Auditory filter nonlinearity across frequency using simultaneous notched-noise masking

Test-retest evaluation of a notched-noise test using consumer-grade mobile audio equipment

OBJECTIVE

The aim of this study was to investigate whether consumer-grade mobile audio equipment can be reliably used as a platform for the notched-noise test, including when the test is conducted outside the laboratory.

DESIGN

Two studies were conducted: Study 1 was a notched-noise masking experiment with three different setups: in a psychoacoustic test booth with a standard laboratory PC; in a psychoacoustic test booth with a mobile device; and in a quiet office room with a mobile device. Study 2 employed the same task as Study 1, but compared circumaural headphones to insert earphones.

STUDY SAMPLE

Nine and ten young, normal-hearing participants completed studies 1 and 2, respectively.

RESULTS

The test-retest accuracy of the notched-noise test on the mobile implementation did not differ from that for the laboratory setup. A possible effect of the earphone design was identified in Study 1, which was corroborated by Study 2, where test-retest variability was smallest when comparing results from experiments conducted using identical acoustic transducers.

CONCLUSIONS

Results and test-retest repeatability comparable to standard laboratory settings for the notched-noise test can be obtained with mobile equipment outside the laboratory.

Extending the Hearing-Aid Speech Perception Index (HASPI): Keywords, sentences, and context

The Hearing-Aid Speech Perception Index version 2 (HASPI v2) is a speech intelligibility metric derived by fitting subject responses scored as the proportion of complete sentences correct. This paper presents an extension of HASPI v2, denoted by HASPI w2, which predicts proportion keywords correct for the same datasets used to derive HASPI v2. The results show that the accuracy of HASPI w2 is nearly identical to that of HASPI v2. The values produced by HASPI w2 and HASPI v2 also allow the comparison of proportion words correct and sentences correct for the same stimuli. Using simulation values for speech in additive noise, a model of context effects for words combined into sentences is developed and accounts for the loss of intelligibility inherent in the impaired auditory periphery. In addition, HASPI w2 and HASPI v2 have a small bias term at poor signal-to-noise ratios; the model for context effects shows that the residual bias is reduced in converting from proportion keywords to sentences correct but is greatly magnified when considering the reverse transformation.

High Spectral and Temporal Acuity in Primary Auditory Cortex of Awake Cats

Most accounts of single- and multi-unit responses in auditory cortex under anesthetized conditions have emphasized V-shaped frequency tuning curves and low-pass sensitivity to rates of repeated sounds. In contrast, single-unit recordings in awake marmosets also show I-shaped and O-shaped response areas having restricted tuning to frequency and (for O units) sound level. That preparation also demonstrates synchrony to moderate click rates and representation of higher click rates by spike rates of non-synchronized tonic responses, neither of which are commonly seen in anesthetized conditions. The spectral and temporal representation observed in the marmoset might reflect special adaptations of that species, might be due to single- rather than multi-unit recording, or might indicate characteristics of awake-versus-anesthetized recording conditions. We studied spectral and temporal representation in the primary auditory cortex of alert cats. We observed V-, I-, and O-shaped response areas like those demonstrated in awake marmosets. Neurons could synchronize to click trains at rates about an octave higher than is usually seen with anesthesia. Representations of click rates by rates of non-synchronized tonic responses exhibited dynamic ranges that covered the entire range of tested click rates. The observation of these spectral and temporal representations in cats demonstrates that they are not unique to primates and, indeed, might be widespread among mammalian species. Moreover, we observed no significant difference in stimulus representation between single- and multi-unit recordings. It appears that the principal factor that has hindered observations of high spectral and temporal acuity in the auditory cortex has been the use of general anesthesia.

Improving Auditory Filter Estimation by Incorporating Absolute Threshold and a Level-dependent Internal Noise

Auditory filter (AF) shape has traditionally been estimated with a combination of a notched-noise (NN) masking experiment and a power spectrum model (PSM) of masking. However, there are several challenges that remain in both the simultaneous and forward masking paradigms. We hypothesized that AF shape estimation would be improved if absolute threshold (AT) and a level-dependent internal noise were explicitly represented in the PSM. To document the interaction between NN threshold and AT in normal hearing (NH) listeners, a large set of NN thresholds was measured at four center frequencies (500, 1000, 2000, and 4000 Hz) with the emphasis on low-level maskers. The proposed PSM, consisting of the compressive gammachirp (cGC) filter and three nonfilter parameters, allowed AF estimation over a wide range of frequencies and levels with fewer coefficients and less error than previous models. The results also provided new insights into the nonfilter parameters. The detector signal-to-noise ratio (K ) was found to be constant across signal frequencies, suggesting that no frequency dependence hypothesis is required in the postfiltering process. The ANSI standard "Hearing Level-0dB" function, i.e., AT of NH listeners, could be applied to the frequency distribution of the noise floor for the best AF estimation. The introduction of a level-dependent internal noise could mitigate the nonlinear effects that occur in the simultaneous NN masking paradigm. The new PSM improves the applicability of the model, particularly when the sound pressure level of the NN threshold is close to AT.

