An alternate approach to constructing distortion product otoacoustic emission (DPOAE) suppression tuning curves.
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Distortion product otoacoustic emissions (DPOAEs) are generated from the nonlinear transduction n cochlear outer hair cells. The transducer function demonstrating a compressive nonlinearity can be estimated from low-frequency modulation of DPOAEs. Experimental results from the gerbils showed that the magnitude of quadratic difference tone (QDT, f2-f1) was either enhanced or suppressed depending on the phase of the low-frequency bias tone. Within one period of the bias tone, QDT magnitudes exhibited two similar modulation patterns, each resembling the absolute value of the second derivative of the transducer function. In the time domain, the center notches of the modulation patterns occurred around the zero crossings of the bias pressure, whereas peaks corresponded to the increase or decrease in bias pressure. Evaluated with respect to the bias pressure, modulated QDT magnitude displayed a double-modulation pattern marked by a separation of the center notches. Loading/unloading of the cochlear transducer or rise/fall in bias pressure shifted the center notch to positive or negative sound pressures, indicating a mechanical hysteresis. These results suggest that QDT arises from the compression that coexists with the active hysteresis in cochlear transduction. Modulation of QDT magnitude reflects the dynamic regulation of cochlear transducer gain and compression.
The acoustical features of kettledrums have been analyzed by means of modal analysis and acoustic radiation (p/v ratio) measurements. Modal analysis of two different kettledrums was undertaken, exciting the system both by a hammer and a shaker. Up to 15 vibrational modes were clearly identified. Acoustic radiation was studied using two ways. Based on previous experiments of other researchers, a new parameter, called intensity of acoustic radiation (IAR), has been defined and measured. Results show a strict relationship between IAR and the frequency response function (FRF, which is the v/F ratio), and IAR also strongly relates the modal pattern to acoustic radiation. Finally, IAR is proposed for vibro-acoustical characterization of kettledrums and other musical instruments such as strings, pianos, and harpsichords.
The amplitude of otoacoustic emissions, which arise on the basilar membrane, is assumed to be proportional to basilar-membrane motion. It should then be possible to assess basilar-membrane motion on the basis of otoacoustic emissions. The present study provides support for this possibility by comparing basilar-membrane motion inferred from emissions to that inferred from psychoacoustic measures. Three psychoacoustic measurements believed to be associated with basilar-membrane motion were investigated: (1) pulsation threshold; (2) loudness functions derived from temporal integration; and (3) loudness functions derived from loudness matches between pure tones and multitone complexes. Results of the psychoacoustic measurements and of the tone-burst otoacoustic emissions led to very similar estimations of basilar-membrane motion. Accordingly, emissions could serve as an excellent tool--one that is objective, noninvasive, and rapid--for estimating relative basilar-membrane motion.
Noise pollution is recognized as a potential danger to marine mammals in general, and to the St. Lawrence beluga in particular. One method of determining the impacts of noise on an animal's communication is to observe a natural and repeatable response of the vocal system to variations in noise level. This is accomplished by observing intensity changes in animal vocalizations in response to environmental noise. One such response observed in humans, songbirds, and some primates is the Lombard vocal response. This response represents a vocal system reaction manifested by changes in vocalization level in direct response to changes in the noise field. In this research, a population of belugas in the St. Lawrence River Estuary was tested to determine whether a Lombard response existed by using hidden Markhov-classified vocalizations as targets for acoustical analyses. Correlation and regression analyses of signals and noise indicated that the phenomenon does exist. Further, results of human subjects experiments [Egan, J. J. (1966), Ph.D. dissertation; Scheifele, P. M. (2003), Ph.D. dissertation], along with previously reported data from other animal species, are similar to those exhibited by the belugas. Overall, findings suggest that typical noise levels in the St. Lawrence River Estuary have a detectable effect on the communication of the beluga.
