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At least 1,477 records · Page 82Linked to original sources
PubMed2000

Auditory discrimination in a sound-producing electric fish (Pollimyrus): tone frequency and click-rate difference detection.

Pollimyrus adspersus is a fish that uses simple sounds for communication and has auditory specializations for sound-pressure detection. The sounds are species-specific, and the sounds of individuals are sufficiently stereotyped that they could mediate individual recognition. Behavioral measurements are presented indicating that Pollimyrus probably can make species and individual discriminations on the basis of acoustic cues. Interclick interval (ICI; 10-40 ms) and frequency (100-1400 Hz) discrimination was assessed using modulations of the fish's electric organ discharge rate in the presence of a target stimulus presented in alternation with an ongoing base stimulus. Tone frequency discrimination was best in the 200-600-Hz range, with the best threshold of 1.7% +/- 0.4% standard error at 500 Hz (or 8.5 Hz +/- 1.9 SE). The just noticeable differences (jnd's) were relatively constant from 100 to 500 Hz (mean 8.7 Hz), then increased at a rate of 13.3 Hz per 100 Hz. For click trains, jnd's increased linearly with ICI. The mean jnd's for 10- and 15-ms ICI were both 300 micros (SE= 0.8 ms at 10-ms ICI, SE= 0.11 ms at 15-ms ICI). The jnd at 20-ms ICI was only 1.1 ms +/- 0.25 SE.

Animals↗
PubMed2001

Orthogonal acoustical factors of a sound field in a bamboo forest.

To investigate the acoustical quality of a sound field in a bamboo forest, acoustical measurements were conducted to obtain orthogonal acoustical factors of the sound field. These results are compared with previous results for a sound field in an ordinary forest [H. Sakai, S. Sato, and Y. Ando, J. Acoust. Soc. Am. 104, 1491-1497 (1998)]. The IACC, which is defined as a maximum value of the normalized interaural cross-correlation function between signals at the ears, was 0.07 (4 kHz) and 0.16 (2 kHz) at positions 20 and 40 m from the source, respectively. These values are much better than those in the previously investigated forest. The measured subsequent reverberation time Tsub was up to 1.5 s in the frequency range above 1 kHz at the position 40 m from the source. For certain music sources with higher frequency components, therefore, sound fields in a bamboo forest have excellent acoustic properties.

Acoustics↗
PubMed2002

Judgments of noticeable differences in sound fields of concert halls caused by intensity variations in early reflections.

In concert halls early reflections combine with the direct sound and with reverberation to determine the subjective rating of a room's acoustics. Of interest is how variations in the amplitudes of these early reflections are related to the subjectively just-noticeable differences (jnd) in several important acoustical parameters for their wide range encountered in existing halls. Investigated were four subjective parameters, apparent source width (ASW), loudness, intimacy and clarity, which are related to the physical measurements, [1-IACC(E3)], G, ITDG, and C80, defined mathematically in Beranek [Concert and Opera Halls: How They Sound (Acoustical Society of America, New York, 1996)]. Forty-eight types of sound fields were chosen in which to make variations in the amplitudes of early reflections and were reproduced electro-acoustically by multiple loudspeakers in an anechoic chamber. The results indicate that ASW and loudness are more sensitive to changes in the levels of early reflections, and were the primary parameters investigated. Although the number of subjects available with enough experience in listening classical music is limited and the measured jnd is an initial estimation, the jnd of [1-IACC(E3)] is measured as 0.065+/-0.015 in variations of sound field structures and the jnd of G was measured as 0.25+/-0.15 dB, which is consistent with the results of previous studies.

Acoustics↗
PubMed2002

The connection between sound production and jet structure of the supersonic impinging jet.