An overview of the HASPI and HASQI metrics for predicting speech intelligibility and speech quality for normal hearing, hearing loss, and hearing aids

Alterations of the speech signal, including additive noise and nonlinear distortion, can reduce speech intelligibility and quality. Hearing aids present an especially complicated situation since these devices may implement nonlinear processing designed to compensate for the hearing loss. Hearing-aid processing is often realized as time-varying multichannel gain adjustments, and may also include frequency reassignment. The challenge in designing metrics for hearing aids and hearing-impaired listeners is to accurately model the perceptual trade-offs between speech audibility and the nonlinear distortion introduced by hearing-aid processing. This paper focuses on the Hearing Aid Speech Perception Index (HASPI) and the Hearing Aid Speech Quality Index (HASQI) as representative metrics for predicting intelligibility and quality. These indices start with a model of the auditory periphery that can be adjusted to represent hearing loss. The peripheral model, the speech features computed from the model outputs, and the procedures used to fit the features to subject data are described. Examples are then presented for using the metrics to measure the effects of additive noise, evaluate noise-suppression processing, and to measure the differences among commercial hearing aids. Open questions and considerations in using these and related metrics are then discussed.

Tonotopic Selectivity in Cats and Humans: Electrophysiology and Psychophysics

We describe a scalp-recorded measure of tonotopic selectivity, the "cortical onset response" (COR) and compare the results between humans and cats. The COR results, in turn, were compared with psychophysical masked-detection thresholds obtained using similar stimuli and obtained from both species. The COR consisted of averaged responses elicited by 50-ms tone-burst probes presented at 1-s intervals against a continuous noise masker. The noise masker had a bandwidth of 1 or 1/8th octave, geometrically centred on 4000 Hz for humans and on 8000 Hz for cats. The probe frequency was either - 0.5, - 0.25, 0, 0.25 or 0.5 octaves re the masker centre frequency. The COR was larger for probe frequencies more distant from the centre frequency of the masker, and this effect was greater for the 1/8th-octave than for the 1-octave masker. This pattern broadly reflected the masked excitation patterns obtained psychophysically with similar stimuli in both species. However, the positive signal-to-noise ratio used to obtain reliable COR measures meant that some aspects of the data differed from those obtained psychophysically, in a way that could be partly explained by the upward spread of the probe's excitation pattern. Our psychophysical measurements also showed that the auditory filter width obtained at 8000 Hz using notched-noise maskers was slightly wider in cat than previous measures from humans. We argue that although conclusions from COR measures differ in some ways from conclusions based on psychophysics, the COR measures provide an objective, noninvasive, valid measure of tonotopic selectivity that does not require training and that may be applied to acoustic and cochlear-implant experiments in humans and laboratory animals.

Diotic and Dichotic Mechanisms of Discrimination Threshold in Musicians and Non-Musicians

The perception of harmonic complexes provides important information for musical and vocal communication. Numerous studies have shown that musical training and expertise are associated with better processing of harmonic complexes, however, it is unclear whether the perceptual improvement associated with musical training is universal to different pitch models. The current study addresses this issue by measuring discrimination thresholds of musicians (n = 20) and non-musicians (n = 18) to diotic (same sound to both ears) and dichotic (different sounds to each ear) sounds of four stimulus types: (1) pure sinusoidal tones, PT; (2) four-harmonic complex tones, CT; (3) iterated rippled noise, IRN; and (4) interaurally correlated broadband noise, called the "Huggins" or "dichotic" pitch, DP. Frequency difference limens (DLF) for each stimulus type were obtained via a three-alternative-forced-choice adaptive task requiring selection of the interval with the highest pitch, yielding the smallest perceptible fundamental frequency (F0) distance (in Hz) between two sounds. Music skill was measured by an online test of musical pitch, melody and timing maintained by the International Laboratory for Brain Music and Sound Research. Musicianship, length of music experience and self-evaluation of musical skill were assessed by questionnaire. Results showed musicians had smaller DLFs in all four conditions with the largest group difference in the dichotic condition. DLF thresholds were related to both subjective and objective musical ability. In addition, subjective self-report of musical ability was shown to be a significant variable in group classification. Taken together, the results suggest that music-related plasticity benefits multiple mechanisms of pitch encoding and that self-evaluation of musicality can be reliably associated with objective measures of perception.

