importance for future research to utilize accurate localization systems, different species, validated sound transmission distances, and to consider different types of behavioral responses to different turbine designs and to the cumulative sound of arrays of multiple turbines.« less

Effects of hydrokinetic turbine sound on the behavior of four species of fish within an experimental mesocosm

DOE Office of Scientific and Technical Information (OSTI.GOV)

Schramm, Michael P.; Bevelhimer, Mark; Scherelis, Constantin

The development of hydrokinetic energy technologies (e.g., tidal turbines) has raised concern over the potential impacts of underwater sound produced by hydrokinetic turbines on fish species likely to encounter these turbines. To assess the potential for behavioral impacts, we exposed four species of fish to varying intensities of recorded hydrokinetic turbine sound in a semi-natural environment. Although we tested freshwater species (redhorse suckers [Moxostoma spp], freshwater drum [Aplondinotus grunniens], largemouth bass [Micropterus salmoides], and rainbow trout [Oncorhynchus mykiss]), these species are also representative of the hearing physiology and sensitivity of estuarine species that would be affected at tidal energy sites.more » Here, we evaluated changes in fish position relative to different intensities of turbine sound as well as trends in location over time with linear mixed-effects and generalized additive mixed models. We also evaluated changes in the proportion of near-source detections relative to sound intensity and exposure time with generalized linear mixed models and generalized additive models. Models indicated that redhorse suckers may respond to sustained turbine sound by increasing distance from the sound source. Freshwater drum models suggested a mixed response to turbine sound, and largemouth bass and rainbow trout models did not indicate any likely responses to turbine sound. Lastly, findings highlight the importance for future research to utilize accurate localization systems, different species, validated sound transmission distances, and to consider different types of behavioral responses to different turbine designs and to the cumulative sound of arrays of multiple turbines.« less

On Identifying the Sound Sources in a Turbulent Flow

NASA Technical Reports Server (NTRS)

Goldstein, M. E.

2008-01-01

A space-time filtering approach is used to divide an unbounded turbulent flow into its radiating and non-radiating components. The result is then used to clarify a number of issues including the possibility of identifying the sources of the sound in such flows. It is also used to investigate the efficacy of some of the more recent computational approaches.

The sound field of a rotating dipole in a plug flow.

PubMed

Wang, Zhao-Huan; Belyaev, Ivan V; Zhang, Xiao-Zheng; Bi, Chuan-Xing; Faranosov, Georgy A; Dowell, Earl H

2018-04-01

An analytical far field solution for a rotating point dipole source in a plug flow is derived. The shear layer of the jet is modelled as an infinitely thin cylindrical vortex sheet and the far field integral is calculated by the stationary phase method. Four numerical tests are performed to validate the derived solution as well as to assess the effects of sound refraction from the shear layer. First, the calculated results using the derived formulations are compared with the known solution for a rotating dipole in a uniform flow to validate the present model in this fundamental test case. After that, the effects of sound refraction for different rotating dipole sources in the plug flow are assessed. Then the refraction effects on different frequency components of the signal at the observer position, as well as the effects of the motion of the source and of the type of source are considered. Finally, the effect of different sound speeds and densities outside and inside the plug flow is investigated. The solution obtained may be of particular interest for propeller and rotor noise measurements in open jet anechoic wind tunnels.

A Robust Sound Source Localization Approach for Microphone Array with Model Errors

NASA Astrophysics Data System (ADS)

Xiao, Hua; Shao, Huai-Zong; Peng, Qi-Cong

In this paper, a robust sound source localization approach is proposed. The approach retains good performance even when model errors exist. Compared with previous work in this field, the contributions of this paper are as follows. First, an improved broad-band and near-field array model is proposed. It takes array gain, phase perturbations into account and is based on the actual positions of the elements. It can be used in arbitrary planar geometry arrays. Second, a subspace model errors estimation algorithm and a Weighted 2-Dimension Multiple Signal Classification (W2D-MUSIC) algorithm are proposed. The subspace model errors estimation algorithm estimates unknown parameters of the array model, i. e., gain, phase perturbations, and positions of the elements, with high accuracy. The performance of this algorithm is improved with the increasing of SNR or number of snapshots. The W2D-MUSIC algorithm based on the improved array model is implemented to locate sound sources. These two algorithms compose the robust sound source approach. The more accurate steering vectors can be provided for further processing such as adaptive beamforming algorithm. Numerical examples confirm effectiveness of this proposed approach.

Acoustic Imaging of Snowpack Physical Properties

NASA Astrophysics Data System (ADS)

Kinar, N. J.; Pomeroy, J. W.

