Sound Source Identification Through Flow Density Measurement and Correlation With Far Field Noise

NASA Technical Reports Server (NTRS)

Panda, J.; Seasholtz, R. G.

2001-01-01

Sound sources in the plumes of unheated round jets, in the Mach number range 0.6 to 1.8, were investigated experimentally using "casuality" approach, where air density fluctuations in the plumes were correlated with the far field noise. The air density was measured using a newly developed Molecular Rayleigh scattering based technique, which did not require any seeding. The reference at the end provides a detailed description of the measurement technique.

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

In-duct identification of a rotating sound source with high spatial resolution

NASA Astrophysics Data System (ADS)

Heo, Yong-Ho; Ih, Jeong-Guon; Bodén, Hans

2015-11-01

To understand and reduce the flow noise generation from in-duct fluid machines, it is necessary to identify the acoustic source characteristics precisely. In this work, a source identification technique, which can identify the strengths and positions of the major sound radiators in the source plane, is studied for an in-duct rotating source. A linear acoustic theory including the effects of evanescent modes and source rotation is formulated based on the modal summation method, which is the underlying theory for the inverse source reconstruction. A validation experiment is conducted on a duct system excited by a loudspeaker in static and rotating conditions, with two different speeds, in the absence of flow. Due to the source rotation, the measured pressure spectra reveal the Doppler effect, and the amount of frequency shift corresponds to the multiplication of the circumferential mode order and the rotation speed. Amplitudes of participating modes are estimated at the shifted frequencies in the stationary reference frame, and the modal amplitude set including the effect of source rotation is collected to investigate the source behavior in the rotating reference frame. By using the estimated modal amplitudes, the near-field pressure is re-calculated and compared with the measured pressure. The obtained maximum relative error is about -25 and -10 dB for rotation speeds at 300 and 600 rev/min, respectively. The spatial distribution of acoustic source parameters is restored from the estimated modal amplitude set. The result clearly shows that the position and magnitude of the main sound source can be identified with high spatial resolution in the rotating reference frame.

Perceptually Salient Regions of the Modulation Power Spectrum for Musical Instrument Identification.

PubMed

Thoret, Etienne; Depalle, Philippe; McAdams, Stephen

2017-01-01

The ability of a listener to recognize sound sources, and in particular musical instruments from the sounds they produce, raises the question of determining the acoustical information used to achieve such a task. It is now well known that the shapes of the temporal and spectral envelopes are crucial to the recognition of a musical instrument. More recently, Modulation Power Spectra (MPS) have been shown to be a representation that potentially explains the perception of musical instrument sounds. Nevertheless, the question of which specific regions of this representation characterize a musical instrument is still open. An identification task was applied to two subsets of musical instruments: tuba, trombone, cello, saxophone, and clarinet on the one hand, and marimba, vibraphone, guitar, harp, and viola pizzicato on the other. The sounds were processed with filtered spectrotemporal modulations with 2D Gaussian windows. The most relevant regions of this representation for instrument identification were determined for each instrument and reveal the regions essential for their identification. The method used here is based on a "molecular approach," the so-called bubbles method. Globally, the instruments were correctly identified and the lower values of spectrotemporal modulations are the most important regions of the MPS for recognizing instruments. Interestingly, instruments that were confused with each other led to non-overlapping regions and were confused when they were filtered in the most salient region of the other instrument. These results suggest that musical instrument timbres are characterized by specific spectrotemporal modulations, information which could contribute to music information retrieval tasks such as automatic source recognition.

Noise source separation of diesel engine by combining binaural sound localization method and blind source separation method

NASA Astrophysics Data System (ADS)

Yao, Jiachi; Xiang, Yang; Qian, Sichong; Li, Shengyang; Wu, Shaowei

2017-11-01

In order to separate and identify the combustion noise and the piston slap noise of a diesel engine, a noise source separation and identification method that combines a binaural sound localization method and blind source separation method is proposed. During a diesel engine noise and vibration test, because a diesel engine has many complex noise sources, a lead covering method was carried out on a diesel engine to isolate other interference noise from the No. 1-5 cylinders. Only the No. 6 cylinder parts were left bare. Two microphones that simulated the human ears were utilized to measure the radiated noise signals 1 m away from the diesel engine. First, a binaural sound localization method was adopted to separate the noise sources that are in different places. Then, for noise sources that are in the same place, a blind source separation method is utilized to further separate and identify the noise sources. Finally, a coherence function method, continuous wavelet time-frequency analysis method, and prior knowledge of the diesel engine are combined to further identify the separation results. The results show that the proposed method can effectively separate and identify the combustion noise and the piston slap noise of a diesel engine. The frequency of the combustion noise and the piston slap noise are respectively concentrated at 4350 Hz and 1988 Hz. Compared with the blind source separation method, the proposed method has superior separation and identification effects, and the separation results have fewer interference components from other noise.

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.

Pollutant source identification model for water pollution incidents in small straight rivers based on genetic algorithm

NASA Astrophysics Data System (ADS)

Zhang, Shou-ping; Xin, Xiao-kang

2017-07-01

Identification of pollutant sources for river pollution incidents is an important and difficult task in the emergency rescue, and an intelligent optimization method can effectively compensate for the weakness of traditional methods. An intelligent model for pollutant source identification has been established using the basic genetic algorithm (BGA) as an optimization search tool and applying an analytic solution formula of one-dimensional unsteady water quality equation to construct the objective function. Experimental tests show that the identification model is effective and efficient: the model can accurately figure out the pollutant amounts or positions no matter single pollution source or multiple sources. Especially when the population size of BGA is set as 10, the computing results are sound agree with analytic results for a single source amount and position identification, the relative errors are no more than 5 %. For cases of multi-point sources and multi-variable, there are some errors in computing results for the reasons that there exist many possible combinations of the pollution sources. But, with the help of previous experience to narrow the search scope, the relative errors of the identification results are less than 5 %, which proves the established source identification model can be used to direct emergency responses.

Computational Acoustic Beamforming for Noise Source Identification for Small Wind Turbines.

PubMed

Ma, Ping; Lien, Fue-Sang; Yee, Eugene

2017-01-01

This paper develops a computational acoustic beamforming (CAB) methodology for identification of sources of small wind turbine noise. This methodology is validated using the case of the NACA 0012 airfoil trailing edge noise. For this validation case, the predicted acoustic maps were in excellent conformance with the results of the measurements obtained from the acoustic beamforming experiment. Following this validation study, the CAB methodology was applied to the identification of noise sources generated by a commercial small wind turbine. The simulated acoustic maps revealed that the blade tower interaction and the wind turbine nacelle were the two primary mechanisms for sound generation for this small wind turbine at frequencies between 100 and 630 Hz.

Automated lung sound analysis for detecting pulmonary abnormalities.

PubMed

Datta, Shreyasi; Dutta Choudhury, Anirban; Deshpande, Parijat; Bhattacharya, Sakyajit; Pal, Arpan

2017-07-01

Identification of pulmonary diseases comprises of accurate auscultation as well as elaborate and expensive pulmonary function tests. Prior arts have shown that pulmonary diseases lead to abnormal lung sounds such as wheezes and crackles. This paper introduces novel spectral and spectrogram features, which are further refined by Maximal Information Coefficient, leading to the classification of healthy and abnormal lung sounds. A balanced lung sound dataset, consisting of publicly available data and data collected with a low-cost in-house digital stethoscope are used. The performance of the classifier is validated over several randomly selected non-overlapping training and validation samples and tested on separate subjects for two separate test cases: (a) overlapping and (b) non-overlapping data sources in training and testing. The results reveal that the proposed method sustains an accuracy of 80% even for non-overlapping data sources in training and testing.

Assessing sound exposure from shipping in coastal waters using a single hydrophone and Automatic Identification System (AIS) data.

PubMed

Merchant, Nathan D; Witt, Matthew J; Blondel, Philippe; Godley, Brendan J; Smith, George H

2012-07-01

Underwater noise from shipping is a growing presence throughout the world's oceans, and may be subjecting marine fauna to chronic noise exposure with potentially severe long-term consequences. The coincidence of dense shipping activity and sensitive marine ecosystems in coastal environments is of particular concern, and noise assessment methodologies which describe the high temporal variability of sound exposure in these areas are needed. We present a method of characterising sound exposure from shipping using continuous passive acoustic monitoring combined with Automatic Identification System (AIS) shipping data. The method is applied to data recorded in Falmouth Bay, UK. Absolute and relative levels of intermittent ship noise contributions to the 24-h sound exposure level are determined using an adaptive threshold, and the spatial distribution of potential ship sources is then analysed using AIS data. This technique can be used to prioritize shipping noise mitigation strategies in coastal marine environments. Copyright © 2012 Elsevier Ltd. All rights reserved.

Computational Acoustic Beamforming for Noise Source Identification for Small Wind Turbines

PubMed Central

Lien, Fue-Sang

2017-01-01

This paper develops a computational acoustic beamforming (CAB) methodology for identification of sources of small wind turbine noise. This methodology is validated using the case of the NACA 0012 airfoil trailing edge noise. For this validation case, the predicted acoustic maps were in excellent conformance with the results of the measurements obtained from the acoustic beamforming experiment. Following this validation study, the CAB methodology was applied to the identification of noise sources generated by a commercial small wind turbine. The simulated acoustic maps revealed that the blade tower interaction and the wind turbine nacelle were the two primary mechanisms for sound generation for this small wind turbine at frequencies between 100 and 630 Hz. PMID:28378012

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-tactile rendering of visual information

NASA Astrophysics Data System (ADS)

Silva, Pubudu Madhawa; Pappas, Thrasyvoulos N.; Atkins, Joshua; West, James E.; Hartmann, William M.

2012-03-01

In previous work, we have proposed a dynamic, interactive system for conveying visual information via hearing and touch. The system is implemented with a touch screen that allows the user to interrogate a two-dimensional (2-D) object layout by active finger scanning while listening to spatialized auditory feedback. Sound is used as the primary source of information for object localization and identification, while touch is used both for pointing and for kinesthetic feedback. Our previous work considered shape and size perception of simple objects via hearing and touch. The focus of this paper is on the perception of a 2-D layout of simple objects with identical size and shape. We consider the selection and rendition of sounds for object identification and localization. We rely on the head-related transfer function for rendering sound directionality, and consider variations of sound intensity and tempo as two alternative approaches for rendering proximity. Subjective experiments with visually-blocked subjects are used to evaluate the effectiveness of the proposed approaches. Our results indicate that intensity outperforms tempo as a proximity cue, and that the overall system for conveying a 2-D layout is quite promising.

Sensing of Particular Speakers for the Construction of Voice Interface Utilized in Noisy Environment

NASA Astrophysics Data System (ADS)

Sawada, Hideyuki; Ohkado, Minoru

Human is able to exchange information smoothly using voice under different situations such as noisy environment in a crowd and with the existence of plural speakers. We are able to detect the position of a source sound in 3D space, extract a particular sound from mixed sounds, and recognize who is talking. By realizing this mechanism with a computer, new applications will be presented for recording a sound with high quality by reducing noise, presenting a clarified sound, and realizing a microphone-free speech recognition by extracting particular sound. The paper will introduce a realtime detection and identification of particular speaker in noisy environment using a microphone array based on the location of a speaker and the individual voice characteristics. The study will be applied to develop an adaptive auditory system of a mobile robot which collaborates with a factory worker.

Improving the Performance of an Auditory Brain-Computer Interface Using Virtual Sound Sources by Shortening Stimulus Onset Asynchrony

PubMed Central

Sugi, Miho; Hagimoto, Yutaka; Nambu, Isao; Gonzalez, Alejandro; Takei, Yoshinori; Yano, Shohei; Hokari, Haruhide; Wada, Yasuhiro

2018-01-01

Recently, a brain-computer interface (BCI) using virtual sound sources has been proposed for estimating user intention via electroencephalogram (EEG) in an oddball task. However, its performance is still insufficient for practical use. In this study, we examine the impact that shortening the stimulus onset asynchrony (SOA) has on this auditory BCI. While very short SOA might improve its performance, sound perception and task performance become difficult, and event-related potentials (ERPs) may not be induced if the SOA is too short. Therefore, we carried out behavioral and EEG experiments to determine the optimal SOA. In the experiments, participants were instructed to direct attention to one of six virtual sounds (target direction). We used eight different SOA conditions: 200, 300, 400, 500, 600, 700, 800, and 1,100 ms. In the behavioral experiment, we recorded participant behavioral responses to target direction and evaluated recognition performance of the stimuli. In all SOA conditions, recognition accuracy was over 85%, indicating that participants could recognize the target stimuli correctly. Next, using a silent counting task in the EEG experiment, we found significant differences between target and non-target sound directions in all but the 200-ms SOA condition. When we calculated an identification accuracy using Fisher discriminant analysis (FDA), the SOA could be shortened by 400 ms without decreasing the identification accuracies. Thus, improvements in performance (evaluated by BCI utility) could be achieved. On average, higher BCI utilities were obtained in the 400 and 500-ms SOA conditions. Thus, auditory BCI performance can be optimized for both behavioral and neurophysiological responses by shortening the SOA. PMID:29535602

33 CFR 164.43 - Automatic Identification System Shipborne Equipment-Prince William Sound.

Code of Federal Regulations, 2012 CFR

2012-07-01

... 33 Navigation and Navigable Waters 2 2012-07-01 2012-07-01 false Automatic Identification System Shipborne Equipment-Prince William Sound. 164.43 Section 164.43 Navigation and Navigable Waters COAST GUARD... Automatic Identification System Shipborne Equipment—Prince William Sound. (a) Until December 31, 2004, each...

33 CFR 164.43 - Automatic Identification System Shipborne Equipment-Prince William Sound.

Code of Federal Regulations, 2013 CFR

2013-07-01

... 33 Navigation and Navigable Waters 2 2013-07-01 2013-07-01 false Automatic Identification System Shipborne Equipment-Prince William Sound. 164.43 Section 164.43 Navigation and Navigable Waters COAST GUARD... Automatic Identification System Shipborne Equipment—Prince William Sound. (a) Until December 31, 2004, each...

33 CFR 164.43 - Automatic Identification System Shipborne Equipment-Prince William Sound.

Code of Federal Regulations, 2014 CFR

2014-07-01

... 33 Navigation and Navigable Waters 2 2014-07-01 2014-07-01 false Automatic Identification System Shipborne Equipment-Prince William Sound. 164.43 Section 164.43 Navigation and Navigable Waters COAST GUARD... Automatic Identification System Shipborne Equipment—Prince William Sound. (a) Until December 31, 2004, each...

Complete de-Dopplerization and acoustic holography for external noise of a high-speed train.

PubMed

Yang, Diange; Wen, Junjie; Miao, Feng; Wang, Ziteng; Gu, Xiaoan; Lian, Xiaomin

2016-09-01

Identification and measurement of moving sound sources are the bases for vehicle noise control. Acoustic holography has been applied in successfully identifying the moving sound source since the 1990s. However, due to the high demand for the accuracy of holographic data, currently the maximum velocity achieved by acoustic holography is just above 100 km/h. The objective of this study was to establish a method based on the complete Morse acoustic model to restore the measured signal in high-speed situations, and to propose a far-field acoustic holography method applicable for high-speed moving sound sources. Simulated comparisons of the proposed far-field acoustic holography with complete Morse model, the acoustic holography with simplified Morse model and traditional delay-and-sum beamforming were conducted. Experiments with a high-speed train running at the speed of 278 km/h validated the proposed far-field acoustic holography. This study extended the applications of acoustic holography to high-speed situations and established the basis for quantitative measurements of far-field acoustic holography.

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.

Identification of environmental sounds and melodies in syndromes of anterior temporal lobe degeneration.

PubMed

Golden, Hannah L; Downey, Laura E; Fletcher, Philip D; Mahoney, Colin J; Schott, Jonathan M; Mummery, Catherine J; Crutch, Sebastian J; Warren, Jason D

2015-05-15

Recognition of nonverbal sounds in semantic dementia and other syndromes of anterior temporal lobe degeneration may determine clinical symptoms and help to define phenotypic profiles. However, nonverbal auditory semantic function has not been widely studied in these syndromes. Here we investigated semantic processing in two key nonverbal auditory domains - environmental sounds and melodies - in patients with semantic dementia (SD group; n=9) and in patients with anterior temporal lobe atrophy presenting with behavioural decline (TL group; n=7, including four cases with MAPT mutations) in relation to healthy older controls (n=20). We assessed auditory semantic performance in each domain using novel, uniform within-modality neuropsychological procedures that determined sound identification based on semantic classification of sound pairs. Both the SD and TL groups showed comparable overall impairments of environmental sound and melody identification; individual patients generally showed superior identification of environmental sounds than melodies, however relative sparing of melody over environmental sound identification also occurred in both groups. Our findings suggest that nonverbal auditory semantic impairment is a common feature of neurodegenerative syndromes with anterior temporal lobe atrophy. However, the profile of auditory domain involvement varies substantially between individuals. Copyright © 2015. Published by Elsevier B.V.

Speech endpoint detection with non-language speech sounds for generic speech processing applications

NASA Astrophysics Data System (ADS)

McClain, Matthew; Romanowski, Brian

2009-05-01

Non-language speech sounds (NLSS) are sounds produced by humans that do not carry linguistic information. Examples of these sounds are coughs, clicks, breaths, and filled pauses such as "uh" and "um" in English. NLSS are prominent in conversational speech, but can be a significant source of errors in speech processing applications. Traditionally, these sounds are ignored by speech endpoint detection algorithms, where speech regions are identified in the audio signal prior to processing. The ability to filter NLSS as a pre-processing step can significantly enhance the performance of many speech processing applications, such as speaker identification, language identification, and automatic speech recognition. In order to be used in all such applications, NLSS detection must be performed without the use of language models that provide knowledge of the phonology and lexical structure of speech. This is especially relevant to situations where the languages used in the audio are not known apriori. We present the results of preliminary experiments using data from American and British English speakers, in which segments of audio are classified as language speech sounds (LSS) or NLSS using a set of acoustic features designed for language-agnostic NLSS detection and a hidden-Markov model (HMM) to model speech generation. The results of these experiments indicate that the features and model used are capable of detection certain types of NLSS, such as breaths and clicks, while detection of other types of NLSS such as filled pauses will require future research.

Development of Sound Localization Strategies in Children with Bilateral Cochlear Implants

PubMed Central

Zheng, Yi; Godar, Shelly P.; Litovsky, Ruth Y.

2015-01-01

Localizing sounds in our environment is one of the fundamental perceptual abilities that enable humans to communicate, and to remain safe. Because the acoustic cues necessary for computing source locations consist of differences between the two ears in signal intensity and arrival time, sound localization is fairly poor when a single ear is available. In adults who become deaf and are fitted with cochlear implants (CIs) sound localization is known to improve when bilateral CIs (BiCIs) are used compared to when a single CI is used. The aim of the present study was to investigate the emergence of spatial hearing sensitivity in children who use BiCIs, with a particular focus on the development of behavioral localization patterns when stimuli are presented in free-field horizontal acoustic space. A new analysis was implemented to quantify patterns observed in children for mapping acoustic space to a spatially relevant perceptual representation. Children with normal hearing were found to distribute their responses in a manner that demonstrated high spatial sensitivity. In contrast, children with BiCIs tended to classify sound source locations to the left and right; with increased bilateral hearing experience, they developed a perceptual map of space that was better aligned with the acoustic space. The results indicate experience-dependent refinement of spatial hearing skills in children with CIs. Localization strategies appear to undergo transitions from sound source categorization strategies to more fine-grained location identification strategies. This may provide evidence for neural plasticity, with implications for training of spatial hearing ability in CI users. PMID:26288142

Development of Sound Localization Strategies in Children with Bilateral Cochlear Implants.

PubMed

Zheng, Yi; Godar, Shelly P; Litovsky, Ruth Y

2015-01-01

Localizing sounds in our environment is one of the fundamental perceptual abilities that enable humans to communicate, and to remain safe. Because the acoustic cues necessary for computing source locations consist of differences between the two ears in signal intensity and arrival time, sound localization is fairly poor when a single ear is available. In adults who become deaf and are fitted with cochlear implants (CIs) sound localization is known to improve when bilateral CIs (BiCIs) are used compared to when a single CI is used. The aim of the present study was to investigate the emergence of spatial hearing sensitivity in children who use BiCIs, with a particular focus on the development of behavioral localization patterns when stimuli are presented in free-field horizontal acoustic space. A new analysis was implemented to quantify patterns observed in children for mapping acoustic space to a spatially relevant perceptual representation. Children with normal hearing were found to distribute their responses in a manner that demonstrated high spatial sensitivity. In contrast, children with BiCIs tended to classify sound source locations to the left and right; with increased bilateral hearing experience, they developed a perceptual map of space that was better aligned with the acoustic space. The results indicate experience-dependent refinement of spatial hearing skills in children with CIs. Localization strategies appear to undergo transitions from sound source categorization strategies to more fine-grained location identification strategies. This may provide evidence for neural plasticity, with implications for training of spatial hearing ability in CI users.

Sound-direction identification with bilateral cochlear implants.

PubMed

Neuman, Arlene C; Haravon, Anita; Sislian, Nicole; Waltzman, Susan B

2007-02-01

The purpose of this study was to compare the accuracy of sound-direction identification in the horizontal plane by bilateral cochlear implant users when localization was measured with pink noise and with speech stimuli. Eight adults who were bilateral users of Nucleus 24 Contour devices participated in the study. All had received implants in both ears in a single surgery. Sound-direction identification was measured in a large classroom by using a nine-loudspeaker array. Localization was tested in three listening conditions (bilateral cochlear implants, left cochlear implant, and right cochlear implant), using two different stimuli (a speech stimulus and pink noise bursts) in a repeated-measures design. Sound-direction identification accuracy was significantly better when using two implants than when using a single implant. The mean root-mean-square error was 29 degrees for the bilateral condition, 54 degrees for the left cochlear implant, and 46.5 degrees for the right cochlear implant condition. Unilateral accuracy was similar for right cochlear implant and left cochlear implant performance. Sound-direction identification performance was similar for speech and pink noise stimuli. The data obtained in this study add to the growing body of evidence that sound-direction identification with bilateral cochlear implants is better than with a single implant. The similarity in localization performance obtained with the speech and pink noise supports the use of either stimulus for measuring sound-direction identification.

[Application of the computer-based respiratory sound analysis system based on Mel-frequency cepstral coefficient and dynamic time warping in healthy children].

PubMed

Yan, W Y; Li, L; Yang, Y G; Lin, X L; Wu, J Z

2016-08-01

We designed a computer-based respiratory sound analysis system to identify pediatric normal lung sound. To verify the validity of the computer-based respiratory sound analysis system. First we downloaded the standard lung sounds from the network database (website: http: //www.easyauscultation.com/lung-sounds-reference-guide) and recorded 3 samples of abnormal loud sound (rhonchi, wheeze and crackles) from three patients of The Department of Pediatrics, the First Affiliated Hospital of Xiamen University. We regarded such lung sounds as"reference lung sounds". The"test lung sounds"were recorded from 29 children form Kindergarten of Xiamen University. we recorded lung sound by portable electronic stethoscope and valid lung sounds were selected by manual identification. We introduced Mel-frequency cepstral coefficient (MFCC) to extract lung sound features and dynamic time warping (DTW) for signal classification. We had 39 standard lung sounds, recorded 58 test lung sounds. This computer-based respiratory sound analysis system was carried out in 58 lung sound recognition, correct identification of 52 times, error identification 6 times. Accuracy was 89.7%. Based on MFCC and DTW, our computer-based respiratory sound analysis system can effectively identify healthy lung sounds of children (accuracy can reach 89.7%), fully embodies the reliability of the lung sounds analysis system.

Judging the similarity of soundscapes does not require categorization: evidence from spliced stimuli.

PubMed

Aucouturier, Jean-Julien; Defreville, Boris

2009-04-01

This study uses an audio signal transformation, splicing, to create an experimental situation where human listeners judge the similarity of audio signals, which they cannot easily categorize. Splicing works by segmenting audio signals into 50-ms frames, then shuffling and concatenating these frames back in random order. Splicing a signal masks the identification of the categories that it normally elicits: For instance, human participants cannot easily identify the sound of cars in a spliced recording of a city street. This study compares human performance on both normal and spliced recordings of soundscapes and music. Splicing is found to degrade human similarity performance significantly less for soundscapes than for music: When two spliced soundscapes are judged similar to one another, the original recordings also tend to sound similar. This establishes that humans are capable of reconstructing consistent similarity relations between soundscapes without relying much on the identification of the natural categories associated with such signals, such as their constituent sound sources. This finding contradicts previous literature and points to new ways to conceptualize the different ways in which humans perceive soundscapes and music.

A study of methods to predict and measure the transmission of sound through the walls of light aircraft

NASA Technical Reports Server (NTRS)

Bernhard, R. J.; Bolton, J. S.

1988-01-01

The objectives are: measurement of dynamic properties of acoustical foams and incorporation of these properties in models governing three-dimensional wave propagation in foams; tests to measure sound transmission paths in the HP137 Jetstream 3; and formulation of a finite element energy model. In addition, the effort to develop a numerical/empirical noise source identification technique was completed. The investigation of a design optimization technique for active noise control was also completed. Monthly progress reports which detail the progress made toward each of the objectives are summarized.

Acoustic Source Localization in Aircraft Interiors Using Microphone Array Technologies

NASA Technical Reports Server (NTRS)

Sklanka, Bernard J.; Tuss, Joel R.; Buehrle, Ralph D.; Klos, Jacob; Williams, Earl G.; Valdivia, Nicolas

2006-01-01

Using three microphone array configurations at two aircraft body stations on a Boeing 777-300ER flight test, the acoustic radiation characteristics of the sidewall and outboard floor system are investigated by experimental measurement. Analysis of the experimental data is performed using sound intensity calculations for closely spaced microphones, PATCH Inverse Boundary Element Nearfield Acoustic Holography, and Spherical Nearfield Acoustic Holography. Each method is compared assessing strengths and weaknesses, evaluating source identification capability for both broadband and narrowband sources, evaluating sources during transient and steady-state conditions, and quantifying field reconstruction continuity using multiple array positions.

Newborn human brain identifies repeated auditory feature conjunctions of low sequential probability.

PubMed

Ruusuvirta, Timo; Huotilainen, Minna; Fellman, Vineta; Näätänen, Risto

2004-11-01

Natural environments are usually composed of multiple sources for sounds. The sounds might physically differ from one another only as feature conjunctions, and several of them might occur repeatedly in the short term. Nevertheless, the detection of rare sounds requires the identification of the repeated ones. Adults have some limited ability to effortlessly identify repeated sounds in such acoustically complex environments, but the developmental onset of this finite ability is unknown. Sleeping newborn infants were presented with a repeated tone carrying six frequent (P = 0.15 each) and six rare (P approximately 0.017 each) conjunctions of its frequency, intensity and duration. Event-related potentials recorded from the infants' scalp were found to shift in amplitude towards positive polarity selectively in response to rare conjunctions. This finding suggests that humans are relatively hard-wired to preattentively identify repeated auditory feature conjunctions even when such conjunctions occur rarely among other similar ones.

Measurement of Correlation Between Flow Density, Velocity, and Density*velocity(sup 2) with Far Field Noise in High Speed Jets

NASA Technical Reports Server (NTRS)

Panda, Jayanta; Seasholtz, Richard G.; Elam, Kristie A.