Neural fluctuation cues for simultaneous notched-noise masking and profile-analysis tasks: Insights from model midbrain responses

Results of simultaneous notched-noise masking are commonly interpreted as reflecting the bandwidth of underlying auditory filters. This interpretation assumes that listeners detect a tone added to notched-noise based on an increase in energy at the output of an auditory filter. Previous work challenged this assumption by showing that randomly and independently varying (roving) the levels of each stimulus interval does not substantially worsen listener thresholds [Lentz, Richards, and Matiasek (1999). J. Acoust. Soc. Am. 106, 2779-2792]. Lentz et al. further challenged this assumption by showing that filter bandwidths based on notched-noise results were different from those based on a profile-analysis task [Green (1983). Am. Psychol. 38, 133-142; (1988). (Oxford University Press, New York)], although these estimates were later reconciled by emphasizing spectral peaks of the profile-analysis stimulus [Lentz (2006). J. Acoust. Soc. Am. 120, 945-956]. Here, a single physiological model is shown to account for performance in fixed- and roving-level notched-noise tasks and the Lentz et al. profile-analysis task. This model depends on peripheral neural fluctuation cues that are transformed into the average rates of model inferior colliculus neurons. Neural fluctuations are influenced by peripheral filters, synaptic adaptation, cochlear amplification, and saturation of inner hair cells, an element not included in previous theories of envelope-based cues for these tasks. Results suggest reevaluation of the interpretation of performance in these paradigms.

Toward Routine Assessments of Auditory Filter Shape

Purpose A Bayesian adaptive procedure, that is, the quick auditory filter (qAF) procedure, has been shown to improve the efficiency for estimating auditory filter shapes of listeners with normal hearing. The current study evaluates the accuracy and test-retest reliability of the qAF procedure for naïve listeners with a variety of ages and hearing status. Method Fifty listeners who were naïve to psychophysical experiments and exhibit wide ranges of age (19-70 years) and hearing threshold (-5 to 70 dB HL at 2 kHz) were recruited. Their auditory filter shapes were estimated for a 15-dB SL target tone at 2 kHz using both the qAF procedure and the traditional threshold-based procedure. The auditory filter model was defined using 3 parameters: (a) the sharpness of the tip portion of the auditory filter, p; (b) the prominence of the low-frequency tail of the filter, 10log( w); and (c) the listener's efficiency in detection, 10log( K). Results The estimated parameters of the auditory filter model were consistent between 2 qAF runs tested on 2 separate days. The parameter estimates from the 2 qAF runs also agreed well with those estimated using the traditional procedure despite being substantially faster. Across the 3 auditory filter estimates, the dependence of the auditory filter parameters on listener age and hearing threshold was consistent across procedures, as well as consistent with previously published estimates. Conclusions The qAF procedure demonstrates satisfactory test-retest reliability and good agreement to the traditional procedure for listeners with a wide range of ages and with hearing status ranging from normal hearing to moderate hearing impairment.

The role of excitation-pattern cues in the detection of frequency shifts in bandpass-filtered complex tones

One task intended to measure sensitivity to temporal fine structure (TFS) involves the discrimination of a harmonic complex tone from a tone in which all harmonics are shifted upwards by the same amount in hertz. Both tones are passed through a fixed bandpass filter centered on the high harmonics to reduce the availability of excitation-pattern cues and a background noise is used to mask combination tones. The role of frequency selectivity in this "TFS1" task was investigated by varying level. Experiment 1 showed that listeners performed more poorly at a high level than at a low level. Experiment 2 included intermediate levels and showed that performance deteriorated for levels above about 57 dB sound pressure level. Experiment 3 estimated the magnitude of excitation-pattern cues from the variation in forward masking of a pure tone as a function of frequency shift in the complex tones. There was negligible variation, except for the lowest level used. The results indicate that the changes in excitation level at threshold for the TFS1 task would be too small to be usable. The results are consistent with the TFS1 task being performed using TFS cues, and with frequency selectivity having an indirect effect on performance via its influence on TFS cues.