2011-12-01

Measurements of snowpack depth, density, structure and temperature have often been conducted by the use of snowpits and invasive measurement devices. Previous research has shown that acoustic waves passing through snow are capable of measuring these properties. An experimental observation device (SAS2, System for the Acoustic Sounding of Snow) was used to autonomously send audible sound waves into the top of the snowpack and to receive and process the waves reflected from the interior and bottom of the snowpack. A loudspeaker and microphone array separated by an offset distance was suspended in the air above the surface of the snowpack. Sound waves produced from a loudspeaker as frequency-swept sequences and maximum length sequences were used as source signals. Up to 24 microphones measured the audible signal from the snowpack. The signal-to-noise ratio was compared between sequences in the presence of environmental noise contributed by wind and reflections from vegetation. Beamforming algorithms were used to reject spurious reflections and to compensate for movement of the sensor assembly during the time of data collection. A custom-designed circuit with digital signal processing hardware implemented an inversion algorithm to relate the reflected sound wave data to snowpack physical properties and to create a two-dimensional image of snowpack stratigraphy. The low power consumption circuit was powered by batteries and through WiFi and Bluetooth interfaces enabled the display of processed data on a mobile device. Acoustic observations were logged to an SD card after each measurement. The SAS2 system was deployed at remote field locations in the Rocky Mountains of Alberta, Canada. Acoustic snow properties data was compared with data collected from gravimetric sampling, thermocouple arrays, radiometers and snowpit observations of density, stratigraphy and crystal structure. Aspects for further research and limitations of the acoustic sensing system are also discussed.

High resolution 1H NMR-based metabonomic study of the auditory cortex analogue of developing chick (Gallus gallus domesticus) following prenatal chronic loud music and noise exposure.

PubMed

Kumar, Vivek; Nag, Tapas Chandra; Sharma, Uma; Mewar, Sujeet; Jagannathan, Naranamangalam R; Wadhwa, Shashi

2014-10-01

Proper functional development of the auditory cortex (ACx) critically depends on early relevant sensory experiences. Exposure to high intensity noise (industrial/traffic) and music, a current public health concern, may disrupt the proper development of the ACx and associated behavior. The biochemical mechanisms associated with such activity dependent changes during development are poorly understood. Here we report the effects of prenatal chronic (last 10 days of incubation), 110dB sound pressure level (SPL) music and noise exposure on metabolic profile of the auditory cortex analogue/field L (AuL) in domestic chicks. Perchloric acid extracts of AuL of post hatch day 1 chicks from control, music and noise groups were subjected to high resolution (700MHz) (1)H NMR spectroscopy. Multivariate regression analysis of the concentration data of 18 metabolites revealed a significant class separation between control and loud sound exposed groups, indicating a metabolic perturbation. Comparison of absolute concentration of metabolites showed that overstimulation with loud sound, independent of spectral characteristics (music or noise) led to extensive usage of major energy metabolites, e.g., glucose, β-hydroxybutyrate and ATP. On the other hand, high glutamine levels and sustained levels of neuromodulators and alternate energy sources, e.g., creatine, ascorbate and lactate indicated a systems restorative measure in a condition of neuronal hyperactivity. At the same time, decreased aspartate and taurine levels in the noise group suggested a differential impact of prenatal chronic loud noise over music exposure. Thus prenatal exposure to loud sound especially noise alters the metabolic activity in the AuL which in turn can affect the functional development and later auditory associated behaviour. Copyright © 2014 Elsevier Ltd. All rights reserved.

Different spatio-temporal electroencephalography features drive the successful decoding of binaural and monaural cues for sound localization.

PubMed

Bednar, Adam; Boland, Francis M; Lalor, Edmund C

2017-03-01

The human ability to localize sound is essential for monitoring our environment and helps us to analyse complex auditory scenes. Although the acoustic cues mediating sound localization have been established, it remains unknown how these cues are represented in human cortex. In particular, it is still a point of contention whether binaural and monaural cues are processed by the same or distinct cortical networks. In this study, participants listened to a sequence of auditory stimuli from different spatial locations while we recorded their neural activity using electroencephalography (EEG). The stimuli were presented over a loudspeaker array, which allowed us to deliver realistic, free-field stimuli in both the horizontal and vertical planes. Using a multivariate classification approach, we showed that it is possible to decode sound source location from scalp-recorded EEG. Robust and consistent decoding was shown for stimuli that provide binaural cues (i.e. Left vs. Right stimuli). Decoding location when only monaural cues were available (i.e. Front vs. Rear and elevational stimuli) was successful for a subset of subjects and showed less consistency. Notably, the spatio-temporal pattern of EEG features that facilitated decoding differed based on the availability of binaural and monaural cues. In particular, we identified neural processing of binaural cues at around 120 ms post-stimulus and found that monaural cues are processed later between 150 and 200 ms. Furthermore, different spatial activation patterns emerged for binaural and monaural cue processing. These spatio-temporal dissimilarities suggest the involvement of separate cortical mechanisms in monaural and binaural acoustic cue processing. © 2017 Federation of European Neuroscience Societies and John Wiley & Sons Ltd.