2002-01-01

To locate noise sources in high-speed jets, the sound pressure fluctuations p', measured at far field locations, were correlated with each of radial velocity v, density rho, and phov(exp 2) fluctuations measured from various points in jet plumes. The experiments follow the cause-and-effect method of sound source identification, where correlation is related to the first, and correlation to the second source terms of Lighthill's equation. Three fully expanded, unheated plumes of Mach number 0.95, 1.4 and 1.8 were studied for this purpose. The velocity and density fluctuations were measured simultaneously using a recently developed, non-intrusive, point measurement technique based on molecular Rayleigh scattering. It was observed that along the jet centerline the density fluctuation spectra S(sub rho) have different shapes than the radial velocity spectra S(sub v), while data obtained from the peripheral shear layer show similarity between the two spectra. Density fluctuations in the jet showed significantly higher correlation, than either rhov(sub 2) or v fluctuations. It is found that a single point correlation from the peak sound emitting region at the end of the potential core can account for nearly 10% of all noise at 30 to the jet axis. The correlation, representing the effectiveness of a longitudinal quadrupole in generating noise 90 to the jet axis, is found to be zero within experimental uncertainty. In contrast rhov(exp 2) fluctuations were better correlated with sound pressure fluctuation at the 30 location. The strongest source of sound is found to lie at the centerline and beyond the end of potential core.

A Survey on the Feasibility of Sound Classification on Wireless Sensor Nodes

PubMed Central

Salomons, Etto L.; Havinga, Paul J. M.

2015-01-01

Wireless sensor networks are suitable to gain context awareness for indoor environments. As sound waves form a rich source of context information, equipping the nodes with microphones can be of great benefit. The algorithms to extract features from sound waves are often highly computationally intensive. This can be problematic as wireless nodes are usually restricted in resources. In order to be able to make a proper decision about which features to use, we survey how sound is used in the literature for global sound classification, age and gender classification, emotion recognition, person verification and identification and indoor and outdoor environmental sound classification. The results of the surveyed algorithms are compared with respect to accuracy and computational load. The accuracies are taken from the surveyed papers; the computational loads are determined by benchmarking the algorithms on an actual sensor node. We conclude that for indoor context awareness, the low-cost algorithms for feature extraction perform equally well as the more computationally-intensive variants. As the feature extraction still requires a large amount of processing time, we present four possible strategies to deal with this problem. PMID:25822142

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.

Acoustic Holography

NASA Astrophysics Data System (ADS)

Kim, Yang-Hann

One of the subtle problems that make noise control difficult for engineers is the invisibility of noise or sound. A visual image of noise often helps to determine an appropriate means for noise control. There have been many attempts to fulfill this rather challenging objective. Theoretical (or numerical) means for visualizing the sound field have been attempted, and as a result, a great deal of progress has been made. However, most of these numerical methods are not quite ready for practical applications to noise control problems. In the meantime, rapid progress with instrumentation has made it possible to use multiple microphones and fast signal-processing systems. Although these systems are not perfect, they are useful. A state-of-the-art system has recently become available, but it still has many problematic issues; for example, how can one implement the visualized noise field. The constructed noise or sound picture always consists of bias and random errors, and consequently, it is often difficult to determine the origin of the noise and the spatial distribution of the noise field. Section 26.2 of this chapter introduces a brief history, which is associated with "sound visualization," acoustic source identification methods and what has been accomplished with a line or surface array. Section 26.2.3 introduces difficulties and recent studies, including de-Dopplerization and de-reverberation methods, both essentialfor visualizing a moving noise source, such as occurs for cars or trains. This section also addresses what produces ambiguity in realizing real sound sources in a room or closed space. Another major issue associated with sound/noise visualization is whether or not we can distinguish between mutual dependencies of noise in space (Sect. 26.2.4); for example, we are asked to answer the question, "Can we see two birds singing or one bird with two beaks?"

Acoustic Holography

NASA Astrophysics Data System (ADS)

Kim, Yang-Hann

One of the subtle problems that make noise control difficult for engineers is the invisibility of noise or sound. A visual image of noise often helps to determine an appropriate means for noise control. There have been many attempts to fulfill this rather challenging objective. Theoretical (or numerical) means for visualizing the sound field have been attempted, and as a result, a great deal of progress has been made. However, most of these numerical methods are not quite ready for practical applications to noise control problems. In the meantime, rapid progress with instrumentation has made it possible to use multiple microphones and fast signal-processing systems. Although these systems are not perfect, they are useful. A state-of-the-art system has recently become available, but it still has many problematic issues; for example, how can one implement the visualized noise field. The constructed noise or sound picture always consists of bias and random errors, and consequently, it is often difficult to determine the origin of the noise and the spatial distribution of the noise field. Section 26.2 of this chapter introduces a brief history, which is associated with sound visualization, acoustic source identification methods and what has been accomplished with a line or surface array. Section 26.2.3 introduces difficulties and recent studies, including de-Dopplerization and de-re verberation methods, both essential for visualizing a moving noise source, such as occurs for cars or trains. This section also addresses what produces ambiguity in realizing real sound sources in a room or closed space. Another major issue associated with sound/noise visualization is whether or not we can distinguish between mutual dependencies of noise in space (Sect. 26.2.4); for example, we are asked to answer the question, Can we see two birds singing or one bird with two beaks?

Frequency-Limiting Effects on Speech and Environmental Sound Identification for Cochlear Implant and Normal Hearing Listeners

PubMed Central

Chang, Son-A; Won, Jong Ho; Kim, HyangHee; Oh, Seung-Ha; Tyler, Richard S.; Cho, Chang Hyun

2018-01-01

Background and Objectives It is important to understand the frequency region of cues used, and not used, by cochlear implant (CI) recipients. Speech and environmental sound recognition by individuals with CI and normal-hearing (NH) was measured. Gradients were also computed to evaluate the pattern of change in identification performance with respect to the low-pass filtering or high-pass filtering cutoff frequencies. Subjects and Methods Frequency-limiting effects were implemented in the acoustic waveforms by passing the signals through low-pass filters (LPFs) or high-pass filters (HPFs) with seven different cutoff frequencies. Identification of Korean vowels and consonants produced by a male and female speaker and environmental sounds was measured. Crossover frequencies were determined for each identification test, where the LPF and HPF conditions show the identical identification scores. Results CI and NH subjects showed changes in identification performance in a similar manner as a function of cutoff frequency for the LPF and HPF conditions, suggesting that the degraded spectral information in the acoustic signals may similarly constraint the identification performance for both subject groups. However, CI subjects were generally less efficient than NH subjects in using the limited spectral information for speech and environmental sound identification due to the inefficient coding of acoustic cues through the CI sound processors. Conclusions This finding will provide vital information in Korean for understanding how different the frequency information is in receiving speech and environmental sounds by CI processor from normal hearing. PMID:29325391

Frequency-Limiting Effects on Speech and Environmental Sound Identification for Cochlear Implant and Normal Hearing Listeners.

PubMed

Chang, Son-A; Won, Jong Ho; Kim, HyangHee; Oh, Seung-Ha; Tyler, Richard S; Cho, Chang Hyun

2017-12-01

It is important to understand the frequency region of cues used, and not used, by cochlear implant (CI) recipients. Speech and environmental sound recognition by individuals with CI and normal-hearing (NH) was measured. Gradients were also computed to evaluate the pattern of change in identification performance with respect to the low-pass filtering or high-pass filtering cutoff frequencies. Frequency-limiting effects were implemented in the acoustic waveforms by passing the signals through low-pass filters (LPFs) or high-pass filters (HPFs) with seven different cutoff frequencies. Identification of Korean vowels and consonants produced by a male and female speaker and environmental sounds was measured. Crossover frequencies were determined for each identification test, where the LPF and HPF conditions show the identical identification scores. CI and NH subjects showed changes in identification performance in a similar manner as a function of cutoff frequency for the LPF and HPF conditions, suggesting that the degraded spectral information in the acoustic signals may similarly constraint the identification performance for both subject groups. However, CI subjects were generally less efficient than NH subjects in using the limited spectral information for speech and environmental sound identification due to the inefficient coding of acoustic cues through the CI sound processors. This finding will provide vital information in Korean for understanding how different the frequency information is in receiving speech and environmental sounds by CI processor from normal hearing.

The impact of the microphone position on the frequency analysis of snoring sounds.

PubMed

Herzog, Michael; Kühnel, Thomas; Bremert, Thomas; Herzog, Beatrice; Hosemann, Werner; Kaftan, Holger

2009-08-01

Frequency analysis of snoring sounds has been reported as a diagnostic tool to differentiate between different sources of snoring. Several studies have been published presenting diverging results of the frequency analyses of snoring sounds. Depending on the position of the used microphones, the results of the frequency analysis of snoring sounds vary. The present study investigated the influence of different microphone positions on the outcome of the frequency analysis of snoring sounds. Nocturnal snoring was recorded simultaneously at six positions (air-coupled: 30 cm middle, 100 cm middle, 30 cm lateral to both sides of the patients' head; body contact: neck and parasternal) in five patients. The used microphones had a flat frequency response and a similar frequency range (10/40 Hz-18 kHz). Frequency analysis was performed by fast Fourier transformation and frequency bands as well as peak intensities (Peaks 1-5) were detected. Air-coupled microphones presented a wider frequency range (60 Hz-10 kHz) compared to contact microphones. The contact microphone at cervical position presented a cut off at frequencies above 300 Hz, whereas the contact microphone at parasternal position revealed a cut off above 100 Hz. On an exemplary base, the study demonstrates that frequencies above 1,000 Hz do appear in complex snoring patterns, and it is emphasised that high frequencies are imported for the interpretation of snoring sounds with respect to the identification of the source of snoring. Contact microphones might be used in screening devices, but for a natural analysis of snoring sounds the use of air-coupled microphones is indispensable.

Heart Sound Biometric System Based on Marginal Spectrum Analysis

PubMed Central

Zhao, Zhidong; Shen, Qinqin; Ren, Fangqin

2013-01-01

This work presents a heart sound biometric system based on marginal spectrum analysis, which is a new feature extraction technique for identification purposes. This heart sound identification system is comprised of signal acquisition, pre-processing, feature extraction, training, and identification. Experiments on the selection of the optimal values for the system parameters are conducted. The results indicate that the new spectrum coefficients result in a significant increase in the recognition rate of 94.40% compared with that of the traditional Fourier spectrum (84.32%) based on a database of 280 heart sounds from 40 participants. PMID:23429515

Observationally constrained modeling of sound in curved ocean internal waves: examination of deep ducting and surface ducting at short range.

PubMed

Duda, Timothy F; Lin, Ying-Tsong; Reeder, D Benjamin

2011-09-01

A study of 400 Hz sound focusing and ducting effects in a packet of curved nonlinear internal waves in shallow water is presented. Sound propagation roughly along the crests of the waves is simulated with a three-dimensional parabolic equation computational code, and the results are compared to measured propagation along fixed 3 and 6 km source/receiver paths. The measurements were made on the shelf of the South China Sea northeast of Tung-Sha Island. Construction of the time-varying three-dimensional sound-speed fields used in the modeling simulations was guided by environmental data collected concurrently with the acoustic data. Computed three-dimensional propagation results compare well with field observations. The simulations allow identification of time-dependent sound forward scattering and ducting processes within the curved internal gravity waves. Strong acoustic intensity enhancement was observed during passage of high-amplitude nonlinear waves over the source/receiver paths, and is replicated in the model. The waves were typical of the region (35 m vertical displacement). Two types of ducting are found in the model, which occur asynchronously. One type is three-dimensional modal trapping in deep ducts within the wave crests (shallow thermocline zones). The second type is surface ducting within the wave troughs (deep thermocline zones). © 2011 Acoustical Society of America

Identification of impact force acting on composite laminated plates using the radiated sound measured with microphones

NASA Astrophysics Data System (ADS)

Atobe, Satoshi; Nonami, Shunsuke; Hu, Ning; Fukunaga, Hisao

2017-09-01

Foreign object impact events are serious threats to composite laminates because impact damage leads to significant degradation of the mechanical properties of the structure. Identification of the location and force history of the impact that was applied to the structure can provide useful information for assessing the structural integrity. This study proposes a method for identifying impact forces acting on CFRP (carbon fiber reinforced plastic) laminated plates on the basis of the sound radiated from the impacted structure. Identification of the impact location and force history is performed using the sound pressure measured with microphones. To devise a method for identifying the impact location from the difference in the arrival times of the sound wave detected with the microphones, the propagation path of the sound wave from the impacted point to the sensor is examined. For the identification of the force history, an experimentally constructed transfer matrix is employed to relate the force history to the corresponding sound pressure. To verify the validity of the proposed method, impact tests are conducted by using a CFRP cross-ply laminate as the specimen, and an impulse hammer as the impactor. The experimental results confirm the validity of the present method for identifying the impact location from the arrival time of the sound wave detected with the microphones. Moreover, the results of force history identification show the feasibility of identifying the force history accurately from the measured sound pressure using the experimental transfer matrix.

Frog sound identification using extended k-nearest neighbor classifier

NASA Astrophysics Data System (ADS)

Mukahar, Nordiana; Affendi Rosdi, Bakhtiar; Athiar Ramli, Dzati; Jaafar, Haryati

2017-09-01

Frog sound identification based on the vocalization becomes important for biological research and environmental monitoring. As a result, different types of feature extractions and classifiers have been employed to evaluate the accuracy of frog sound identification. This paper presents a frog sound identification with Extended k-Nearest Neighbor (EKNN) classifier. The EKNN classifier integrates the nearest neighbors and mutual sharing of neighborhood concepts, with the aims of improving the classification performance. It makes a prediction based on who are the nearest neighbors of the testing sample and who consider the testing sample as their nearest neighbors. In order to evaluate the classification performance in frog sound identification, the EKNN classifier is compared with competing classifier, k -Nearest Neighbor (KNN), Fuzzy k -Nearest Neighbor (FKNN) k - General Nearest Neighbor (KGNN)and Mutual k -Nearest Neighbor (MKNN) on the recorded sounds of 15 frog species obtained in Malaysia forest. The recorded sounds have been segmented using Short Time Energy and Short Time Average Zero Crossing Rate (STE+STAZCR), sinusoidal modeling (SM), manual and the combination of Energy (E) and Zero Crossing Rate (ZCR) (E+ZCR) while the features are extracted by Mel Frequency Cepstrum Coefficient (MFCC). The experimental results have shown that the EKNCN classifier exhibits the best performance in terms of accuracy compared to the competing classifiers, KNN, FKNN, GKNN and MKNN for all cases.

Microbiological quality of Puget Sound Basin streams and identification of contaminant sources

USGS Publications Warehouse

Embrey, S.S.

2001-01-01

Fecal coliforms, Escherichia coli, enterococci, and somatic coliphages were detected in samples from 31 sites on streams draining urban and agricultural regions of the Puget Sound Basin Lowlands. Densities of bacteria in 48 and 71 percent of the samples exceeded U.S. Environmental Protection Agency's freshwater recreation criteria for Escherichia coli and enterococci, respectively, and 81 percent exceeded Washington State fecal coliform standards. Male-specific coliphages were detected in samples from 15 sites. Male-specific F+RNA coliphages isolated from samples taken at South Fork Thornton and Longfellow Creeks were serotyped as Group II, implicating humans as potential contaminant sources. These two sites are located in residential, urban areas. F+RNA coliphages in samples from 10 other sites, mostly in agricultural or rural areas, were serotyped as Group I, implicating non-human animals as likely sources. Chemicals common to wastewater, including fecal sterols, were detected in samples from several urban streams, and also implicate humans, at least in part, as possible sources of fecal bacteria and viruses to the streams.

Sleep duration predicts behavioral and neural differences in adult speech sound learning.

PubMed

Earle, F Sayako; Landi, Nicole; Myers, Emily B

2017-01-01

Sleep is important for memory consolidation and contributes to the formation of new perceptual categories. This study examined sleep as a source of variability in typical learners' ability to form new speech sound categories. We trained monolingual English speakers to identify a set of non-native speech sounds at 8PM, and assessed their ability to identify and discriminate between these sounds immediately after training, and at 8AM on the following day. We tracked sleep duration overnight, and found that light sleep duration predicted gains in identification performance, while total sleep duration predicted gains in discrimination ability. Participants obtained an average of less than 6h of sleep, pointing to the degree of sleep deprivation as a potential factor. Behavioral measures were associated with ERP indexes of neural sensitivity to the learned contrast. These results demonstrate that the relative success in forming new perceptual categories depends on the duration of post-training sleep. Copyright © 2016 Elsevier Ireland Ltd. All rights reserved.

Sleep duration predicts behavioral and neural differences in adult speech sound learning

PubMed Central

Earle, F. Sayako; Landi, Nicole; Myers, Emily B.

2016-01-01

Sleep is important for memory consolidation and contributes to the formation of new perceptual categories. This study examined sleep as a source of variability in typical learners’ ability to form new speech sound categories. We trained monolingual English speakers to identify a set of non-native speech sounds at 8PM, and assessed their ability to identify and discriminate between these sounds immediately after training, and at 8AM on the following day. We tracked sleep duration overnight, and found that light sleep duration predicted gains in identification performance, while total sleep duration predicted gains in discrimination ability. Participants obtained an average of less than 6 hours of sleep, pointing to the degree of sleep deprivation as a potential factor. Behavioral measures were associated with ERP indexes of neural sensitivity to the learned contrast. These results demonstrate that the relative success in forming new perceptual categories depends on the duration of post-training sleep. PMID:27793703

Stochastic sediment property inversion in Shallow Water 06.

PubMed

Michalopoulou, Zoi-Heleni

2017-11-01

Received time-series at a short distance from the source allow the identification of distinct paths; four of these are direct, surface and bottom reflections, and sediment reflection. In this work, a Gibbs sampling method is used for the estimation of the arrival times of these paths and the corresponding probability density functions. The arrival times for the first three paths are then employed along with linearization for the estimation of source range and depth, water column depth, and sound speed in the water. Propagating densities of arrival times through the linearized inverse problem, densities are also obtained for the above parameters, providing maximum a posteriori estimates. These estimates are employed to calculate densities and point estimates of sediment sound speed and thickness using a non-linear, grid-based model. Density computation is an important aspect of this work, because those densities express the uncertainty in the inversion for sediment properties.

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.

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.

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.

Listening to sounds from an exploding meteor and oceanic waves

NASA Astrophysics Data System (ADS)

Evers, L. G.; Haak, H. W.

Low frequency sound (infrasound) measurements have been selected within the Comprehensive Nuclear-Test-Ban Treaty (CTBT) as a technique to detect and identify possible nuclear explosions. The Seismology Division of the Royal Netherlands Meteorological Institute (KNMI) operates since 1999 an experimental infrasound array of 16 micro-barometers. Here we show the rare detection and identification of an exploding meteor above Northern Germany on November 8th, 1999 with data from the Deelen Infrasound Array (DIA). At the same time, sound was radiated from the Atlantic Ocean, South of Iceland, due to the atmospheric coupling of standing ocean waves, called microbaroms. Occurring with only 0.04 Hz difference in dominant frequency, DIA proved to be able to discriminate between the physically different sources of infrasound through its unique lay-out and instruments. The explosive power of the meteor being 1.5 kT TNT is in the range of nuclear explosions and therefore relevant to the CTBT.

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

When Hearing the Bark Helps to Identify the Dog: Semantically-Congruent Sounds Modulate the Identification of Masked Pictures

ERIC Educational Resources Information Center

Chen, Yi-Chuan; Spence, Charles

2010-01-01

We report a series of experiments designed to assess the effect of audiovisual semantic congruency on the identification of visually-presented pictures. Participants made unspeeded identification responses concerning a series of briefly-presented, and then rapidly-masked, pictures. A naturalistic sound was sometimes presented together with the…

A critical review of the potential impacts of marine seismic surveys on fish & invertebrates.

PubMed

Carroll, A G; Przeslawski, R; Duncan, A; Gunning, M; Bruce, B

2017-01-15

Marine seismic surveys produce high intensity, low-frequency impulsive sounds at regular intervals, with most sound produced between 10 and 300Hz. Offshore seismic surveys have long been considered to be disruptive to fisheries, but there are few ecological studies that target commercially important species, particularly invertebrates. This review aims to summarise scientific studies investigating the impacts of low-frequency sound on marine fish and invertebrates, as well as to critically evaluate how such studies may apply to field populations exposed to seismic operations. We focus on marine seismic surveys due to their associated unique sound properties (i.e. acute, low-frequency, mobile source locations), as well as fish and invertebrates due to the commercial value of many species in these groups. The main challenges of seismic impact research are the translation of laboratory results to field populations over a range of sound exposure scenarios and the lack of sound exposure standardisation which hinders the identification of response thresholds. An integrated multidisciplinary approach to manipulative and in situ studies is the most effective way to establish impact thresholds in the context of realistic exposure levels, but if that is not practical the limitations of each approach must be carefully considered. Crown Copyright © 2016. Published by Elsevier Ltd. All rights reserved.

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.

Parallel Processing of Large Scale Microphone Arrays for Sound Capture

NASA Astrophysics Data System (ADS)

Jan, Ea-Ee.

1995-01-01

Performance of microphone sound pick up is degraded by deleterious properties of the acoustic environment, such as multipath distortion (reverberation) and ambient noise. The degradation becomes more prominent in a teleconferencing environment in which the microphone is positioned far away from the speaker. Besides, the ideal teleconference should feel as easy and natural as face-to-face communication with another person. This suggests hands-free sound capture with no tether or encumbrance by hand-held or body-worn sound equipment. Microphone arrays for this application represent an appropriate approach. This research develops new microphone array and signal processing techniques for high quality hands-free sound capture in noisy, reverberant enclosures. The new techniques combine matched-filtering of individual sensors and parallel processing to provide acute spatial volume selectivity which is capable of mitigating the deleterious effects of noise interference and multipath distortion. The new method outperforms traditional delay-and-sum beamformers which provide only directional spatial selectivity. The research additionally explores truncated matched-filtering and random distribution of transducers to reduce complexity and improve sound capture quality. All designs are first established by computer simulation of array performance in reverberant enclosures. The simulation is achieved by a room model which can efficiently calculate the acoustic multipath in a rectangular enclosure up to a prescribed order of images. It also calculates the incident angle of the arriving signal. Experimental arrays were constructed and their performance was measured in real rooms. Real room data were collected in a hard-walled laboratory and a controllable variable acoustics enclosure of similar size, approximately 6 x 6 x 3 m. An extensive speech database was also collected in these two enclosures for future research on microphone arrays. The simulation results are shown to be consistent with the real room data. Localization of sound sources has been explored using cross-power spectrum time delay estimation and has been evaluated using real room data under slightly, moderately and highly reverberant conditions. To improve the accuracy and reliability of the source localization, an outlier detector that removes incorrect time delay estimation has been invented. To provide speaker selectivity for microphone array systems, a hands-free speaker identification system has been studied. A recently invented feature using selected spectrum information outperforms traditional recognition methods. Measured results demonstrate the capabilities of speaker selectivity from a matched-filtered array. In addition, simulation utilities, including matched -filtering processing of the array and hands-free speaker identification, have been implemented on the massively -parallel nCube super-computer. This parallel computation highlights the requirements for real-time processing of array signals.

21 CFR 876.4590 - Interlocking urethral sound.

Code of Federal Regulations, 2010 CFR

2010-04-01

... 21 Food and Drugs 8 2010-04-01 2010-04-01 false Interlocking urethral sound. 876.4590 Section 876...) MEDICAL DEVICES GASTROENTEROLOGY-UROLOGY DEVICES Surgical Devices § 876.4590 Interlocking urethral sound. (a) Identification. An interlocking urethral sound is a device that consists of two metal sounds...

21 CFR 876.4590 - Interlocking urethral sound.

Code of Federal Regulations, 2014 CFR

2014-04-01

... 21 Food and Drugs 8 2014-04-01 2014-04-01 false Interlocking urethral sound. 876.4590 Section 876...) MEDICAL DEVICES GASTROENTEROLOGY-UROLOGY DEVICES Surgical Devices § 876.4590 Interlocking urethral sound. (a) Identification. An interlocking urethral sound is a device that consists of two metal sounds...

21 CFR 876.4590 - Interlocking urethral sound.

Code of Federal Regulations, 2012 CFR

2012-04-01

... 21 Food and Drugs 8 2012-04-01 2012-04-01 false Interlocking urethral sound. 876.4590 Section 876...) MEDICAL DEVICES GASTROENTEROLOGY-UROLOGY DEVICES Surgical Devices § 876.4590 Interlocking urethral sound. (a) Identification. An interlocking urethral sound is a device that consists of two metal sounds...

21 CFR 876.4590 - Interlocking urethral sound.

Code of Federal Regulations, 2013 CFR

2013-04-01

... 21 Food and Drugs 8 2013-04-01 2013-04-01 false Interlocking urethral sound. 876.4590 Section 876...) MEDICAL DEVICES GASTROENTEROLOGY-UROLOGY DEVICES Surgical Devices § 876.4590 Interlocking urethral sound. (a) Identification. An interlocking urethral sound is a device that consists of two metal sounds...

21 CFR 876.4590 - Interlocking urethral sound.

Code of Federal Regulations, 2011 CFR

2011-04-01

... 21 Food and Drugs 8 2011-04-01 2011-04-01 false Interlocking urethral sound. 876.4590 Section 876...) MEDICAL DEVICES GASTROENTEROLOGY-UROLOGY DEVICES Surgical Devices § 876.4590 Interlocking urethral sound. (a) Identification. An interlocking urethral sound is a device that consists of two metal sounds...

Auditory and visual localization accuracy in young children and adults.

PubMed

Martin, Karen; Johnstone, Patti; Hedrick, Mark

2015-06-01

This study aimed to measure and compare sound and light source localization ability in young children and adults who have normal hearing and normal/corrected vision in order to determine the extent to which age, type of stimuli, and stimulus order affects sound localization accuracy. Two experiments were conducted. The first involved a group of adults only. The second involved a group of 30 children aged 3 to 5 years. Testing occurred in a sound-treated booth containing a semi-circular array of 15 loudspeakers set at 10° intervals from -70° to 70° azimuth. Each loudspeaker had a tiny light bulb and a small picture fastened underneath. Seven of the loudspeakers were used to randomly test sound and light source identification. The sound stimulus was the word "baseball". The light stimulus was a flashing of a light bulb triggered by the digital signal of the word "baseball". Each participant was asked to face 0° azimuth, and identify the location of the test stimulus upon presentation. Adults used a computer mouse to click on an icon; children responded by verbally naming or walking toward the picture underneath the corresponding loudspeaker or light. A mixed experimental design using repeated measures was used to determine the effect of age and stimulus type on localization accuracy in children and adults. A mixed experimental design was used to compare the effect of stimulus order (light first/last) and varying or fixed intensity sound on localization accuracy in children and adults. Localization accuracy was significantly better for light stimuli than sound stimuli for children and adults. Children, compared to adults, showed significantly greater localization errors for audition. Three-year-old children had significantly greater sound localization errors compared to 4- and 5-year olds. Adults performed better on the sound localization task when the light localization task occurred first. Young children can understand and attend to localization tasks, but show poorer localization accuracy than adults in sound localization. This may be a reflection of differences in sensory modality development and/or central processes in young children, compared to adults. Copyright © 2015 Elsevier Ireland Ltd. All rights reserved.

On the importance of measurement system calibration for underwater passive monitoring

NASA Astrophysics Data System (ADS)

Miqueleti, S. A.; Costa-Félix, R. P. B.

2016-07-01

The underwater passive acoustic monitoring of sound in oceans is growing in recent years and has served as a source of information on marine life and the interference of human activities on the environment. The recordings are used for species identification and prevention of potential adverse effects of vessel traffic, sonar and offshore activities as a whole. However, not much attention is given to the calibration of the hydrophone used to ensure the validity of the information collected. The resulting sound depends on the input audio, and the transfer function of the intensity of the input signal. This paper presents an assessment of how the lack of calibration of hydroacoustic systems might compromise the evaluation of the marine environment.

Underwater sound radiation patterns of contemporary merchant ships

NASA Astrophysics Data System (ADS)

Gassmann, M.; Wiggins, S. M.; Hildebrand, J. A.