Rapid estimation of high-parameter auditory-filter shapes

A Bayesian adaptive procedure, the quick-auditory-filter (qAF) procedure, was used to estimate auditory-filter shapes that were asymmetric about their peaks. In three experiments, listeners who were naive to psychoacoustic experiments detected a fixed-level, pure-tone target presented with a spectrally notched noise masker. The qAF procedure adaptively manipulated the masker spectrum level and the position of the masker notch, which was optimized for the efficient estimation of the five parameters of an auditory-filter model. Experiment I demonstrated that the qAF procedure provided a convergent estimate of the auditory-filter shape at 2 kHz within 150 to 200 trials (approximately 15 min to complete) and, for a majority of listeners, excellent test-retest reliability. In experiment II, asymmetric auditory filters were estimated for target frequencies of 1 and 4 kHz and target levels of 30 and 50 dB sound pressure level. The estimated filter shapes were generally consistent with published norms, especially at the low target level. It is known that the auditory-filter estimates are narrower for forward masking than simultaneous masking due to peripheral suppression, a result replicated in experiment III using fewer than 200 qAF trials.

On the controversy about the sharpness of human cochlear tuning

In signal processing terms, the operation of the mammalian cochlea in the inner ear may be likened to a bank of filters. Based on otoacoustic emission evidence, it has been recently claimed that cochlear tuning is sharper for human than for other mammals. The claim was corroborated with a behavioral method that involves the masking of pure tones with forward notched noises (NN). Using this method, it has been further claimed that human cochlear tuning is sharper than suggested by earlier behavioral studies. These claims are controversial. Here, we contribute to the controversy by theoretically assessing the accuracy of the NN method at inferring the bandwidth (BW) of nonlinear cochlear filters. Behavioral forward masking was mimicked using a computer model of the squared basilar membrane response followed by a temporal integrator. Isoresponse and isolevel versions of the forward masking NN method were applied to infer the already known BW of the cochlear filter used in the model. We show that isolevel methods were overall more accurate than isoresponse methods. We also show that BWs for NNs and sinusoids equate only for isolevel methods and when the levels of the two stimuli are appropriately scaled. Lastly, we show that the inferred BW depends on the method version (isolevel BW was twice as broad as isoresponse BW at 40 dB SPL) and on the stimulus level (isoresponse and isolevel BW decreased and increased, respectively, with increasing level over the level range where cochlear responses went from linear to compressive). We suggest that the latter may contribute to explaining the reported differences in cochlear tuning across behavioral studies and species. We further suggest that given the well-established nonlinear nature of cochlear responses, even greater care must be exercised when using a single BW value to describe and compare cochlear tuning.

Restoration of loudness summation and differential loudness growth in hearing-impaired listeners

When normal-hearing (NH) listeners compare the loudness of narrowband and wideband sounds presented at identical sound pressure levels, the wideband sound will most often be perceived as louder than the narrowband sound, a phenomenon referred to as loudness summation. Hearing-impaired (HI) listeners typically show less-than-normal loudness summation, due to reduced cochlear compressive gain and degraded frequency selectivity. In the present study, loudness summation at 1 and 3 kHz was estimated monaurally for five NH and eight HI listeners by matching the loudness of narrowband and wideband noise stimuli. The loudness summation was measured as a function both of noise bandwidth and level. The HI listeners were tested unaided and aided using three different compression systems to investigate the possibility of restoring loudness summation in these listeners. A compression system employing level-dependent compression channels yielded the most promising outcome. The present results inform the development of future loudness models and advanced compensation strategies for the hearing impaired.

Notionally steady background noise acts primarily as a modulation masker of speech

Stone et al. [J. Acoust. Soc Am. 130, 2874-2881 (2011)], using vocoder processing, showed that the envelope modulations of a notionally steady noise were more effective than the envelope energy as a masker of speech. Here the same effect is demonstrated using non-vocoded signals. Speech was filtered into 28 channels. A masker centered on each channel was added to the channel signal at a target-to-background ratio of -5 or -10 dB. Maskers were sinusoids or noise bands with bandwidth 1/3 or 1 ERB(N) (ERB(N) being the bandwidth of "normal" auditory filters), synthesized with Gaussian (GN) or low-noise (LNN) statistics. To minimize peripheral interactions between maskers, odd-numbered channels were presented to one ear and even to the other. Speech intelligibility was assessed in the presence of each "steady" masker and that masker 100% sinusoidally amplitude modulated (SAM) at 8 Hz. Intelligibility decreased with increasing envelope fluctuation of the maskers. Masking release, the difference in intelligibility between the SAM and its "steady" counterpart, increased with bandwidth from near-zero to around 50 percentage points for the 1-ERB(N) GN. It is concluded that the sinusoidal and GN maskers behaved primarily as energetic and modulation maskers, respectively.