Subglottal pressure is one of the main voice control factors, controlling vocal loudness. In this investigation the effects of subglottal pressure variation on the voice source in untrained female and male voices phonating at a low, a middle, and a high fundamental frequency are analyzed. The subjects produced a series of /pae/ syllables at varied degrees of vocal loudness, attempting to keep pitch constant. Subglottal pressure was estimated from the oral pressure during the /p/ occlusion. Ten subglottal pressure values, approximately equidistantly spaced within the pressure range used, were identified, and the voice source of the vowels following these pressure values was analyzed by inverse filtering the airflow signal as captured by a Rothenberg mask. The maximum flow declination rate (MFDR) was found to increase linearly with subglottal pressure, but a given subglottal pressure produced lower values for female than for male voices. The closed quotient increased quickly with subglottal pressure at low pressures and slowly at high pressures, such that the relationship can be approximated by a power function. For a given subglottal pressure value, female voices reached lower values of closed quotient than male voices.
Three methods were applied to determine which decision model best accounted for same-different judgments about the amplitude of acoustic sinusoids. The methods were (1) analysis of the shape of the receiver operating characteristic; (2) analysis of an observer's decision space; and (3) a correlation method based on the conditional-on-single-stimulus procedure. In one experiment, observers rated their confidence that a pair of 1-kHz sinusoids was drawn from the same or from different sets. The two sets had identical Gaussian distributions of decibel amplitudes, but their mean amplitudes differed by 3 dB. The differencing model, in which observers base their decision on the absolute difference between the two observations, accounted for the data better than the independent-observation model, in which observers compute independent likelihood ratios for each observation. A second experiment added to every trial an interval that contained a 1-kHz sinusoid whose amplitude equaled the grand mean of both stimulus sets. Despite this additional information, which is needed to adopt the independent-observation model, the differencing model again better accounted for the data.
The purpose of this paper is to provide insight into how speech is processed by the auditory system, by quantifying the nature of nonsense speech sound confusions. (1) The Miller and Nicely [J. Acoust. Soc. Am. 27(2), 338-352 (1955)] confusion matrix (CM) data are analyzed by plotting the CM elements Si,j(SNR) as a function of the signal-to-noise ratio (SNR). This allows for the robust clustering of perceptual feature (event) groups, not robustly defined by a single CM table, where clusters depend on the sound order. (2) The SNR is then re-expressed as an articulation index (AI), and used as the independent variable. The normalized log scores log(1-Si,i(AI)) and log(Si,j(AI)), j not equal to i, then become linear functions of AI, on log-error versus AI plots. This linear dependence may be interpreted as an extension of the band-independence model of Fletcher. (3) The model formula for the average score for the finite-alphabet case Pc(AI,H)= sigma(N)i=1Si,i/N is then modified to include the effect of entropy H. Due to the grouping of sounds with increased SNR (and AI), the sound-group entropy Hg plays a key role in this performance measure. (4) A parametric model for the confusions Si,j(AI,Hg) is then described, which characterizes the confusions between competing sounds within a group.
The nonlinear auditory phenomena of compression, suppression, and distortion are known to have a cochlear-mechanical origin. An instantaneous nonlinear transfer function is often assumed to underlie these phenomena, but there are experimental indications that auditory nonlinearity is sluggish rather than instantaneous. This study analyzes the consequences of such sluggishness, using automatic gain control (AGC) as a model noninstantaneous nonlinearity. The distinctive characteristic of AGC, its delayed action, is shown to produce a number of observable and measurable effects that distinguish AGC from instantaneous nonlinearities. A major class of such AGC-specific effects concerns the phase of aural distortion products. For example, the phase of the cancellation tone in the classical psychoacoustic cancellation paradigm is linearly related to the frequency spacing of the primary tones in an AGC, as opposed to the square-law relationship produced by an instantaneous nonlinearity. These and other predictions are confronted with experimental data from the literature. The impact of putative AGC-related delays on the interpretation of distortion product otoacoustic emissions (DPOAEs) is discussed. Detailed suggestions are made for experiments specifically aimed at determining whether cochlear nonlinearity is instantaneous or delayed.