2016-12-01

Merchant ships radiate underwater sound as an unintended by-product of their operation and as consequence contribute significantly to low-frequency, man-made noise in the ocean. Current measurement standards for the description of underwater sound from ships (ISO 17208-1:2016 and ANSI S12.64-2009) require nominal hydrophone depths of 15°, 30° and 45° at the starboard and portside of the test vessel.To opportunistically study the underwater sound of contemporary merchant ships that were tracked by the Automatic Identification System (AIS), an array of seven high-frequency acoustic recording packages (HARPs) with a sampling frequency of 200 kHz was deployed in the Santa Barbara Channel in the primary outgoing shipping lane for the port of Los Angeles and Long Beach. The vertical and horizontal aperture of the array allowed for starboard and portside measurements at all standard-required nominal hydrophone depths in addition to measurements taken at the keel aspect. Based on these measurements, frequency-dependent radiation patterns of contemporary merchant ships were estimated and used to evaluate current standards for computing ship source levels.

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.

Prototype electronic stethoscope vs. conventional stethoscope for auscultation of heart sounds.

PubMed

Kelmenson, Daniel A; Heath, Janae K; Ball, Stephanie A; Kaafarani, Haytham M A; Baker, Elisabeth M; Yeh, Daniel D; Bittner, Edward A; Eikermann, Matthias; Lee, Jarone

2014-08-01

In an effort to decrease the spread of hospital-acquired infections, many hospitals currently use disposable plastic stethoscopes in patient rooms. As an alternative, this study examines a prototype electronic stethoscope that does not break the isolation barrier between clinician and patient and may also improve the diagnostic accuracy of the stethoscope exam. This study aimed to investigate whether the new prototype electronic stethoscope improved auscultation of heart sounds compared to the standard conventional isolation stethoscope. In a controlled, non-blinded, cross-over study, clinicians were randomized to identify heart sounds with both the prototype electronic stethoscope and a conventional stethoscope. The primary outcome was the score on a 10-question heart sound identification test. In total, 41 clinicians completed the study. Subjects performed significantly better in the identification of heart sounds when using the prototype electronic stethoscope (median = 9 [7-10] vs. 8 [6-9] points, p value <0.0001). Subjects also significantly preferred the prototype electronic stethoscope. Clinicians using a new prototype electronic stethoscope achieved greater accuracy in identification of heart sounds and also universally favoured the new device, compared to the conventional stethoscope.

Correlation between Identification Accuracy and Response Confidence for Common Environmental Sounds

DTIC Science & Technology

set of environmental sounds with stimulus control and precision. The present study is one in a series of efforts to provide a baseline evaluation of a...sounds from six broad categories: household items, alarms, animals, human generated, mechanical, and vehicle sounds. Each sound was presented five times

Advanced Systems for Monitoring Underwater Sounds

NASA Technical Reports Server (NTRS)

Lane, Michael; Van Meter, Steven; Gilmore, Richard Grant; Sommer, Keith

2007-01-01

The term "Passive Acoustic Monitoring System" (PAMS) describes a developmental sensing-and-data-acquisition system for recording underwater sounds. The sounds (more precisely, digitized and preprocessed versions from acoustic transducers) are subsequently analyzed by a combination of data processing and interpretation to identify and/or, in some cases, to locate the sources of those sounds. PAMS was originally designed to locate the sources such as fish of species that one knows or seeks to identify. The PAMS unit could also be used to locate other sources, for example, marine life, human divers, and/or vessels. The underlying principles of passive acoustic sensing and analyzing acoustic-signal data in conjunction with temperature and salinity data are not new and not unique to PAMS. Part of the uniqueness of the PAMS design is that it is the first deep-sea instrumentation design to provide a capability for studying soniferous marine animals (especially fish) over the wide depth range described below. The uniqueness of PAMS also lies partly in a synergistic combination of advanced sensing, packaging, and data-processing design features with features adapted from proven marine instrumentation systems. This combination affords a versatility that enables adaptation to a variety of undersea missions using a variety of sensors. The interpretation of acoustic data can include visual inspection of power-spectrum plots for identification of spectral signatures of known biological species or artificial sources. Alternatively or in addition, data analysis could include determination of relative times of arrival of signals at different acoustic sensors arrayed at known locations. From these times of arrival, locations of acoustic sources (and errors in those locations) can be estimated. Estimates of relative locations of sources and sensors can be refined through analysis of the attenuation of sound in the intervening water in combination with water-temperature and salinity data acquired by instrumentation systems other than PAMS. A PAMS is packaged as a battery-powered unit, mated with external sensors, that can operate in the ocean at any depth from 2 m to 1 km. A PAMS includes a pressure housing, a deep-sea battery, a hydrophone (which is one of the mating external sensors), and an external monitor and keyboard box. In addition to acoustic transducers, external sensors can include temperature probes and, potentially, underwater cameras. The pressure housing contains a computer that includes a hard drive, DC-to- DC power converters, a post-amplifier board, a sound card, and a universal serial bus (USB) 4-port hub.

Children with Developmental Disabilities: The Effect of Sound Field Amplification on Word Identification.

ERIC Educational Resources Information Center

Flexer, Carol; And Others

1990-01-01

Using sound field amplification which increased the intensity of the teacher's voice by 10 decibels, 9 primary-level children with developmental disabilities made fewer errors on a word identification task, were more relaxed, and responded more quickly than without amplification. (Author/JDD)

21 CFR 884.2900 - Fetal stethoscope.

Code of Federal Regulations, 2010 CFR

2010-04-01

... Fetal stethoscope. (a) Identification. A fetal stethoscope is a device used for listening to fetal heart sounds. It is designed to transmit the fetal heart sounds not only through sound channels by air...

21 CFR 884.2900 - Fetal stethoscope.

Code of Federal Regulations, 2011 CFR

2011-04-01

... Fetal stethoscope. (a) Identification. A fetal stethoscope is a device used for listening to fetal heart sounds. It is designed to transmit the fetal heart sounds not only through sound channels by air...

21 CFR 884.2900 - Fetal stethoscope.

Code of Federal Regulations, 2013 CFR

2013-04-01

... Fetal stethoscope. (a) Identification. A fetal stethoscope is a device used for listening to fetal heart sounds. It is designed to transmit the fetal heart sounds not only through sound channels by air...

21 CFR 884.2900 - Fetal stethoscope.

Code of Federal Regulations, 2014 CFR

2014-04-01

... Fetal stethoscope. (a) Identification. A fetal stethoscope is a device used for listening to fetal heart sounds. It is designed to transmit the fetal heart sounds not only through sound channels by air...

21 CFR 884.2900 - Fetal stethoscope.

Code of Federal Regulations, 2012 CFR

2012-04-01

... Fetal stethoscope. (a) Identification. A fetal stethoscope is a device used for listening to fetal heart sounds. It is designed to transmit the fetal heart sounds not only through sound channels by air...

Selective Listening Point Audio Based on Blind Signal Separation and Stereophonic Technology

NASA Astrophysics Data System (ADS)

Niwa, Kenta; Nishino, Takanori; Takeda, Kazuya

A sound field reproduction method is proposed that uses blind source separation and a head-related transfer function. In the proposed system, multichannel acoustic signals captured at distant microphones are decomposed to a set of location/signal pairs of virtual sound sources based on frequency-domain independent component analysis. After estimating the locations and the signals of the virtual sources by convolving the controlled acoustic transfer functions with each signal, the spatial sound is constructed at the selected point. In experiments, a sound field made by six sound sources is captured using 48 distant microphones and decomposed into sets of virtual sound sources. Since subjective evaluation shows no significant difference between natural and reconstructed sound when six virtual sources and are used, the effectiveness of the decomposing algorithm as well as the virtual source representation are confirmed.

A method for evaluating the relation between sound source segregation and masking

PubMed Central

Lutfi, Robert A.; Liu, Ching-Ju

2011-01-01

Sound source segregation refers to the ability to hear as separate entities two or more sound sources comprising a mixture. Masking refers to the ability of one sound to make another sound difficult to hear. Often in studies, masking is assumed to result from a failure of segregation, but this assumption may not always be correct. Here a method is offered to identify the relation between masking and sound source segregation in studies and an example is given of its application. PMID:21302979

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

The Application of Acoustic Measurements and Audio Recordings for Diagnosis of In-Flight Hardware Anomalies

NASA Technical Reports Server (NTRS)

Welsh, David; Denham, Samuel; Allen, Christopher

2011-01-01

In many cases, an initial symptom of hardware malfunction is unusual or unexpected acoustic noise. Many industries such as automotive, heating and air conditioning, and petro-chemical processing use noise and vibration data along with rotating machinery analysis techniques to identify noise sources and correct hardware defects. The NASA/Johnson Space Center Acoustics Office monitors the acoustic environment of the International Space Station (ISS) through periodic sound level measurement surveys. Trending of the sound level measurement survey results can identify in-flight hardware anomalies. The crew of the ISS also serves as a "detection tool" in identifying unusual hardware noises; in these cases the spectral analysis of audio recordings made on orbit can be used to identify hardware defects that are related to rotating components such as fans, pumps, and compressors. In this paper, three examples of the use of sound level measurements and audio recordings for the diagnosis of in-flight hardware anomalies are discussed: identification of blocked inter-module ventilation (IMV) ducts, diagnosis of abnormal ISS Crew Quarters rack exhaust fan noise, and the identification and replacement of a defective flywheel assembly in the Treadmill with Vibration Isolation (TVIS) hardware. In each of these examples, crew time was saved by identifying the off nominal component or condition that existed and in directing in-flight maintenance activities to address and correct each of these problems.

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

21 CFR 870.2860 - Heart sound transducer.

Code of Federal Regulations, 2010 CFR

2010-04-01

... 21 Food and Drugs 8 2010-04-01 2010-04-01 false Heart sound transducer. 870.2860 Section 870.2860...) MEDICAL DEVICES CARDIOVASCULAR DEVICES Cardiovascular Monitoring Devices § 870.2860 Heart sound transducer. (a) Identification. A heart sound transducer is an external transducer that exhibits a change in...

21 CFR 870.2860 - Heart sound transducer.

Code of Federal Regulations, 2014 CFR

2014-04-01

... 21 Food and Drugs 8 2014-04-01 2014-04-01 false Heart sound transducer. 870.2860 Section 870.2860...) MEDICAL DEVICES CARDIOVASCULAR DEVICES Cardiovascular Monitoring Devices § 870.2860 Heart sound transducer. (a) Identification. A heart sound transducer is an external transducer that exhibits a change in...

21 CFR 870.2860 - Heart sound transducer.

Code of Federal Regulations, 2013 CFR

2013-04-01

... 21 Food and Drugs 8 2013-04-01 2013-04-01 false Heart sound transducer. 870.2860 Section 870.2860...) MEDICAL DEVICES CARDIOVASCULAR DEVICES Cardiovascular Monitoring Devices § 870.2860 Heart sound transducer. (a) Identification. A heart sound transducer is an external transducer that exhibits a change in...

21 CFR 870.2860 - Heart sound transducer.

Code of Federal Regulations, 2011 CFR

2011-04-01

... 21 Food and Drugs 8 2011-04-01 2011-04-01 false Heart sound transducer. 870.2860 Section 870.2860...) MEDICAL DEVICES CARDIOVASCULAR DEVICES Cardiovascular Monitoring Devices § 870.2860 Heart sound transducer. (a) Identification. A heart sound transducer is an external transducer that exhibits a change in...

21 CFR 870.2860 - Heart sound transducer.

Code of Federal Regulations, 2012 CFR

2012-04-01

... 21 Food and Drugs 8 2012-04-01 2012-04-01 false Heart sound transducer. 870.2860 Section 870.2860...) MEDICAL DEVICES CARDIOVASCULAR DEVICES Cardiovascular Monitoring Devices § 870.2860 Heart sound transducer. (a) Identification. A heart sound transducer is an external transducer that exhibits a change in...

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.

Hierarchical neurocomputations underlying concurrent sound segregation: connecting periphery to percept.

PubMed

Bidelman, Gavin M; Alain, Claude

2015-02-01

Natural soundscapes often contain multiple sound sources at any given time. Numerous studies have reported that in human observers, the perception and identification of concurrent sounds is paralleled by specific changes in cortical event-related potentials (ERPs). Although these studies provide a window into the cerebral mechanisms governing sound segregation, little is known about the subcortical neural architecture and hierarchy of neurocomputations that lead to this robust perceptual process. Using computational modeling, scalp-recorded brainstem/cortical ERPs, and human psychophysics, we demonstrate that a primary cue for sound segregation, i.e., harmonicity, is encoded at the auditory nerve level within tens of milliseconds after the onset of sound and is maintained, largely untransformed, in phase-locked activity of the rostral brainstem. As then indexed by auditory cortical responses, (in)harmonicity is coded in the signature and magnitude of the cortical object-related negativity (ORN) response (150-200 ms). The salience of the resulting percept is then captured in a discrete, categorical-like coding scheme by a late negativity response (N5; ~500 ms latency), just prior to the elicitation of a behavioral judgment. Subcortical activity correlated with cortical evoked responses such that weaker phase-locked brainstem responses (lower neural harmonicity) generated larger ORN amplitude, reflecting the cortical registration of multiple sound objects. Studying multiple brain indices simultaneously helps illuminate the mechanisms and time-course of neural processing underlying concurrent sound segregation and may lead to further development and refinement of physiologically driven models of auditory scene analysis. Copyright © 2014 Elsevier Ltd. All rights reserved.

Transfer of auditory perceptual learning with spectrally reduced speech to speech and nonspeech tasks: implications for cochlear implants.

PubMed

Loebach, Jeremy L; Pisoni, David B; Svirsky, Mario A

2009-12-01

The objective of this study was to assess whether training on speech processed with an eight-channel noise vocoder to simulate the output of a cochlear implant would produce transfer of auditory perceptual learning to the recognition of nonspeech environmental sounds, the identification of speaker gender, and the discrimination of talkers by voice. Twenty-four normal-hearing subjects were trained to transcribe meaningful English sentences processed with a noise vocoder simulation of a cochlear implant. An additional 24 subjects served as an untrained control group and transcribed the same sentences in their unprocessed form. All subjects completed pre- and post-test sessions in which they transcribed vocoded sentences to provide an assessment of training efficacy. Transfer of perceptual learning was assessed using a series of closed set, nonlinguistic tasks: subjects identified talker gender, discriminated the identity of pairs of talkers, and identified ecologically significant environmental sounds from a closed set of alternatives. Although both groups of subjects showed significant pre- to post-test improvements, subjects who transcribed vocoded sentences during training performed significantly better at post-test than those in the control group. Both groups performed equally well on gender identification and talker discrimination. Subjects who received explicit training on the vocoded sentences, however, performed significantly better on environmental sound identification than the untrained subjects. Moreover, across both groups, pre-test speech performance and, to a higher degree, post-test speech performance, were significantly correlated with environmental sound identification. For both groups, environmental sounds that were characterized as having more salient temporal information were identified more often than environmental sounds that were characterized as having more salient spectral information. Listeners trained to identify noise-vocoded sentences showed evidence of transfer of perceptual learning to the identification of environmental sounds. In addition, the correlation between environmental sound identification and sentence transcription indicates that subjects who were better able to use the degraded acoustic information to identify the environmental sounds were also better able to transcribe the linguistic content of novel sentences. Both trained and untrained groups performed equally well ( approximately 75% correct) on the gender-identification task, indicating that training did not have an effect on the ability to identify the gender of talkers. Although better than chance, performance on the talker discrimination task was poor overall ( approximately 55%), suggesting that either explicit training is required to discriminate talkers' voices reliably or that additional information (perhaps spectral in nature) not present in the vocoded speech is required to excel in such tasks. Taken together, the results suggest that although transfer of auditory perceptual learning with spectrally degraded speech does occur, explicit task-specific training may be necessary for tasks that cannot rely on temporal information alone.

Transfer of Auditory Perceptual Learning with Spectrally Reduced Speech to Speech and Nonspeech Tasks: Implications for Cochlear Implants

PubMed Central

Loebach, Jeremy L.; Pisoni, David B.; Svirsky, Mario A.

2009-01-01

Objective The objective of this study was to assess whether training on speech processed with an 8-channel noise vocoder to simulate the output of a cochlear implant would produce transfer of auditory perceptual learning to the recognition of non-speech environmental sounds, the identification of speaker gender, and the discrimination of talkers by voice. Design Twenty-four normal hearing subjects were trained to transcribe meaningful English sentences processed with a noise vocoder simulation of a cochlear implant. An additional twenty-four subjects served as an untrained control group and transcribed the same sentences in their unprocessed form. All subjects completed pre- and posttest sessions in which they transcribed vocoded sentences to provide an assessment of training efficacy. Transfer of perceptual learning was assessed using a series of closed-set, nonlinguistic tasks: subjects identified talker gender, discriminated the identity of pairs of talkers, and identified ecologically significant environmental sounds from a closed set of alternatives. Results Although both groups of subjects showed significant pre- to posttest improvements, subjects who transcribed vocoded sentences during training performed significantly better at posttest than subjects in the control group. Both groups performed equally well on gender identification and talker discrimination. Subjects who received explicit training on the vocoded sentences, however, performed significantly better on environmental sound identification than the untrained subjects. Moreover, across both groups, pretest speech performance, and to a higher degree posttest speech performance, were significantly correlated with environmental sound identification. For both groups, environmental sounds that were characterized as having more salient temporal information were identified more often than environmental sounds that were characterized as having more salient spectral information. Conclusions Listeners trained to identify noise-vocoded sentences showed evidence of transfer of perceptual learning to the identification of environmental sounds. In addition, the correlation between environmental sound identification and sentence transcription indicates that subjects who were better able to utilize the degraded acoustic information to identify the environmental sounds were also better able to transcribe the linguistic content of novel sentences. Both trained and untrained groups performed equally well (~75% correct) on the gender identification task, indicating that training did not have an effect on the ability to identify the gender of talkers. Although better than chance, performance on the talker discrimination task was poor overall (~55%), suggesting that either explicit training is required to reliably discriminate talkers’ voices, or that additional information (perhaps spectral in nature) not present in the vocoded speech is required to excel in such tasks. Taken together, the results suggest that while transfer of auditory perceptual learning with spectrally degraded speech does occur, explicit task-specific training may be necessary for tasks that cannot rely on temporal information alone. PMID:19773659

Directional acoustic measurements by laser Doppler velocimeters. [for jet aircraft noise

NASA Technical Reports Server (NTRS)

Mazumder, M. K.; Overbey, R. L.; Testerman, M. K.

1976-01-01

Laser Doppler velocimeters (LDVs) were used as velocity microphones to measure sound pressure level in the range of 90-130 db, spectral components, and two-point cross correlation functions for acoustic noise source identification. Close agreement between LDV and microphone data is observed. It was concluded that directional sensitivity and the ability to measure remotely make LDVs useful tools for acoustic measurement where placement of any physical probe is difficult or undesirable, as in the diagnosis of jet aircraft noise.

[The application of cybernetic modeling methods for the forensic medical personality identification based on the voice and sounding speech characteristics].

PubMed

Kaganov, A Sh; Kir'yanov, P A

2015-01-01

The objective of the present publication was to discuss the possibility of application of cybernetic modeling methods to overcome the apparent discrepancy between two kinds of the speech records, viz. initial ones (e.g. obtained in the course of special investigation activities) and the voice prints obtained from the persons subjected to the criminalistic examination. The paper is based on the literature sources and the materials of original criminalistics expertises performed by the authors.

PROTAX-Sound: A probabilistic framework for automated animal sound identification

PubMed Central

Somervuo, Panu; Ovaskainen, Otso

2017-01-01

Autonomous audio recording is stimulating new field in bioacoustics, with a great promise for conducting cost-effective species surveys. One major current challenge is the lack of reliable classifiers capable of multi-species identification. We present PROTAX-Sound, a statistical framework to perform probabilistic classification of animal sounds. PROTAX-Sound is based on a multinomial regression model, and it can utilize as predictors any kind of sound features or classifications produced by other existing algorithms. PROTAX-Sound combines audio and image processing techniques to scan environmental audio files. It identifies regions of interest (a segment of the audio file that contains a vocalization to be classified), extracts acoustic features from them and compares with samples in a reference database. The output of PROTAX-Sound is the probabilistic classification of each vocalization, including the possibility that it represents species not present in the reference database. We demonstrate the performance of PROTAX-Sound by classifying audio from a species-rich case study of tropical birds. The best performing classifier achieved 68% classification accuracy for 200 bird species. PROTAX-Sound improves the classification power of current techniques by combining information from multiple classifiers in a manner that yields calibrated classification probabilities. PMID:28863178

PROTAX-Sound: A probabilistic framework for automated animal sound identification.

PubMed

de Camargo, Ulisses Moliterno; Somervuo, Panu; Ovaskainen, Otso

2017-01-01

Autonomous audio recording is stimulating new field in bioacoustics, with a great promise for conducting cost-effective species surveys. One major current challenge is the lack of reliable classifiers capable of multi-species identification. We present PROTAX-Sound, a statistical framework to perform probabilistic classification of animal sounds. PROTAX-Sound is based on a multinomial regression model, and it can utilize as predictors any kind of sound features or classifications produced by other existing algorithms. PROTAX-Sound combines audio and image processing techniques to scan environmental audio files. It identifies regions of interest (a segment of the audio file that contains a vocalization to be classified), extracts acoustic features from them and compares with samples in a reference database. The output of PROTAX-Sound is the probabilistic classification of each vocalization, including the possibility that it represents species not present in the reference database. We demonstrate the performance of PROTAX-Sound by classifying audio from a species-rich case study of tropical birds. The best performing classifier achieved 68% classification accuracy for 200 bird species. PROTAX-Sound improves the classification power of current techniques by combining information from multiple classifiers in a manner that yields calibrated classification probabilities.

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.

Evaluation of selective attention in patients with misophonia.

PubMed

Silva, Fúlvia Eduarda da; Sanchez, Tanit Ganz

2018-03-21

Misophonia is characterized by the aversion to very selective sounds, which evoke a strong emotional reaction. It has been inferred that misophonia, as well as tinnitus, is associated with hyperconnectivity between auditory and limbic systems. Individuals with bothersome tinnitus may have selective attention impairment, but it has not been demonstrated in case of misophonia yet. To characterize a sample of misophonic subjects and compare it with two control groups, one with tinnitus individuals (without misophonia) and the other with asymptomatic individuals (without misophonia and without tinnitus), regarding the selective attention. We evaluated 40 normal-hearing participants: 10 with misophonia, 10 with tinnitus (without misophonia) and 20 without tinnitus and without misophonia. In order to evaluate the selective attention, the dichotic sentence identification test was applied in three situations: firstly, the Brazilian Portuguese test was applied. Then, the same test was applied, combined with two competitive sounds: chewing sound (representing a sound that commonly triggers misophonia), and white noise (representing a common type of tinnitus which causes discomfort to patients). The dichotic sentence identification test with chewing sound, showed that the average of correct responses differed between misophonia and without tinnitus and without misophonia (p=0.027) and between misophonia and tinnitus (without misophonia) (p=0.002), in both cases lower in misophonia. Both, the dichotic sentence identification test alone, and with white noise, failed to show differences in the average of correct responses among the three groups (p≥0.452). The misophonia participants presented a lower percentage of correct responses in the dichotic sentence identification test with chewing sound; suggesting that individuals with misophonia may have selective attention impairment when they are exposed to sounds that trigger this condition. Copyright © 2018 Associação Brasileira de Otorrinolaringologia e Cirurgia Cérvico-Facial. Published by Elsevier Editora Ltda. All rights reserved.

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.

The aeromagnetic method as a tool to identify Cenozoic magmatism in the West Antarctic Rift System beneath the West Antarctic Ice Sheet: a review; Thiel subglacial volcano as possible source of the ash layer in the WAISCOR

USGS Publications Warehouse

Behrendt, John C.

2013-01-01

The West Antarctic Ice Sheet (WAIS) flows through the volcanically active West Antarctic Rift System (WARS). The aeromagnetic method has been the most useful geophysical tool for identification of subglacial volcanic rocks, since 1959–64 surveys, particularly combined with 1978 radar ice-sounding. The unique 1991–97 Central West Antarctica (CWA) aerogeophysical survey covering 354,000 km2 over the WAIS, (5-km line-spaced, orthogonal lines of aeromagnetic, radar ice-sounding, and aerogravity measurements), still provides invaluable information on subglacial volcanic rocks, particularly combined with the older aeromagnetic profiles. These data indicate numerous 100–>1000 nT, 5–50-km width, shallow-source, magnetic anomalies over an area greater than 1.2 × 106 km2, mostly from subglacial volcanic sources. I interpreted the CWA anomalies as defining about 1000 “volcanic centers” requiring high remanent normal magnetizations in the present field direction. About 400 anomaly sources correlate with bed topography. At least 80% of these sources have less than 200 m relief at the WAIS bed. They appear modified by moving ice, requiring a younger age than the WAIS (about 25 Ma). Exposed volcanoes in the WARS are The present rapid changes resulting from global warming, could be accelerated by subglacial volcanism.

Research on Humpback and Blue Whales off California, Oregon and Washington in 2001

DTIC Science & Technology

2002-12-01

northern Puget Sound , Grays Harbor, off Oregon, and Monterey Bay (Tables 4). Photographic identification from whale-watch boats was most extensive in...whales but some coverage for humpback whales was also conducted. • Photographic identification of gray whales was conducted in northern Puget Sound ...Farallones 53 3 1 4 N California 63 1 1 Oregon 72 1 1 Central Washigton 75 1 1 1 1 4 N Washington/BC 76 3 1 4 Puget Sound 79 1 1 Grand Total 5 3 8 21 13

Overnight consolidation promotes generalization across talkers in the identification of nonnative speech sounds.

PubMed

Earle, F Sayako; Myers, Emily B

2015-01-01

This investigation explored the generalization of phonetic learning across talkers following training on a nonnative (Hindi dental and retroflex) contrast. Participants were trained in two groups, either in the morning or in the evening. Discrimination and identification performance was assessed in the trained talker and an untrained talker three times over 24 h following training. Results suggest that overnight consolidation promotes generalization across talkers in identification, but not necessarily discrimination, of nonnative speech sounds.

On-road and wind-tunnel measurement of motorcycle helmet noise.

PubMed

Kennedy, J; Carley, M; Walker, I; Holt, N

2013-09-01

The noise source mechanisms involved in motorcycling include various aerodynamic sources and engine noise. The problem of noise source identification requires extensive data acquisition of a type and level that have not previously been applied. Data acquisition on track and on road are problematic due to rider safety constraints and the portability of appropriate instrumentation. One way to address this problem is the use of data from wind tunnel tests. The validity of these measurements for noise source identification must first be demonstrated. In order to achieve this extensive wind tunnel tests have been conducted and compared with the results from on-track measurements. Sound pressure levels as a function of speed were compared between on track and wind tunnel tests and were found to be comparable. Spectral conditioning techniques were applied to separate engine and wind tunnel noise from aerodynamic noise and showed that the aerodynamic components were equivalent in both cases. The spectral conditioning of on-track data showed that the contribution of engine noise to the overall noise is a function of speed and is more significant than had previously been thought. These procedures form a basis for accurate experimental measurements of motorcycle noise.

Active room compensation for sound reinforcement using sound field separation techniques.

PubMed

Heuchel, Franz M; Fernandez-Grande, Efren; Agerkvist, Finn T; Shabalina, Elena

2018-03-01

This work investigates how the sound field created by a sound reinforcement system can be controlled at low frequencies. An indoor control method is proposed which actively absorbs the sound incident on a reflecting boundary using an array of secondary sources. The sound field is separated into incident and reflected components by a microphone array close to the secondary sources, enabling the minimization of reflected components by means of optimal signals for the secondary sources. The method is purely feed-forward and assumes constant room conditions. Three different sound field separation techniques for the modeling of the reflections are investigated based on plane wave decomposition, equivalent sources, and the Spatial Fourier transform. Simulations and an experimental validation are presented, showing that the control method performs similarly well at enhancing low frequency responses with the three sound separation techniques. Resonances in the entire room are reduced, although the microphone array and secondary sources are confined to a small region close to the reflecting wall. Unlike previous control methods based on the creation of a plane wave sound field, the investigated method works in arbitrary room geometries and primary source positions.

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.

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.

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.

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.

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.