The role of time and place cues in the detection of frequency modulation by hearing-impaired listeners

Frequency modulation detection limens (FMDLs) were measured for five hearing-impaired (HI) subjects for carrier frequencies f(c) = 1000, 4000, and 6000 Hz, using modulation frequencies f(m) = 2 and 10 Hz and levels of 20 dB sensation level and 90 dB SPL. FMDLs were smaller for f(m) = 10 than for f(m) = 2 Hz for the two higher f(c), but not for f(c) = 1000 Hz. FMDLs were also determined with additional random amplitude modulation (AM), to disrupt excitation-pattern cues. The disruptive effect was larger for f(m) = 10 than for f(m) = 2 Hz. The smallest disruption occurred for f(m) = 2 Hz and f(c) = 1000 Hz. AM detection thresholds for normal-hearing and HI subjects were measured for the same f(c) and f(m) values. Performance was better for the HI subjects for both f(m). AM detection was much better for f(m)  = 10 than for f(m) = 2 Hz. Additional tests showed that most HI subjects could discriminate temporal fine structure (TFS) at 800 Hz. The results are consistent with the idea that, for f(m) = 2 Hz and f(c) = 1000 Hz, frequency modulation (FM) detection was partly based on the use of TFS information. For higher carrier frequencies and for all carrier frequencies with f(m) = 10 Hz, FM detection was probably based on place cues.

Across-channel timing differences as a potential code for the frequency of pure tones

When a pure tone or low-numbered harmonic is presented to a listener, the resulting travelling wave in the cochlea slows down at the portion of the basilar membrane (BM) tuned to the input frequency due to the filtering properties of the BM. This slowing is reflected in the phase of the response of neurons across the auditory nerve (AN) array. It has been suggested that the auditory system exploits these across-channel timing differences to encode the pitch of both pure tones and resolved harmonics in complex tones. Here, we report a quantitative analysis of previously published data on the response of guinea pig AN fibres, of a range of characteristic frequencies, to pure tones of different frequencies and levels. We conclude that although the use of across-channel timing cues provides an a priori attractive and plausible means of encoding pitch, many of the most obvious metrics for using that cue produce pitch estimates that are strongly influenced by the overall level and therefore are unlikely to provide a straightforward means for encoding the pitch of pure tones.

A new method of calculating auditory excitation patterns and loudness for steady sounds

A new method for calculating auditory excitation patterns and loudness for steady sounds is described. The method is based on a nonlinear filterbank in which each filter is the sum of a broad passive filter and a sharp active filter. All filters have a rounded-exponential shape. For each center frequency (CF), the gain of the active filter is controlled by the output of the passive filter. The parameters of the model were derived from large sets of previously published notched-noise masking data obtained from human subjects. Excitation patterns derived using the new filterbank include the effects of basilar membrane compression. Loudness can be calculated as the area under the excitation pattern when plotted in intensity-like units on an ERB(N)-number (Cam) scale; no transformation from excitation to specific loudness is required. The method predicts the standard equal-loudness contours and loudness as a function of bandwidth with good accuracy. With some additional assumptions, the method also gives reasonably accurate predictions of partial loudness.

Effect of level on the discrimination of harmonic and frequency-shifted complex tones at high frequencies

Moore and Sęk [J. Acoust. Soc. Am. 125, 3186-3193 (2009)] measured discrimination of a harmonic complex tone and a tone in which all harmonics were shifted upwards by the same amount in Hertz. Both tones were passed through a fixed bandpass filter and a background noise was used to mask combination tones. Performance was well above chance when the fundamental frequency was 800 Hz, and all audible components were above 8000 Hz. Moore and Sęk argued that this suggested the use of temporal fine structure information at high frequencies. However, the task may have been performed using excitation-pattern cues. To test this idea, performance on a similar task was measured as a function of level. The auditory filters broaden with increasing level, so performance based on excitation-pattern cues would be expected to worsen as level increases. The results did not show such an effect, suggesting that the task was not performed using excitation-pattern cues.