An experimental investigation into the sound-producing characteristics of moderately and highly underexpanded supersonic impinging jets exhausting from a round convergent nozzle is presented. The production of large plate tones by impingement on a square plate with a side dimension equal to 12 nozzle exit diameters is studied using random and phase-locked shadowgraph photography. Discrete frequency sound is produced in the near-wall region of the jet when a Mach disk occurs upstream of the standoff shock wave. Tones cease when the plate distance is approximately 2.2 free-jet cell lengths and the first and second shock waves are located in the free-jet positions. The production of impulsive sound appears to be associated with the collapse of the standoff shock wave during a portion of the oscillation cycle. Results from unsteady plate-pressure measurements indicate that plane-wave motion occurs in the impingement region and a secondary pressure maximum is observed on the plate adjacent to the flow region where sound appears to originate.

Aircraft↗
PubMed2003

Derivation of an optimal directivity pattern for sweet spot widening in stereo sound reproduction.

In this paper the correction of the degradation of the stereophonic illusion during sound reproduction due to off-center listening is investigated. The main idea is that the directivity pattern of a loudspeaker array should have a well-defined shape such that a good stereo reproduction is achieved in a large listening area. Therefore, a mathematical description to derive an optimal directivity pattern l(opt) that achieves sweet spot widening in a large listening area for stereophonic sound applications is described. This optimal directivity pattern is based on parametrized time/intensity trading data coming from psycho-acoustic experiments within a wide listening area. After the study, the required digital FIR filters are determined by means of a least-squares optimization method for a given stereo base setup (two pair of drivers for the loudspeaker arrays and 2.5-m distance between loudspeakers), which radiate sound in a broad range of listening positions in accordance with the derived l(opt). Informal listening tests have shown that the l(opt) worked as predicted by the theoretical simulations. They also demonstrated the correct central sound localization for speech and music for a number of listening positions. This application is referred to as "Position-Independent (PI) stereo."

Acoustics↗
PubMed2003

The influence of large-scale seafloor slope and average bottom sound speed on low-grazing-angle monostatic acoustic scattering.

Variations in large-scale seafloor slope and average seabed sound speed account for a significant portion of the variations in scattering intensity observed in low-grazing-angle monostatic reverberation. Numerical modeling using a finite-difference solution to the elastic wave equation is used to quantify the effect of these large-scale parameters in interpretations of reverberation data. For hard rough seafloor (e.g., basalt), the results of the modeling suggest that the monostatic backscattering strength increases with increasing large-scale seafloor slope up to a slope of about 15 degrees dipping toward the incident direction. Once the grazing angle of the incident wavefield exceeds the critical grazing angle for the flat reference seafloor the backscattering intensity increases only slowly with increasing grazing angle. Similarly, average subseafloor sound speed has a significant effect. Seafloor with low sound speeds characteristic of soft bottoms (e.g., sediment) generate significantly weaker backscatter signals than seafloor with sound speeds characteristic of hard bottoms (e.g., basalt). The difference is that the shear waves can always be passed into soft bottoms because even for a flat seafloor there is no shear wave critical grazing angle.

Acoustics↗
PubMed2003

Application of loudness models to sound processing for cochlear implants.

A new paradigm for processing sound signals for multiple-electrode cochlear implants is introduced, and results are presented from an initial psychophysical evaluation of its effect on the perceived loudness of complex sounds. A real-time processing scheme based on this paradigm, called SpeL, has been developed primarily to improve control of loudness for implant users. SpeL differs from previous schemes in several ways. Most importantly, it incorporates a published numerical model which predicts the loudness perceived by implant users for complex patterns of pulsatile electric stimulation as a function of the pulses' physical parameters. This model is controlled by the output of a corresponding model that estimates the loudness perceived by normally hearing listeners for complex sounds. The latter model produces an estimate of the specific loudness arising from an acoustic signal. In SpeL, the specific loudness function, which describes the contribution to total loudness of each of a number of frequency bands (or cochlear positions), is converted to a pattern of electric stimulation on an appropriate set of electrodes. By application of the loudness model for electric stimulation, this pattern is designed to produce a specific loudness function for the implant user which approximates that produced by the normal-hearing model for the same input signal. The results of loudness magnitude estimation experiments with five users of the SpeL scheme confirmed that the psychophysical functions relating overall loudness perceived to input sound level for five complex acoustic signals were, on average, very similar to those for normal hearing.