Automatic adventitious respiratory sound analysis: A systematic review

PubMed Central

Bowyer, Stuart; Rodriguez-Villegas, Esther

2017-01-01

Background Automatic detection or classification of adventitious sounds is useful to assist physicians in diagnosing or monitoring diseases such as asthma, Chronic Obstructive Pulmonary Disease (COPD), and pneumonia. While computerised respiratory sound analysis, specifically for the detection or classification of adventitious sounds, has recently been the focus of an increasing number of studies, a standardised approach and comparison has not been well established. Objective To provide a review of existing algorithms for the detection or classification of adventitious respiratory sounds. This systematic review provides a complete summary of methods used in the literature to give a baseline for future works. Data sources A systematic review of English articles published between 1938 and 2016, searched using the Scopus (1938-2016) and IEEExplore (1984-2016) databases. Additional articles were further obtained by references listed in the articles found. Search terms included adventitious sound detection, adventitious sound classification, abnormal respiratory sound detection, abnormal respiratory sound classification, wheeze detection, wheeze classification, crackle detection, crackle classification, rhonchi detection, rhonchi classification, stridor detection, stridor classification, pleural rub detection, pleural rub classification, squawk detection, and squawk classification. Study selection Only articles were included that focused on adventitious sound detection or classification, based on respiratory sounds, with performance reported and sufficient information provided to be approximately repeated. Data extraction Investigators extracted data about the adventitious sound type analysed, approach and level of analysis, instrumentation or data source, location of sensor, amount of data obtained, data management, features, methods, and performance achieved. Data synthesis A total of 77 reports from the literature were included in this review. 55 (71.43%) of the studies focused on wheeze, 40 (51.95%) on crackle, 9 (11.69%) on stridor, 9 (11.69%) on rhonchi, and 18 (23.38%) on other sounds such as pleural rub, squawk, as well as the pathology. Instrumentation used to collect data included microphones, stethoscopes, and accelerometers. Several references obtained data from online repositories or book audio CD companions. Detection or classification methods used varied from empirically determined thresholds to more complex machine learning techniques. Performance reported in the surveyed works were converted to accuracy measures for data synthesis. Limitations Direct comparison of the performance of surveyed works cannot be performed as the input data used by each was different. A standard validation method has not been established, resulting in different works using different methods and performance measure definitions. Conclusion A review of the literature was performed to summarise different analysis approaches, features, and methods used for the analysis. The performance of recent studies showed a high agreement with conventional non-automatic identification. This suggests that automated adventitious sound detection or classification is a promising solution to overcome the limitations of conventional auscultation and to assist in the monitoring of relevant diseases. PMID:28552969

Asymmetric cultural effects on perceptual expertise underlie an own-race bias for voices

PubMed Central

Perrachione, Tyler K.; Chiao, Joan Y.; Wong, Patrick C.M.

2009-01-01

The own-race bias in memory for faces has been a rich source of empirical work on the mechanisms of person perception. This effect is thought to arise because the face-perception system differentially encodes the relevant structural dimensions of features and their configuration based on experiences with different groups of faces. However, the effects of sociocultural experiences on person perception abilities in other identity-conveying modalities like audition have not been explored. Investigating an own-race bias in the auditory domain provides a unique opportunity for studying whether person identification is a modality-independent construct and how it is sensitive to asymmetric cultural experiences. Here we show that an own-race bias in talker identification arises from asymmetric experience with different spoken dialects. When listeners categorized voices by race (White or Black), a subset of the Black voices were categorized as sounding White, while the opposite case was unattested. Acoustic analyses indicated listeners' perceptions about race were consistent with differences in specific phonetic and phonological features. In a subsequent person-identification experiment, the Black voices initially categorized as sounding White elicited an own-race bias from White listeners, but not from Black listeners. These effects are inconsistent with person-perception models that strictly analogize faces and voices based on recognition from only structural features. Our results demonstrate that asymmetric exposure to spoken dialect, independent from talkers' physical characteristics, affects auditory perceptual expertise for talker identification. Person perception thus additionally relies on socioculturally-acquired dynamic information, which may be represented by different mechanisms in different sensory modalities. PMID:19782970

The influence of the level formants on the perception of synthetic vowel sounds

NASA Astrophysics Data System (ADS)

Kubzdela, Henryk; Owsianny, Mariuz

A computer model of a generator of periodic complex sounds simulating consonants was developed. The system makes possible independent regulation of the level of each of the formants and instant generation of the sound. A trapezoid approximates the curve of the spectrum within the range of the formant. In using this model, each person in a group of six listeners experimentally selected synthesis parameters for six sounds that to him seemed optimal approximations of Polish consonants. From these, another six sounds were selected that were identified by a majority of the six persons and several additional listeners as being best qualified to serve as prototypes of Polish consonants. These prototypes were then used to randomly create sounds with various combinations at the level of the second and third formant and these were presented to seven listeners for identification. The results of the identifications are presented in table form in three variants and are described from the point of view of the requirements of automatic recognition of consonants in continuous speech.

DETECTION AND IDENTIFICATION OF SPEECH SOUNDS USING CORTICAL ACTIVITY PATTERNS

PubMed Central

Centanni, T.M.; Sloan, A.M.; Reed, A.C.; Engineer, C.T.; Rennaker, R.; Kilgard, M.P.

2014-01-01

We have developed a classifier capable of locating and identifying speech sounds using activity from rat auditory cortex with an accuracy equivalent to behavioral performance without the need to specify the onset time of the speech sounds. This classifier can identify speech sounds from a large speech set within 40 ms of stimulus presentation. To compare the temporal limits of the classifier to behavior, we developed a novel task that requires rats to identify individual consonant sounds from a stream of distracter consonants. The classifier successfully predicted the ability of rats to accurately identify speech sounds for syllable presentation rates up to 10 syllables per second (up to 17.9 ± 1.5 bits/sec), which is comparable to human performance. Our results demonstrate that the spatiotemporal patterns generated in primary auditory cortex can be used to quickly and accurately identify consonant sounds from a continuous speech stream without prior knowledge of the stimulus onset times. Improved understanding of the neural mechanisms that support robust speech processing in difficult listening conditions could improve the identification and treatment of a variety of speech processing disorders. PMID:24286757

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.

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

Statistics of natural reverberation enable perceptual separation of sound and space

PubMed Central

Traer, James; McDermott, Josh H.

2016-01-01

In everyday listening, sound reaches our ears directly from a source as well as indirectly via reflections known as reverberation. Reverberation profoundly distorts the sound from a source, yet humans can both identify sound sources and distinguish environments from the resulting sound, via mechanisms that remain unclear. The core computational challenge is that the acoustic signatures of the source and environment are combined in a single signal received by the ear. Here we ask whether our recognition of sound sources and spaces reflects an ability to separate their effects and whether any such separation is enabled by statistical regularities of real-world reverberation. To first determine whether such statistical regularities exist, we measured impulse responses (IRs) of 271 spaces sampled from the distribution encountered by humans during daily life. The sampled spaces were diverse, but their IRs were tightly constrained, exhibiting exponential decay at frequency-dependent rates: Mid frequencies reverberated longest whereas higher and lower frequencies decayed more rapidly, presumably due to absorptive properties of materials and air. To test whether humans leverage these regularities, we manipulated IR decay characteristics in simulated reverberant audio. Listeners could discriminate sound sources and environments from these signals, but their abilities degraded when reverberation characteristics deviated from those of real-world environments. Subjectively, atypical IRs were mistaken for sound sources. The results suggest the brain separates sound into contributions from the source and the environment, constrained by a prior on natural reverberation. This separation process may contribute to robust recognition while providing information about spaces around us. PMID:27834730

Statistics of natural reverberation enable perceptual separation of sound and space.

PubMed

Traer, James; McDermott, Josh H

2016-11-29

In everyday listening, sound reaches our ears directly from a source as well as indirectly via reflections known as reverberation. Reverberation profoundly distorts the sound from a source, yet humans can both identify sound sources and distinguish environments from the resulting sound, via mechanisms that remain unclear. The core computational challenge is that the acoustic signatures of the source and environment are combined in a single signal received by the ear. Here we ask whether our recognition of sound sources and spaces reflects an ability to separate their effects and whether any such separation is enabled by statistical regularities of real-world reverberation. To first determine whether such statistical regularities exist, we measured impulse responses (IRs) of 271 spaces sampled from the distribution encountered by humans during daily life. The sampled spaces were diverse, but their IRs were tightly constrained, exhibiting exponential decay at frequency-dependent rates: Mid frequencies reverberated longest whereas higher and lower frequencies decayed more rapidly, presumably due to absorptive properties of materials and air. To test whether humans leverage these regularities, we manipulated IR decay characteristics in simulated reverberant audio. Listeners could discriminate sound sources and environments from these signals, but their abilities degraded when reverberation characteristics deviated from those of real-world environments. Subjectively, atypical IRs were mistaken for sound sources. The results suggest the brain separates sound into contributions from the source and the environment, constrained by a prior on natural reverberation. This separation process may contribute to robust recognition while providing information about spaces around us.

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.

21 CFR 870.1875 - Stethoscope.

Code of Federal Regulations, 2010 CFR

2010-04-01

...) Identification. A manual stethoscope is a mechanical device used to project the sounds associated with the heart... stethoscope is an electrically amplified device used to project the sounds associated with the heart, arteries...

21 CFR 870.1875 - Stethoscope.

Code of Federal Regulations, 2011 CFR

2011-04-01

...) Identification. A manual stethoscope is a mechanical device used to project the sounds associated with the heart... stethoscope is an electrically amplified device used to project the sounds associated with the heart, arteries...

21 CFR 870.1875 - Stethoscope.

Code of Federal Regulations, 2012 CFR

2012-04-01

...) Identification. A manual stethoscope is a mechanical device used to project the sounds associated with the heart... stethoscope is an electrically amplified device used to project the sounds associated with the heart, arteries...

21 CFR 870.1875 - Stethoscope.

Code of Federal Regulations, 2014 CFR

2014-04-01

...) Identification. A manual stethoscope is a mechanical device used to project the sounds associated with the heart... stethoscope is an electrically amplified device used to project the sounds associated with the heart, arteries...

21 CFR 870.1875 - Stethoscope.

Code of Federal Regulations, 2013 CFR

2013-04-01

...) Identification. A manual stethoscope is a mechanical device used to project the sounds associated with the heart... stethoscope is an electrically amplified device used to project the sounds associated with the heart, arteries...

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.

21 CFR 870.2390 - Phonocardiograph.

Code of Federal Regulations, 2010 CFR

2010-04-01

...) Identification. A phonocardiograph is a device used to amplify or condition the signal from a heart sound... display of the heart sounds. (b) Classification. Class I (general controls). The device is exempt from the...

21 CFR 870.2390 - Phonocardiograph.

Code of Federal Regulations, 2011 CFR

2011-04-01

...) Identification. A phonocardiograph is a device used to amplify or condition the signal from a heart sound... display of the heart sounds. (b) Classification. Class I (general controls). The device is exempt from the...

21 CFR 870.2390 - Phonocardiograph.

Code of Federal Regulations, 2013 CFR

2013-04-01

...) Identification. A phonocardiograph is a device used to amplify or condition the signal from a heart sound... display of the heart sounds. (b) Classification. Class I (general controls). The device is exempt from the...

21 CFR 870.2390 - Phonocardiograph.

Code of Federal Regulations, 2014 CFR

2014-04-01

...) Identification. A phonocardiograph is a device used to amplify or condition the signal from a heart sound... display of the heart sounds. (b) Classification. Class I (general controls). The device is exempt from the...

21 CFR 870.2390 - Phonocardiograph.

Code of Federal Regulations, 2012 CFR

2012-04-01

...) Identification. A phonocardiograph is a device used to amplify or condition the signal from a heart sound... display of the heart sounds. (b) Classification. Class I (general controls). The device is exempt from the...

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.

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.

A Syllable Segmentation, Letter-Sound, and Initial-Sound Intervention with Students Who Are Deaf or Hard of Hearing and Use Sign Language

ERIC Educational Resources Information Center

Tucci, Stacey L.; Easterbrooks, Susan R.

2015-01-01

This study investigated children's acquisition of three aspects of an early literacy curriculum, "Foundations for Literacy" ("Foundations"), designed specifically for prekindergarten students who are deaf or hard of hearing (DHH): syllable segmentation, identification of letter-sound correspondences, and initial-sound…

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.

Global Marine Gravity and Bathymetry at 1-Minute Resolution

NASA Astrophysics Data System (ADS)

Sandwell, D. T.; Smith, W. H.

2008-12-01

We have developed global gravity and bathymetry grids at 1-minute resolution. Three approaches are used to reduce the error in the satellite-derived marine gravity anomalies. First, we have retracked the raw waveforms from the ERS-1 and Geosat/GM missions resulting in improvements in range precision of 40% and 27%, respectively. Second, we have used the recently published EGM2008 global gravity model as a reference field to provide a seamless gravity transition from land to ocean. Third we have used a biharmonic spline interpolation method to construct residual vertical deflection grids. Comparisons between shipboard gravity and the global gravity grid show errors ranging from 2.0 mGal in the Gulf of Mexico to 4.0 mGal in areas with rugged seafloor topography. The largest errors occur on the crests of narrow large seamounts. The bathymetry grid is based on prediction from satellite gravity and available ship soundings. Global soundings were assembled from a wide variety of sources including NGDC/GEODAS, NOAA Coastal Relief, CCOM, IFREMER, JAMSTEC, NSF Polar Programs, UKHO, LDEO, HIG, SIO and numerous miscellaneous contributions. The National Geospatial-intelligence Agency and other volunteering hydrographic offices within the International Hydrographic Organization provided global significant shallow water (< 300 m) soundings derived from their nautical charts. All soundings were converted to a common format and were hand-edited in relation to a smooth bathymetric model. Land elevations and shoreline location are based on a combination SRTM30, GTOPO30, and ICESAT data. A new feature of the bathymetry grid is a matching grid of source identification number that enables one to establish the origin of the depth estimate in each grid cell. Both the gravity and bathymetry grids are freely available.

Spatial and identity negative priming in audition: evidence of feature binding in auditory spatial memory.

PubMed

Mayr, Susanne; Buchner, Axel; Möller, Malte; Hauke, Robert

2011-08-01

Two experiments are reported with identical auditory stimulation in three-dimensional space but with different instructions. Participants localized a cued sound (Experiment 1) or identified a sound at a cued location (Experiment 2). A distractor sound at another location had to be ignored. The prime distractor and the probe target sound were manipulated with respect to sound identity (repeated vs. changed) and location (repeated vs. changed). The localization task revealed a symmetric pattern of partial repetition costs: Participants were impaired on trials with identity-location mismatches between the prime distractor and probe target-that is, when either the sound was repeated but not the location or vice versa. The identification task revealed an asymmetric pattern of partial repetition costs: Responding was slowed down when the prime distractor sound was repeated as the probe target, but at another location; identity changes at the same location were not impaired. Additionally, there was evidence of retrieval of incompatible prime responses in the identification task. It is concluded that feature binding of auditory prime distractor information takes place regardless of whether the task is to identify or locate a sound. Instructions determine the kind of identity-location mismatch that is detected. Identity information predominates over location information in auditory memory.

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.

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.

Hearing Scenes: A Neuromagnetic Signature of Auditory Source and Reverberant Space Separation

PubMed Central

Oliva, Aude

2017-01-01

Abstract Perceiving the geometry of surrounding space is a multisensory process, crucial to contextualizing object perception and guiding navigation behavior. Humans can make judgments about surrounding spaces from reverberation cues, caused by sounds reflecting off multiple interior surfaces. However, it remains unclear how the brain represents reverberant spaces separately from sound sources. Here, we report separable neural signatures of auditory space and source perception during magnetoencephalography (MEG) recording as subjects listened to brief sounds convolved with monaural room impulse responses (RIRs). The decoding signature of sound sources began at 57 ms after stimulus onset and peaked at 130 ms, while space decoding started at 138 ms and peaked at 386 ms. Importantly, these neuromagnetic responses were readily dissociable in form and time: while sound source decoding exhibited an early and transient response, the neural signature of space was sustained and independent of the original source that produced it. The reverberant space response was robust to variations in sound source, and vice versa, indicating a generalized response not tied to specific source-space combinations. These results provide the first neuromagnetic evidence for robust, dissociable auditory source and reverberant space representations in the human brain and reveal the temporal dynamics of how auditory scene analysis extracts percepts from complex naturalistic auditory signals. PMID:28451630

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.

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.

Techniques and instrumentation for the measurement of transient sound energy flux

NASA Astrophysics Data System (ADS)

Watkinson, P. S.; Fahy, F. J.

1983-12-01

The evaluation of sound intensity distributions, and sound powers, of essentially continuous sources such as automotive engines, electric motors, production line machinery, furnaces, earth moving machinery and various types of process plants were studied. Although such systems are important sources of community disturbance and, to a lesser extent, of industrial health hazard, the most serious sources of hearing hazard in industry are machines operating on an impact principle, such as drop forges, hammers and punches. Controlled experiments to identify major noise source regions and mechanisms are difficult because it is normally impossible to install them in quiet, anechoic environments. The potential for sound intensity measurement to provide a means of overcoming these difficulties has given promising results, indicating the possibility of separation of directly radiated and reverberant sound fields. However, because of the complexity of transient sound fields, a fundamental investigation is necessary to establish the practicability of intensity field decomposition, which is basic to source characterization techniques.

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.

Objective approach for analysis of noise source characteristics and acoustic conditions in noisy computerized embroidery workrooms.

PubMed

Aliabadi, Mohsen; Golmohammadi, Rostam; Mansoorizadeh, Muharram

2014-03-01

It is highly important to analyze the acoustic properties of workrooms in order to identify best noise control measures from the standpoint of noise exposure limits. Due to the fact that sound pressure is dependent upon environments, it cannot be a suitable parameter for determining the share of workroom acoustic characteristics in producing noise pollution. This paper aims to empirically analyze noise source characteristics and acoustic properties of noisy embroidery workrooms based on special parameters. In this regard, reverberation time as the special room acoustic parameter in 30 workrooms was measured based on ISO 3382-2. Sound power quantity of embroidery machines was also determined based on ISO 9614-3. Multiple linear regression was employed for predicting reverberation time based on acoustic features of the workrooms using MATLAB software. The results showed that the measured reverberation times in most of the workrooms were approximately within the ranges recommended by ISO 11690-1. Similarity between reverberation time values calculated by the Sabine formula and measured values was relatively poor (R (2) = 0.39). This can be due to the inaccurate estimation of the acoustic influence of furniture and formula preconditions. Therefore, this value cannot be considered representative of an actual acoustic room. However, the prediction performance of the regression method with root mean square error (RMSE) = 0.23 s and R (2) = 0.69 is relatively acceptable. Because the sound power of the embroidery machines was relatively high, these sources get the highest priority when it comes to applying noise controls. Finally, an objective approach for the determination of the share of workroom acoustic characteristics in producing noise could facilitate the identification of cost-effective noise controls.

IDENTIFICATION OF TOXICANTS IN WHOLE MARINE SEDIMENTS: METHODS AND RESULTS

EPA Science Inventory

Identification of stressors in aquatic systems is critical to sound assessment and management of our nation's waterways. Information from stressor identification can be useful in designing effective sediment remediation methods, assessing options for sediment disposal, allowing m...

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.

Interior sound field control using generalized singular value decomposition in the frequency domain.

PubMed

Pasco, Yann; Gauthier, Philippe-Aubert; Berry, Alain; Moreau, Stéphane

2017-01-01

The problem of controlling a sound field inside a region surrounded by acoustic control sources is considered. Inspired by the Kirchhoff-Helmholtz integral, the use of double-layer source arrays allows such a control and avoids the modification of the external sound field by the control sources by the approximation of the sources as monopole and radial dipole transducers. However, the practical implementation of the Kirchhoff-Helmholtz integral in physical space leads to large numbers of control sources and error sensors along with excessive controller complexity in three dimensions. The present study investigates the potential of the Generalized Singular Value Decomposition (GSVD) to reduce the controller complexity and separate the effect of control sources on the interior and exterior sound fields, respectively. A proper truncation of the singular basis provided by the GSVD factorization is shown to lead to effective cancellation of the interior sound field at frequencies below the spatial Nyquist frequency of the control sources array while leaving the exterior sound field almost unchanged. Proofs of concept are provided through simulations achieved for interior problems by simulations in a free field scenario with circular arrays and in a reflective environment with square arrays.

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.

Acoustic characterization of a nonlinear vibroacoustic absorber at low frequencies and high sound levels

NASA Astrophysics Data System (ADS)

Chauvin, A.; Monteil, M.; Bellizzi, S.; Côte, R.; Herzog, Ph.; Pachebat, M.

2018-03-01

A nonlinear vibroacoustic absorber (Nonlinear Energy Sink: NES), involving a clamped thin membrane made in Latex, is assessed in the acoustic domain. This NES is here considered as an one-port acoustic system, analyzed at low frequencies and for increasing excitation levels. This dynamic and frequency range requires a suitable experimental technique, which is presented first. It involves a specific impedance tube able to deal with samples of sufficient size, and reaching high sound levels with a guaranteed linear response thank's to a specific acoustic source. The identification method presented here requires a single pressure measurement, and is calibrated from a set of known acoustic loads. The NES reflection coefficient is then estimated at increasing source levels, showing its strong level dependency. This is presented as a mean to understand energy dissipation. The results of the experimental tests are first compared to a nonlinear viscoelastic model of the membrane absorber. In a second step, a family of one degree of freedom models, treated as equivalent Helmholtz resonators is identified from the measurements, allowing a parametric description of the NES behavior over a wide range of levels.

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

The Incongruency Advantage for Environmental Sounds Presented in Natural Auditory Scenes

ERIC Educational Resources Information Center

Gygi, Brian; Shafiro, Valeriy

2011-01-01

The effect of context on the identification of common environmental sounds (e.g., dogs barking or cars honking) was tested by embedding them in familiar auditory background scenes (street ambience, restaurants). Initial results with subjects trained on both the scenes and the sounds to be identified showed a significant advantage of about five…

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.

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.

AN OVERVIEW OF TOXICANT IDENTIFICATION IN SEDIMENTS AND DREDGED MATERIALS

EPA Science Inventory

The identification of toxicants affecting aquatic benthic systems is critical to sound assessment and management of our nation?s waterways. Identification of toxicants can be useful in designing effective sediment remediation plans and reasonable options for sediment disposal. K...

IDENTIFICATION OF STRESSORS IN TOXIC SEDIMENTS: WHOLE SEDIMENT AND INSTITIAL WATER RESULTS

EPA Science Inventory

Identification of stressors in aquatic systems is critical to sound assessment and management of our nation's waterways. Information from stressor identification can be useful in designing effective sediment remediation methods, assessing options for sediment disposal, allowing m...

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.

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.

Identification and Remediation of Phonological and Motor Errors in Acquired Sound Production Impairment

PubMed Central

Gagnon, Bernadine; Miozzo, Michele

2017-01-01

Purpose This study aimed to test whether an approach to distinguishing errors arising in phonological processing from those arising in motor planning also predicts the extent to which repetition-based training can lead to improved production of difficult sound sequences. Method Four individuals with acquired speech production impairment who produced consonant cluster errors involving deletion were examined using a repetition task. We compared the acoustic details of productions with deletion errors in target consonant clusters to singleton consonants. Changes in accuracy over the course of the study were also compared. Results Two individuals produced deletion errors consistent with a phonological locus of the errors, and 2 individuals produced errors consistent with a motoric locus of the errors. The 2 individuals who made phonologically driven errors showed no change in performance on a repetition training task, whereas the 2 individuals with motoric errors improved in their production of both trained and untrained items. Conclusions The results extend previous findings about a metric for identifying the source of sound production errors in individuals with both apraxia of speech and aphasia. In particular, this work may provide a tool for identifying predominant error types in individuals with complex deficits. PMID:28655044

Modeling and observations of an elevated, moving infrasonic source: Eigenray methods.

PubMed

Blom, Philip; Waxler, Roger

2017-04-01

The acoustic ray tracing relations are extended by the inclusion of auxiliary parameters describing variations in the spatial ray coordinates and eikonal vector due to changes in the initial conditions. Computation of these parameters allows one to define the geometric spreading factor along individual ray paths and assists in identification of caustic surfaces so that phase shifts can be easily identified. A method is developed leveraging the auxiliary parameters to identify propagation paths connecting specific source-receiver geometries, termed eigenrays. The newly introduced method is found to be highly efficient in cases where propagation is non-planar due to horizontal variations in the propagation medium or the presence of cross winds. The eigenray method is utilized in analysis of infrasonic signals produced by a multi-stage sounding rocket launch with promising results for applications of tracking aeroacoustic sources in the atmosphere and specifically to analysis of motor performance during dynamic tests.

Toward a Nonspeech Test of Auditory Cognition: Semantic Context Effects in Environmental Sound Identification in Adults of Varying Age and Hearing Abilities

PubMed Central

Sheft, Stanley; Norris, Molly; Spanos, George; Radasevich, Katherine; Formsma, Paige; Gygi, Brian

2016-01-01

Objective Sounds in everyday environments tend to follow one another as events unfold over time. The tacit knowledge of contextual relationships among environmental sounds can influence their perception. We examined the effect of semantic context on the identification of sequences of environmental sounds by adults of varying age and hearing abilities, with an aim to develop a nonspeech test of auditory cognition. Method The familiar environmental sound test (FEST) consisted of 25 individual sounds arranged into ten five-sound sequences: five contextually coherent and five incoherent. After hearing each sequence, listeners identified each sound and arranged them in the presentation order. FEST was administered to young normal-hearing, middle-to-older normal-hearing, and middle-to-older hearing-impaired adults (Experiment 1), and to postlingual cochlear-implant users and young normal-hearing adults tested through vocoder-simulated implants (Experiment 2). Results FEST scores revealed a strong positive effect of semantic context in all listener groups, with young normal-hearing listeners outperforming other groups. FEST scores also correlated with other measures of cognitive ability, and for CI users, with the intelligibility of speech-in-noise. Conclusions Being sensitive to semantic context effects, FEST can serve as a nonspeech test of auditory cognition for diverse listener populations to assess and potentially improve everyday listening skills. PMID:27893791

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

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.

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.

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

Sound production due to large-scale coherent structures. [and identification of noise mechanisms in turbulent shear flow

NASA Technical Reports Server (NTRS)

Gatski, T. B.

1979-01-01

The sound due to the large-scale (wavelike) structure in an infinite free turbulent shear flow is examined. Specifically, a computational study of a plane shear layer is presented, which accounts, by way of triple decomposition of the flow field variables, for three distinct component scales of motion (mean, wave, turbulent), and from which the sound - due to the large-scale wavelike structure - in the acoustic field can be isolated by a simple phase average. The computational approach has allowed for the identification of a specific noise production mechanism, viz the wave-induced stress, and has indicated the effect of coherent structure amplitude and growth and decay characteristics on noise levels produced in the acoustic far field.

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

IDENTIFICATION AND EVALUATION OF STRESSORS IN TOXIC SEDIMENTS AND DREDGED MATERIALS

EPA Science Inventory

Identification of stressors in aquatic systems is critical to sound assessment and management of our nation's waterways for a number of reasons. Identification of specific classes of toxicants (or stressors) can be useful in designing effective sediment remediation methods and re...

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.

Effects of the Exxon Valdez oil spill on the abundance and distribution of humpback whales (`megaptera novaeangliae`) in Prince William Sound. Marine mammal study number 1. Exxon Valdez oil spill state/federal natural resource damage assessment final report

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

Dahlheim, M.E.; Von Ziegesar, O.

1993-12-01

Photo-identification studies of Prince William Sound humpback whales were conducted from May to September in 1989 and 1990 to assess the impact on the spill on humpback whale life history and ecology. In 1989, concurrent studies were conducted in Southeast Alaska on humpback whales to determine if whales avoided contaminated waters of Prince William Sound and moved to other northern feeding areas. In 1989, photographic analysis of Prince William Sound humpbacks resulted in the identification of 59 whales. In 1990, 66 whales were documented. The increase in whale sightings may have been due to the increase in effort during themore » 1989 and 1990 season. Because of the difference in survey effort before and after the spill, it is impossible to determine if there was a difference in the number of humpback whales using the Sound. Distribution varied among years and may be related to prey distribution.« less

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.