Psychophysical tuning curves for frequencies below 100 Hz

Psychophysical tuning curves (PTCs) were measured for sinusoidal signals with frequency f(s) = 31.5, 40, 50, 63, and 80 Hz, using sinusoidal and narrowband-noise maskers. For the former, conditions were included where a pair of beating tones was added to reduce the use of cues related to beats. Estimates of each subject's middle-ear transfer function (METF) were obtained from equal-loudness contours measured from 20 to 160 Hz. With decreasing f(s), the PTCs became progressively broadened and markedly asymmetrical, with shallow upper skirts and steep lower skirts. For the sinusoidal maskers, the tips were more irregular than for narrowband-noise maskers or when beating tones were added. For f(s) = 31.5 and 40 Hz, the tips of the PTCs always fell above f(s). Allowing for the METF so as to infer underlying filter shapes resulted in flatter lower skirts, especially below 40 Hz, and reduced the frequency at the tips for f(s) between 31.5 and 50 Hz; however, the tips did not fall below 40 to 50 Hz. The bandwidths of the PTCs increased with decreasing f(s) below 80 Hz. However, bandwidths remained roughly constant if the METF was included as part of auditory filtering for frequencies below 40 Hz.

Auditory filter shapes and high-frequency hearing in adults who have impaired speech in noise performance despite clinically normal audiograms

Some individuals complain of hearing difficulties in the presence of background noise even in the absence of clinically significant hearing loss (obscure auditory dysfunction). Previous evidence suggests that these listeners have impaired frequency resolution, but there has been no thorough characterization of auditory filter shapes in this population. Here, the filter shapes of adults (n = 14) who self-reported speech recognition problems in noise and performed poorly on a sentence-in-noise perception test despite having clinically normal audiograms were compared to those of controls (n = 10). The filter shapes were evaluated using a 2-kHz probe with a fixed level of 30, 40, or 50 dB sound pressure level (SPL) and notched-noise simultaneous maskers that were varied in level to determine the masker level necessary to just mask the probe. The filters of the impaired group were significantly wider than those of controls at all probe levels owing to an unusual broadening of the upper slope of the filter. In addition, absolute thresholds were statistically indistinguishable between the groups at the standard audiometric frequencies, but were elevated in the impaired listeners at higher frequencies. These results strengthen the idea that this population has a variety of hearing deficits that go undetected by standard audiometry.

Resolvability of components in complex tones and implications for theories of pitch perception

This paper reviews methods that have been used to estimate the resolvability of individual partials in harmonic and inharmonic complex tones and considers the implications of the results for theories of pitch perception. The methods include: requiring comparisons of the pitch of an isolated pure tone and a partial within a complex tone as a measure of the ability to "hear out" that partial; considering the magnitude of ripples in the calculated excitation pattern of a complex tone; using a complex tone as a forward masker and using ripples in the masking pattern to estimate resolvability; measuring sensitivity to the relative phase of the components within complex tones. The measures are broadly consistent in indicating that harmonics with numbers up to about five are well resolved, but that resolution decreases for higher harmonics. Most measures suggest that harmonics with numbers above eight are poorly, if at all, resolved. However, there are uncertainties associated with each method that make the exact upper limit of resolvability uncertain. Evidence is presented suggesting a partial dissociation between resolution in the excitation pattern and the ability to hear out a partial. It is proposed that the latter requires information from temporal fine structure (phase locking).

Frequency selectivity for frequencies below 100 Hz: comparisons with mid-frequencies

Auditory filter shapes were derived for signal frequencies (f(s)) between 50 and 1000 Hz, using the notched-noise method. The masker spectrum level (N(0)) was 50 dB (re 20 μPa). For f(s) = 63 and 50 Hz, measurements were also made with N(0) = 62 dB for the lower band. The data were fitted using a rounded-exponential filter model, with special consideration of the filtering effects of the middle-ear transfer function (METF) at low frequencies. The results showed: (1) For very low values of f(s), the lower skirts of the filters were only well defined when N(0) = 62 dB for the lower band; (2) the sharpness of both sides of the filters decreased with decreasing f(s); (3) the dynamic range of the filters decreased with decreasing f(s); (4) the equivalent rectangular bandwidth of the filters decreased with decreasing f(s) down to f(s) = 80 Hz, but increased for f(s) below that; (5) the assumed METF, which includes the shunt effect of the helicotrema for frequencies below 50 Hz, increasingly influenced the low-frequency skirt of the filters as f(s) decreased; and (6) detection efficiency worsened with decreasing f(s) for f(s) between 100 and 500 Hz, but improved slightly below that.