Classification of large acoustic datasets using machine learning and crowdsourcing: application to whale calls.

PubMed

Shamir, Lior; Yerby, Carol; Simpson, Robert; von Benda-Beckmann, Alexander M; Tyack, Peter; Samarra, Filipa; Miller, Patrick; Wallin, John

2014-02-01

Vocal communication is a primary communication method of killer and pilot whales, and is used for transmitting a broad range of messages and information for short and long distance. The large variation in call types of these species makes it challenging to categorize them. In this study, sounds recorded by audio sensors carried by ten killer whales and eight pilot whales close to the coasts of Norway, Iceland, and the Bahamas were analyzed using computer methods and citizen scientists as part of the Whale FM project. Results show that the computer analysis automatically separated the killer whales into Icelandic and Norwegian whales, and the pilot whales were separated into Norwegian long-finned and Bahamas short-finned pilot whales, showing that at least some whales from these two locations have different acoustic repertoires that can be sensed by the computer analysis. The citizen science analysis was also able to separate the whales to locations by their sounds, but the separation was somewhat less accurate compared to the computer method.

Device for recording the 20 Hz - 200 KHz sound frequency spectrum using teletransmission

NASA Technical Reports Server (NTRS)

Baciu, I.

1974-01-01

The device described consists of two distinct parts: (1) The sound pickup system consisting of the wide-frequency band condenser microphone which contains in the same assembly the frequency-modulated oscillator and the output stage. Being transistorized and small, this system can be easily moved, so that sounds can be picked up even in places that are difficult to reach with larger devices. (2) The receiving and recording part is separate and can be at a great distance from the sound pickup system. This part contains a 72 MHz input stage, a frequency changer that gives an intermediate frequency of 30 MHz and a multichannel analyzer coupled to an oscilloscope and a recorder.

Active Exhaust Silencing Systen For the Management of Auxillary Power Unit Sound Signatures

DTIC Science & Technology

2014-08-01

conceptual mass-less pistons are introduced into the system before and after the injection site, such that they will move exactly with the plane wave...Unit Sound Signatures, Helminen, et al. Page 2 of 7 either the primary source or the injected source. It is assumed that the pistons are ‘close...source, it causes both pistons to move identically. The pressures induced by the flow on the pistons do not affect the flow generated by the

The rotary subwoofer: a controllable infrasound source.

PubMed

Park, Joseph; Garcés, Milton; Thigpen, Bruce

2009-04-01

The rotary subwoofer is a novel acoustic transducer capable of projecting infrasonic signals at high sound pressure levels. The projector produces higher acoustic particle velocities than conventional transducers which translate into higher radiated sound pressure levels. This paper characterizes measured performance of a rotary subwoofer and presents a model to predict sound pressure levels.

Physics of thermo-acoustic sound generation

NASA Astrophysics Data System (ADS)

Daschewski, M.; Boehm, R.; Prager, J.; Kreutzbruck, M.; Harrer, A.

2013-09-01

We present a generalized analytical model of thermo-acoustic sound generation based on the analysis of thermally induced energy density fluctuations and their propagation into the adjacent matter. The model provides exact analytical prediction of the sound pressure generated in fluids and solids; consequently, it can be applied to arbitrary thermal power sources such as thermophones, plasma firings, laser beams, and chemical reactions. Unlike existing approaches, our description also includes acoustic near-field effects and sound-field attenuation. Analytical results are compared with measurements of sound pressures generated by thermo-acoustic transducers in air for frequencies up to 1 MHz. The tested transducers consist of titanium and indium tin oxide coatings on quartz glass and polycarbonate substrates. The model reveals that thermo-acoustic efficiency increases linearly with the supplied thermal power and quadratically with thermal excitation frequency. Comparison of the efficiency of our thermo-acoustic transducers with those of piezoelectric-based airborne ultrasound transducers using impulse excitation showed comparable sound pressure values. The present results show that thermo-acoustic transducers can be applied as broadband, non-resonant, high-performance ultrasound sources.

3-D localization of virtual sound sources: effects of visual environment, pointing method, and training.

PubMed

Majdak, Piotr; Goupell, Matthew J; Laback, Bernhard

2010-02-01

The ability to localize sound sources in three-dimensional space was tested in humans. In Experiment 1, naive subjects listened to noises filtered with subject-specific head-related transfer functions. The tested conditions included the pointing method (head or manual pointing) and the visual environment (VE; darkness or virtual VE). The localization performance was not significantly different between the pointing methods. The virtual VE significantly improved the horizontal precision and reduced the number of front-back confusions. These results show the benefit of using a virtual VE in sound localization tasks. In Experiment 2, subjects were provided with sound localization training. Over the course of training, the performance improved for all subjects, with the largest improvements occurring during the first 400 trials. The improvements beyond the first 400 trials were smaller. After the training, there was still no significant effect of pointing method, showing that the choice of either head- or manual-pointing method plays a minor role in sound localization performance. The results of Experiment 2 reinforce the importance of perceptual training for at least 400 trials in sound localization studies.