Aged↗
PubMed2004

Categorization and discrimination of nonspeech sounds: differences between steady-state and rapidly-changing acoustic cues.

Different patterns of performance across vowels and consonants in tests of categorization and discrimination indicate that vowels tend to be perceived more continuously, or less categorically, than consonants. The present experiments examined whether analogous differences in perception would arise in nonspeech sounds that share critical transient acoustic cues of consonants and steady-state spectral cues of simplified synthetic vowels. Listeners were trained to categorize novel nonspeech sounds varying along a continuum defined by a steady-state cue, a rapidly-changing cue, or both cues. Listeners' categorization of stimuli varying on the rapidly changing cue showed a sharp category boundary and posttraining discrimination was well predicted from the assumption of categorical perception. Listeners more accurately discriminated but less accurately categorized steady-state nonspeech stimuli. When listeners categorized stimuli defined by both rapidly-changing and steady-state cues, discrimination performance was accurate and the categorization function exhibited a sharp boundary. These data are similar to those found in experiments with dynamic vowels, which are defined by both steady-state and rapidly-changing acoustic cues. A general account for the speech and nonspeech patterns is proposed based on the supposition that the perceptual trace of rapidly-changing sounds decays faster than the trace of steady-state sounds.

Acoustic Stimulation↗
PubMed2004

Hearing-aid automatic gain control adapting to two sound sources in the environment, using three time constants.

A hearing aid AGC algorithm is presented that uses a richer representation of the sound environment than previous algorithms. The proposed algorithm is designed to (1) adapt slowly (in approximately 10 s) between different listening environments, e.g., when the user leaves a single talker lecture for a multi-babble coffee-break; (2) switch rapidly (about 100 ms) between different dominant sound sources within one listening situation, such as the change from the user's own voice to a distant speaker's voice in a quiet conference room; (3) instantly reduce gain for strong transient sounds and then quickly return to the previous gain setting; and (4) not change the gain in silent pauses but instead keep the gain setting of the previous sound source. An acoustic evaluation showed that the algorithm worked as intended. The algorithm was evaluated together with a reference algorithm in a pilot field test. When evaluated by nine users in a set of speech recognition tests, the algorithm showed similar results to the reference algorithm.

Acoustics↗
PubMed2004

Drilling and operational sounds from an oil production island in the ice-covered Beaufort sea.

Recordings of sounds underwater and in air, and of iceborne vibrations, were obtained at Northstar Island, an artificial gravel island in the Beaufort Sea near Prudhoe Bay (Alaska). The aim was to document the levels, characteristics, and range dependence of sounds and vibrations produced by drilling and oil production during the winter, when the island was surrounded by shore-fast ice. Drilling produced the highest underwater broadband (10-10,000 Hz) levels (maximum= 124 dB re: 1 microPa at 1 km), and mainly affected 700-1400 Hz frequencies. In contrast, drilling did not increase broadband levels in air or ice relative to levels during other island activities. Production did not increase broadband levels for any of the sensors. In all media, broadband levels decreased by approximately 20 dB/tenfold change in distance. Background levels underwater were reached by 9.4 km during drilling and 3-4 km without. In the air and ice, background levels were reached 5-10 km and 2-10 km from Northstar, respectively, depending on the wind but irrespective of drilling. A comparison of the recorded sounds with harbor and ringed seal audiograms showed that Northstar sounds were probably audible to seals, at least intermittently, out to approximately 1.5 km in water and approximately 5 km in air.

Acoustics↗
PubMed2005

Are high perches in the blackcap Sylvia atricapilla song or listening posts? A sound transmission study.