Mechanisms underlying the detection of frequency modulation

Frequency modulation detection limens (FMDLs) were measured for carrier frequencies (f(c)) of 1000, 4000, and 6000 Hz, using modulation frequencies (f(m)) of 2 and 10 Hz and levels of 20 and 60 dB sensation level (SL), both with and without random amplitude modulation (AM), applied in all intervals of a forced-choice trial. The AM was intended to disrupt excitation-pattern cues. At 60 dB SL, the deleterious effect of the AM was smaller for f(m) = 2 than for f(m) = 10 Hz for f(c) = 1000 and 4000 Hz, respectively, while for f(c) = 6000 Hz the deleterious effect was large and similar for the two values of f(m). This is consistent with the idea that, for f(c) below about 5000 Hz and f(m) = 2 Hz, frequency modulation can be detected via changes in phase locking over time. However, at 20 dB SL, the deleterious effect of the added AM for f(c) = 1000 and 4000 Hz was similar for the two values of f(m), while for f(c) = 6000 Hz, the deleterious effect of the AM was greater for f(m) = 10 than for f(m) = 2 Hz. It is suggested that, at low SLs, the auditory filters become relatively sharp and phase locking weakens, so that excitation-pattern cues influence FMDLs even for low f(c) and low f(m).

Isoresponse versus isoinput estimates of cochlear filter tuning

The tuning of a linear filter may be inferred from the filter's isoresponse (e.g., tuning curves) or isoinput (e.g., isolevel curves) characteristics. This paper provides a theoretical demonstration that for nonlinear filters with compressive response characteristics like those of the basilar membrane, isoresponse measures can suggest strikingly sharper tuning than isoinput measures. The practical significance of this phenomenon is demonstrated by inferring the 3-dB-down bandwidths (BW(3dB)) of human auditory filters at 500 and 4,000 Hz from behavioral isoresponse and isoinput measures obtained with sinusoidal and notched noise forward maskers. Inferred cochlear responses were compressive for the two types of maskers. Consistent with expectations, low-level BW(3dB) estimates obtained from isoresponse conditions were considerably narrower than those obtained from isolevel conditions: 69 vs. 174 Hz, respectively, at 500 Hz, and 280 vs. 464 Hz, respectively, at 4,000 Hz. Furthermore, isoresponse BW(3dB) decreased with increasing level while corresponding isolevel estimates remained approximately constant at 500 Hz or increased slightly at 4 kHz. It is suggested that comparisons between isoresponse supra-threshold human tuning and threshold animal neural tuning should be made with caution.

Diagnosing cochlear dead regions in children

OBJECTIVE

A dead region (DR) is defined as a region in the cochlea where inner hair cells and/or neurons are functioning so poorly that a tone producing peak vibration in this region is detected by off-frequency listening, i.e., via a place on the basilar membrane with a characteristic frequency different from that of the tone. The presence of a DR can have a significant effect on the perception of speech. People with and without DRs may differ in the benefit obtained from amplification and require different hearing aid settings. The Threshold Equalizing Noise (TEN) test and psychophysical tuning curves (PTCs) are two procedures used to identify a DR in adults. Because diagnosing a DR involves measuring masked thresholds, and there are reports in the literature that young children perform poorly compared with adults in background noise, it may be possible that the criteria used with adults may not be appropriate when testing children. Therefore, the aim of this study was to evaluate the consistency of the fast-PTC and TEN tests in diagnosing a DR in hearing-impaired children. In addition, the masked thresholds for normal-hearing children were measured with different TEN levels to assess whether any age-related effect in children compared with adults may occur.

DESIGN

Participants were divided into two groups: eight normal-hearing children (16 ears) and 12 hearing-impaired children (21 ears), aged 7 to 13 yr. TEN is based on measuring masked threshold in TEN. For normal-hearing participants, the masked thresholds were measured for five levels of noise (30, 40, 50, 60, and 70 dB per averaged equivalent rectangular bandwidth). For hearing-impaired participants, the level of the TEN was selected separately for each ear based on the highest acceptable level minus 5 dB. The TEN test results in hearing-impaired children were further validated by measuring fast-PTCs. The fast-PTC technique involves measuring the level of the narrowband noise masker needed to mask the signal. The center frequency of the masker sweeps across the required frequency range.