Optical and Acoustic Sensor-Based 3D Ball Motion Estimation for Ball Sport Simulators †.

PubMed

Seo, Sang-Woo; Kim, Myunggyu; Kim, Yejin

2018-04-25

Estimation of the motion of ball-shaped objects is essential for the operation of ball sport simulators. In this paper, we propose an estimation system for 3D ball motion, including speed and angle of projection, by using acoustic vector and infrared (IR) scanning sensors. Our system is comprised of three steps to estimate a ball motion: sound-based ball firing detection, sound source localization, and IR scanning for motion analysis. First, an impulsive sound classification based on the mel-frequency cepstrum and feed-forward neural network is introduced to detect the ball launch sound. An impulsive sound source localization using a 2D microelectromechanical system (MEMS) microphones and delay-and-sum beamforming is presented to estimate the firing position. The time and position of a ball in 3D space is determined from a high-speed infrared scanning method. Our experimental results demonstrate that the estimation of ball motion based on sound allows a wider activity area than similar camera-based methods. Thus, it can be practically applied to various simulations in sports such as soccer and baseball.

Turbine Sound May Influence the Metamorphosis Behaviour of Estuarine Crab Megalopae

PubMed Central

Pine, Matthew K.; Jeffs, Andrew G.; Radford, Craig A.

2012-01-01

It is now widely accepted that a shift towards renewable energy production is needed in order to avoid further anthropogenically induced climate change. The ocean provides a largely untapped source of renewable energy. As a result, harvesting electrical power from the wind and tides has sparked immense government and commercial interest but with relatively little detailed understanding of the potential environmental impacts. This study investigated how the sound emitted from an underwater tidal turbine and an offshore wind turbine would influence the settlement and metamorphosis of the pelagic larvae of estuarine brachyuran crabs which are ubiquitous in most coastal habitats. In a laboratory experiment the median time to metamorphosis (TTM) for the megalopae of the crabs Austrohelice crassa and Hemigrapsus crenulatus was significantly increased by at least 18 h when exposed to either tidal turbine or sea-based wind turbine sound, compared to silent control treatments. Contrastingly, when either species were subjected to natural habitat sound, observed median TTM decreased by approximately 21–31% compared to silent control treatments, 38–47% compared to tidal turbine sound treatments, and 46–60% compared to wind turbine sound treatments. A lack of difference in median TTM in A. crassa between two different source levels of tidal turbine sound suggests the frequency composition of turbine sound is more relevant in explaining such responses rather than sound intensity. These results show that estuarine mudflat sound mediates natural metamorphosis behaviour in two common species of estuarine crabs, and that exposure to continuous turbine sound interferes with this natural process. These results raise concerns about the potential ecological impacts of sound generated by renewable energy generation systems placed in the nearshore environment. PMID:23240063

Turbine sound may influence the metamorphosis behaviour of estuarine crab megalopae.

PubMed

Pine, Matthew K; Jeffs, Andrew G; Radford, Craig A

2012-01-01

It is now widely accepted that a shift towards renewable energy production is needed in order to avoid further anthropogenically induced climate change. The ocean provides a largely untapped source of renewable energy. As a result, harvesting electrical power from the wind and tides has sparked immense government and commercial interest but with relatively little detailed understanding of the potential environmental impacts. This study investigated how the sound emitted from an underwater tidal turbine and an offshore wind turbine would influence the settlement and metamorphosis of the pelagic larvae of estuarine brachyuran crabs which are ubiquitous in most coastal habitats. In a laboratory experiment the median time to metamorphosis (TTM) for the megalopae of the crabs Austrohelice crassa and Hemigrapsus crenulatus was significantly increased by at least 18 h when exposed to either tidal turbine or sea-based wind turbine sound, compared to silent control treatments. Contrastingly, when either species were subjected to natural habitat sound, observed median TTM decreased by approximately 21-31% compared to silent control treatments, 38-47% compared to tidal turbine sound treatments, and 46-60% compared to wind turbine sound treatments. A lack of difference in median TTM in A. crassa between two different source levels of tidal turbine sound suggests the frequency composition of turbine sound is more relevant in explaining such responses rather than sound intensity. These results show that estuarine mudflat sound mediates natural metamorphosis behaviour in two common species of estuarine crabs, and that exposure to continuous turbine sound interferes with this natural process. These results raise concerns about the potential ecological impacts of sound generated by renewable energy generation systems placed in the nearshore environment.