The ability to understand speech in a multi-source environment containing informational masking may depend on the perceptual arrangement of signal and masker objects in space. In normal-hearing listeners, Arbogast et al. [J. Acoust. Soc. Am. 112, 2086-2098 (2002)] found an 18-dB spatial release from a primarily informational masker, compared to 7 dB for a primarily energetic masker. This article extends the earlier work to include the study of listeners with sensorineural hearing loss. Listeners performed closed-set speech recognition in two spatial conditions: 0 degrees and 90 degrees separation between signal and masker. Three maskers were tested: (1) the different-band sentence masker was designed to be primarily informational; (2) the different-band noise masker was a control for the different-band sentence; and (3) the same-band noise masker was designed to be primarily energetic. The spatial release from the different-band sentence was larger than for the other maskers, but was smaller (10 dB) for the hearing-impaired group than for the normal-hearing group (15 dB). The smaller benefit for the hearing-impaired listeners can be partially explained by masker sensation level. However, the results suggest that hearing-impaired listeners can use the perceptual effect of spatial separation to improve speech recognition in the presence of a primarily informational masker.
The speech intelligibility index (SII) (ANSI S3.5-1997) provides a means for estimating speech intelligibility under conditions of additive stationary noise or bandwidth reduction. The SII concept for estimating intelligibility is extended in this paper to include broadband peak-clipping and center-clipping distortion, with the coherence between the input and output signals used to estimate the noise and distortion effects. The speech intelligibility predictions using the new procedure are compared with intelligibility scores obtained from normal-hearing and hearing-impaired subjects for conditions of additive noise and peak-clipping and center-clipping distortion. The most effective procedure divides the speech signal into low-, mid-, and high-level regions, computes the coherence SII separately for the signal segments in each region, and then estimates intelligibility from a weighted combination of the three coherence SII values.
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Loudness of interaurally correlated narrow- and broadband noises was investigated using a loudness estimation paradigm (with two anchors) presented via headphones. Throughout the experiments (most performed by 12 subjects), the results from both anchors agreed very well. In the first experiment, third-octave-band noises centered around 250, 710, or 2000 Hz, as well as broadband red (-10 dB/oct), pink (-3 dB/oct), and blue (+10 dB/oct) noises, with interaural level differences of delta L = 0, 4, 10, 20, and infinity dB, were presented as test signals while the same sound presented monaurally or diotically served as anchor. The binaurally summed loudness at delta L = 0 dB was found to be larger than the loudness of a monaural signal of the same SPL by a factor of about 1.5 and decreased with increasing delta L. No dependence of this behavior on frequency, level, or spectral shape was found. In a second experiment, abutting frequency bands of varying width were alternately presented to the subject's left and right ears with the overall spectrum encompassing the whole audio range. The binaural loudness was larger for fewer but broader frequency bands. In a third experiment, uniform exciting noise was switched between the two ears at various speeds. Increasing the switching frequency gave rise to an increase in loudness of about 20%. All results are discussed from the viewpoint of the use of the standardized loudness meter. At this point, there is no evidence that any significant systematic errors due to single-channel evaluation (in contrast to the human two-channel processing) are made by measuring loudness using these meters.
A series of experiments on the detectability of vowels in isolation has been completed. Stimuli consisted of three sets of ten vowels: one synthetic, one from a male talker, and one from a female talker. Vowel durations ranged from 20-160 ms for each of the sets. Thresholds for detecting the vowels in isolation were obtained from well-trained, normal-hearing listeners using an adaptive-tracking paradigm. For a given duration, detection thresholds for vowels calibrated for equal rms sound pressure at the earphones differed by 22 dB across the 30 vowels. In addition, an orderly decrease in vowel thresholds was obtained for increased duration, as predicted from previous data on temporal integration. Several different analyses were performed in an attempt to explain the differential detectability across the 30 vowels. Analyses accounting for audibility reduced threshold variability significantly, but vowel thresholds still ranged over 15 dB. Vowel spectra were subsequently modeled as excitation patterns, and several detection hypotheses were examined. A simple average of excitation levels across excited critical bands provided the best prediction of the level variations needed to maintain threshold-level loudness across all vowels.