RESULTS

The masked thresholds in TEN measured for normal-hearing children were usually below and never higher than 5 dB above TEN level per averaged equivalent rectangular bandwidth. This suggests that no age-related effect on masked threshold in children compared with adults was observed. All hearing-impaired children were able to perform the TEN test and fast-PTCs. The results of the two tests were consistent in 17 of 21 ears (81%): eight ears did not show evidence of a DR and nine ears did. In three ears, the criteria for a DR were met on the TEN test, but there was no evidence of a DR on the fast-PTC test. In one ear, the TEN test did not show evidence of DRs at two frequencies, whereas fast-PTCs did.

CONCLUSIONS

The results of this study suggest that DRs can be detected in children using the fast-PTC technique and the TEN test interpreted with the adult criteria, which are the most appropriate in terms of specificity and sensitivity. However, in cases in which the masked threshold is 10 to 15 dB above the TEN level, it is recommended to confirm DR diagnosis with fast-PTC measurement.

Measurement of the binaural auditory filter using a detection task

The spectral resolution of the binaural system was measured using a tone-detection task in a binaural analog of the notched-noise technique. Three listeners performed 2-interval, 2-alternative, forced choice tasks with a 500-ms out-of-phase signal within 500 ms of broadband masking noise consisting of an "outer" band of either interaurally uncorrelated or anticorrelated noise, and an "inner" band of interaurally correlated noise. Three signal frequencies were tested (250, 500, and 750 Hz), and the asymmetry of the filter was measured by keeping the signal at a constant frequency and moving the correlated noise band relative to the signal. Thresholds were taken for bandwidths of correlated noise ranging from 0 to 400 Hz. The equivalent rectangular bandwidth of the binaural filter was found to increase with signal frequency, and estimates tended to be larger than monaural bandwidths measured for the same listeners using equivalent techniques.

Spatiotemporal reconstruction of the auditory steady-state response to frequency modulation using magnetoencephalography

The aim of this study was to investigate the mechanisms involved in the perception of perceptually salient frequency modulation (FM) using auditory steady-state responses (ASSRs) measured with magnetoencephalography (MEG). Previous MEG studies using frequency-modulated amplitude modulation as stimuli (Luo et al., 2006, 2007) suggested that a phase modulation encoding mechanism exists for low (<5 Hz) FM modulation frequencies but additional amplitude modulation encoding is required for faster FM modulation frequencies. In this study single-cycle sinusoidal FM stimuli were used to generate the ASSR. The stimulus was either an unmodulated 1-kHz sinusoid or a 1-kHz sinusoid that was frequency-modulated with a repetition rate of 4, 8, or 12 Hz. The fast Fourier transform (FFT) of each MEG channel was calculated to obtain the phase and magnitude of the ASSR in sensor-space and multivariate Hotelling's T(2) statistics were used to determine the statistical significance of ASSRs. MEG beamformer analyses were used to localise the ASSR sources. Virtual electrode analyses were used to reconstruct the time series at each source. FFTs of the virtual electrode time series were calculated to obtain the amplitude and phase characteristics of each source identified in the beamforming analyses. Multivariate Hotelling's T(2) statistics were used to determine the statistical significance of these reconstructed ASSRs. The results suggest that the ability of auditory cortex to phase-lock to FM is dependent on the FM pulse rate and that the ASSR to FM is lateralised to the right hemisphere.

Comparison of the roex and gammachirp filters as representations of the auditory filter

Although the rounded-exponential (roex) filter has been successfully used to represent the magnitude response of the auditory filter, recent studies with the roex(p, w, t) filter reveal two serious problems: the fits to notched-noise masking data are somewhat unstable unless the filter is reduced to a physically unrealizable form, and there is no time-domain version of the roex(p, w, t) filter to support modeling of the perception of complex sounds. This paper describes a compressive gammachirp (cGC) filter with the same architecture as the roex(p, w, t) which can be implemented in the time domain. The gain and asymmetry of this parallel cGC filter are shown to be comparable to those of the roex(p, w, t) filter, but the fits to masking data are still somewhat unstable. The roex(p, w, t) and parallel cGC filters were also compared with the cascade cGC filter [Patterson et al., J. Acoust. Soc. Am. 114, 1529-1542 (2003)], which was found to provide an equivalent fit with 25% fewer coefficients. Moreover, the fits were stable. The advantage of the cascade cGC filter appears to derive from its parsimonious representation of the high-frequency side of the filter. It is concluded that cGC filters offer better prospects than roex filters for the representation of the auditory filter.