Birds often sing from high perches referred to as song posts. However, birds also listen and keep a lookout from these perches. We used a sound transmission experiment to investigate the changes for receiving and sending conditions that a territorial songbird may experience by moving upwards in the vegetation. Representative song elements of the blackcap Sylvia atricapilla were transmitted in a forest habitat in spring using a complete factorial design with natural transmission distances and speaker and microphone heights. Four aspects of sound degradation were quantified: signal-to-noise ratio, excess attenuation, distortion within the sounds determined as a blur ratio, and prolongation of the sounds with "tails" of echoes determined as a tail-to-signal ratio. All four measures indicated that degradation decreased with speaker and microphone height. However, the decrease was considerably higher for the microphone than for the speaker. This suggests that choosing high perches in a forest at spring results in more benefits to blackcaps in terms of improved communication conditions when they act as receivers than as senders.

Amplifiers, Electronic↗
PubMed2005

Classification and identification of recorded and synthesized impact sounds by practiced listeners, musicians, and nonmusicians.

Three experiments were conducted to test the viability of a low-parameter modal model for synthesizing impact sounds to be used in commercial and psychoacoustic research. The model was constrained to have four physically based parameters dictating the amplitude, frequency, and decay of modes. The values of these parameters were selected by ear to roughly match the recordings of ten different resonant objects suspended by hand and struck with different mallets. In experiment 1, neither 35 professional musicians nor 187 college undergraduates could identify which of the two matched sounds was the real recording with better than chance accuracy, though significantly better than chance performance was obtained when modal parameters were selected without the previously imposed physical constraints. In experiment 2, the undergraduates identified the source corresponding to the recorded and synthesized sounds with the same level of accuracy and largely the same pattern of errors. Finally, experiment 3 showed highly practiced listeners to be largely insensitive to changes in the acoustic waveform resulting from an increase in the number of free parameters used in the modal model beyond 3. The results suggest that low-parameter, modal models might be exploited meaningfully in many commercial and research applications involving human perception of impact sounds.

Adult↗
PubMed2006

Vocal tract resonances and the sound of the Australian didjeridu (yidaki) I. experiment.

The didjeridu, or yidaki, is a simple tube about 1.5 m long, played with the lips, as in a tuba, but mostly producing just a tonal, rhythmic drone sound. The acoustic impedance spectra of performers' vocal tracts were measured while they played and compared with the radiated sound spectra. When the tongue is close to the hard palate, the vocal tract impedance has several maxima in the range 1-3 kHz. These maxima, if sufficiently large, produce minima in the spectral envelope of the sound because the corresponding frequency components of acoustic current in the flow entering the instrument are small. In the ranges between the impedance maxima, the lower impedance of the tract allows relatively large acoustic current components that correspond to strong formants in the radiated sound. Broad, weak formants can also be observed when groups of even or odd harmonics coincide with bore resonances. Schlieren photographs of the jet entering the instrument and high speed video images of the player's lips show that the lips are closed for about half of each cycle, thus generating high levels of upper harmonics of the lip frequency. Examples of the spectra of "circular breathing" and combined playing and vocalization are shown.

Acoustics↗
PubMed2006

Directional loudness in an anechoic sound field, head-related transfer functions, and binaural summation.

The effect of sound incidence angle on loudness was investigated using real sound sources positioned in an anechoic chamber. Eight normal-hearing listeners produced loudness matches between a frontal reference location and seven sources placed at other directions, both in the horizontal and median planes. Matches were obtained via a two-interval, adaptive forced-choice (2AFC) procedure for three center frequencies (0.4, 1, and 5 kHz) and two overall levels (45 and 65 dB SPL). The results showed that loudness is not constant over sound incidence angles, with directional sensitivity varying over a range of up to 10 dB, exhibiting considerable frequency dependence, but only minor effects of overall level. The pattern of results varied substantially between subjects, but was largely accounted for by variations in individual head-related transfer functions. Modeling of binaural loudness based on the at-ear signals favored a sound-power summation model, according to which the maximum binaural gain is only 3 dB, over competing models based on larger gains, or on the summation of monaural loudness indices.

Adult↗
PubMed2006

Experimental evaluation of radiosity for room sound-field prediction.