Gap detection thresholds were measured by forced-choice procedure for conditions where the duration of a silent gap was varied adaptively between pairs of sinusoidal markers of the same or different frequency. Frequencies of the first sinusoid in a pair of markers ranged from F1 = 500 to 4000 Hz. Second-sinusoid marker frequencies F2 included F1 = F2, and usually frequencies 2%, 5%, 24%, and 50% higher than F1. In preliminary studies the role of presentation level (E/N0) on gap detection was considered. Preliminary data revealed confounding extraneous factors arising from gating transients and from overall stimulus (i.e., markers + gap) and/or masker duration cues. In the main experiments, the contributions of these extraneous cues were evaluated with experimental designs aimed at identifying and minimizing the confounding roles of these cues in gap detection. For conditions where extraneous gating transient cues were minimized (by presenting the sinusoidal markers in a continuous noise masker with random onset phase for the second sinusoid in every pair of markers) and overall stimulus duration cues were diminished (by randomizing the duration of each marker independently), gap detection thresholds increased from 5 to 90 ms as the frequency separation between F1 and F2 was increased by half an octave. When the gap detection thresholds were treated as filter attenuation values by normalizing and converting the data into decibels, the data were closely fit by the roex filter model. On average, the listeners' performances were modeled well by a constant-percentage (7%) bandwidth filter centered on F1.
Either an interaural phase shift or level difference was introduced to a narrow section of broadband noise in order to measure the acuity of the binaural system to segregate a narrowband from a broadband stimulus. Listeners were asked to indicate whether this dichotic noise or a totally diotic noise was presented in a single-interval procedure. Thresholds for interaural phase and level differences were estimated from four point psychometric functions. These thresholds were determined for three bandwidths of interaurally altered noise (2, 10, and 100 Hz) centered at four center frequencies (200, 500, 1000, and 1600 Hz). Thresholds were lowest when the interaurally altered band of noise was centered at 500 Hz, and thresholds increased as the bandwidth of the interaurally altered noise decreased. Performance did not exceed 75% correct when either an interaural phase shift (180 degrees) or interaural level difference (50 dB) was introduced to a 100 Hz band of noise centered at frequencies higher than 1600 Hz. In a second set of conditions, performance was measured when both an interaural phase shift and level difference were presented in a 10-Hz-wide band of noise centered at 500 Hz. A version of the Durlach E-C model was able to account for a great deal of the data. The results are discussed in terms of the Huggins dichotic pitch.
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A detailed measurement of distortion product otoacoustic emission (DPOAE) fine structure was used to extract estimates of the two major components believed to contribute to the overall DPOAE level in the ear canal. A fixed-ratio paradigm was used to record DPOAE fine structure from three normal-hearing ears over a range of 400 Hz for 12 different stimulus-frequency ratios between 1.053 and 1.36 and stimulus levels between 45 and 75 dB SPL. Inverse Fourier transforms of the amplitude and phase data were filtered to extract the early component from the generator region of maximum stimulus overlap and the later component reflected from the characteristic frequency region of the DPOAE. After filtering, the data were returned to the frequency domain to evaluate the impact of the stimulus-frequency ratio and stimulus level on the relative levels of the components. Although there were significant differences between data from different ears some consistent patterns could be detected. The component from the overlap region of the stimulus tones exhibits a bandpass shape, with the maximum occurring at a ratio of 1.2. The mean data from the DPOAE characteristic frequency region also exhibits a bandpass shape but is less sharply tuned and exhibits greater variety across ears and stimulus levels. The component from the DPOAE characteristic frequency region is dominant at ratios narrower than approximately 1.1 (the transition varies between ears). The relative levels of the two components are highly variable at ratios greater than 1.3 and highly dependent on the stimulus level. The reflection component is larger at all ratios at the lowest stimulus level tested (45/45 dB SPL). We discuss the factors shaping DPOAE-component behavior and some cursory implications for the choice of stimulus parameters to be used in clinical protocols.