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

Causal Uncertainty in the Identification of Environmental Sounds

DTIC Science & Technology

1986-11-01

importance of particular stimulus properties (Chaney & Webster, 1966; Howard, 1977; Mackie, Wylie, Ridihalgh, Shultz, & Seltzer, 1981; Talamo , 1982; Warren...207). Berlin: Abakon Verlagsgesellschaft, 183-207. Talamo , .J. D. C. (1982). The perception of machinery indicator sounds. ErgonoriS_ 225, 41-51

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.

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

Hybrid CFD/CAA Modeling for Liftoff Acoustic Predictions

NASA Technical Reports Server (NTRS)

Strutzenberg, Louise L.; Liever, Peter A.

2011-01-01

This paper presents development efforts at the NASA Marshall Space flight Center to establish a hybrid Computational Fluid Dynamics and Computational Aero-Acoustics (CFD/CAA) simulation system for launch vehicle liftoff acoustics environment analysis. Acoustic prediction engineering tools based on empirical jet acoustic strength and directivity models or scaled historical measurements are of limited value in efforts to proactively design and optimize launch vehicles and launch facility configurations for liftoff acoustics. CFD based modeling approaches are now able to capture the important details of vehicle specific plume flow environment, identifY the noise generation sources, and allow assessment of the influence of launch pad geometric details and sound mitigation measures such as water injection. However, CFD methodologies are numerically too dissipative to accurately capture the propagation of the acoustic waves in the large CFD models. The hybrid CFD/CAA approach combines the high-fidelity CFD analysis capable of identifYing the acoustic sources with a fast and efficient Boundary Element Method (BEM) that accurately propagates the acoustic field from the source locations. The BEM approach was chosen for its ability to properly account for reflections and scattering of acoustic waves from launch pad structures. The paper will present an overview of the technology components of the CFD/CAA framework and discuss plans for demonstration and validation against test data.

In-duct identification of fluid-borne source with high spatial resolution

NASA Astrophysics Data System (ADS)

Heo, Yong-Ho; Ih, Jeong-Guon; Bodén, Hans

2014-11-01

Source identification of acoustic characteristics of in-duct fluid machinery is required for coping with the fluid-borne noise. By knowing the acoustic pressure and particle velocity field at the source plane in detail, the sound generation mechanism of a fluid machine can be understood. The identified spatial distribution of the strength of major radiators would be useful for the low noise design. Conventional methods for measuring the source in a wide duct have not been very helpful in investigating the source properties in detail because their spatial resolution is improper for the design purpose. In this work, an inverse method to estimate the source parameters with a high spatial resolution is studied. The theoretical formulation including the evanescent modes and near-field measurement data is given for a wide duct. After validating the proposed method to a duct excited by an acoustic driver, an experiment on a duct system driven by an air blower is conducted in the presence of flow. A convergence test for the evanescent modes is performed to find the necessary number of modes to regenerate the measured pressure field precisely. By using the converged modal amplitudes, very-close near-field pressure to the source is regenerated and compared with the measured pressure, and the maximum error was -16.3 dB. The source parameters are restored from the converged modal amplitudes. Then, the distribution of source parameters on the driver and the blower is clearly revealed with a high spatial resolution for kR<1.84 in which range only plane waves can propagate to far field in a duct. Measurement using a flush mounted sensor array is discussed, and the removal of pure radial modes in the modeling is suggested.

Separation of non-stationary multi-source sound field based on the interpolated time-domain equivalent source method

NASA Astrophysics Data System (ADS)

Bi, Chuan-Xing; Geng, Lin; Zhang, Xiao-Zheng

2016-05-01

In the sound field with multiple non-stationary sources, the measured pressure is the sum of the pressures generated by all sources, and thus cannot be used directly for studying the vibration and sound radiation characteristics of every source alone. This paper proposes a separation model based on the interpolated time-domain equivalent source method (ITDESM) to separate the pressure field belonging to every source from the non-stationary multi-source sound field. In the proposed method, ITDESM is first extended to establish the relationship between the mixed time-dependent pressure and all the equivalent sources distributed on every source with known location and geometry information, and all the equivalent source strengths at each time step are solved by an iterative solving process; then, the corresponding equivalent source strengths of one interested source are used to calculate the pressure field generated by that source alone. Numerical simulation of two baffled circular pistons demonstrates that the proposed method can be effective in separating the non-stationary pressure generated by every source alone in both time and space domains. An experiment with two speakers in a semi-anechoic chamber further evidences the effectiveness of the proposed method.

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.

Sound transmission loss of windows on high speed trains

NASA Astrophysics Data System (ADS)

Zhang, Yumei; Xiao, Xinbiao; Thompson, David; Squicciarini, Giacomo; Wen, Zefeng; Li, Zhihui; Wu, Yue

2016-09-01

The window is one of the main components of the high speed train car body structure through which noise can be transmitted. To study the windows’ acoustic properties, the vibration of one window of a high speed train has been measured for a running speed of 250 km/h. The corresponding interior noise and the noise in the wheel-rail area have been measured simultaneously. The experimental results show that the window vibration velocity has a similar spectral shape to the interior noise. Interior noise source identification further indicates that the window makes a contribution to the interior noise. Improvement of the window's Sound Transmission Loss (STL) can reduce the interior noise from this transmission path. An STL model of the window is built based on wave propagation and modal superposition methods. From the theoretical results, the window's STL property is studied and several factors affecting it are investigated, which provide indications for future low noise design of high speed train windows.

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

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.

Comparison of speech perception performance between Sprint/Esprit 3G and Freedom processors in children implanted with nucleus cochlear implants.

PubMed

Santarelli, Rosamaria; Magnavita, Vincenzo; De Filippi, Roberta; Ventura, Laura; Genovese, Elisabetta; Arslan, Edoardo

2009-04-01

To compare speech perception performance in children fitted with previous generation Nucleus sound processor, Sprint or Esprit 3G, and the Freedom, the most recently released system from the Cochlear Corporation that features a larger input dynamic range. Prospective intrasubject comparative study. University Medical Center. Seventeen prelingually deafened children who had received the Nucleus 24 cochlear implant and used the Sprint or Esprit 3G sound processor. Cochlear implantation with Cochlear device. Speech perception was evaluated at baseline (Sprint, n = 11; Esprit 3G, n = 6) and after 1 month's experience with the Freedom sound processor. Identification and recognition of disyllabic words and identification of vowels were performed via recorded voice in quiet (70 dB [A]), in the presence of background noise at various levels of signal-to-noise ratio (+10, +5, 0, -5) and at a soft presentation level (60 dB [A]). Consonant identification and recognition of disyllabic words, trisyllabic words, and sentences were evaluated in live voice. Frequency discrimination was measured in a subset of subjects (n = 5) by using an adaptive, 3-interval, 3-alternative, forced-choice procedure. Identification of disyllabic words administered at a soft presentation level showed a significant increase when switching to the Freedom compared with the previously worn processor in children using the Sprint or Esprit 3G. Identification and recognition of disyllabic words in the presence of background noise as well as consonant identification and sentence recognition increased significantly for the Freedom compared with the previously worn device only in children fitted with the Sprint. Frequency discrimination was significantly better when switching to the Freedom compared with the previously worn processor. Serial comparisons revealed that that speech perception performance evaluated in children aged 5 to 15 years was superior with the Freedom than previous generations of Nucleus sound processors. These differences are deemed to ensue from an increased input dynamic range, a feature that offers potentially enhanced phonemic discrimination.

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.

Developing the STS sound pollution unit for enhancing students' applying knowledge among science technology engineering and mathematics

NASA Astrophysics Data System (ADS)

Jumpatong, Sutthaya; Yuenyong, Chokchai

2018-01-01

STEM education suggested that students should be enhanced to learn science with integration between Science, Technology, Engineering and Mathematics. To help Thai students make sense of relationship between Science, Technology, Engineering and Mathematics, this paper presents learning activities of STS Sound Pollution. The developing of STS Sound Pollution is a part of research that aimed to enhance students' perception of the relationship between Science Technology Engineering and Mathematics. This paper will discuss how to develop Sound Pollution through STS approach in framework of Yuenyong (2006) where learning activities were provided based on 5 stages. These included (1) identification of social issues, (2) identification of potential solutions, (3) need for knowledge, (4) decisionmaking, and (5) socialization stage. The learning activities could be highlighted as following. First stage, we use video clip of `Problem of people about Sound Pollution'. Second stage, students will need to identification of potential solutions by design Home/Factory without noisy. The need of scientific and other knowledge will be proposed for various alternative solutions. Third stage, students will gain their scientific knowledge through laboratory and demonstration of sound wave. Fourth stage, students have to make decision for the best solution of designing safety Home/Factory based on their scientific knowledge and others (e.g. mathematics, economics, art, value, and so on). Finally, students will present and share their Design Safety Home/Factory in society (e.g. social media or exhibition) in order to validate their ideas and redesigning. The paper, then, will discuss how those activities would allow students' applying knowledge of science technology engineering, mathematics and others (art, culture and value) for their possible solution of the STS issues.

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

43 CFR 4750.2-1 - Health and identification requirements.

Code of Federal Regulations, 2014 CFR

2014-10-01

... 43 Public Lands: Interior 2 2014-10-01 2014-10-01 false Health and identification requirements... CONTROL OF WILD FREE-ROAMING HORSES AND BURROS Private Maintenance § 4750.2-1 Health and identification... animal's soundness and good health, determine its age and sex, and administer immunizations, worming...

43 CFR 4750.2-1 - Health and identification requirements.

Code of Federal Regulations, 2012 CFR

2012-10-01

... 43 Public Lands: Interior 2 2012-10-01 2012-10-01 false Health and identification requirements... CONTROL OF WILD FREE-ROAMING HORSES AND BURROS Private Maintenance § 4750.2-1 Health and identification... animal's soundness and good health, determine its age and sex, and administer immunizations, worming...

43 CFR 4750.2-1 - Health and identification requirements.

Code of Federal Regulations, 2011 CFR

2011-10-01

... 43 Public Lands: Interior 2 2011-10-01 2011-10-01 false Health and identification requirements... CONTROL OF WILD FREE-ROAMING HORSES AND BURROS Private Maintenance § 4750.2-1 Health and identification... animal's soundness and good health, determine its age and sex, and administer immunizations, worming...

43 CFR 4750.2-1 - Health and identification requirements.

Code of Federal Regulations, 2013 CFR

2013-10-01

... 43 Public Lands: Interior 2 2013-10-01 2013-10-01 false Health and identification requirements... CONTROL OF WILD FREE-ROAMING HORSES AND BURROS Private Maintenance § 4750.2-1 Health and identification... animal's soundness and good health, determine its age and sex, and administer immunizations, worming...

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

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.

Identification and Evaluation of Underground Obstacle Sensor Embodiments for Their Applicability to the Combat Engineers’ Rapid Excavation Mission(s),

DTIC Science & Technology

1980-01-01

November 1976. 11. Ohio State University, Electroscience Laboratory, Electromagnetic Pulse Sounding for Geological Surveying with Application in Rock...Peters, L. and Moffatt, D. L., Electromagnetic Pulse Sounding for Geological Surveying with Application in Rock Mechanics and Rapid Excavation... Electromagnetic Pulse Sounding for Geolog- ical Surveying with Application in Rock Mechanics and Rapid Excava- tion Program, Ohio State University, Report

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.

Echo Source Discrimination in Airborne Radar Sounding Data for Mars Analog Studies, Dry Valleys, Antarctica

NASA Technical Reports Server (NTRS)

Holt, J. W.; Blankenship, D. D.; Peters, M. E.; Kempf, S. D.; Morse, D. L.; Williams, B. J.

2003-01-01

The recent identification of features on Mars exhibiting morphologies consistent with ice/rock mixtures, near-surface ice bodies and near-surface liquid water [1,2], and the importance of such features to the search for water on Mars, highlights the need for appropriate terrestrial analogs in order to prepare for upcoming radar missions targeting these and other water-related features. Climatic, hydrological, and geological conditions in the McMurdo Dry Valleys of Antarctica are analogous in many ways to those on Mars, and a number of ice-related features in the Dry Valleys may have direct morphologic and compositional counterparts on Mars.

Sound reduction of air compressors using a systematic approach

NASA Astrophysics Data System (ADS)

Moylan, Justin Tharp

The noise emitted by portable electric air compressors can often be a nuisance or potentially hazardous to the operator or others nearby. Therefore, reducing the noise of these air compressors is desired. This research focuses on compressors with a reciprocating piston design as this is the most common type of pump design for portable compressors. An experimental setup was developed to measure the sound and vibration of the air compressors, including testing inside a semi-anechoic chamber. The design of a quiet air compressor was performed in four stages: 1) Teardown and benchmarking of air compressors, 2) Identification and isolation of noise sources, 3) Development of individual means to quiet noise sources, 4) Selection and testing of integrated solutions. The systematic approach and results for each of these stages will be discussed. Two redesigned solutions were developed and measured to be approximately 65% quieter than the previous unmodified compressor. An additional analysis was performed on the solutions selected by the participants involved in the selection process. This analysis involved determining which of the design criteria each participant considered most important when selecting solutions. The results from each participant were then compared to their educational background and experience and correlations were identified. The correlations discovered suggest that educational background and experience may be key determinants for the preference models developed.

Visually-guided attention enhances target identification in a complex auditory scene.

PubMed

Best, Virginia; Ozmeral, Erol J; Shinn-Cunningham, Barbara G

2007-06-01

In auditory scenes containing many similar sound sources, sorting of acoustic information into streams becomes difficult, which can lead to disruptions in the identification of behaviorally relevant targets. This study investigated the benefit of providing simple visual cues for when and/or where a target would occur in a complex acoustic mixture. Importantly, the visual cues provided no information about the target content. In separate experiments, human subjects either identified learned birdsongs in the presence of a chorus of unlearned songs or recalled strings of spoken digits in the presence of speech maskers. A visual cue indicating which loudspeaker (from an array of five) would contain the target improved accuracy for both kinds of stimuli. A cue indicating which time segment (out of a possible five) would contain the target also improved accuracy, but much more for birdsong than for speech. These results suggest that in real world situations, information about where a target of interest is located can enhance its identification, while information about when to listen can also be helpful when targets are unfamiliar or extremely similar to their competitors.

Visually-guided Attention Enhances Target Identification in a Complex Auditory Scene

PubMed Central

Ozmeral, Erol J.; Shinn-Cunningham, Barbara G.

2007-01-01

In auditory scenes containing many similar sound sources, sorting of acoustic information into streams becomes difficult, which can lead to disruptions in the identification of behaviorally relevant targets. This study investigated the benefit of providing simple visual cues for when and/or where a target would occur in a complex acoustic mixture. Importantly, the visual cues provided no information about the target content. In separate experiments, human subjects either identified learned birdsongs in the presence of a chorus of unlearned songs or recalled strings of spoken digits in the presence of speech maskers. A visual cue indicating which loudspeaker (from an array of five) would contain the target improved accuracy for both kinds of stimuli. A cue indicating which time segment (out of a possible five) would contain the target also improved accuracy, but much more for birdsong than for speech. These results suggest that in real world situations, information about where a target of interest is located can enhance its identification, while information about when to listen can also be helpful when targets are unfamiliar or extremely similar to their competitors. PMID:17453308

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.

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

The Central Role of Recognition in Auditory Perception: A Neurobiological Model

ERIC Educational Resources Information Center

McLachlan, Neil; Wilson, Sarah

2010-01-01

The model presents neurobiologically plausible accounts of sound recognition (including absolute pitch), neural plasticity involved in pitch, loudness and location information integration, and streaming and auditory recall. It is proposed that a cortical mechanism for sound identification modulates the spectrotemporal response fields of inferior…

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.

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

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.

Quantifying chemical reactions by using mixing analysis.

PubMed

Jurado, Anna; Vázquez-Suñé, Enric; Carrera, Jesús; Tubau, Isabel; Pujades, Estanislao

2015-01-01

This work is motivated by a sound understanding of the chemical processes that affect the organic pollutants in an urban aquifer. We propose an approach to quantify such processes using mixing calculations. The methodology consists of the following steps: (1) identification of the recharge sources (end-members) and selection of the species (conservative and non-conservative) to be used, (2) identification of the chemical processes and (3) evaluation of mixing ratios including the chemical processes. This methodology has been applied in the Besòs River Delta (NE Barcelona, Spain), where the River Besòs is the main aquifer recharge source. A total number of 51 groundwater samples were collected from July 2007 to May 2010 during four field campaigns. Three river end-members were necessary to explain the temporal variability of the River Besòs: one river end-member is from the wet periods (W1) and two are from dry periods (D1 and D2). This methodology has proved to be useful not only to compute the mixing ratios but also to quantify processes such as calcite and magnesite dissolution, aerobic respiration and denitrification undergone at each observation point. Copyright © 2014 Elsevier B.V. All rights reserved.

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

Minimal Pair Distinctions and Intelligibility in Preschool Children with and without Speech Sound Disorders

ERIC Educational Resources Information Center

Hodge, Megan M.; Gotzke, Carrie L.

2011-01-01

Listeners' identification of young children's productions of minimally contrastive words and predictive relationships between accurately identified words and intelligibility scores obtained from a 100-word spontaneous speech sample were determined for 36 children with typically developing speech (TDS) and 36 children with speech sound disorders…

Children's Auditory Perception of Movement of Traffic Sounds.

ERIC Educational Resources Information Center

Pfeffer, K.; Barnecutt, P.

1996-01-01

Examined children's auditory perception of traffic sounds, focusing on identification of vehicle movement. Subjects were 60 children of 5, 8, and 11 years. Results indicated that the auditory perception of movement was a problem area for children, especially five-year olds. Discussed the role of attention-demanding characteristics of some traffic…

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.

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.

HAZARDOUS WASTE IDENTIFICATION

EPA Science Inventory

This research is in direct support of the regulatory reform efforts under the Hazarous Waste Identification (HWIR) and is related to the development of national "exit levels" based on sound scientific data and models. Research focuses on developing a systems approach to modelin...

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.

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

Blind separation of incoherent and spatially disjoint sound sources

NASA Astrophysics Data System (ADS)

Dong, Bin; Antoni, Jérôme; Pereira, Antonio; Kellermann, Walter

2016-11-01

Blind separation of sound sources aims at reconstructing the individual sources which contribute to the overall radiation of an acoustical field. The challenge is to reach this goal using distant measurements when all sources are operating concurrently. The working assumption is usually that the sources of interest are incoherent - i.e. statistically orthogonal - so that their separation can be approached by decorrelating a set of simultaneous measurements, which amounts to diagonalizing the cross-spectral matrix. Principal Component Analysis (PCA) is traditionally used to this end. This paper reports two new findings in this context. First, a sufficient condition is established under which "virtual" sources returned by PCA coincide with true sources; it stipulates that the sources of interest should be not only incoherent but also spatially orthogonal. A particular case of this instance is met by spatially disjoint sources - i.e. with non-overlapping support sets. Second, based on this finding, a criterion that enforces both statistical and spatial orthogonality is proposed to blindly separate incoherent sound sources which radiate from disjoint domains. This criterion can be easily incorporated into acoustic imaging algorithms such as beamforming or acoustical holography to identify sound sources of different origins. The proposed methodology is validated on laboratory experiments. In particular, the separation of aeroacoustic sources is demonstrated in a wind tunnel.

Assessing the Underwater Acoustics of the World's Largest Vibration Hammer (OCTA-KONG) and Its Potential Effects on the Indo-Pacific Humpbacked Dolphin (Sousa chinensis)

PubMed Central

Wang, Zhitao; Wu, Yuping; Duan, Guoqin; Cao, Hanjiang; Liu, Jianchang; Wang, Kexiong; Wang, Ding

2014-01-01

Anthropogenic noise in aquatic environments is a worldwide concern due to its potential adverse effects on the environment and aquatic life. The Hongkong-Zhuhai-Macao Bridge is currently under construction in the Pearl River Estuary, a hot spot for the Indo-Pacific humpbacked dolphin (Sousa chinensis) in China. The OCTA-KONG, the world's largest vibration hammer, is being used during this construction project to drive or extract steel shell piles 22 m in diameter. This activity poses a substantial threat to marine mammals, and an environmental assessment is critically needed. The underwater acoustic properties of the OCTA-KONG were analyzed, and the potential impacts of the underwater acoustic energy on Sousa, including auditory masking and physiological impacts, were assessed. The fundamental frequency of the OCTA-KONG vibration ranged from 15 Hz to 16 Hz, and the noise increments were below 20 kHz, with a dominant frequency and energy below 10 kHz. The resulting sounds are most likely detectable by Sousa over distances of up to 3.5 km from the source. Although Sousa clicks do not appear to be adversely affected, Sousa whistles are susceptible to auditory masking, which may negatively impact this species' social life. Therefore, a safety zone with a radius of 500 m is proposed. Although the zero-to-peak source level (SL) of the OCTA-KONG was lower than the physiological damage level, the maximum root-mean-square SL exceeded the cetacean safety exposure level on several occasions. Moreover, the majority of the unweighted cumulative source sound exposure levels (SSELs) and the cetacean auditory weighted cumulative SSELs exceeded the acoustic threshold levels for the onset of temporary threshold shift, a type of potentially recoverable auditory damage resulting from prolonged sound exposure. These findings may aid in the identification and design of appropriate mitigation methods, such as the use of air bubble curtains, “soft start” and “power down” techniques. PMID:25338113

Assessing the underwater acoustics of the world's largest vibration hammer (OCTA-KONG) and its potential effects on the Indo-Pacific humpbacked dolphin (Sousa chinensis).

PubMed

Wang, Zhitao; Wu, Yuping; Duan, Guoqin; Cao, Hanjiang; Liu, Jianchang; Wang, Kexiong; Wang, Ding

2014-01-01

Anthropogenic noise in aquatic environments is a worldwide concern due to its potential adverse effects on the environment and aquatic life. The Hongkong-Zhuhai-Macao Bridge is currently under construction in the Pearl River Estuary, a hot spot for the Indo-Pacific humpbacked dolphin (Sousa chinensis) in China. The OCTA-KONG, the world's largest vibration hammer, is being used during this construction project to drive or extract steel shell piles 22 m in diameter. This activity poses a substantial threat to marine mammals, and an environmental assessment is critically needed. The underwater acoustic properties of the OCTA-KONG were analyzed, and the potential impacts of the underwater acoustic energy on Sousa, including auditory masking and physiological impacts, were assessed. The fundamental frequency of the OCTA-KONG vibration ranged from 15 Hz to 16 Hz, and the noise increments were below 20 kHz, with a dominant frequency and energy below 10 kHz. The resulting sounds are most likely detectable by Sousa over distances of up to 3.5 km from the source. Although Sousa clicks do not appear to be adversely affected, Sousa whistles are susceptible to auditory masking, which may negatively impact this species' social life. Therefore, a safety zone with a radius of 500 m is proposed. Although the zero-to-peak source level (SL) of the OCTA-KONG was lower than the physiological damage level, the maximum root-mean-square SL exceeded the cetacean safety exposure level on several occasions. Moreover, the majority of the unweighted cumulative source sound exposure levels (SSELs) and the cetacean auditory weighted cumulative SSELs exceeded the acoustic threshold levels for the onset of temporary threshold shift, a type of potentially recoverable auditory damage resulting from prolonged sound exposure. These findings may aid in the identification and design of appropriate mitigation methods, such as the use of air bubble curtains, "soft start" and "power down" techniques.

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

Assessment of injuries and recovery monitoring of Prince William Sound killer whales using photo-identification techniques. Restoration project 93042/94092. Exxon Valdez oil spill restoration project final report

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

Dahlheim, M.E.

1994-04-01

Photo-identification studies of individual killer whales inhabitating Prince William Sound were collected from 1989-91 and from July to September 1993 to determine the impact of the spill on whale abundance and distribution (1989-1991) and monitor recovery (1993). Concurrent photo-identification studies were also conducted in Southeast Alaska to determine if PWS killer whales were displaced to other areas between 1989 and 1991. Despite increased effort, the number of encounters with PWS killer whales appears to be decreasing. The authors assume, that the whales are dead from natural causes, a result of interactions with fisheries, from the spill, or a combination ofmore » these causes.« less

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.

TIES: EVERYTHING YOU WANTED TO KNOW BUT WERE AFRAID TO ASK

EPA Science Inventory

Identification of stressors in aquatic systems is critical to sound assessment and management of our nation's waterways. Information from stressor identification can be useful in designing effective sediment remediation methods, assessing options for sediment disposal, allowing m...

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.

Source fields reconstruction with 3D mapping by means of the virtual acoustic volume concept

NASA Astrophysics Data System (ADS)

Forget, S.; Totaro, N.; Guyader, J. L.; Schaeffer, M.

2016-10-01

This paper presents the theoretical framework of the virtual acoustic volume concept and two related inverse Patch Transfer Functions (iPTF) identification methods (called u-iPTF and m-iPTF depending on the chosen boundary conditions for the virtual volume). They are based on the application of Green's identity on an arbitrary closed virtual volume defined around the source. The reconstruction of sound source fields combines discrete acoustic measurements performed at accessible positions around the source with the modal behavior of the chosen virtual acoustic volume. The mode shapes of the virtual volume can be computed by a Finite Element solver to handle the geometrical complexity of the source. As a result, it is possible to identify all the acoustic source fields at the real surface of an irregularly shaped structure and irrespective of its acoustic environment. The m-iPTF method is introduced for the first time in this paper. Conversely to the already published u-iPTF method, the m-iPTF method needs only acoustic pressure and avoids particle velocity measurements. This paper is focused on its validation, both with numerical computations and by experiments on a baffled oil pan.

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.

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.

[Music therapy in adults with cochlear implants : Effects on music perception and subjective sound quality].

PubMed

Hutter, E; Grapp, M; Argstatter, H

2016-12-01

People with severe hearing impairments and deafness can achieve good speech comprehension using a cochlear implant (CI), although music perception often remains impaired. A novel concept of music therapy for adults with CI was developed and evaluated in this study. This study included 30 adults with a unilateral CI following postlingual deafness. The subjective sound quality of the CI was rated using the hearing implant sound quality index (HISQUI) and musical tests for pitch discrimination, melody recognition and timbre identification were applied. As a control 55 normally hearing persons also completed the musical tests. In comparison to normally hearing subjects CI users showed deficits in the perception of pitch, melody and timbre. Specific effects of therapy were observed in the subjective sound quality of the CI, in pitch discrimination into a high and low pitch range and in timbre identification, while general learning effects were found in melody recognition. Music perception shows deficits in CI users compared to normally hearing persons. After individual music therapy in the rehabilitation process, improvements in this delicate area could be achieved.

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.

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.

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.

Soundscape and Noise Exposure Monitoring in a Marine Protected Area Using Shipping Data and Time-Lapse Footage.

PubMed

Merchant, Nathan D; Pirotta, Enrico; Barton, Tim R; Thompson, Paul M

2016-01-01

We review recent work that developed new techniques for underwater noise assessment that integrate acoustic monitoring with automatic identification system (AIS) shipping data and time-lapse video, meteorological, and tidal data. Two sites were studied within the Moray Firth Special Area of Conservation (SAC) for bottlenose dolphins, where increased shipping traffic is expected from construction of offshore wind farms outside the SAC. Noise exposure varied markedly between the sites, and natural and anthropogenic contributions were characterized using multiple data sources. At one site, AIS-operating vessels accounted for total cumulative sound exposure (0.1-10 kHz), suggesting that noise modeling using the AIS would be feasible.