An acoustical radiosity model was evaluated for how it performs in predicting real room sound fields. This was done by comparing radiosity predictions with experimental results for three existing rooms--a squash court, a classroom, and an office. Radiosity predictions were also compared with those by ray tracing--a "reference" prediction model--for both specular and diffuse surface reflection. Comparisons were made for detailed and discretized echograms, sound-decay curves, sound-propagation curves, and the variations with frequency of four room-acoustical parameters--EDT, RT, D50, and C80. In general, radiosity and diffuse ray tracing gave very similar predictions. Predictions by specular ray tracing were often very different. Radiosity agreed well with experiment in some cases, less well in others. Definitive conclusions regarding the accuracy with which the rooms were modeled, or the accuracy of the radiosity approach, were difficult to draw. The results suggest that radiosity predicts room sound fields with some accuracy, at least as well as diffuse ray tracing and, in general, better than specular ray tracing. The predictions of detailed echograms are less accurate, those of derived room-acoustical parameters more accurate. The results underline the need to develop experimental methods for accurately characterizing the absorptive and reflective characteristics of room surfaces, possible including phase.

Acoustics↗
PubMed1983

Localization of sound in rooms.

This paper is concerned with the localization of sources of sounds by human listeners in rooms. It presents the results of source-identification experiments designed to determine whether the ability to localize sound in a room depends upon the room acoustics, and how it depends upon the nature of the source signal. The experiments indicate that the localization of impulsive sounds, with strong attack transients, is independent of the room reverberation time, though it may depend upon the room geometry. For sounds without attack transients, localization improves monotonically with the spectral density of the source. Localization of continuous broadband noise does depend upon room reverberation time, and we propose the concept of direct signal to reverberant noise ratio to study that effect. Source identification experiments reveal certain localization biases, invisible to minimum-audible-angle experiments, and of uncertain origin. Appendices to this paper develop the statistics of the source-identification paradigm and show how they relate to the minimum audible angle.

Acoustics↗
PubMed1984

Speechreading supplemented with frequency-selective sound-pressure information.

The benefit of supplementing speechreading with frequency-selective sound-pressure information was studied by auditorily presenting this information to normal-hearing listeners. The sound-pressure levels in one or two frequency bands of the speech signal with center frequencies of 500, 1600, and 3160 Hz, respectively, and with 1- or 1/3-oct bandwidth were used to amplitude-modulate pure-tone carriers with frequencies equal to the center frequencies of the filter bands. Short sentences were presented to 18 normal-hearing listeners under the conditions of speechreading-only and speechreading combined with the sound-pressure information. The mean number of correctly perceived syllables increased from 22.8% for speechreading-only to 65.7% when sound-pressure information was supplied in a single 1-oct band at 500 Hz and to 86.7% with two 1-oct bands at 500 and 3160 Hz, respectively. The latter signal scored only 26.7% correct syllables without accompanying visual information.

Adolescent↗
PubMed1989

Informational processing of complex sound. I: Intensity discrimination.

This paper reports on some initial experiments using the sample discrimination paradigm to investigate normal-hearing listeners' ability to process information in complex, nonspeech sounds. An important feature of the sample discrimination experiment is that the value of the difference to be discriminated randomly varies from trial to trial. It is this variation that yields potential information. In the present study, listeners heard a pair of multitone complexes (or sequences) on each trial. The individual levels of the tones were drawn from two normal distributions differing only in mean. The listener's task was to identify the sound having the higher mean tone level. For an ideal observer in these experiments, performance in d' grows as the square root n, where n is the number of tones. Obtained d' grew more nearly as the cube root of n regardless of whether the tones were played sequentially or simultaneously or whether they were increased in number from high frequencies to low or from low frequencies to high. A preliminary model is proposed in which discrimination performance depends predominantly on the information content of the sounds and is largely independent of the physical dimensions along which the sounds vary. Information content is defined in terms of the variance of the underlying stimulus distributions and a stimulus equivocation factor that is derived from the data. Based on this model, transmitted information is estimated to be between 1.0 and 2.6 bits.

Acoustic Stimulation↗