A System for Heart Sounds Classification

PubMed Central

Redlarski, Grzegorz; Gradolewski, Dawid; Palkowski, Aleksander

2014-01-01

The future of quick and efficient disease diagnosis lays in the development of reliable non-invasive methods. As for the cardiac diseases – one of the major causes of death around the globe – a concept of an electronic stethoscope equipped with an automatic heart tone identification system appears to be the best solution. Thanks to the advancement in technology, the quality of phonocardiography signals is no longer an issue. However, appropriate algorithms for auto-diagnosis systems of heart diseases that could be capable of distinguishing most of known pathological states have not been yet developed. The main issue is non-stationary character of phonocardiography signals as well as a wide range of distinguishable pathological heart sounds. In this paper a new heart sound classification technique, which might find use in medical diagnostic systems, is presented. It is shown that by combining Linear Predictive Coding coefficients, used for future extraction, with a classifier built upon combining Support Vector Machine and Modified Cuckoo Search algorithm, an improvement in performance of the diagnostic system, in terms of accuracy, complexity and range of distinguishable heart sounds, can be made. The developed system achieved accuracy above 93% for all considered cases including simultaneous identification of twelve different heart sound classes. The respective system is compared with four different major classification methods, proving its reliability. PMID:25393113

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

Vocalisation sound pattern identification in young broiler chickens.

PubMed

Fontana, I; Tullo, E; Scrase, A; Butterworth, A

2016-09-01

In this study, we describe the monitoring of young broiler chicken vocalisation, with sound recorded and assessed at regular intervals throughout the life of the birds from day 1 to day 38, with a focus on the first week of life. We assess whether there are recognisable, and even predictable, vocalisation patterns based on frequency and sound spectrum analysis, which can be observed in birds at different ages and stages of growth within the relatively short life of the birds in commercial broiler production cycles. The experimental trials were carried out in a farm where the broiler where reared indoor, and audio recording procedures carried out over 38 days. The recordings were made using two microphones connected to a digital recorder, and the sonic data was collected in situations without disturbance of the animals beyond that created by the routine activities of the farmer. Digital files of 1 h duration were cut into short files of 10 min duration, and these sound recordings were analysed and labelled using audio analysis software. Analysis of these short sound files showed that the key vocalisation frequency and patterns changed in relation to increasing age and the weight of the broilers. Statistical analysis showed a significant correlation (P<0.001) between the frequency of vocalisation and the age of the birds. Based on the identification of specific frequencies of the sounds emitted, in relation to age and weight, it is proposed that there is potential for audio monitoring and comparison with 'anticipated' sound patterns to be used to evaluate the status of farmed broiler chicken.

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.

Automatic adventitious respiratory sound analysis: A systematic review.

PubMed

Pramono, Renard Xaviero Adhi; Bowyer, Stuart; Rodriguez-Villegas, Esther

2017-01-01

Automatic detection or classification of adventitious sounds is useful to assist physicians in diagnosing or monitoring diseases such as asthma, Chronic Obstructive Pulmonary Disease (COPD), and pneumonia. While computerised respiratory sound analysis, specifically for the detection or classification of adventitious sounds, has recently been the focus of an increasing number of studies, a standardised approach and comparison has not been well established. To provide a review of existing algorithms for the detection or classification of adventitious respiratory sounds. This systematic review provides a complete summary of methods used in the literature to give a baseline for future works. A systematic review of English articles published between 1938 and 2016, searched using the Scopus (1938-2016) and IEEExplore (1984-2016) databases. Additional articles were further obtained by references listed in the articles found. Search terms included adventitious sound detection, adventitious sound classification, abnormal respiratory sound detection, abnormal respiratory sound classification, wheeze detection, wheeze classification, crackle detection, crackle classification, rhonchi detection, rhonchi classification, stridor detection, stridor classification, pleural rub detection, pleural rub classification, squawk detection, and squawk classification. Only articles were included that focused on adventitious sound detection or classification, based on respiratory sounds, with performance reported and sufficient information provided to be approximately repeated. Investigators extracted data about the adventitious sound type analysed, approach and level of analysis, instrumentation or data source, location of sensor, amount of data obtained, data management, features, methods, and performance achieved. A total of 77 reports from the literature were included in this review. 55 (71.43%) of the studies focused on wheeze, 40 (51.95%) on crackle, 9 (11.69%) on stridor, 9 (11.69%) on rhonchi, and 18 (23.38%) on other sounds such as pleural rub, squawk, as well as the pathology. Instrumentation used to collect data included microphones, stethoscopes, and accelerometers. Several references obtained data from online repositories or book audio CD companions. Detection or classification methods used varied from empirically determined thresholds to more complex machine learning techniques. Performance reported in the surveyed works were converted to accuracy measures for data synthesis. Direct comparison of the performance of surveyed works cannot be performed as the input data used by each was different. A standard validation method has not been established, resulting in different works using different methods and performance measure definitions. A review of the literature was performed to summarise different analysis approaches, features, and methods used for the analysis. The performance of recent studies showed a high agreement with conventional non-automatic identification. This suggests that automated adventitious sound detection or classification is a promising solution to overcome the limitations of conventional auscultation and to assist in the monitoring of relevant diseases.

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

Assessment of injuries to killer whales in Prince William Sound. Marine mammal study number 2. Exxon Valdez oil spill state/federal natural resource damage assessment final report

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

Dahlheim, M.E.; Matkin, C.O.

1993-12-01

Photo-identification studies of individual killer whales inhabiting Prince William Sound were collected from 1989-91 to determine the impact of the spill on whale abundance and distribution. Concurrent photo-identification studies were also conducted in Southeast Alaska to determine if PWS killer whales were displaced to other areas. Despite increased effort, the number of encounters with PWS killer whales appears to be decreasing. The authors assume, that the whales are dead from natural causes, a result of interactions with fisheries, from the spill, or a combination of these causes.

Normalized inverse characterization of sound absorbing rigid porous media.

PubMed

Zieliński, Tomasz G

2015-06-01

This paper presents a methodology for the inverse characterization of sound absorbing rigid porous media, based on standard measurements of the surface acoustic impedance of a porous sample. The model parameters need to be normalized to have a robust identification procedure which fits the model-predicted impedance curves with the measured ones. Such a normalization provides a substitute set of dimensionless (normalized) parameters unambiguously related to the original model parameters. Moreover, two scaling frequencies are introduced, however, they are not additional parameters and for different, yet reasonable, assumptions of their values, the identification procedure should eventually lead to the same solution. The proposed identification technique uses measured and computed impedance curves for a porous sample not only in the standard configuration, that is, set to the rigid termination piston in an impedance tube, but also with air gaps of known thicknesses between the sample and the piston. Therefore, all necessary analytical formulas for sound propagation in double-layered media are provided. The methodology is illustrated by one numerical test and by two examples based on the experimental measurements of the acoustic impedance and absorption of porous ceramic samples of different thicknesses and a sample of polyurethane foam.

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.

The Crossmodal Facilitation of Visual Object Representations by Sound: Evidence from the Backward Masking Paradigm

ERIC Educational Resources Information Center

Chen, Yi-Chuan; Spence, Charles

2011-01-01

We report a series of experiments designed to demonstrate that the presentation of a sound can facilitate the identification of a concomitantly presented visual target letter in the backward masking paradigm. Two visual letters, serving as the target and its mask, were presented successively at various interstimulus intervals (ISIs). The results…

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.

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

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.

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.

Diversity of fish sound types in the Pearl River Estuary, China

PubMed Central

Wang, Zhi-Tao; Nowacek, Douglas P.; Akamatsu, Tomonari; Liu, Jian-Chang; Duan, Guo-Qin; Cao, Han-Jiang

2017-01-01

Background Repetitive species-specific sound enables the identification of the presence and behavior of soniferous species by acoustic means. Passive acoustic monitoring has been widely applied to monitor the spatial and temporal occurrence and behavior of calling species. Methods Underwater biological sounds in the Pearl River Estuary, China, were collected using passive acoustic monitoring, with special attention paid to fish sounds. A total of 1,408 suspected fish calls comprising 18,942 pulses were qualitatively analyzed using a customized acoustic analysis routine. Results We identified a diversity of 66 types of fish sounds. In addition to single pulse, the sounds tended to have a pulse train structure. The pulses were characterized by an approximate 8 ms duration, with a peak frequency from 500 to 2,600 Hz and a majority of the energy below 4,000 Hz. The median inter-pulsepeak interval (IPPI) of most call types was 9 or 10 ms. Most call types with median IPPIs of 9 ms and 10 ms were observed at times that were exclusive from each other, suggesting that they might be produced by different species. According to the literature, the two section signal types of 1 + 1 and 1 + N10 might belong to big-snout croaker (Johnius macrorhynus), and 1 + N19 might be produced by Belanger’s croaker (J. belangerii). Discussion Categorization of the baseline ambient biological sound is an important first step in mapping the spatial and temporal patterns of soniferous fishes. The next step is the identification of the species producing each sound. The distribution pattern of soniferous fishes will be helpful for the protection and management of local fishery resources and in marine environmental impact assessment. Since the local vulnerable Indo-Pacific humpback dolphin (Sousa chinensis) mainly preys on soniferous fishes, the fine-scale distribution pattern of soniferous fishes can aid in the conservation of this species. Additionally, prey and predator relationships can be observed when a database of species-identified sounds is completed. PMID:29085746

Diversity of fish sound types in the Pearl River Estuary, China.

PubMed

Wang, Zhi-Tao; Nowacek, Douglas P; Akamatsu, Tomonari; Wang, Ke-Xiong; Liu, Jian-Chang; Duan, Guo-Qin; Cao, Han-Jiang; Wang, Ding

2017-01-01

Repetitive species-specific sound enables the identification of the presence and behavior of soniferous species by acoustic means. Passive acoustic monitoring has been widely applied to monitor the spatial and temporal occurrence and behavior of calling species. Underwater biological sounds in the Pearl River Estuary, China, were collected using passive acoustic monitoring, with special attention paid to fish sounds. A total of 1,408 suspected fish calls comprising 18,942 pulses were qualitatively analyzed using a customized acoustic analysis routine. We identified a diversity of 66 types of fish sounds. In addition to single pulse, the sounds tended to have a pulse train structure. The pulses were characterized by an approximate 8 ms duration, with a peak frequency from 500 to 2,600 Hz and a majority of the energy below 4,000 Hz. The median inter-pulsepeak interval (IPPI) of most call types was 9 or 10 ms. Most call types with median IPPIs of 9 ms and 10 ms were observed at times that were exclusive from each other, suggesting that they might be produced by different species. According to the literature, the two section signal types of 1 + 1 and 1 + N 10 might belong to big-snout croaker ( Johnius macrorhynus ), and 1 + N 19 might be produced by Belanger's croaker ( J. belangerii ). Categorization of the baseline ambient biological sound is an important first step in mapping the spatial and temporal patterns of soniferous fishes. The next step is the identification of the species producing each sound. The distribution pattern of soniferous fishes will be helpful for the protection and management of local fishery resources and in marine environmental impact assessment. Since the local vulnerable Indo-Pacific humpback dolphin ( Sousa chinensis ) mainly preys on soniferous fishes, the fine-scale distribution pattern of soniferous fishes can aid in the conservation of this species. Additionally, prey and predator relationships can be observed when a database of species-identified sounds is completed.

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

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.

Differential Neural Contributions to Native- and Foreign-Language Talker Identification

ERIC Educational Resources Information Center

Perrachione, Tyler K.; Pierrehumbert, Janet B.; Wong, Patrick C. M.

2009-01-01

Humans are remarkably adept at identifying individuals by the sound of their voice, a behavior supported by the nervous system's ability to integrate information from voice and speech perception. Talker-identification abilities are significantly impaired when listeners are unfamiliar with the language being spoken. Recent behavioral studies…

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

Training guide for bird identification in Pacific Northwest Douglas-fir forests.

Treesearch

Andrew B. Carey; Valen E. Castellano; Christopher Chappell; Robert Kuntz; Richard W. Lundquist; Bruce G. Marcot; S. Kim Nelson; Paul Sulllivan; [Technical Compilers].

1990-01-01

Bird calls and songs vary regionally, and some birds emit a variety of sounds. Existing guides are inadequate for training observers to do detailed surveys of bird communities, because more than 90 percent of birds detected are identified by the sounds they emit. This guide summarizes existing guides and adds the observations of the compilers and other technicians who...

Launch summary for 1978 - 1982. [sounding rockets, space probes, and satellites

NASA Technical Reports Server (NTRS)

Hills, H. K.

1984-01-01

Data pertinent to the launching of space probes, soundings rockets, and satellites presented in tables include launch date, time, and site; agency rocket identification; sponsoring country or countries; instruments carried for experiments; the peak altitude achieved by the rockets; and the apoapsis and periapsis for satellites. The experimenter or institution involved in the launching is also cited.

Experimental and theoretical identification of a four- acoustic-inputs/two-vibration-outputs hearing system

NASA Astrophysics Data System (ADS)

Balaji, P. A.

1999-07-01

A cricket's ear is a directional acoustic sensor. It has a remarkable level of sensitivity to the direction of sound propagation in a narrow frequency bandwidth of 4-5 KHz. Because of its complexity, the directional sensitivity has long intrigued researchers. The cricket's ear is a four-acoustic-inputs/two-vibration-outputs system. In this dissertation, this system is examined in depth, both experimentally and theoretically, with a primary goal to understand the mechanics involved in directional hearing. Experimental identification of the system is done by using random signal processing techniques. Theoretical identification of the system is accomplished by analyzing sound transmission through complex trachea of the ear. Finally, a description of how the cricket achieves directional hearing sensitivity is proposed. The fundamental principle involved in directional heating of the cricket has been utilized to design a device to obtain a directional signal from non- directional inputs.

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.

A comparative study of electronic stethoscopes for cardiac auscultation.

PubMed

Pinto, C; Pereira, D; Ferreira-Coimbra, J; Portugues, J; Gama, V; Coimbra, M

2017-07-01

There are several electronic stethoscopes available on the market today, with a very high potential for healthcare namely telemedicine, assisted decision and education. However, there are no recent comparatives studies published about the recording quality of auscultation sounds. In this study we aim to: a) define a ranking, according to experts opinion of 6 of the most relevant electronic stethoscopes on the market today; b) verify if there are any relations between a stethoscope's performance and the type of pathology present; c) analyze if some pathologies are more easily identified than others when using electronic auscultation. Our methodology consisted in creating two study groups: the first group included 18 cardiologists and cardiology house officers, acting as the gold standard of this work. The second included 30 medical students. Using a database of heart sounds recorded in real hospital environments, we applied questionnaires to observers from each group. The first group listened to 60 cardiac auscultations recorded by the 6 stethoscopes, and each one was asked to identify the pathological sound present: aortic stenosis, mitral regurgitation or normal. The second group was asked to choose, between two auscultation recordings, using as criteria the best sound quality for the identification of pathological sounds. Results include a total of 1080 evaluations, in which 72% of cases were correctly diagnosed. A detailed breakdown of these results is presented in this paper. As conclusions, results showed that the impact of the differences between stethoscopes is very small, given that we did not find statistically significant differences between all pairs of stethoscopes. Normal sounds showed to be easier to identify than pathological sounds, but we did not find differences between stethoscopes in this identification.

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.

Comprehensive measures of sound exposures in cinemas using smart phones.

PubMed

Huth, Markus E; Popelka, Gerald R; Blevins, Nikolas H

2014-01-01

Sensorineural hearing loss from sound overexposure has a considerable prevalence. Identification of sound hazards is crucial, as prevention, due to a lack of definitive therapies, is the sole alternative to hearing aids. One subjectively loud, yet little studied, potential sound hazard is movie theaters. This study uses smart phones to evaluate their applicability as a widely available, validated sound pressure level (SPL) meter. Therefore, this study measures sound levels in movie theaters to determine whether sound levels exceed safe occupational noise exposure limits and whether sound levels in movie theaters differ as a function of movie, movie theater, presentation time, and seat location within the theater. Six smart phones with an SPL meter software application were calibrated with a precision SPL meter and validated as an SPL meter. Additionally, three different smart phone generations were measured in comparison to an integrating SPL meter. Two different movies, an action movie and a children's movie, were measured six times each in 10 different venues (n = 117). To maximize representativeness, movies were selected focusing on large release productions with probable high attendance. Movie theaters were selected in the San Francisco, CA, area based on whether they screened both chosen movies and to represent the largest variety of theater proprietors. Measurements were analyzed in regard to differences between theaters, location within the theater, movie, as well as presentation time and day as indirect indicator of film attendance. The smart phone measurements demonstrated high accuracy and reliability. Overall, sound levels in movie theaters do not exceed safe exposure limits by occupational standards. Sound levels vary significantly across theaters and demonstrated statistically significant higher sound levels and exposures in the action movie compared to the children's movie. Sound levels decrease with distance from the screen. However, no influence on time of day or day of the week as indirect indicator of film attendance could be found. Calibrated smart phones with an appropriate software application as used in this study can be utilized as a validated SPL meter. Because of the wide availability, smart phones in combination with the software application can provide high quantity recreational sound exposure measurements, which can facilitate the identification of potential noise hazards. Sound levels in movie theaters decrease with distance to the screen, but do not exceed safe occupational noise exposure limits. Additionally, there are significant differences in sound levels across movie theaters and movies, but not in presentation time.

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 influence of linguistic experience on pitch perception in speech and nonspeech sounds

NASA Astrophysics Data System (ADS)

Bent, Tessa; Bradlow, Ann R.; Wright, Beverly A.

2003-04-01

How does native language experience with a tone or nontone language influence pitch perception? To address this question 12 English and 13 Mandarin listeners participated in an experiment involving three tasks: (1) Mandarin tone identification-a clearly linguistic task where a strong effect of language background was expected, (2) pure-tone and pulse-train frequency discrimination-a clearly nonlinguistic auditory discrimination task where no effect of language background was expected, and (3) pitch glide identification-a nonlinguistic auditory categorization task where some effect of language background was expected. As anticipated, Mandarin listeners identified Mandarin tones significantly more accurately than English listeners (Task 1) and the two groups' pure-tone and pulse-train frequency discrimination thresholds did not differ (Task 2). For pitch glide identification (Task 3), Mandarin listeners made more identification errors: in comparison with English listeners, Mandarin listeners more frequently misidentified falling pitch glides as level, and more often misidentified level pitch ``glides'' with relatively high frequencies as rising and those with relatively low frequencies as falling. Thus, it appears that the effect of long-term linguistic experience can extend beyond lexical tone category identification in syllables to pitch class identification in certain nonspeech sounds. [Work supported by Sigma Xi and NIH.

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.

Determining the speed of sound in the air by sound wave interference

NASA Astrophysics Data System (ADS)

Silva, Abel A.

2017-07-01

Mechanical waves propagate through material media. Sound is an example of a mechanical wave. In fluids like air, sound waves propagate through successive longitudinal perturbations of compression and decompression. Audible sound frequencies for human ears range from 20 to 20 000 Hz. In this study, the speed of sound v in the air is determined using the identification of maxima of interference from two synchronous waves at frequency f. The values of v were correct to 0 °C. The experimental average value of {\\bar{ν }}\\exp =336 +/- 4 {{m}} {{{s}}}-1 was found. It is 1.5% larger than the reference value. The standard deviation of 4 m s-1 (1.2% of {\\bar{ν }}\\exp ) is an improved value by the use of the concept of the central limit theorem. The proposed procedure to determine the speed of sound in the air aims to be an academic activity for physics classes of scientific and technological courses in college.

Comparison of the identification and ease of use of two alarm sound sets by critical and acute care nurses with little or no music training: a laboratory study.

PubMed

Atyeo, J; Sanderson, P M

2015-07-01

The melodic alarm sound set for medical electrical equipment that was recommended in the International Electrotechnical Commission's IEC 60601-1-8 standard has proven difficult for clinicians to learn and remember, especially clinicians with little prior formal music training. An alarm sound set proposed by Patterson and Edworthy in 1986 might improve performance for such participants. In this study, 31 critical and acute care nurses with less than one year of formal music training identified alarm sounds while they calculated drug dosages. Sixteen nurses used the IEC and 15 used the Patterson-Edworthy alarm sound set. The mean (SD) percentage of alarms correctly identified by nurses was 51.3 (25.6)% for the IEC alarm set and 72.1 (18.8)% for the Patterson-Edworthy alarms (p = 0.016). Nurses using the Patterson-Edworthy alarm sound set reported that it was easier to distinguish between alarm sounds than did nurses using the IEC alarm sound set (p = 0.015). Principles used to construct the Patterson-Edworthy alarm sounds should be adopted for future alarm sound sets. © 2015 The Association of Anaesthetists of Great Britain and Ireland.

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.

a New Approach to Physiologic Triggering in Medical Imaging Using Multiple Heart Sounds Alone.

NASA Astrophysics Data System (ADS)

Groch, Mark Walter

A new method for physiological synchronization of medical image acquisition using both the first and second heart sound has been developed. Heart sounds gating (HSG) circuitry has been developed which identifies, individually, both the first (S1) and second (S2) heart sounds from their timing relationship alone, and provides two synchronization points during the cardiac cycle. Identification of first and second heart sounds from their timing relationship alone and application to medical imaging has, heretofore, not been performed in radiology or nuclear medicine. The heart sounds are obtained as conditioned analog signals from a piezoelectric transducer microphone placed on the patient's chest. The timing relationships between the S1 to S2 pulses and the S2 to S1 pulses are determined using a logic scheme capable of distinguishing the S1 and S2 pulses from the heart sounds themselves, using their timing relationships, and the assumption that initially the S1-S2 interval will be shorter than the S2-S1 interval. Digital logic circuitry is utilized to continually track the timing intervals and extend the S1/S2 identification to heart rates up to 200 beats per minute (where the S1-S2 interval is not shorter than the S2-S1 interval). Clinically, first heart sound gating may be performed to assess the systolic ejection portion of the cardiac cycle, with S2 gating utilized for reproduction of the diastolic filling portion of the cycle. One application of HSG used for physiologic synchronization is in multigated blood pool (MGBP) imaging in nuclear medicine. Heart sounds gating has been applied to twenty patients who underwent analysis of ventricular function in Nuclear Medicine, and compared to conventional ECG gated MGBP. Left ventricular ejection fractions calculated from MGBP studies using a S1 and a S2 heart sound trigger correlated well with conventional ECG gated acquisitions in patients adequately gated by HSG and ECG. Heart sounds gating provided superior definition of the diastolic filling phase of the cardiac cycle by qualitative assessment of the left ventricular volume time -activity curves. Heart sounds physiological synchronization has potential to be used in other imaging modalities, such as magnetic resonance imaging, where the ECG is distorted due to the electromagnetic environment within the imager.

A data base describing low-gravity fluids and materials processing experiments

NASA Technical Reports Server (NTRS)

Winter, C. A.; Jones, J. C.

1992-01-01

A data base documenting information on approximately 600 fluids and materials processing experiments performed in a low-gravity environment has been prepared at NASA Marshall Space Flight Center (MSFC). The compilation was designed to document all such experimental efforts performed: (1) on U.S. manned space vehicles; (2) on payloads deployed from U.S. manned space vehicles; and (3) on all domestic and international sounding rocket programs (excluding those of the U.S.S.R. and China). Identification of major (reported) sources of significant anomalies during 100 of the experiments is reported and discussed. Further, a preliminary summary of the number of these 100 investigations which experienced an anomaly affecting a certain percentage of the experimental results/objectives is presented.

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.

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.

Initial Integration of Noise Prediction Tools for Acoustic Scattering Effects

NASA Technical Reports Server (NTRS)

Nark, Douglas M.; Burley, Casey L.; Tinetti, Ana; Rawls, John W.

2008-01-01

This effort provides an initial glimpse at NASA capabilities available in predicting the scattering of fan noise from a non-conventional aircraft configuration. The Aircraft NOise Prediction Program, Fast Scattering Code, and the Rotorcraft Noise Model were coupled to provide increased fidelity models of scattering effects on engine fan noise sources. The integration of these codes led to the identification of several keys issues entailed in applying such multi-fidelity approaches. In particular, for prediction at noise certification points, the inclusion of distributed sources leads to complications with the source semi-sphere approach. Computational resource requirements limit the use of the higher fidelity scattering code to predict radiated sound pressure levels for full scale configurations at relevant frequencies. And, the ability to more accurately represent complex shielding surfaces in current lower fidelity models is necessary for general application to scattering predictions. This initial step in determining the potential benefits/costs of these new methods over the existing capabilities illustrates a number of the issues that must be addressed in the development of next generation aircraft system noise prediction tools.

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

Listeners' identification and discrimination of digitally manipulated sounds as prolongations.

PubMed

Kawai, Norimune; Healey, E Charles; Carrell, Thomas D

2007-08-01

The present study had two main purposes. One was to examine if listeners perceive gradually increasing durations of a voiceless fricative categorically ("fluent" versus "stuttered") or continuously (gradient perception from fluent to stuttered). The second purpose was to investigate whether there are gender differences in how listeners perceive various duration of sounds as "prolongations." Forty-four listeners were instructed to rate the duration of the // in the word "shape" produced by a normally fluent speaker. The target word was embedded in the middle of an experimental phrase and the initial // sound was digitally manipulated to create a range of fluent to stuttered sounds. This was accomplished by creating 20 ms stepwise increments for sounds ranging from 120 to 500 ms in duration. Listeners were instructed to give a rating of 1 for a fluent word and a rating of 100 for a stuttered word. The results showed listeners perceived the range of sounds continuously. Also, there was a significant gender difference in that males rated fluent sounds higher than females but female listeners rated stuttered sounds higher than males. The implications of these results are discussed.

12 CFR Appendix J to Part 41 - Interagency Guidelines on Identity Theft Detection, Prevention, and Mitigation

Code of Federal Regulations, 2010 CFR

2010-01-01

... risks to customers or to the safety and soundness of the financial institution or creditor from identity... unusual use of, or other suspicious activity related to, a covered account; and (5) Notice from customers... policies and procedures regarding identification and verification set forth in the Customer Identification...

16 CFR Appendix A to Part 681 - Interagency Guidelines on Identity Theft Detection, Prevention, and Mitigation

Code of Federal Regulations, 2010 CFR

2010-01-01

... risks to customers or to the safety and soundness of the financial institution or creditor from identity... unusual use of, or other suspicious activity related to, a covered account; and (5) Notice from customers... policies and procedures regarding identification and verification set forth in the Customer Identification...

Information Feedback: Contributions to Learning and Performance in Perceptual Identification Training.

ERIC Educational Resources Information Center

Abrams, Alvin J.; Cook, Richard L.

In training people to perform auditory identification tasks (e.g., training students to identify sound characteristics in a sonar classification task), it is important to know whether or not training procedures are merely sustaining performance during training or whether they enhance learning of the task. Often an incorrect assumption is made that…

33 CFR 164.43 - Automatic Identification System Shipborne Equipment-Prince William Sound.

Code of Federal Regulations, 2010 CFR

2010-07-01

... (AISSE) system consisting of a: (1) Twelve-channel all-in-view Differential Global Positioning System (d... to indicate to shipboard personnel that the U.S. Coast Guard dGPS system cannot provide the required... 33 Navigation and Navigable Waters 2 2010-07-01 2010-07-01 false Automatic Identification System...

Effect of Dialect on Identification and Severity of Speech Impairment in Indigenous Australian Children

ERIC Educational Resources Information Center

Toohill, Bethany J.; Mcleod, Sharynne; Mccormack, Jane

2012-01-01

This study investigated the effect of dialectal difference on identification and rating of severity of speech impairment in children from Indigenous Australian backgrounds. The speech of 15 Indigenous Australian children identified by their parents/caregivers and teachers as having "difficulty talking and making speech sounds" was…

Call recognition and individual identification of fish vocalizations based on automatic speech recognition: An example with the Lusitanian toadfish.

PubMed

Vieira, Manuel; Fonseca, Paulo J; Amorim, M Clara P; Teixeira, Carlos J C

2015-12-01

The study of acoustic communication in animals often requires not only the recognition of species specific acoustic signals but also the identification of individual subjects, all in a complex acoustic background. Moreover, when very long recordings are to be analyzed, automatic recognition and identification processes are invaluable tools to extract the relevant biological information. A pattern recognition methodology based on hidden Markov models is presented inspired by successful results obtained in the most widely known and complex acoustical communication signal: human speech. This methodology was applied here for the first time to the detection and recognition of fish acoustic signals, specifically in a stream of round-the-clock recordings of Lusitanian toadfish (Halobatrachus didactylus) in their natural estuarine habitat. The results show that this methodology is able not only to detect the mating sounds (boatwhistles) but also to identify individual male toadfish, reaching an identification rate of ca. 95%. Moreover this method also proved to be a powerful tool to assess signal durations in large data sets. However, the system failed in recognizing other sound types.

Identification of Noise Sources During Rocket Engine Test Firings and a Rocket Launch Using a Microphone Phased-Array

NASA Technical Reports Server (NTRS)

Panda, Jayanta; Mosher, Robert N.; Porter, Barry J.

2013-01-01

A 70 microphone, 10-foot by 10-foot, microphone phased array was built for use in the harsh environment of rocket launches. The array was setup at NASA Wallops launch pad 0A during a static test firing of Orbital Sciences' Antares engines, and again during the first launch of the Antares vehicle. It was placed 400 feet away from the pad, and was hoisted on a scissor lift 40 feet above ground. The data sets provided unprecedented insight into rocket noise sources. The duct exit was found to be the primary source during the static test firing; the large amount of water injected beneath the nozzle exit and inside the plume duct quenched all other sources. The maps of the noise sources during launch were found to be time-dependent. As the engines came to full power and became louder, the primary source switched from the duct inlet to the duct exit. Further elevation of the vehicle caused spilling of the hot plume, resulting in a distributed noise map covering most of the pad. As the entire plume emerged from the duct, and the ondeck water system came to full power, the plume itself became the loudest noise source. These maps of the noise sources provide vital insight for optimization of sound suppression systems for future Antares launches.

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 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…

Development of a Korotkov sound processor for automatic identification of auscultatory events. I - Specification of preprocessing bandpass filters

NASA Technical Reports Server (NTRS)

Golden, D. P., Jr.; Wolthuis, R. A.; Hoffler, G. W.; Gowen, R. J.

1974-01-01

Frequency bands that best discriminate the Korotkov sounds at systole and at diastole from the sounds immediately preceding these events are defined. Korotkov sound data were recorded from five normotensive subjects during orthostatic stress (lower body negative pressure) and bicycle ergometry. A spectral analysis of the seven Korotkov sounds centered about the systolic and diastolic auscultatory events revealed that a maximum increase in amplitude at the systolic transition occurred in the 18-26-Hz band, while a maximum decrease in amplitude at the diastolic transition occurred in the 40-60-Hz band. These findings were remarkably consistent across subjects and test conditions. These passbands are included in the design specifications for an automatic blood pressure measuring system used in conjuction with medical experiments during NASA's Skylab program.

The Past, Present and Future of the Meteorological Phenomena Identification Near the Ground (mPING) Project

NASA Astrophysics Data System (ADS)

Elmore, K. L.

2016-12-01

The Metorological Phenomemna Identification NeartheGround (mPING) project is an example of a crowd-sourced, citizen science effort to gather data of sufficeint quality and quantity needed by new post processing methods that use machine learning. Transportation and infrastructure are particularly sensitive to precipitation type in winter weather. We extract attributes from operational numerical forecast models and use them in a random forest to generate forecast winter precipitation types. We find that random forests applied to forecast soundings are effective at generating skillful forecasts of surface ptype with consideralbly more skill than the current algorithms, especuially for ice pellets and freezing rain. We also find that three very different forecast models yuield similar overall results, showing that random forests are able to extract essentially equivalent information from different forecast models. We also show that the random forest for each model, and each profile type is unique to the particular forecast model and that the random forests developed using a particular model suffer significant degradation when given attributes derived from a different model. This implies that no single algorithm can perform well across all forecast models. Clearly, random forests extract information unavailable to "physically based" methods because the physical information in the models does not appear as we expect. One intersting result is that results from the classic "warm nose" sounding profile are, by far, the most sensitive to the particular forecast model, but this profile is also the one for which random forests are most skillful. Finally, a method for calibrarting probabilties for each different ptype using multinomial logistic regression is shown.

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.

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.

Perception of touch quality in piano tones.

PubMed

Goebl, Werner; Bresin, Roberto; Fujinaga, Ichiro

2014-11-01

Both timbre and dynamics of isolated piano tones are determined exclusively by the speed with which the hammer hits the strings. This physical view has been challenged by pianists who emphasize the importance of the way the keyboard is touched. This article presents empirical evidence from two perception experiments showing that touch-dependent sound components make sounds with identical hammer velocities but produced with different touch forms clearly distinguishable. The first experiment focused on finger-key sounds: musicians could identify pressed and struck touches. When the finger-key sounds were removed from the sounds, the effect vanished, suggesting that these sounds were the primary identification cue. The second experiment looked at key-keyframe sounds that occur when the key reaches key-bottom. Key-bottom impact was identified from key motion measured by a computer-controlled piano. Musicians were able to discriminate between piano tones that contain a key-bottom sound from those that do not. However, this effect might be attributable to sounds associated with the mechanical components of the piano action. In addition to the demonstrated acoustical effects of different touch forms, visual and tactile modalities may play important roles during piano performance that influence the production and perception of musical expression on the piano.

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.

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.

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.

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.

Sound vibration signal processing for detection and identification detonation (knock) to optimize performance Otto engine

NASA Astrophysics Data System (ADS)

Sujono, A.; Santoso, B.; Juwana, W. E.

2016-03-01

Problems of detonation (knock) on Otto engine (petrol engine) is completely unresolved problem until now, especially if want to improve the performance. This research did sound vibration signal processing engine with a microphone sensor, for the detection and identification of detonation. A microphone that can be mounted is not attached to the cylinder block, that's high temperature, so that its performance will be more stable, durable and inexpensive. However, the method of analysis is not very easy, because a lot of noise (interference). Therefore the use of new methods of pattern recognition, through filtration, and the regression function normalized envelope. The result is quite good, can achieve a success rate of about 95%.

Further Progress in Noise Source Identification in High Speed Jets via Causality Principle

NASA Technical Reports Server (NTRS)

Panda, J.; Seasholtz, R. G.; Elam, K. A.

2004-01-01

To locate noise sources in high-speed jets, the sound pressure fluctuations p/, measured at far field locations, were correlated with each of density p, axial velocity u, radial velocity v, puu and pvv fluctuations measured from various points in fully expanded, unheated plumes of Mach number 0.95, 1.4 and 1.8. The velocity and density fluctuations were measured simultaneously using a recently developed, non-intrusive, point measurement technique based on molecular Rayleigh scattering (Seasholtz, Panda, and Elam, AIAA Paper 2002-0827). The technique uses a continuous wave, narrow line-width laser, Fabry-Perot interferometer and photon counting electronics. The far field sound pressure fluctuations at 30 to the jet axis provided the highest correlation coefficients with all flow variables. The correlation coefficients decreased sharply with increased microphone polar angle, and beyond about 60 all correlation mostly fell below the experimental noise floor. Among all correlations < puu; p/> showed the highest values. Interestingly, , in all respects, were very similar to . The and correlations with 90deg microphone fell below the noise floor. By moving the laser probe at various locations in the jet it was found that the strongest noise source lies downstream of the end of the potential core and extends many diameters beyond. Correlation measurement from the lip shear layer showed a Mach number dependency. While significant correlations were measured in Mach 1.8 jet, values were mostly below the noise floor for subsonic Mach 0.95 jet. Various additional analyses showed that fluctuations from large organized structures mostly contributed to the measured correlation, while that from small scale structures fell below the noise floor.

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

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.

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 Influence of Environmental Sound Training on the Perception of Spectrally Degraded Speech and Environmental Sounds

PubMed Central

Sheft, Stanley; Gygi, Brian; Ho, Kim Thien N.

2012-01-01

Perceptual training with spectrally degraded environmental sounds results in improved environmental sound identification, with benefits shown to extend to untrained speech perception as well. The present study extended those findings to examine longer-term training effects as well as effects of mere repeated exposure to sounds over time. Participants received two pretests (1 week apart) prior to a week-long environmental sound training regimen, which was followed by two posttest sessions, separated by another week without training. Spectrally degraded stimuli, processed with a four-channel vocoder, consisted of a 160-item environmental sound test, word and sentence tests, and a battery of basic auditory abilities and cognitive tests. Results indicated significant improvements in all speech and environmental sound scores between the initial pretest and the last posttest with performance increments following both exposure and training. For environmental sounds (the stimulus class that was trained), the magnitude of positive change that accompanied training was much greater than that due to exposure alone, with improvement for untrained sounds roughly comparable to the speech benefit from exposure. Additional tests of auditory and cognitive abilities showed that speech and environmental sound performance were differentially correlated with tests of spectral and temporal-fine-structure processing, whereas working memory and executive function were correlated with speech, but not environmental sound perception. These findings indicate generalizability of environmental sound training and provide a basis for implementing environmental sound training programs for cochlear implant (CI) patients. PMID:22891070

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.

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

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

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.

Visual Speech Fills in Both Discrimination and Identification of Non-Intact Auditory Speech in Children

ERIC Educational Resources Information Center

Jerger, Susan; Damian, Markus F.; McAlpine, Rachel P.; Abdi, Herve

2018-01-01

To communicate, children must discriminate and identify speech sounds. Because visual speech plays an important role in this process, we explored how visual speech influences phoneme discrimination and identification by children. Critical items had intact visual speech (e.g. baez) coupled to non-intact (excised onsets) auditory speech (signified…

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.

Acoustic constituents of prosodic typology

NASA Astrophysics Data System (ADS)

Komatsu, Masahiko

Different languages sound different, and considerable part of it derives from the typological difference of prosody. Although such difference is often referred to as lexical accent types (stress accent, pitch accent, and tone; e.g. English, Japanese, and Chinese respectively) and rhythm types (stress-, syllable-, and mora-timed rhythms; e.g. English, Spanish, and Japanese respectively), it is unclear whether these types are determined in terms of acoustic properties, The thesis intends to provide a potential basis for the description of prosody in terms of acoustics. It argues for the hypothesis that the source component of the source-filter model (acoustic features) approximately corresponds to prosody (linguistic features) through several experimental-phonetic studies. The study consists of four parts. (1) Preliminary experiment: Perceptual language identification tests were performed using English and Japanese speech samples whose frequency spectral information (i.e. non-source component) is heavily reduced. The results indicated that humans can discriminate languages with such signals. (2) Discussion on the linguistic information that the source component contains: This part constitutes the foundation of the argument of the thesis. Perception tests of consonants with the source signal indicated that the source component carries the information on broad categories of phonemes that contributes to the creation of rhythm. (3) Acoustic analysis: The speech samples of Chinese, English, Japanese, and Spanish, differing in prosodic types, were analyzed. These languages showed difference in acoustic characteristics of the source component. (4) Perceptual experiment: A language identification test for the above four languages was performed using the source signal with its acoustic features parameterized. It revealed that humans can discriminate prosodic types solely with the source features and that the discrimination is easier as acoustic information increases. The series of studies showed the correspondence of the source component to prosodic features. In linguistics, prosodic types have not been discussed purely in terms of acoustics; they are usually related to the function of prosody or phonological units such as phonemes. The present thesis focuses on acoustics and makes a contribution to establishing the crosslinguistic description system of prosody.

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.

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

NPSNET: Aural cues for virtual world immersion

NASA Astrophysics Data System (ADS)

Dahl, Leif A.

1992-09-01

NPSNET is a low-cost visual and aural simulation system designed and implemented at the Naval Postgraduate School. NPSNET is an example of a virtual world simulation environment that incorporates real-time aural cues through software-hardware interaction. In the current implementation of NPSNET, a graphics workstation functions in the sound server role which involves sending and receiving networked sound message packets across a Local Area Network, composed of multiple graphics workstations. The network messages contain sound file identification information that is transmitted from the sound server across an RS-422 protocol communication line to a serial to Musical Instrument Digital Interface (MIDI) converter. The MIDI converter, in turn relays the sound byte to a sampler, an electronic recording and playback device. The sampler correlates the hexadecimal input to a specific note or stored sound and sends it as an audio signal to speakers via an amplifier. The realism of a simulation is improved by involving multiple participant senses and removing external distractions. This thesis describes the incorporation of sound as aural cues, and the enhancement they provide in the virtual simulation environment of NPSNET.

Wheeze sound analysis using computer-based techniques: a systematic review.

PubMed

Ghulam Nabi, Fizza; Sundaraj, Kenneth; Chee Kiang, Lam; Palaniappan, Rajkumar; Sundaraj, Sebastian

2017-10-31

Wheezes are high pitched continuous respiratory acoustic sounds which are produced as a result of airway obstruction. Computer-based analyses of wheeze signals have been extensively used for parametric analysis, spectral analysis, identification of airway obstruction, feature extraction and diseases or pathology classification. While this area is currently an active field of research, the available literature has not yet been reviewed. This systematic review identified articles describing wheeze analyses using computer-based techniques on the SCOPUS, IEEE Xplore, ACM, PubMed and Springer and Elsevier electronic databases. After a set of selection criteria was applied, 41 articles were selected for detailed analysis. The findings reveal that 1) computerized wheeze analysis can be used for the identification of disease severity level or pathology, 2) further research is required to achieve acceptable rates of identification on the degree of airway obstruction with normal breathing, 3) analysis using combinations of features and on subgroups of the respiratory cycle has provided a pathway to classify various diseases or pathology that stem from airway obstruction.

The central role of recognition in auditory perception: a neurobiological model.

PubMed

McLachlan, Neil; Wilson, Sarah

2010-01-01

The model presents neurobiologically plausible accounts of sound recognition (including absolute pitch), neural plasticity involved in pitch, loudness and location information integration, and streaming and auditory recall. It is proposed that a cortical mechanism for sound identification modulates the spectrotemporal response fields of inferior colliculus neurons and regulates the encoding of the echoic trace in the thalamus. Identification involves correlation of sequential spectral slices of the stimulus-driven neural activity with stored representations in association with multimodal memories, verbal lexicons, and contextual information. Identities are then consolidated in auditory short-term memory and bound with attribute information (usually pitch, loudness, and direction) that has been integrated according to the identities' spectral properties. Attention to, or recall of, a particular identity will excite a particular sequence in the identification hierarchies and so lead to modulation of thalamus and inferior colliculus neural spectrotemporal response fields. This operates as an adaptive filter for identities, or their attributes, and explains many puzzling human auditory behaviors, such as the cocktail party effect, selective attention, and continuity illusions.

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.

Adventitious sounds identification and extraction using temporal-spectral dominance-based features.

PubMed

Jin, Feng; Krishnan, Sridhar Sri; Sattar, Farook

2011-11-01

Respiratory sound (RS) signals carry significant information about the underlying functioning of the pulmonary system by the presence of adventitious sounds (ASs). Although many studies have addressed the problem of pathological RS classification, only a limited number of scientific works have focused on the analysis of the evolution of symptom-related signal components in joint time-frequency (TF) plane. This paper proposes a new signal identification and extraction method for various ASs based on instantaneous frequency (IF) analysis. The presented TF decomposition method produces a noise-resistant high definition TF representation of RS signals as compared to the conventional linear TF analysis methods, yet preserving the low computational complexity as compared to those quadratic TF analysis methods. The discarded phase information in conventional spectrogram has been adopted for the estimation of IF and group delay, and a temporal-spectral dominance spectrogram has subsequently been constructed by investigating the TF spreads of the computed time-corrected IF components. The proposed dominance measure enables the extraction of signal components correspond to ASs from noisy RS signal at high noise level. A new set of TF features has also been proposed to quantify the shapes of the obtained TF contours, and therefore strongly, enhances the identification of multicomponents signals such as polyphonic wheezes. An overall accuracy of 92.4±2.9% for the classification of real RS recordings shows the promising performance of the presented method.

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.

Source localization of turboshaft engine broadband noise using a three-sensor coherence method

NASA Astrophysics Data System (ADS)

Blacodon, Daniel; Lewy, Serge

2015-03-01

Turboshaft engines can become the main source of helicopter noise at takeoff. Inlet radiation mainly comes from the compressor tones, but aft radiation is more intricate: turbine tones usually are above the audible frequency range and do not contribute to the weighted sound levels; jet is secondary and radiates low noise levels. A broadband component is the most annoying but its sources are not well known (it is called internal or core noise). Present study was made in the framework of the European project TEENI (Turboshaft Engine Exhaust Noise Identification). Its main objective was to localize the broadband sources in order to better reduce them. Several diagnostic techniques were implemented by the various TEENI partners. As regards ONERA, a first attempt at separating sources was made in the past with Turbomeca using a three-signal coherence method (TSM) to reject background non-acoustic noise. The main difficulty when using TSM is the assessment of the frequency range where the results are valid. This drawback has been circumvented in the TSM implemented in TEENI. Measurements were made on a highly instrumented Ardiden turboshaft engine in the Turbomeca open-air test bench. Two engine powers (approach and takeoff) were selected to apply TSM. Two internal pressure probes were located in various cross-sections, either behind the combustion chamber (CC), the high-pressure turbine (HPT), the free-turbine first stage (TL), or in four nozzle sections. The third transducer was a far-field microphone located around the maximum of radiation, at 120° from the intake centerline. The key result is that coherence increases from CC to HPT and TL, then decreases in the nozzle up to the exit. Pressure fluctuations from HPT and TL are very coherent with the far-field acoustic spectra up to 700 Hz. They are thus the main acoustic source and can be attributed to indirect combustion noise (accuracy decreases above 700 Hz because coherence is lower, but far-field sound spectra also are much lower above 700 Hz).

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.

Moving microphone arrays to reduce spatial aliasing in the beamforming technique: theoretical background and numerical investigation.

PubMed

Cigada, Alfredo; Lurati, Massimiliano; Ripamonti, Francesco; Vanali, Marcello

2008-12-01

This paper introduces a measurement technique aimed at reducing or possibly eliminating the spatial aliasing problem in the beamforming technique. Beamforming main disadvantages are a poor spatial resolution, at low frequency, and the spatial aliasing problem, at higher frequency, leading to the identification of false sources. The idea is to move the microphone array during the measurement operation. In this paper, the proposed approach is theoretically and numerically investigated by means of simple sound propagation models, proving its efficiency in reducing the spatial aliasing. A number of different array configurations are numerically investigated together with the most important parameters governing this measurement technique. A set of numerical results concerning the case of a planar rotating array is shown, together with a first experimental validation of the method.

Development of an alarm sound database and simulator.

PubMed

Takeuchi, Akihiro; Hirose, Minoru; Shinbo, Toshiro; Imai, Megumi; Mamorita, Noritaka; Ikeda, Noriaki

2006-10-01

The purpose of this study was to develop an interactive software package of alarm sounds to present, recognize and share problems about alarm sounds among medical staff and medical manufactures. The alarm sounds were recorded in variable alarm conditions in a WAV file. The alarm conditions were arbitrarily induced by modifying attachments of various medical devices. The software package that integrated an alarm sound database and simulator was used to assess the ability to identify the monitor that sounded the alarm for the medical staff. Eighty alarm sound files (40MB in total) were recorded from 41 medical devices made by 28 companies. There were three pairs of similar alarm sounds that could not easily be distinguished, two alarm sounds which had a different priority, either low or high. The alarm sound database was created in an Excel file (ASDB.xls 170 kB, 40 MB with photos), and included a list of file names that were hyperlinked to alarm sound files. An alarm sound simulator (AlmSS) was constructed with two modules for simultaneously playing alarm sound files and for designing new alarm sounds. The AlmSS was used in the assessing procedure to determine whether 19 clinical engineers could identify 13 alarm sounds only by their distinctive sounds. They were asked to choose from a list of devices and to rate the priority of each alarm. The overall correct identification rate of the alarm sounds was 48%, and six characteristic alarm sounds were correctly recognized by beetween 63% to 100% of the subjects. The overall recognition rate of the alarm sound priority was only 27%. We have developed an interactive software package of alarm sounds by integrating the database and the alarm sound simulator (URL: http://info.ahs.kitasato-u.ac.jp/tkweb/alarm/asdb.html ). The AlmSS was useful for replaying multiple alarm sounds simultaneously and designing new alarm sounds interactively.

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.

Acoustic detection of Oryctes rhinoceros (Coleoptera: Scarabaeidae: Dynastinae) and Nasutitermes luzonicus (Isoptera: Termitidae) in palm trees in urban Guam.

PubMed

Mankin, R W; Moore, A

2010-08-01

Adult and larval Oryctes rhinoceros (L.) (Coleoptera: Scarabaeidae: Dynastinae) were acoustically detected in live and dead palm trees and logs in recently invaded areas of Guam, along with Nasutitermes luzonicus Oshima (Isoptera: Termitidae), and other small, sound-producing invertebrates and invertebrates. The low-frequency, long-duration sound-impulse trains produced by large, active O. rhinoceros and the higher frequency, shorter impulse trains produced by feeding N. luzonicus had distinctive spectral and temporal patterns that facilitated their identification and discrimination from background noise, as well as from roaches, earwigs, and other small sound-producing organisms present in the trees and logs. The distinctiveness of the O. rhinoceros sounds enables current usage of acoustic detection as a tactic in Guam's ongoing O. rhinoceros eradication program.

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.

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.

Bayesian identification of acoustic impedance in treated ducts.

PubMed

Buot de l'Épine, Y; Chazot, J-D; Ville, J-M

2015-07-01

The noise reduction of a liner placed in the nacelle of a turbofan engine is still difficult to predict due to the lack of knowledge of its acoustic impedance that depends on grazing flow profile, mode order, and sound pressure level. An eduction method, based on a Bayesian approach, is presented here to adjust an impedance model of the liner from sound pressures measured in a rectangular treated duct under multimodal propagation and flow. The cost function is regularized with prior information provided by Guess's [J. Sound Vib. 40, 119-137 (1975)] impedance of a perforated plate. The multi-parameter optimization is achieved with an Evolutionary-Markov-Chain-Monte-Carlo algorithm.

The Audible Human Project: Modeling Sound Transmission in the Lungs and Torso

NASA Astrophysics Data System (ADS)

Dai, Zoujun

Auscultation has been used qualitatively by physicians for hundreds of years to aid in the monitoring and diagnosis of pulmonary diseases. Alterations in the structure and function of the pulmonary system that occur in disease or injury often give rise to measurable changes in lung sound production and transmission. Numerous acoustic measurements have revealed the differences of breath sounds and transmitted sounds in the lung under normal and pathological conditions. Compared to the extensive cataloging of lung sound measurements, the mechanism of sound transmission in the pulmonary system and how it changes with alterations of lung structural and material properties has received less attention. A better understanding of sound transmission and how it is altered by injury and disease might improve interpretation of lung sound measurements, including new lung imaging modalities that are based on an array measurement of the acoustic field on the torso surface via contact sensors or are based on a 3-dimensional measurement of the acoustic field throughout the lungs and torso using magnetic resonance elastography. A long-term goal of the Audible Human Project (AHP ) is to develop a computational acoustic model that would accurately simulate generation, transmission and noninvasive measurement of sound and vibration within the pulmonary system and torso caused by both internal (e.g. respiratory function) and external (e.g. palpation) sources. The goals of this dissertation research, fitting within the scope of the AHP, are to develop specific improved theoretical understandings, computational algorithms and experimental methods aimed at transmission and measurement. The research objectives undertaken in this dissertation are as follows. (1) Improve theoretical modeling and experimental identification of viscoelasticity in soft biological tissues. (2) Develop a poroviscoelastic model for lung tissue vibroacoustics. (3) Improve lung airway acoustics modeling and its coupling to the lung parenchyma; and (4) Develop improved techniques in array acoustic measurement on the torso surface of sound transmitted through the pulmonary system and torso. Tissue Viscoelasticity. Two experimental identification approaches of shear viscoelasticity were used. The first approach is to directly estimate the frequency-dependent surface wave speed and then to optimize the coefficients in an assumed viscoelastic model type. The second approach is to measure the complex-valued frequency response function (FRF) between the excitation location and points at known radial distances. The FRF has embedded in it frequency-dependent information about both surface wave phase speed and attenuation that can be used to directly estimate the complex shear modulus. The coefficients in an assumed viscoelastic tissue model type can then be optimized. Poroviscoelasticity Model for Lung Vibro-acoustics. A poroviscoelastic model based on Biot theory of wave propagation in porous media was used for compression waves in the lungs. This model predicts a fast compression wave speed close to the one predicted by the effective medium theory at low frequencies and an additional slow compression wave due to the out of phase motion of the air and the lung parenchyma. Both compression wave speeds vary with frequency. The fast compression wave speed and attenuation were measured on an excised pig lung under two different transpulmonary pressures. Good agreement was achieved between the experimental observation and theoretical predictions. Sound Transmission in Airways and Coupling to Lung Parenchyma. A computer generated airway tree was simplified to 255 segments and integrated into the lung geometry from the Visible Human Male for numerical simulations. Acoustic impedance boundary conditions were applied at the ends of the terminal segments to represent the unmodeled downstream airway segments. Experiments were also carried out on a preserved pig lung and similar trends of lung surface velocity distribution were observed between the experiments and simulations. This approach provides a feasible way of simplifying the airway tree and greatly reduces the computation time. Acoustic Measurements of Sound Transmission in Human Subjects. Scanning laser Doppler vibrometry (SLDV) was used as a gold standard for transmitted sound measurements on a human subject. A low cost piezodisk sensor array was also constructed as an alternative to SLDV. The advantages and disadvantages of each technique are discussed.

Commissioning Cornell OSTs for SRF cavity testing at Jlab

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

Eremeev, Grigory

2011-07-01

Understanding the current quench limitations in SRF cavities is a topic essential for any SRF accelerator that requires high fields. This understanding crucially depends on correct and precise quench identification. Second sound quench detection in superfluid liquid helium with oscillating superleak transducers is a technique recently applied at Cornell University as a fast and versatile method for quench identification in SRF cavities. Having adopted Cornell design, we report in this contribution on our experience with OST for quench identification in different cavities at JLab.

The Incongruency Advantage for Environmental Sounds Presented in Natural Auditory Scenes

PubMed Central

Gygi, Brian; Shafiro, Valeriy

2011-01-01

The effect of context on the identification of common environmental sounds (e.g., dogs barking or cars honking) was tested by embedding them in familiar auditory background scenes (street ambience, restaurants). Initial results with subjects trained on both the scenes and the sounds to be identified showed a significant advantage of about 5 percentage points better accuracy for sounds that were contextually incongruous with the background scene (e.g., a rooster crowing in a hospital). Further studies with naïve (untrained) listeners showed that this Incongruency Advantage (IA) is level-dependent: there is no advantage for incongruent sounds lower than a Sound/Scene ratio (So/Sc) of −7.5 dB, but there is about 5 percentage points better accuracy for sounds with greater So/Sc. Testing a new group of trained listeners on a larger corpus of sounds and scenes showed that the effect is robust and not confined to specific stimulus set. Modeling using spectral-temporal measures showed that neither analyses based on acoustic features, nor semantic assessments of sound-scene congruency can account for this difference, indicating the Incongruency Advantage is a complex effect, possibly arising from the sensitivity of the auditory system to new and unexpected events, under particular listening conditions. PMID:21355664

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

The ability of cochlear implant users to use temporal envelope cues recovered from speech frequency modulation.

PubMed

Won, Jong Ho; Lorenzi, Christian; Nie, Kaibao; Li, Xing; Jameyson, Elyse M; Drennan, Ward R; Rubinstein, Jay T

2012-08-01

Previous studies have demonstrated that normal-hearing listeners can understand speech using the recovered "temporal envelopes," i.e., amplitude modulation (AM) cues from frequency modulation (FM). This study evaluated this mechanism in cochlear implant (CI) users for consonant identification. Stimuli containing only FM cues were created using 1, 2, 4, and 8-band FM-vocoders to determine if consonant identification performance would improve as the recovered AM cues become more available. A consistent improvement was observed as the band number decreased from 8 to 1, supporting the hypothesis that (1) the CI sound processor generates recovered AM cues from broadband FM, and (2) CI users can use the recovered AM cues to recognize speech. The correlation between the intact and the recovered AM components at the output of the sound processor was also generally higher when the band number was low, supporting the consonant identification results. Moreover, CI subjects who were better at using recovered AM cues from broadband FM cues showed better identification performance with intact (unprocessed) speech stimuli. This suggests that speech perception performance variability in CI users may be partly caused by differences in their ability to use AM cues recovered from FM speech cues.

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.

How learning to abstract shapes neural sound representations

PubMed Central

Ley, Anke; Vroomen, Jean; Formisano, Elia

2014-01-01

The transformation of acoustic signals into abstract perceptual representations is the essence of the efficient and goal-directed neural processing of sounds in complex natural environments. While the human and animal auditory system is perfectly equipped to process the spectrotemporal sound features, adequate sound identification and categorization require neural sound representations that are invariant to irrelevant stimulus parameters. Crucially, what is relevant and irrelevant is not necessarily intrinsic to the physical stimulus structure but needs to be learned over time, often through integration of information from other senses. This review discusses the main principles underlying categorical sound perception with a special focus on the role of learning and neural plasticity. We examine the role of different neural structures along the auditory processing pathway in the formation of abstract sound representations with respect to hierarchical as well as dynamic and distributed processing models. Whereas most fMRI studies on categorical sound processing employed speech sounds, the emphasis of the current review lies on the contribution of empirical studies using natural or artificial sounds that enable separating acoustic and perceptual processing levels and avoid interference with existing category representations. Finally, we discuss the opportunities of modern analyses techniques such as multivariate pattern analysis (MVPA) in studying categorical sound representations. With their increased sensitivity to distributed activation changes—even in absence of changes in overall signal level—these analyses techniques provide a promising tool to reveal the neural underpinnings of perceptually invariant sound representations. PMID:24917783

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.

Sound field separation with sound pressure and particle velocity measurements.

PubMed

Fernandez-Grande, Efren; Jacobsen, Finn; Leclère, Quentin

2012-12-01

In conventional near-field acoustic holography (NAH) it is not possible to distinguish between sound from the two sides of the array, thus, it is a requirement that all the sources are confined to only one side and radiate into a free field. When this requirement cannot be fulfilled, sound field separation techniques make it possible to distinguish between outgoing and incoming waves from the two sides, and thus NAH can be applied. In this paper, a separation method based on the measurement of the particle velocity in two layers and another method based on the measurement of the pressure and the velocity in a single layer are proposed. The two methods use an equivalent source formulation with separate transfer matrices for the outgoing and incoming waves, so that the sound from the two sides of the array can be modeled independently. A weighting scheme is proposed to account for the distance between the equivalent sources and measurement surfaces and for the difference in magnitude between pressure and velocity. Experimental and numerical studies have been conducted to examine the methods. The double layer velocity method seems to be more robust to noise and flanking sound than the combined pressure-velocity method, although it requires an additional measurement surface. On the whole, the separation methods can be useful when the disturbance of the incoming field is significant. Otherwise the direct reconstruction is more accurate and straightforward.

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.

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.

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.

Neural plasticity associated with recently versus often heard objects.

PubMed

Bourquin, Nathalie M-P; Spierer, Lucas; Murray, Micah M; Clarke, Stephanie

2012-09-01

In natural settings the same sound source is often heard repeatedly, with variations in spectro-temporal and spatial characteristics. We investigated how such repetitions influence sound representations and in particular how auditory cortices keep track of recently vs. often heard objects. A set of 40 environmental sounds was presented twice, i.e. as prime and as repeat, while subjects categorized the corresponding sound sources as living vs. non-living. Electrical neuroimaging analyses were applied to auditory evoked potentials (AEPs) comparing primes vs. repeats (effect of presentation) and the four experimental sections. Dynamic analysis of distributed source estimations revealed i) a significant main effect of presentation within the left temporal convexity at 164-215 ms post-stimulus onset; and ii) a significant main effect of section in the right temporo-parietal junction at 166-213 ms. A 3-way repeated measures ANOVA (hemisphere×presentation×section) applied to neural activity of the above clusters during the common time window confirmed the specificity of the left hemisphere for the effect of presentation, but not that of the right hemisphere for the effect of section. In conclusion, spatio-temporal dynamics of neural activity encode the temporal history of exposure to sound objects. Rapidly occurring plastic changes within the semantic representations of the left hemisphere keep track of objects heard a few seconds before, independent of the more general sound exposure history. Progressively occurring and more long-lasting plastic changes occurring predominantly within right hemispheric networks, which are known to code for perceptual, semantic and spatial aspects of sound objects, keep track of multiple exposures. Copyright © 2012 Elsevier Inc. All rights reserved.

Numerical Modelling of the Sound Fields in Urban Streets with Diffusely Reflecting Boundaries

NASA Astrophysics Data System (ADS)

KANG, J.

2002-12-01

A radiosity-based theoretical/computer model has been developed to study the fundamental characteristics of the sound fields in urban streets resulting from diffusely reflecting boundaries, and to investigate the effectiveness of architectural changes and urban design options on noise reduction. Comparison between the theoretical prediction and the measurement in a scale model of an urban street shows very good agreement. Computations using the model in hypothetical rectangular streets demonstrate that though the boundaries are diffusely reflective, the sound attenuation along the length is significant, typically at 20-30 dB/100 m. The sound distribution in a cross-section is generally even unless the cross-section is very close to the source. In terms of the effectiveness of architectural changes and urban design options, it has been shown that over 2-4 dB extra attenuation can be obtained either by increasing boundary absorption evenly or by adding absorbent patches on the façades or the ground. Reducing building height has a similar effect. A gap between buildings can provide about 2-3 dB extra sound attenuation, especially in the vicinity of the gap. The effectiveness of air absorption on increasing sound attenuation along the length could be 3-9 dB at high frequencies. If a treatment is effective with a single source, it is also effective with multiple sources. In addition, it has been demonstrated that if the façades in a street are diffusely reflective, the sound field of the street does not change significantly whether the ground is diffusely or geometrically reflective.

Applications of Hilbert Spectral Analysis for Speech and Sound Signals

NASA Technical Reports Server (NTRS)

Huang, Norden E.

2003-01-01

A new method for analyzing nonlinear and nonstationary data has been developed, and the natural applications are to speech and sound signals. The key part of the method is the Empirical Mode Decomposition method with which any complicated data set can be decomposed into a finite and often small number of Intrinsic Mode Functions (IMF). An IMF is defined as any function having the same numbers of zero-crossing and extrema, and also having symmetric envelopes defined by the local maxima and minima respectively. The IMF also admits well-behaved Hilbert transform. This decomposition method is adaptive, and, therefore, highly efficient. Since the decomposition is based on the local characteristic time scale of the data, it is applicable to nonlinear and nonstationary processes. With the Hilbert transform, the Intrinsic Mode Functions yield instantaneous frequencies as functions of time, which give sharp identifications of imbedded structures. This method invention can be used to process all acoustic signals. Specifically, it can process the speech signals for Speech synthesis, Speaker identification and verification, Speech recognition, and Sound signal enhancement and filtering. Additionally, as the acoustical signals from machinery are essentially the way the machines are talking to us. Therefore, the acoustical signals, from the machines, either from sound through air or vibration on the machines, can tell us the operating conditions of the machines. Thus, we can use the acoustic signal to diagnosis the problems of machines.

Recurrent oil sheens at the deepwater horizon disaster site fingerprinted with synthetic hydrocarbon drilling fluids.

PubMed

Aeppli, Christoph; Reddy, Christopher M; Nelson, Robert K; Kellermann, Matthias Y; Valentine, David L

2013-08-06

We used alkenes commonly found in synthetic drilling-fluids to identify sources of oil sheens that were first observed in September 2012 close to the Deepwater Horizon (DWH) disaster site, more than two years after the Macondo well (MW) was sealed. While explorations of the sea floor by BP confirmed that the well was sound, they identified the likely source as leakage from an 80-ton cofferdam, abandoned during the operation to control the MW in May 2010. We acquired sheen samples and cofferdam oil and analyzed them using comprehensive two-dimensional gas chromatography. This allowed for the identification of drilling-fluid C16- to C18-alkenes in sheen samples that were absent in cofferdam oil. Furthermore, the spatial pattern of evaporative losses of sheen oil alkanes indicated that oil surfaced closer to the DWH wreckage than the cofferdam site. Last, ratios of alkenes and oil hydrocarbons pointed to a common source of oil found in sheen samples and recovered from oil-covered DWH debris collected shortly after the explosion. These lines of evidence suggest that the observed sheens do not originate from the MW, cofferdam, or from natural seeps. Rather, the likely source is oil in tanks and pits on the DWH wreckage, representing a finite oil volume for leakage.

Locating arbitrarily time-dependent sound sources in three dimensional space in real time.

PubMed

Wu, Sean F; Zhu, Na

2010-08-01

This paper presents a method for locating arbitrarily time-dependent acoustic sources in a free field in real time by using only four microphones. This method is capable of handling a wide variety of acoustic signals, including broadband, narrowband, impulsive, and continuous sound over the entire audible frequency range, produced by multiple sources in three dimensional (3D) space. Locations of acoustic sources are indicated by the Cartesian coordinates. The underlying principle of this method is a hybrid approach that consists of modeling of acoustic radiation from a point source in a free field, triangulation, and de-noising to enhance the signal to noise ratio (SNR). Numerical simulations are conducted to study the impacts of SNR, microphone spacing, source distance and frequency on spatial resolution and accuracy of source localizations. Based on these results, a simple device that consists of four microphones mounted on three mutually orthogonal axes at an optimal distance, a four-channel signal conditioner, and a camera is fabricated. Experiments are conducted in different environments to assess its effectiveness in locating sources that produce arbitrarily time-dependent acoustic signals, regardless whether a sound source is stationary or moves in space, even toward behind measurement microphones. Practical limitations on this method are discussed.

NESSTI: Norms for Environmental Sound Stimuli

PubMed Central

Hocking, Julia; Dzafic, Ilvana; Kazovsky, Maria; Copland, David A.

2013-01-01

In this paper we provide normative data along multiple cognitive and affective variable dimensions for a set of 110 sounds, including living and manmade stimuli. Environmental sounds are being increasingly utilized as stimuli in the cognitive, neuropsychological and neuroimaging fields, yet there is no comprehensive set of normative information for these type of stimuli available for use across these experimental domains. Experiment 1 collected data from 162 participants in an on-line questionnaire, which included measures of identification and categorization as well as cognitive and affective variables. A subsequent experiment collected response times to these sounds. Sounds were normalized to the same length (1 second) in order to maximize usage across multiple paradigms and experimental fields. These sounds can be freely downloaded for use, and all response data have also been made available in order that researchers can choose one or many of the cognitive and affective dimensions along which they would like to control their stimuli. Our hope is that the availability of such information will assist researchers in the fields of cognitive and clinical psychology and the neuroimaging community in choosing well-controlled environmental sound stimuli, and allow comparison across multiple studies. PMID:24023866

Common humpback whale (Megaptera novaeangliae) sound types for passive acoustic monitoring.

PubMed

Stimpert, Alison K; Au, Whitlow W L; Parks, Susan E; Hurst, Thomas; Wiley, David N

2011-01-01

Humpback whales (Megaptera novaeangliae) are one of several baleen whale species in the Northwest Atlantic that coexist with vessel traffic and anthropogenic noise. Passive acoustic monitoring strategies can be used in conservation management, but the first step toward understanding the acoustic behavior of a species is a good description of its acoustic repertoire. Digital acoustic tags (DTAGs) were placed on humpback whales in the Stellwagen Bank National Marine Sanctuary to record and describe the non-song sounds being produced in conjunction with foraging activities. Peak frequencies of sounds were generally less than 1 kHz, but ranged as high as 6 kHz, and sounds were generally less than 1 s in duration. Cluster analysis distilled the dataset into eight groups of sounds with similar acoustic properties. The two most stereotyped and distinctive types ("wops" and "grunts") were also identified aurally as candidates for use in passive acoustic monitoring. This identification of two of the most common sound types will be useful for moving forward conservation efforts on this Northwest Atlantic feeding ground.

An approach for automatic classification of grouper vocalizations with passive acoustic monitoring.

PubMed

Ibrahim, Ali K; Chérubin, Laurent M; Zhuang, Hanqi; Schärer Umpierre, Michelle T; Dalgleish, Fraser; Erdol, Nurgun; Ouyang, B; Dalgleish, A

2018-02-01

Grouper, a family of marine fishes, produce distinct vocalizations associated with their reproductive behavior during spawning aggregation. These low frequencies sounds (50-350 Hz) consist of a series of pulses repeated at a variable rate. In this paper, an approach is presented for automatic classification of grouper vocalizations from ambient sounds recorded in situ with fixed hydrophones based on weighted features and sparse classifier. Group sounds were labeled initially by humans for training and testing various feature extraction and classification methods. In the feature extraction phase, four types of features were used to extract features of sounds produced by groupers. Once the sound features were extracted, three types of representative classifiers were applied to categorize the species that produced these sounds. Experimental results showed that the overall percentage of identification using the best combination of the selected feature extractor weighted mel frequency cepstral coefficients and sparse classifier achieved 82.7% accuracy. The proposed algorithm has been implemented in an autonomous platform (wave glider) for real-time detection and classification of group vocalizations.

A full-spectrum analysis of high-speed train interior noise under multi-physical-field coupling excitations

NASA Astrophysics Data System (ADS)

Zheng, Xu; Hao, Zhiyong; Wang, Xu; Mao, Jie

2016-06-01

High-speed-railway-train interior noise at low, medium, and high frequencies could be simulated by finite element analysis (FEA) or boundary element analysis (BEA), hybrid finite element analysis-statistical energy analysis (FEA-SEA) and statistical energy analysis (SEA), respectively. First, a new method named statistical acoustic energy flow (SAEF) is proposed, which can be applied to the full-spectrum HST interior noise simulation (including low, medium, and high frequencies) with only one model. In an SAEF model, the corresponding multi-physical-field coupling excitations are firstly fully considered and coupled to excite the interior noise. The interior noise attenuated by sound insulation panels of carriage is simulated through modeling the inflow acoustic energy from the exterior excitations into the interior acoustic cavities. Rigid multi-body dynamics, fast multi-pole BEA, and large-eddy simulation with indirect boundary element analysis are first employed to extract the multi-physical-field excitations, which include the wheel-rail interaction forces/secondary suspension forces, the wheel-rail rolling noise, and aerodynamic noise, respectively. All the peak values and their frequency bands of the simulated acoustic excitations are validated with those from the noise source identification test. Besides, the measured equipment noise inside equipment compartment is used as one of the excitation sources which contribute to the interior noise. Second, a full-trimmed FE carriage model is firstly constructed, and the simulated modal shapes and frequencies agree well with the measured ones, which has validated the global FE carriage model as well as the local FE models of the aluminum alloy-trim composite panel. Thus, the sound transmission loss model of any composite panel has indirectly been validated. Finally, the SAEF model of the carriage is constructed based on the accurate FE model and stimulated by the multi-physical-field excitations. The results show that the trend of the simulated 1/3 octave band sound pressure spectrum agrees well with that of the on-site-measured one. The deviation between the simulated and measured overall sound pressure level (SPL) is 2.6 dB(A) and well controlled below the engineering tolerance limit, which has validated the SAEF model in the full-spectrum analysis of the high speed train interior noise.

200 kHz Commercial Sonar Systems Generate Lower Frequency Side Lobes Audible to Some Marine Mammals

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

Deng, Zhiqun; Southall, Brandon; Carlson, Thomas J.

2014-04-15

The spectral properties of pulses transmitted by three commercially available 200 kHz echo sounders were measured to assess the possibility that sound energy in below the center (carrier) frequency might be heard by marine mammals. The study found that all three sounders generated sound at frequencies below the center frequency and within the hearing range of some marine mammals and that this sound was likely detectable by the animals over limited ranges. However, at standard operating source levels for the sounders, the sound below the center frequency was well below potentially harmful levels. It was concluded that the sounds generatedmore » by the sounders could affect the behavior of marine mammals within fairly close proximity to the sources and that that the blanket exclusion of echo sounders from environmental impact analysis based solely on the center frequency output in relation to the range of marine mammal hearing should be reconsidered.« less

Blue whales respond to simulated mid-frequency military sonar.

PubMed

Goldbogen, Jeremy A; Southall, Brandon L; DeRuiter, Stacy L; Calambokidis, John; Friedlaender, Ari S; Hazen, Elliott L; Falcone, Erin A; Schorr, Gregory S; Douglas, Annie; Moretti, David J; Kyburg, Chris; McKenna, Megan F; Tyack, Peter L

2013-08-22

Mid-frequency military (1-10 kHz) sonars have been associated with lethal mass strandings of deep-diving toothed whales, but the effects on endangered baleen whale species are virtually unknown. Here, we used controlled exposure experiments with simulated military sonar and other mid-frequency sounds to measure behavioural responses of tagged blue whales (Balaenoptera musculus) in feeding areas within the Southern California Bight. Despite using source levels orders of magnitude below some operational military systems, our results demonstrate that mid-frequency sound can significantly affect blue whale behaviour, especially during deep feeding modes. When a response occurred, behavioural changes varied widely from cessation of deep feeding to increased swimming speed and directed travel away from the sound source. The variability of these behavioural responses was largely influenced by a complex interaction of behavioural state, the type of mid-frequency sound and received sound level. Sonar-induced disruption of feeding and displacement from high-quality prey patches could have significant and previously undocumented impacts on baleen whale foraging ecology, individual fitness and population health.

The Effects of Ambient Conditions on Helicopter Rotor Source Noise Modeling

NASA Technical Reports Server (NTRS)

Schmitz, Frederic H.; Greenwood, Eric

2011-01-01

A new physics-based method called Fundamental Rotorcraft Acoustic Modeling from Experiments (FRAME) is used to demonstrate the change in rotor harmonic noise of a helicopter operating at different ambient conditions. FRAME is based upon a non-dimensional representation of the governing acoustic and performance equations of a single rotor helicopter. Measured external noise is used together with parameter identification techniques to develop a model of helicopter external noise that is a hybrid between theory and experiment. The FRAME method is used to evaluate the main rotor harmonic noise of a Bell 206B3 helicopter operating at different altitudes. The variation with altitude of Blade-Vortex Interaction (BVI) noise, known to be a strong function of the helicopter s advance ratio, is dependent upon which definition of airspeed is flown by the pilot. If normal flight procedures are followed and indicated airspeed (IAS) is held constant, the true airspeed (TAS) of the helicopter increases with altitude. This causes an increase in advance ratio and a decrease in the speed of sound which results in large changes to BVI noise levels. Results also show that thickness noise on this helicopter becomes more intense at high altitudes where advancing tip Mach number increases because the speed of sound is decreasing and advance ratio increasing for the same indicated airspeed. These results suggest that existing measurement-based empirically derived helicopter rotor noise source models may give incorrect noise estimates when they are used at conditions where data were not measured and may need to be corrected for mission land-use planning purposes.

77 FR 42279 - Takes of Marine Mammals Incidental to Specified Activities; Taking Marine Mammals Incidental to a...

Federal Register 2010, 2011, 2012, 2013, 2014

2012-07-18

... sound waves emanating from the pile, thereby reducing the sound energy. A confined bubble curtain... physically block sound waves and they prevent air bubbles from migrating away from the pile. The literature... acoustic pressure wave propagates out from a source, was estimated as so-called ``practical spreading loss...

Physics and Psychophysics of High-Fidelity Sound. Part III: The Components of a Sound-Reproducing System: Amplifiers and Loudspeakers.

ERIC Educational Resources Information Center

Rossing, Thomas D.

1980-01-01

Described are the components for a high-fidelity sound-reproducing system which focuses on various program sources, the amplifier, and loudspeakers. Discussed in detail are amplifier power and distortion, air suspension, loudspeaker baffles and enclosures, bass-reflex enclosure, drone cones, rear horn and acoustic labyrinth enclosures, horn…

Auditory enhancement of increments in spectral amplitude stems from more than one source.

PubMed

Carcagno, Samuele; Semal, Catherine; Demany, Laurent

2012-10-01

A component of a test sound consisting of simultaneous pure tones perceptually "pops out" if the test sound is preceded by a copy of itself with that component attenuated. Although this "enhancement" effect was initially thought to be purely monaural, it is also observable when the test sound and the precursor sound are presented contralaterally (i.e., to opposite ears). In experiment 1, we assessed the magnitude of ipsilateral and contralateral enhancement as a function of the time interval between the precursor and test sounds (10, 100, or 600 ms). The test sound, randomly transposed in frequency from trial to trial, was followed by a probe tone, either matched or mismatched in frequency to the test sound component which was the target of enhancement. Listeners' ability to discriminate matched probes from mismatched probes was taken as an index of enhancement magnitude. The results showed that enhancement decays more rapidly for ipsilateral than for contralateral precursors, suggesting that ipsilateral enhancement and contralateral enhancement stem from at least partly different sources. It could be hypothesized that, in experiment 1, contralateral precursors were effective only because they provided attentional cues about the target tone frequency. In experiment 2, this hypothesis was tested by presenting the probe tone before the precursor sound rather than after the test sound. Although the probe tone was then serving as a frequency cue, contralateral precursors were again found to produce enhancement. This indicates that contralateral enhancement cannot be explained by cuing alone and is a genuine sensory phenomenon.

Experiments to investigate the acoustic properties of sound propagation

NASA Astrophysics Data System (ADS)

Dagdeviren, Omur E.

2018-07-01

Propagation of sound waves is one of the fundamental concepts in physics. Some of the properties of sound propagation such as attenuation of sound intensity with increasing distance are familiar to everybody from the experiences of daily life. However, the frequency dependence of sound propagation and the effect of acoustics in confined environments are not straightforward to estimate. In this article, we propose experiments, which can be conducted in a classroom environment with commonly available devices such as smartphones and laptops to measure sound intensity level as a function of the distance between the source and the observer and frequency of the sound. Our experiments and deviations from the theoretical calculations can be used to explain basic concepts of sound propagation and acoustics to a diverse population of students.

The Robustness of Acoustic Analogies

NASA Technical Reports Server (NTRS)

Freund, J. B.; Lele, S. K.; Wei, M.

2004-01-01

Acoustic analogies for the prediction of flow noise are exact rearrangements of the flow equations N(right arrow q) = 0 into a nominal sound source S(right arrow q) and sound propagation operator L such that L(right arrow q) = S(right arrow q). In practice, the sound source is typically modeled and the propagation operator inverted to make predictions. Since the rearrangement is exact, any sufficiently accurate model of the source will yield the correct sound, so other factors must determine the merits of any particular formulation. Using data from a two-dimensional mixing layer direct numerical simulation (DNS), we evaluate the robustness of two analogy formulations to different errors intentionally introduced into the source. The motivation is that since S can not be perfectly modeled, analogies that are less sensitive to errors in S are preferable. Our assessment is made within the framework of Goldstein's generalized acoustic analogy, in which different choices of a base flow used in constructing L give different sources S and thus different analogies. A uniform base flow yields a Lighthill-like analogy, which we evaluate against a formulation in which the base flow is the actual mean flow of the DNS. The more complex mean flow formulation is found to be significantly more robust to errors in the energetic turbulent fluctuations, but its advantage is less pronounced when errors are made in the smaller scales.

Sound exposure changes European seabass behaviour in a large outdoor floating pen: Effects of temporal structure and a ramp-up procedure.

PubMed

Neo, Y Y; Hubert, J; Bolle, L; Winter, H V; Ten Cate, C; Slabbekoorn, H

2016-07-01

Underwater sound from human activities may affect fish behaviour negatively and threaten the stability of fish stocks. However, some fundamental understanding is still lacking for adequate impact assessments and potential mitigation strategies. For example, little is known about the potential contribution of the temporal features of sound, the efficacy of ramp-up procedures, and the generalisability of results from indoor studies to the outdoors. Using a semi-natural set-up, we exposed European seabass in an outdoor pen to four treatments: 1) continuous sound, 2) intermittent sound with a regular repetition interval, 3) irregular repetition intervals and 4) a regular repetition interval with amplitude 'ramp-up'. Upon sound exposure, the fish increased swimming speed and depth, and swam away from the sound source. The behavioural readouts were generally consistent with earlier indoor experiments, but the changes and recovery were more variable and were not significantly influenced by sound intermittency and interval regularity. In addition, the 'ramp-up' procedure elicited immediate diving response, similar to the onset of treatment without a 'ramp-up', but the fish did not swim away from the sound source as expected. Our findings suggest that while sound impact studies outdoors increase ecological and behavioural validity, the inherently higher variability also reduces resolution that may be counteracted by increasing sample size or looking into different individual coping styles. Our results also question the efficacy of 'ramp-up' in deterring marine animals, which warrants more investigation. Copyright © 2016 Elsevier Ltd. All rights reserved.

Sound Waves Induce Neural Differentiation of Human Bone Marrow-Derived Mesenchymal Stem Cells via Ryanodine Receptor-Induced Calcium Release and Pyk2 Activation.

PubMed

Choi, Yura; Park, Jeong-Eun; Jeong, Jong Seob; Park, Jung-Keug; Kim, Jongpil; Jeon, Songhee

2016-10-01

Mesenchymal stem cells (MSCs) have shown considerable promise as an adaptable cell source for use in tissue engineering and other therapeutic applications. The aims of this study were to develop methods to test the hypothesis that human MSCs could be differentiated using sound wave stimulation alone and to find the underlying mechanism. Human bone marrow (hBM)-MSCs were stimulated with sound waves (1 kHz, 81 dB) for 7 days and the expression of neural markers were analyzed. Sound waves induced neural differentiation of hBM-MSC at 1 kHz and 81 dB but not at 1 kHz and 100 dB. To determine the signaling pathways involved in the neural differentiation of hBM-MSCs by sound wave stimulation, we examined the Pyk2 and CREB phosphorylation. Sound wave induced an increase in the phosphorylation of Pyk2 and CREB at 45 min and 90 min, respectively, in hBM-MSCs. To find out the upstream activator of Pyk2, we examined the intracellular calcium source that was released by sound wave stimulation. When we used ryanodine as a ryanodine receptor antagonist, sound wave-induced calcium release was suppressed. Moreover, pre-treatment with a Pyk2 inhibitor, PF431396, prevented the phosphorylation of Pyk2 and suppressed sound wave-induced neural differentiation in hBM-MSCs. These results suggest that specific sound wave stimulation could be used as a neural differentiation inducer of hBM-MSCs.

Study on acoustical properties of sintered bronze porous material for transient exhaust noise of pneumatic system

NASA Astrophysics Data System (ADS)

Li, Jingxiang; Zhao, Shengdun; Ishihara, Kunihiko

2013-05-01

A novel approach is presented to study the acoustical properties of sintered bronze material, especially used to suppress the transient noise generated by the pneumatic exhaust of pneumatic friction clutch and brake (PFC/B) systems. The transient exhaust noise is impulsive and harmful due to the large sound pressure level (SPL) that has high-frequency. In this paper, the exhaust noise is related to the transient impulsive exhaust, which is described by a one-dimensional aerodynamic model combining with a pressure drop expression of the Ergun equation. A relation of flow parameters and sound source is set up. Additionally, the piston acoustic source approximation of sintered bronze silencer with cylindrical geometry is presented to predict SPL spectrum at a far-field observation point. A semi-phenomenological model is introduced to analyze the sound propagation and reduction in the sintered bronze materials assumed as an equivalent fluid with rigid frame. Experiment results under different initial cylinder pressures are shown to corroborate the validity of the proposed aerodynamic model. In addition, the calculated sound pressures according to the equivalent sound source are compared with the measured noise signals both in time-domain and frequency-domain. Influences of porosity of the sintered bronze material are also discussed.